From msc at freeswitch.org Fri Feb 1 00:06:26 2013 From: msc at freeswitch.org (Michael Collins) Date: Thu, 31 Jan 2013 13:06:26 -0800 Subject: [Freeswitch-users] Execute app in sync mode when using event sockets In-Reply-To: References: Message-ID: Are you using the Java ESL library? If so it should have what you need to do sync vs. async on sending execute commands to the event socket. If you are not using ESL I would advise you to do so since it has solved this problem (and others) already. More info here: http://wiki.freeswitch.org/wiki/Event_Socket_Library -MC On Thu, Jan 31, 2013 at 11:18 AM, Hector Geraldino < Hector.Geraldino at ipsoft.com> wrote: > Hi,**** > > ** ** > > I have a java app that uses mod_event_socket (inbound mode) to control > FreeSWITCH. I can send api commands which, by default, are synchronous, so > the thread locks until the command is successfully executed. The problem > I?m facing is when I have to execute commands that are in other modules > (like playback, or speak). As they are not api commands, I must send an > execute appName + arguments, which by nature is executed in async mode.*** > * > > ** ** > > What I want to achieve is to execute an speak command and wait until the > TTS operation ends (blocking the thread). This sounds like a trivial use > case: play some audio (TTS, file) and then execute the rest of the logic. > It is easy to accomplish on the dialplan, but I just can?t figure out how > to do it on an ESL application. Here?s what I?ve tried so far:**** > > ** ** > > + send execute speak command + api sleep and wait for the > CHANNEL_EXECUTE_COMPLETE (to simulate a lock [sleep] and continue [break > after event is received]), but the event arrives after the sleep command > completes. **** > > + use uuid_broadcast uuid speak::args, doesn?t work**** > > ** ** > > Any hints, ideas are welcomed.**** > > ** ** > > Thank you!**** > > Hector**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130131/ff1a8bcd/attachment.html From sdevoy at bizfocused.com Fri Feb 1 00:07:16 2013 From: sdevoy at bizfocused.com (Sean Devoy) Date: Thu, 31 Jan 2013 16:07:16 -0500 Subject: [Freeswitch-users] simple install question In-Reply-To: References: <091801cdffef$6a4229c0$3ec67d40$@bizfocused.com> Message-ID: <097d01cdfff6$f3531b00$d9f95100$@bizfocused.com> Thanks everyone. I will update the wiki IF this works! No answer on stopping the running FS though? Sean From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi Marcus Sent: Thursday, January 31, 2013 3:32 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] simple install question It would be nice to have instructions for this somewhere... some people tar up their old build directory, or at least the compiled files. That way if the build/install goes bonkers, they can untar and re-install their old binaries. -Avi On Thu, Jan 31, 2013 at 10:24 PM, Nick Vines wrote: Stolen from an email from Ken. "If you need to switch between branches, you _must_ git clean -fdx and rm all the existing binaries in your existing FreeSWITCH install for the mod lib and bin directories or you will most likely end up with build skew and crazy weird segfaults..." I think it is possible to install while freeswitch is running, but highly advised against. Just do it at a 3am maintenance outage. You should be able to run the bootstrap and configure before you shutdown your FS server to reduce downtime. On the base rackspace cloud instance (512mb version), it takes about 10 min total from shutdown to startup of FS. On Thu, Jan 31, 2013 at 12:13 PM, Sean Devoy wrote: Hi, I want to switch from the stable release version to the git head and rebuild in hopes of ridding myself of the delay in connection of the audio. On almost all my calls I must hesitate about 1 second before saying hello or the caller does not hear me. I have seen many discussions saying that this is probably fixed in the git root. So here are my questions: The procedure appears to be: Log in to root cd /usr/local/src/freeswitch git checkout master Which responds: Branch master set up to track remote branch master from origin. Switched to a new branch 'master' Then just: make current Is that correct? Do I need to stop the running FS while running this? If so, how long will I be down? (ball park it) I realize this is a RTFM kind of question, but I can't risk an outage. I need confirmation from someone who understands. Thanks, Sean _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130131/2f7b5f2f/attachment.html From krice at freeswitch.org Fri Feb 1 00:10:56 2013 From: krice at freeswitch.org (Ken Rice) Date: Thu, 31 Jan 2013 15:10:56 -0600 Subject: [Freeswitch-users] simple install question In-Reply-To: <097d01cdfff6$f3531b00$d9f95100$@bizfocused.com> Message-ID: If you are reinstalling FreeSWITCH I would stop it while you are doing so... Sometimes you can get away with leaving it running but it will crash if it tries to access a lib or something that?s not there any longer K On 1/31/13 3:07 PM, "Sean Devoy" wrote: > Thanks everyone. I will update the wiki IF this works! > > No answer on stopping the running FS though? > > Sean > > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi Marcus > Sent: Thursday, January 31, 2013 3:32 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] simple install question > > > It would be nice to have instructions for this somewhere... > > some people tar up their old build directory, or at least the compiled files. > That way if the build/install goes bonkers, they can untar and re-install > their old binaries. > > > -Avi > > > On Thu, Jan 31, 2013 at 10:24 PM, Nick Vines wrote: > > Stolen from an email from Ken. > > > > "If you need to switch between branches, you _must_ git clean -fdx and rm all > the existing binaries in your existing FreeSWITCH install for the mod lib > and bin directories or you will most likely end up with build skew and crazy > weird segfaults..." > > > > I think it is possible to install while freeswitch is running, but highly > advised against. Just do it at a 3am maintenance outage. You should be able to > run the bootstrap and configure before you shutdown your FS server to reduce > downtime. > > > > On the base rackspace cloud instance (512mb version), it takes about 10 min > total from shutdown to startup of FS. > > > > On Thu, Jan 31, 2013 at 12:13 PM, Sean Devoy wrote: >> >> Hi, >> >> I want to switch from the stable release version to the git head and rebuild >> in hopes of ridding myself of the delay in connection of the audio. On >> almost all my calls I must hesitate about 1 second before saying hello or the >> caller does not hear me. I have seen many discussions saying that this is >> probably fixed in the git root. So here are my questions: >> >> The procedure appears to be: >> Log in to root >> cd /usr/local/src/freeswitch >> git checkout master >> Which responds: Branch master set up to track remote branch master from >> origin. >> Switched to a new branch 'master' >> Then just: >> make current >> >> Is that correct? >> Do I need to stop the running FS while running this? >> If so, how long will I be down? (ball park it) >> >> I realize this is a RTFM kind of question, but I can?t risk an outage. I >> need confirmation from someone who understands. >> >> Thanks, >> Sean >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130131/ccc586c9/attachment-0001.html From msc at freeswitch.org Fri Feb 1 00:16:13 2013 From: msc at freeswitch.org (Michael Collins) Date: Thu, 31 Jan 2013 13:16:13 -0800 Subject: [Freeswitch-users] simple install question In-Reply-To: <097d01cdfff6$f3531b00$d9f95100$@bizfocused.com> References: <091801cdffef$6a4229c0$3ec67d40$@bizfocused.com> <097d01cdfff6$f3531b00$d9f95100$@bizfocused.com> Message-ID: Personally I think you're okay to leave FS running through the bootstrap, configure and make. I would not, though, leave FS running through the make install. That's just me personally. I'm curious to hear what others have to say. -MC On Thu, Jan 31, 2013 at 1:07 PM, Sean Devoy wrote: > Thanks everyone. I will update the wiki IF this works!**** > > ** ** > > No answer on stopping the running FS though?**** > > ** ** > > Sean**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Avi Marcus > *Sent:* Thursday, January 31, 2013 3:32 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] simple install question**** > > ** ** > > It would be nice to have instructions for this somewhere...**** > > some people tar up their old build directory, or at least the compiled > files. That way if the build/install goes bonkers, they can untar and > re-install their old binaries.**** > > > **** > > -Avi**** > > ** ** > > On Thu, Jan 31, 2013 at 10:24 PM, Nick Vines wrote:*** > * > > Stolen from an email from Ken. **** > > ** ** > > "If you need to switch between branches, you _must_ git clean -fdx and rm > all**** > > the existing binaries in your existing FreeSWITCH install for the mod lib > and bin directories or you will most likely end up with build skew and > crazy > weird segfaults..."**** > > ** ** > > I think it is possible to install while freeswitch is running, but highly > advised against. Just do it at a 3am maintenance outage. You should be able > to run the bootstrap and configure before you shutdown your FS server to > reduce downtime. **** > > ** ** > > On the base rackspace cloud instance (512mb version), it takes about 10 > min total from shutdown to startup of FS. **** > > ** ** > > On Thu, Jan 31, 2013 at 12:13 PM, Sean Devoy > wrote:**** > > Hi,**** > > **** > > I want to switch from the stable release version to the git head and > rebuild in hopes of ridding myself of the delay in connection of the > audio. On almost all my calls I must hesitate about 1 second before saying > hello or the caller does not hear me. I have seen many discussions saying > that this is probably fixed in the git root. So here are my questions:*** > * > > **** > > The procedure appears to be:**** > > Log in to root**** > > cd /usr/local/src/freeswitch**** > > git checkout master**** > > Which responds: Branch master set up to track remote branch master from > origin.**** > > Switched to a new branch 'master'**** > > Then just: **** > > make current**** > > **** > > Is that correct?**** > > Do I need to stop the running FS while running this?**** > > If so, how long will I be down? (ball park it)**** > > **** > > I realize this is a RTFM kind of question, but I can?t risk an outage. I > need confirmation from someone who understands.**** > > **** > > Thanks,**** > > Sean**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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**** > > ** ** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130131/ff68826d/attachment.html From msc at freeswitch.org Fri Feb 1 00:24:53 2013 From: msc at freeswitch.org (Michael Collins) Date: Thu, 31 Jan 2013 13:24:53 -0800 Subject: [Freeswitch-users] Low Volume levels on ip phones In-Reply-To: <1359611396.16353.95.camel@mythtv.toddbailey.net> References: <1341685164.3454.51.camel@mythtv> <1341748446.53554.YahooMailNeo@web39304.mail.mud.yahoo.com> <1341812640.3767.6.camel@mythtv> <1341823757.69136.YahooMailNeo@web39303.mail.mud.yahoo.com> <1341853685.4860.3.camel@mythtv> <1359611396.16353.95.camel@mythtv.toddbailey.net> Message-ID: There's not a whole lot you have left that you can do. There is one thing to try: http://wiki.freeswitch.org/wiki/Mod_commands#uuid_audio The max you can set it to is 4 but I suppose that's better than what you've currently got. -MC On Wed, Jan 30, 2013 at 9:49 PM, Todd Bailey wrote: > Hello All, > > Is there an easy way to boost volume levels going to a ip phone? > I have the volume set to max and the volume levels is still very quiet? > I already adjusted the ata a spa3102 to boost the pstn to spa gain to > max which helped a little but not by much > > The primary phones are polycom 501's > > > thanks > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130131/543f1227/attachment.html From krice at freeswitch.org Fri Feb 1 00:26:23 2013 From: krice at freeswitch.org (Ken Rice) Date: Thu, 31 Jan 2013 15:26:23 -0600 Subject: [Freeswitch-users] FreeSWITCH Mirror on Github Message-ID: Hey Guy, As many of you know there is a Github mirror of the GIT repo at https://github.com/FreeSWITCH/FreeSWITCH . This is intended as a READ ONLY mirror of our master repos. I have noticed that some people are actually doing push request and opening issues there. Please do not do that. The Core dev team rarely (if ever) reviews the other functions that Github provides. If you have an issue or a possible patch, please open a jira and attach the patch. If you have already patched your local copy, using the ?git diff? command to get the patch will work, just pipe its output to a .txt file and attach to your Jira. Thanks Ken -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130131/c04bebcb/attachment-0001.html From Hector.Geraldino at ipsoft.com Fri Feb 1 00:34:46 2013 From: Hector.Geraldino at ipsoft.com (Hector Geraldino) Date: Thu, 31 Jan 2013 21:34:46 +0000 Subject: [Freeswitch-users] Execute app in sync mode when using event sockets In-Reply-To: References: Message-ID: Well, I'm not using the Java ESL library but the Java ESL Client library (which doesn't require the native libraries and is pure java-based). Using this lib, when I send an execute command (using the SendMsg + execute: appName + execute-app-args: arguments) the thread isn't being locked, so that's my problem. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Thursday, January 31, 2013 4:06 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Execute app in sync mode when using event sockets Are you using the Java ESL library? If so it should have what you need to do sync vs. async on sending execute commands to the event socket. If you are not using ESL I would advise you to do so since it has solved this problem (and others) already. More info here: http://wiki.freeswitch.org/wiki/Event_Socket_Library -MC On Thu, Jan 31, 2013 at 11:18 AM, Hector Geraldino > wrote: Hi, I have a java app that uses mod_event_socket (inbound mode) to control FreeSWITCH. I can send api commands which, by default, are synchronous, so the thread locks until the command is successfully executed. The problem I'm facing is when I have to execute commands that are in other modules (like playback, or speak). As they are not api commands, I must send an execute appName + arguments, which by nature is executed in async mode. What I want to achieve is to execute an speak command and wait until the TTS operation ends (blocking the thread). This sounds like a trivial use case: play some audio (TTS, file) and then execute the rest of the logic. It is easy to accomplish on the dialplan, but I just can't figure out how to do it on an ESL application. Here's what I've tried so far: + send execute speak command + api sleep and wait for the CHANNEL_EXECUTE_COMPLETE (to simulate a lock [sleep] and continue [break after event is received]), but the event arrives after the sleep command completes. + use uuid_broadcast uuid speak::args, doesn't work Any hints, ideas are welcomed. Thank you! Hector _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130131/9d078f0d/attachment.html From msc at freeswitch.org Fri Feb 1 00:34:53 2013 From: msc at freeswitch.org (Michael Collins) Date: Thu, 31 Jan 2013 13:34:53 -0800 Subject: [Freeswitch-users] Recording / concurrent calls and performance In-Reply-To: References: Message-ID: Yes, it will take some resources, like disk i/o and maybe a little CPU depending on the read/write codecs. However, I don't think there's an easy way to quantify how much it will slow things down without just loading up the system with some calls and see what happens. -MC On Tue, Jan 29, 2013 at 3:51 PM, Malay Thakershi wrote: > Hello, > > Does recording audio sessions slow down performance of concurrent calls / > FreeSwitch? > > I am doing: > application="record_session" > in the dial plan before transferring the call to managed DLL. > > Thanks for help. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130131/7ab14dc2/attachment.html From msc at freeswitch.org Fri Feb 1 01:13:11 2013 From: msc at freeswitch.org (Michael Collins) Date: Thu, 31 Jan 2013 14:13:11 -0800 Subject: [Freeswitch-users] Execute app in sync mode when using event sockets In-Reply-To: References: Message-ID: That's too bad. Not sure what else you can do except maybe go into a loop after you send your execute command and then check every event that comes back and ignore everything until you get the execute complete event for the specific command you sent. If you have issued "myevents" that will be easier, but in any case you'll need to come up with a way to handle all that. Or you could use ESL. :) -MC On Thu, Jan 31, 2013 at 1:34 PM, Hector Geraldino < Hector.Geraldino at ipsoft.com> wrote: > Well, I?m not using the Java ESL library but the Java ESL Client library > (which doesn?t require the native libraries and is pure java-based).**** > > ** ** > > Using this lib, when I send an execute command (using the SendMsg + > execute: appName + execute-app-args: arguments) the thread isn?t being > locked, so that?s my problem.**** > > ** ** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael > Collins > *Sent:* Thursday, January 31, 2013 4:06 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Execute app in sync mode when using > event sockets**** > > ** ** > > Are you using the Java ESL library? If so it should have what you need to > do sync vs. async on sending execute commands to the event socket. If you > are not using ESL I would advise you to do so since it has solved this > problem (and others) already. > > More info here: > http://wiki.freeswitch.org/wiki/Event_Socket_Library > > -MC > > **** > > On Thu, Jan 31, 2013 at 11:18 AM, Hector Geraldino < > Hector.Geraldino at ipsoft.com> wrote:**** > > Hi,**** > > **** > > I have a java app that uses mod_event_socket (inbound mode) to control > FreeSWITCH. I can send api commands which, by default, are synchronous, so > the thread locks until the command is successfully executed. The problem > I?m facing is when I have to execute commands that are in other modules > (like playback, or speak). As they are not api commands, I must send an > execute appName + arguments, which by nature is executed in async mode.*** > * > > **** > > What I want to achieve is to execute an speak command and wait until the > TTS operation ends (blocking the thread). This sounds like a trivial use > case: play some audio (TTS, file) and then execute the rest of the logic. > It is easy to accomplish on the dialplan, but I just can?t figure out how > to do it on an ESL application. Here?s what I?ve tried so far:**** > > **** > > + send execute speak command + api sleep and wait for the > CHANNEL_EXECUTE_COMPLETE (to simulate a lock [sleep] and continue [break > after event is received]), but the event arrives after the sleep command > completes. **** > > + use uuid_broadcast uuid speak::args, doesn?t work**** > > **** > > Any hints, ideas are welcomed.**** > > **** > > Thank you!**** > > Hector**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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**** > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130131/5c838de6/attachment-0001.html From schoch+freeswitch.org at xwin32.com Fri Feb 1 01:19:59 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Thu, 31 Jan 2013 14:19:59 -0800 Subject: [Freeswitch-users] Non-numeric extension - bad idea? Message-ID: We have a main number and extensions, but some people have their own DID number that rings on line 2 (of a 2-line Polycom phone), with a separate voice mailbox. This "line 2" is not supposed to be an extension, so in its directory entry, I have the "effective_caller_id_number" set to the DID number. I don't want this line to be a real extension, so I set the user id to be a descriptive short name, not a number. Is this a bad idea? The problem is that it's impossible to check voice mail on that line because the user can't dial the user id. Would it be better to use the 10-digit number as the user id? Or should I use an extension number that starts with a different digit? That would solve the voice mail problem as well, because the dialplan passes the caller-id to the voicemail check application. -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130131/a51ebac5/attachment.html From bdfoster at endigotech.com Fri Feb 1 01:30:48 2013 From: bdfoster at endigotech.com (Brian Foster) Date: Thu, 31 Jan 2013 17:30:48 -0500 Subject: [Freeswitch-users] Non-numeric extension - bad idea? In-Reply-To: References: Message-ID: By default we'll take the last 4 of the DID and make that the extension number. DID just gets redirected to the extension number. Basically everything's on one line. Effective caller id number is set to internal 4 digit extension, outbound caller id number is set to DID. Sent from my iPhone On Jan 31, 2013, at 5:19 PM, Steven Schoch wrote: > We have a main number and extensions, but some people have their own DID number that rings on line 2 (of a 2-line Polycom phone), with a separate voice mailbox. > > This "line 2" is not supposed to be an extension, so in its directory entry, I have the "effective_caller_id_number" set to the DID number. > > I don't want this line to be a real extension, so I set the user id to be a descriptive short name, not a number. Is this a bad idea? The problem is that it's impossible to check voice mail on that line because the user can't dial the user id. > > Would it be better to use the 10-digit number as the user id? Or should I use an extension number that starts with a different digit? That would solve the voice mail problem as well, because the dialplan passes the caller-id to the voicemail check application. > > -- > Steve > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 schoch+freeswitch.org at xwin32.com Fri Feb 1 01:34:43 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Thu, 31 Jan 2013 14:34:43 -0800 Subject: [Freeswitch-users] DID providers - any thoughts? (jan 2013) In-Reply-To: References: Message-ID: On Wed, Jan 30, 2013 at 4:59 AM, Cal Leeming [Simplicity Media Ltd] < cal.leeming at simplicitymedialtd.co.uk> wrote: > > Flowroute - Any else able to offer some reviews on this company - > specifically relating to US inbound?? > I'm just starting my VoIP venture and I have chosen Flowroute as our provider. So far, mostly as a test, I have transferred one of our AT&T lines to Flowroute. It worked very smoothly. One thing I discovered, which you all probably know, but I'm new with VoIP, was that when I called the number from a POTS line, it went through the PSTN, through Flowroute, through FreeSwitch, and rang on my Polycom phone even before I heard the first ring from the calling phone. Also, the Flowroute tech support team responds very quickly and helpfully. No complaints. -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130131/5774cab5/attachment.html From eburke at edge-net.net Fri Feb 1 01:43:34 2013 From: eburke at edge-net.net (Eli Burke) Date: Thu, 31 Jan 2013 17:43:34 -0500 Subject: [Freeswitch-users] problems with freeswitch + zrtp in proxy-media mode In-Reply-To: References: Message-ID: I'm trying to use Freeswitch with an open source voip client called Linphone, but cannot manage to establish functioning ZRTP sessions. The problem *seems* to be related to the RTP stream's SSRC. I'm trying to conclusively determine if the problem lies with Linphone, Freeswitch, or one of the underlying zrtp implementations (libzrtp and zrtpcpp). Here's what I've determined so far: 1) ZRTP support is compiled and functioning. I can establish ZRTP sessions between two iPhones running Groundwire. 2) linphone + ZRTP works correctly if I make a one legged call to a Freeswitch echo extension 3) linphone + ZRTP works correctly if I turn on bypass-media 4) linphone gets a CRC error when trying to verify the ZRTP packet on any two-legged call 5) disabling the CRC check results in a successful ZRTP session, but the audio is extremely garbled Is Freeswitch modifying the ZRTP packets when running in proxy mode? If so, how is it that some clients are able to handle this situation? Thanks for any suggestions, Eli From schoch+freeswitch.org at xwin32.com Fri Feb 1 04:35:58 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Thu, 31 Jan 2013 17:35:58 -0800 Subject: [Freeswitch-users] FAX problems In-Reply-To: <5102AF93.2000201@coppice.org> References: <20130119191943.ca09ab6d@mail.tritonwest.net> <5102AF93.2000201@coppice.org> Message-ID: I'm still struggling with sending FAXes from a Windows 7 system. I believe my requirements are simple: The ability to send a document (like a Word document or a QuickBooks invoice) directly from the Windows system to a FAX number. I don't want to have to use a physical modem and an ATA device. I don't want to run Windows in "Test" mode in order to use an unsigned driver. I don't mind paying some money for a commercial software product. The setup is simple: I have FreeSwitch installed on a Linux machine, on our LAN. Also on the LAN I have several Windows 7 machines and an ATA device (an OBi100). An analog FAX machine is plugged into the ATA. What works: I can send a fax from the FreeSwitch machine using txfax. I can send a fax through the ATA and the FreeSwitch machine using the analog FAX machine. What doesn't: I tried www.t38faxvoip.com through FreeSwitch. It doesn't work. I tried www.t38faxvoip.com directly to Flowroute. That didn't work either. I called the analog fax machine through the ATA and FreeSwitch from www.t38faxvoip.com. I tried www.faxback.com. Same results. The people at www.t38faxvoip.com at least responded to my emails, but said they didn't know what was happening after looking at my Wireshark dump. All of the tests from Windows behaved the same. The called FAX machine answered, but never completed the handshake. The software reported that the call was answered by something other than a FAX machine. Yes, we can live with printing out the document and sticking it in the FAX machine, but it sure would be nice to be able to save some paper. Can't anyone solve my problem? -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130131/8be14a7c/attachment.html From itsusama at gmail.com Fri Feb 1 04:44:50 2013 From: itsusama at gmail.com (Usama Zaidi) Date: Fri, 1 Feb 2013 06:44:50 +0500 Subject: [Freeswitch-users] Tweaking end_pointing on mod_conference Message-ID: Hey Hi, I wanted to know if there's a way to tweak audio end-pointing on mod_conference, the duration of speech/silence it waits for before throwing out a speaking/stopped speaking event, and not just the energy-level. Thanks. -Usama -- I'd love to change the world, but they wont gimme the source code to it -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130201/cfa15949/attachment.html From anthony.minessale at gmail.com Fri Feb 1 05:31:02 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 31 Jan 2013 20:31:02 -0600 Subject: [Freeswitch-users] [CS_NEW] [WRONG_CALL_STATE] with incoming calls In-Reply-To: <510AA553.9060105@targointernet.com> References: <51092BBA.8030001@targointernet.com> <510A865C.505@targointernet.com> <06E0038D-469F-47F1-8449-C6B590AAECA0@gmail.com> <510AA553.9060105@targointernet.com> Message-ID: the best way to change the ip using the default config is to explicitly set local_ip_v4 in vars.xml and leave all the other settings as default. The trace is missing the siptrace which is necessary to diagnose the problem and its better as an attachment so it does not flood the screen. On Thu, Jan 31, 2013 at 11:09 AM, Frederick Pruneau < frederick at targointernet.com> wrote: > I changed sip-ip and rtp-ip settings for my ip address. I restarted > freeswitch. The problem is still there. > > My FS server is not behind nat. > > Thanks for your help! > > Fred > > Le 2013-01-31 11:46, Steven Ayre a ?crit : > > Check the sip-ip rtp-ip settings on your Sofia profiles. They may be using > a variable such as local_ip4, if so check that too. > > Did you restart FS or at least the relevant profiles after changing the > IP? > > Note that if you're not setting it explicitly it may be autodetecting > your IP, in which case that might not happen correctly if your server now > has multiple IPs. Or using the old one if FS wasn't restarted. > > Also remember Sofia cannot listen on 0.0.0.0 > > If your FS is behind NAT then that also may imply a few other settings > need checking too (ext-sip-ip etc). > > Steve > > > > On 31 Jan 2013, at 14:57, Frederick Pruneau > wrote: > > Hi Anthony, > > First of all, thanks for your reply! I appreciate your help! > > I checked conf files that could have changed since the new ip address has > been set. I tried to rollback as far as I can but did not find the option > that makes incoming calls not working. I checked also my network settings. > Everything seems to be good. I'm probably missing something in freeswitch > config files. > > Is there a file (or files) that I need to modify and put my new IP? > > Thanks again. > > Fred > > Le 2013-01-30 20:07, Anthony Minessale a ?crit : > > That means when the call was challenged for auth, it never replied with > the new INVITE with www-auth header. > You may have some network settings wrong making the followup message go to > the wrong host. > > > > On Wed, Jan 30, 2013 at 8:18 AM, Frederick Pruneau < > frederick at targointernet.com> wrote: > >> Hi guys! >> >> I have a problem with my freeswitch server and I need some help. All >> incoming calls are not working since yesterday. All I did is to change >> the IP address. The only error message I have is this one: >> >> [WARNING] switch_core_state_machine.c:514 >> 7b24f8fa-6ae6-11e2-a6be-25229cd2cfbb >> sofia/internal/XXX at XXX.XXX.XXX.XXX:5060 Abandoned >> [NOTICE] switch_core_state_machine.c:517 Hangup >> sofia/internal/XXX at XXX.XXX.XXX.XXX:5060 [CS_NEW] [WRONG_CALL_STATE] >> >> What does Wrong call state mean and what can I do to fix this issue? >> >> Everything else is working. I can make outgoing calls and call other >> extensions. >> >> Thanks in advance. >> >> Fred >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130131/af9951f6/attachment-0001.html From mike at jerris.com Fri Feb 1 05:34:44 2013 From: mike at jerris.com (Michael Jerris) Date: Thu, 31 Jan 2013 21:34:44 -0500 Subject: [Freeswitch-users] [CS_NEW] [WRONG_CALL_STATE] with incoming calls In-Reply-To: <510AA553.9060105@targointernet.com> References: <51092BBA.8030001@targointernet.com> <510A865C.505@targointernet.com> <06E0038D-469F-47F1-8449-C6B590AAECA0@gmail.com> <510AA553.9060105@targointernet.com> Message-ID: <315B88CA-4918-4DDF-91FF-384C0C937690@jerris.com> I would grep the entire config directory for the old IP address to be sure. Also, are you running the latest release or something older? On Jan 31, 2013, at 12:09 PM, Frederick Pruneau wrote: > I changed sip-ip and rtp-ip settings for my ip address. I restarted freeswitch. The problem is still there. > > My FS server is not behind nat. > > Thanks for your help! > > Fred > > Le 2013-01-31 11:46, Steven Ayre a ?crit : >> Check the sip-ip rtp-ip settings on your Sofia profiles. They may be using a variable such as local_ip4, if so check that too. >> >> Did you restart FS or at least the relevant profiles after changing the IP? >> >> Note that if you're not setting it explicitly it may be autodetecting your IP, in which case that might not happen correctly if your server now has multiple IPs. Or using the old one if FS wasn't restarted. >> >> Also remember Sofia cannot listen on 0.0.0.0 >> >> If your FS is behind NAT then that also may imply a few other settings need checking too (ext-sip-ip etc). >> >> Steve >> >> >> >> On 31 Jan 2013, at 14:57, Frederick Pruneau wrote: >> >>> Hi Anthony, >>> >>> First of all, thanks for your reply! I appreciate your help! >>> >>> I checked conf files that could have changed since the new ip address has been set. I tried to rollback as far as I can but did not find the option that makes incoming calls not working. I checked also my network settings. Everything seems to be good. I'm probably missing something in freeswitch config files. >>> >>> Is there a file (or files) that I need to modify and put my new IP? >>> >>> Thanks again. >>> >>> Fred >>> >>> Le 2013-01-30 20:07, Anthony Minessale a ?crit : >>>> That means when the call was challenged for auth, it never replied with the new INVITE with www-auth header. >>>> You may have some network settings wrong making the followup message go to the wrong host. >>>> >>>> >>>> >>>> On Wed, Jan 30, 2013 at 8:18 AM, Frederick Pruneau wrote: >>>> Hi guys! >>>> >>>> I have a problem with my freeswitch server and I need some help. All >>>> incoming calls are not working since yesterday. All I did is to change >>>> the IP address. The only error message I have is this one: >>>> >>>> [WARNING] switch_core_state_machine.c:514 >>>> 7b24f8fa-6ae6-11e2-a6be-25229cd2cfbb >>>> sofia/internal/XXX at XXX.XXX.XXX.XXX:5060 Abandoned >>>> [NOTICE] switch_core_state_machine.c:517 Hangup >>>> sofia/internal/XXX at XXX.XXX.XXX.XXX:5060 [CS_NEW] [WRONG_CALL_STATE] >>>> >>>> What does Wrong call state mean and what can I do to fix this issue? >>>> >>>> Everything else is working. I can make outgoing calls and call other >>>> extensions. >>>> >>>> Thanks in advance. >>>> >>>> Fred >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>>> >>>> >>>> >>>> -- >>>> Anthony Minessale II >>>> >>>> FreeSWITCH http://www.freeswitch.org/ >>>> ClueCon http://www.cluecon.com/ >>>> Twitter: http://twitter.com/FreeSWITCH_wire >>>> >>>> AIM: anthm >>>> MSN:anthony_minessale at hotmail.com >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>> IRC: irc.freenode.net #freeswitch >>>> >>>> FreeSWITCH Developer Conference >>>> sip:888 at conference.freeswitch.org >>>> googletalk:conf+888 at conference.freeswitch.org >>>> pstn:+19193869900 >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130131/625ff35b/attachment.html From mike at jerris.com Fri Feb 1 05:37:26 2013 From: mike at jerris.com (Michael Jerris) Date: Thu, 31 Jan 2013 21:37:26 -0500 Subject: [Freeswitch-users] ICTFAX installation in FS In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF23559FE@Mail-Kilo.squay.com> References: <592A9CF93E12394E8472A6CC66E66BF23559FE@Mail-Kilo.squay.com> Message-ID: I have never heard of ictfax before, but it appears they have forums of their own. I would try posting there. http://forum.ictfax.org/ On Jan 31, 2013, at 12:21 PM, Archana Venugopan wrote: > Hi, > > Am trying to install ICTFAX. As per this site ictfax-20-installation-guide I followed the instructions. But got stuck in the below procedures > > 1. Create a symbolic link for /usr/ictfax in /var/www/html > ln -s /usr/ictfax /var/www/html/ictfax > > ln: creating symbolic link `/var/www/html/ictfax': No such file or directory. There is no ictfax directory in the folder. So what to do? > > 2. Now visit http://DOMAIN.COM/ictfax and follow the installation instructions for ICTFax (drupal based) front end installation > > When I click on the link its giving 404 error. > > Can anyone please guide me on how to proceed further on this. Many thanks > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130131/c8fef7cd/attachment-0001.html From ahe.sanath at gmail.com Fri Feb 1 07:07:54 2013 From: ahe.sanath at gmail.com (Sanath Prasanna) Date: Fri, 1 Feb 2013 09:37:54 +0530 Subject: [Freeswitch-users] Messaging & video calls using IMSDROID SIP client Message-ID: Hi, Recently I configured customized IMSDROID SIP client with Freeswitch Server. (sms-mod enabled) I can get calls between 2 IMSDROID clients through Freeswitch. I want to extend that to messaging & video calls. So if some one doing that. pls help to configure Freeswitch Server & IMSDROID client sides. Br, Sanath -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130201/508bbfef/attachment.html From royj at yandex.ru Fri Feb 1 09:20:57 2013 From: royj at yandex.ru (royj) Date: Fri, 1 Feb 2013 10:20:57 +0400 Subject: [Freeswitch-users] problems with freeswitch + zrtp in proxy-media mode In-Reply-To: References: Message-ID: <20130201102057.dcb2957ab4de53f956ea7457@yandex.ru> Is there in profile On Thu, 31 Jan 2013 17:43:34 -0500 Eli Burke wrote: > I'm trying to use Freeswitch with an open source voip client called Linphone, but cannot manage to establish functioning ZRTP sessions. The problem *seems* to be related to the RTP stream's SSRC. I'm trying to conclusively determine if the problem lies with Linphone, Freeswitch, or one of the underlying zrtp implementations (libzrtp and zrtpcpp). > > Here's what I've determined so far: > 1) ZRTP support is compiled and functioning. I can establish ZRTP sessions between two iPhones running Groundwire. > 2) linphone + ZRTP works correctly if I make a one legged call to a Freeswitch echo extension > 3) linphone + ZRTP works correctly if I turn on bypass-media > 4) linphone gets a CRC error when trying to verify the ZRTP packet on any two-legged call > 5) disabling the CRC check results in a successful ZRTP session, but the audio is extremely garbled > > Is Freeswitch modifying the ZRTP packets when running in proxy mode? If so, how is it that some clients are able to handle this situation? > > Thanks for any suggestions, > Eli > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 vinay.in333 at gmail.com Fri Feb 1 09:23:14 2013 From: vinay.in333 at gmail.com (Vinay V) Date: Fri, 1 Feb 2013 11:53:14 +0530 Subject: [Freeswitch-users] Lync and Openfire Integration Message-ID: Hi All, We are in process of integrating Microsoft Lync with Openfire (Ignite Realtime). Using the default procedure documents available in the Google to do the same did not help us to accomplish it. *Default procedure like using Microsoft XMPP gateway.* * * *My present test lab is having 2 Microsoft Domain Controllers.* * * *Two domain controllers are smart.lab and test.lab. * * * *Domain controller smart.lab contains MS LYNC 2010.* * * *Domain controller test.lab contains Openfire.* * * *MS Lync edge is having two nics (one internal and one external) and it?s not joined to domain. MS OCS XMPP GW 2007 R2 server is having one nic and even this is not joined to domain.* * * *Tried a lot but was not successful.* After browsing a lot in Google came across Freeswitch application which will help us to proceed further. Through the blogs came to know that XMPP servers can be integrated with Freeswitch. As I am new to this this tool, unable to understand the concepts. If any anybody have tried integrating Lync with Openfire using Freeswitch are requested to provide me the inputs (Docs/URL) to proceed further. Thanks & Regards * Adarsh P M* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130201/b24e9edb/attachment.html From steveayre at gmail.com Fri Feb 1 13:54:39 2013 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 1 Feb 2013 10:54:39 +0000 Subject: [Freeswitch-users] simple install question In-Reply-To: References: <091801cdffef$6a4229c0$3ec67d40$@bizfocused.com> <097d01cdfff6$f3531b00$d9f95100$@bizfocused.com> Message-ID: Depending on the impact on CPU and disk i/o anyway. Should be fine if you have multiple cores and don't do a parallel build. Or if its low usage eg office PBX. You could also build once on one server and then distribute to all others using Debian .deb or CentOS RPM packages. If in doubt for a clean build directory just delete/rename/backup the current one and do a new git clone without selecting a branch for checkout (default branch is master). Steve On 31 Jan 2013, at 21:16, Michael Collins wrote: > Personally I think you're okay to leave FS running through the bootstrap, configure and make. I would not, though, leave FS running through the make install. That's just me personally. I'm curious to hear what others have to say. > > -MC > > On Thu, Jan 31, 2013 at 1:07 PM, Sean Devoy wrote: >> Thanks everyone. I will update the wiki IF this works! >> >> >> >> No answer on stopping the running FS though? >> >> >> >> Sean >> >> >> >> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi Marcus >> Sent: Thursday, January 31, 2013 3:32 PM >> To: FreeSWITCH Users Help >> Subject: Re: [Freeswitch-users] simple install question >> >> >> >> It would be nice to have instructions for this somewhere... >> >> some people tar up their old build directory, or at least the compiled files. That way if the build/install goes bonkers, they can untar and re-install their old binaries. >> >> >> >> -Avi >> >> >> >> On Thu, Jan 31, 2013 at 10:24 PM, Nick Vines wrote: >> >> Stolen from an email from Ken. >> >> >> >> "If you need to switch between branches, you _must_ git clean -fdx and rm all >> >> the existing binaries in your existing FreeSWITCH install for the mod lib >> and bin directories or you will most likely end up with build skew and crazy >> weird segfaults..." >> >> >> >> I think it is possible to install while freeswitch is running, but highly advised against. Just do it at a 3am maintenance outage. You should be able to run the bootstrap and configure before you shutdown your FS server to reduce downtime. >> >> >> >> On the base rackspace cloud instance (512mb version), it takes about 10 min total from shutdown to startup of FS. >> >> >> >> On Thu, Jan 31, 2013 at 12:13 PM, Sean Devoy wrote: >> >> Hi, >> >> >> >> I want to switch from the stable release version to the git head and rebuild in hopes of ridding myself of the delay in connection of the audio. On almost all my calls I must hesitate about 1 second before saying hello or the caller does not hear me. I have seen many discussions saying that this is probably fixed in the git root. So here are my questions: >> >> >> >> The procedure appears to be: >> >> Log in to root >> >> cd /usr/local/src/freeswitch >> >> git checkout master >> >> Which responds: Branch master set up to track remote branch master from origin. >> >> Switched to a new branch 'master' >> >> Then just: >> >> make current >> >> >> >> Is that correct? >> >> Do I need to stop the running FS while running this? >> >> If so, how long will I be down? (ball park it) >> >> >> >> I realize this is a RTFM kind of question, but I can?t risk an outage. I need confirmation from someone who understands. >> >> >> >> Thanks, >> >> Sean >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130201/a4290693/attachment-0001.html From steveayre at gmail.com Fri Feb 1 14:04:31 2013 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 1 Feb 2013 11:04:31 +0000 Subject: [Freeswitch-users] FreeSWITCH Mirror on Github In-Reply-To: References: Message-ID: <335647C8-86F0-40BA-8BB6-F21476E71E07@gmail.com> Who owns the account? 'The World's First Cross-Platform Scalable FREE Multi-Protocol' Perhaps it'd be worth putting a note to that effect in that blurb ^ at the top of the github page, directing them to the correct place. -Steve On 31 Jan 2013, at 21:26, Ken Rice wrote: > Hey Guy, > > As many of you know there is a Github mirror of the GIT repo at https://github.com/FreeSWITCH/FreeSWITCH . > > This is intended as a READ ONLY mirror of our master repos. I have noticed that some people are actually doing push request and opening issues there. Please do not do that. The Core dev team rarely (if ever) reviews the other functions that Github provides. If you have an issue or a possible patch, please open a jira and attach the patch. > > If you have already patched your local copy, using the ?git diff? command to get the patch will work, just pipe its output to a .txt file and attach to your Jira. > > Thanks > Ken > > -- > Ken > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > irc.freenode.net #freeswitch > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130201/dcfabf79/attachment.html From leonardo at daitangroup.com Fri Feb 1 14:42:22 2013 From: leonardo at daitangroup.com (Leonardo N. S. Pereira) Date: Fri, 1 Feb 2013 09:42:22 -0200 Subject: [Freeswitch-users] I need to Inject some string when sending an SMS via chat command to avoid a concurrency problem In-Reply-To: <506F0C5B.6000204@daitangroup.com> References: <506DF2BB.3010801@daitangroup.com> <506EE95F.10307@daitangroup.com> <506F0C5B.6000204@daitangroup.com> Message-ID: <510BAA1E.10204@daitangroup.com> I'm sending the SMS via mod_socket interface using the following command: bgapi chat sip|+17778882233|external/+17778882234 at sms-proxy.com|SMS test The SMS is sent successfully. The problem is that I need to inject an string to be retrieved when the BACKGROUND_JOB event is received. This is the scenario: #1 Send an SMS using BGAPI #2 create a new record in database using the JOB_ID as key #3 As soon as the BACKGROUND_JOB event is received, update the database. The problem is that sometimes, the event is received before the record has being created. Is there any way to inject an kind of "trakcing_id" when sending an SMS that will be received in BG event? (just in case, for calls I had the same problem, and I used the parameter 'origination_channel_name' to handle the problem) Thanks in advance. Leo From shouldbeq931 at gmail.com Fri Feb 1 15:53:03 2013 From: shouldbeq931 at gmail.com (shouldbe q931) Date: Fri, 1 Feb 2013 12:53:03 +0000 Subject: [Freeswitch-users] FAX problems In-Reply-To: References: <20130119191943.ca09ab6d@mail.tritonwest.net> <5102AF93.2000201@coppice.org> Message-ID: On Fri, Feb 1, 2013 at 1:35 AM, Steven Schoch < schoch+freeswitch.org at xwin32.com> wrote: > I'm still struggling with sending FAXes from a Windows 7 system. I > believe my requirements are simple: The ability to send a document (like a > Word document or a QuickBooks invoice) directly from the Windows system to > a FAX number. I don't want to have to use a physical modem and an ATA > device. I don't want to run Windows in "Test" mode in order to use an > unsigned driver. I don't mind paying some money for a commercial software > product. > > The setup is simple: I have FreeSwitch installed on a Linux machine, on > our LAN. Also on the LAN I have several Windows 7 machines and an ATA > device (an OBi100). An analog FAX machine is plugged into the ATA. > > What works: > I can send a fax from the FreeSwitch machine using txfax. > I can send a fax through the ATA and the FreeSwitch machine using the > analog FAX machine. > > What doesn't: > I tried www.t38faxvoip.com through FreeSwitch. It doesn't work. > I tried www.t38faxvoip.com directly to Flowroute. That didn't work > either. > I called the analog fax machine through the ATA and FreeSwitch from > www.t38faxvoip.com. > I tried www.faxback.com. Same results. > > The people at www.t38faxvoip.com at least responded to my emails, but > said they didn't know what was happening after looking at my Wireshark dump. > > All of the tests from Windows behaved the same. The called FAX machine > answered, but never completed the handshake. The software reported that > the call was answered by something other than a FAX machine. > > Yes, we can live with printing out the document and sticking it in the FAX > machine, but it sure would be nice to be able to save some paper. Can't > anyone solve my problem? > > -- > Steve > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > As a replacement for the faxback plugin. how about using hylafax+ with t38modem ? http://hylafax.sourceforge.net/ or the commercial "fork" from iFax ? http://www.ifax.com/products/hylafax/fax-over-ip.html -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130201/af6c2c68/attachment.html From steveayre at gmail.com Fri Feb 1 17:42:43 2013 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 1 Feb 2013 14:42:43 +0000 Subject: [Freeswitch-users] I need to Inject some string when sending an SMS via chat command to avoid a concurrency problem In-Reply-To: <510BAA1E.10204@daitangroup.com> References: <506DF2BB.3010801@daitangroup.com> <506EE95F.10307@daitangroup.com> <506F0C5B.6000204@daitangroup.com> <510BAA1E.10204@daitangroup.com> Message-ID: You could try blocking receiving events during the DB update, or if you cannot do that try storing the BACKGROUND_JOB event in a hash in memory so that when the record creation succeeds it checks the hash to see if the event has already been received. You can also use a hash to see if the record is currently being inserted or not to save checking if it's in the database. -Steve On 1 February 2013 11:42, Leonardo N. S. Pereira wrote: > I'm sending the SMS via mod_socket interface using the following command: > bgapi chat sip|+17778882233|external/+17778882234 at sms-proxy.com|SMS > test > The SMS is sent successfully. > The problem is that I need to inject an string to be retrieved when the > BACKGROUND_JOB event is received. > > This is the scenario: > #1 Send an SMS using BGAPI > #2 create a new record in database using the JOB_ID as key > #3 As soon as the BACKGROUND_JOB event is received, update the database. > > The problem is that sometimes, the event is received before the record > has being created. > > > Is there any way to inject an kind of "trakcing_id" when sending an SMS > that will be received in BG event? (just in case, for calls I had the > same problem, and I used the parameter 'origination_channel_name' to > handle the problem) > > Thanks in advance. > Leo > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130201/babe4070/attachment-0001.html From sertys at gmail.com Fri Feb 1 17:54:12 2013 From: sertys at gmail.com (Daniel Ivanov) Date: Fri, 1 Feb 2013 15:54:12 +0100 Subject: [Freeswitch-users] Messaging & video calls using IMSDROID SIP client In-Reply-To: References: Message-ID: I havent tried ims, but SIMPLE messaging and video calls are natural extensions to the sip protocol. Especially SIMPLE is just a matter of client handling. Non-transcode video in h264 is easy and native as well. Just try messaging and video calling out of the box and see what happpens. On Feb 1, 2013 6:11 AM, "Sanath Prasanna" wrote: > Hi, > Recently I configured customized IMSDROID SIP client with Freeswitch > Server. (sms-mod enabled) I can get calls between 2 IMSDROID clients > through Freeswitch. > I want to extend that to messaging & video calls. So if some one doing > that. pls help to configure Freeswitch Server & IMSDROID client sides. > Br, > Sanath > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130201/debdd12e/attachment.html From sdevoy at bizfocused.com Fri Feb 1 20:20:14 2013 From: sdevoy at bizfocused.com (Sean Devoy) Date: Fri, 1 Feb 2013 12:20:14 -0500 Subject: [Freeswitch-users] Site Specific Errors (Polycom? Comcast Router? FS?) Message-ID: <0e8f01ce00a0$66206710$32613530$@bizfocused.com> Hi, I am having intermittent problems with one of my new customers. These problems have not been reported by other customers using the same FS server, but that does not mean there may not be a FS solution. I appreciate any suggestions you folks have. I just did a GIT update yesterday from head. The site is unique in that it uses a Comcast Business Router/Modem/Gateway and is our only site with Polycom 330 phones. We were forced to set the local (phone) SIP port to a unique number across the board. That us each line on each phone has a unique port on the NAT'ed Router. As I mentioned this is our first attempt at using Polycom phones. The periodic problems are: Cross-Talk: Users report hearing other conversations in the office on their call. Googling produced a proposed suggestion of putting each phone's connection on a SWITCH to isolate rtp traffic. We removed their hub and put in a layer 2 switch. Several Users have reported that as they are dialing if another call comes in to their phone from FS they immediately get a busy signal. One user reported a call dropped when another call came in to his phone. I mentioned cross-talk only because it may mean something to you. I am hoping that issue is resolved. These are all intermittent, and our staff has not been able to reproduce at the customer site! Thanks again. Sean -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130201/2ae20163/attachment.html From eduardonunesp at gmail.com Fri Feb 1 20:53:06 2013 From: eduardonunesp at gmail.com (Eduardo Nunes Pereira) Date: Fri, 1 Feb 2013 15:53:06 -0200 Subject: [Freeswitch-users] ESL and Event Socket with secure connection Message-ID: How is the best way to implement connection between ESL and Event Socket with secure connection, something like HTTPS. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130201/f2484ec1/attachment.html From roger.castaldo at gmail.com Fri Feb 1 21:01:48 2013 From: roger.castaldo at gmail.com (Roger Castaldo) Date: Fri, 1 Feb 2013 13:01:48 -0500 Subject: [Freeswitch-users] ESL and Event Socket with secure connection In-Reply-To: References: Message-ID: If you are not using ESL on the server itself I would suggest using ssl or ssh tunneling to handle it, or even a vpn connection would work as well. If you are running on linux servers, ssl tunneling and ssh tunneling are relatively easy to setup. Windows, don't know. On Fri, Feb 1, 2013 at 12:53 PM, Eduardo Nunes Pereira < eduardonunesp at gmail.com> wrote: > How is the best way to implement connection between ESL and Event Socket > with secure connection, something like HTTPS. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130201/50d14151/attachment.html From mustafin.aleksandr at gmail.com Fri Feb 1 14:18:22 2013 From: mustafin.aleksandr at gmail.com (=?UTF-8?B?0JDQu9C10LrRgdCw0L3QtNGAINCc0YPRgdGC0LDRhNC40L0=?=) Date: Fri, 01 Feb 2013 17:18:22 +0600 Subject: [Freeswitch-users] Choice of codecs for incoming call Message-ID: <510BA47E.2070906@gmail.com> Hello. My FS receive INVITE with 2 codecs in SDP: PCMA and PCMU. After hunting dialplan, this call forwarding to IVR, and I see next message in log: 2013-01-31 01:23:22.388780 [DEBUG] sofia_glue.c:5094 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2013-01-31 01:23:22.388780 [DEBUG] sofia_glue.c:3077 Set Codec sofia/internal/9089080908 at PSTN PCMU/8000 20 ms 160 samples 64000 bits 2013-01-31 01:23:22.388780 [DEBUG] switch_core_codec.c:111 sofia/internal/9089080908 at PSTN Original read codec set to PCMU:0 And FS send 200 OK to remote side with one codec in SDP: PCMU. My parameters: In internal.xml (profile) In vars.xml This is normal behavior?How can I manage set of codec in inbound calls? Best regards, Alexander. From eburke at edge-net.net Fri Feb 1 17:15:19 2013 From: eburke at edge-net.net (Eli Burke) Date: Fri, 1 Feb 2013 09:15:19 -0500 Subject: [Freeswitch-users] problems with freeswitch + zrtp in proxy-media mode In-Reply-To: References: Message-ID: <2DBB65A0-2551-4FE8-88E0-CB4624325733@edge-net.net> Yes, inbound-zrtp-passthru is true. I think it gets set automatically when you turn on ZRTP. The config I'm testing with is a stock 'enable_zrtp=yes' build of HEAD circa last week, with proxy-media=true added to Local_Extension in the default dial plan. -Eli On Feb 1, 2013, at 6:00 AM, royj at yandex.ru wrote: > Is there in profile > > On Thu, 31 Jan 2013 17:43:34 -0500 > Eli Burke wrote: > >> I'm trying to use Freeswitch with an open source voip client called Linphone, but cannot manage to establish functioning ZRTP sessions. The problem *seems* to be related to the RTP stream's SSRC. I'm trying to conclusively determine if the problem lies with Linphone, Freeswitch, or one of the underlying zrtp implementations (libzrtp and zrtpcpp). >> >> Here's what I've determined so far: >> 1) ZRTP support is compiled and functioning. I can establish ZRTP sessions between two iPhones running Groundwire. >> 2) linphone + ZRTP works correctly if I make a one legged call to a Freeswitch echo extension >> 3) linphone + ZRTP works correctly if I turn on bypass-media >> 4) linphone gets a CRC error when trying to verify the ZRTP packet on any two-legged call >> 5) disabling the CRC check results in a successful ZRTP session, but the audio is extremely garbled >> >> Is Freeswitch modifying the ZRTP packets when running in proxy mode? If so, how is it that some clients are able to handle this situation? From frederick at targointernet.com Fri Feb 1 17:49:39 2013 From: frederick at targointernet.com (Frederick Pruneau) Date: Fri, 01 Feb 2013 09:49:39 -0500 Subject: [Freeswitch-users] [CS_NEW] [WRONG_CALL_STATE] with incoming calls In-Reply-To: References: <51092BBA.8030001@targointernet.com> <510A865C.505@targointernet.com> <06E0038D-469F-47F1-8449-C6B590AAECA0@gmail.com> <510AA553.9060105@targointernet.com> Message-ID: <510BD603.50303@targointernet.com> I think I have enabled siptrace correctly in my log file. Let me know if i'm wrong because it's the first time I enable this option. It was enable in the console but not in log file. I have hide the first three numbers of my ip address. IP addresses look like this in log file : XXX.XXX.XXX.10. I tried to grep the old IP in config directory but got no result. I'm running the latest release. Thanks! Fred Le 2013-01-31 21:31, Anthony Minessale a ?crit : > the best way to change the ip using the default config is to > explicitly set local_ip_v4 in vars.xml and leave all the other > settings as default. > > The trace is missing the siptrace which is necessary to diagnose the > problem and its better as an attachment so it does not flood the screen. > > > > On Thu, Jan 31, 2013 at 11:09 AM, Frederick Pruneau > > wrote: > > I changed sip-ip and rtp-ip settings for my ip address. I > restarted freeswitch. The problem is still there. > > My FS server is not behind nat. > > Thanks for your help! > > Fred > > Le 2013-01-31 11:46, Steven Ayre a ?crit : >> Check the sip-ip rtp-ip settings on your Sofia profiles. They may >> be using a variable such as local_ip4, if so check that too. >> >> Did you restart FS or at least the relevant profiles after >> changing the IP? >> >> Note that if you're not setting it explicitly it may be >> autodetecting your IP, in which case that might not happen >> correctly if your server now has multiple IPs. Or using the old >> one if FS wasn't restarted. >> >> Also remember Sofia cannot listen on 0.0.0.0 >> >> If your FS is behind NAT then that also may imply a few other >> settings need checking too (ext-sip-ip etc). >> >> Steve >> >> >> >> On 31 Jan 2013, at 14:57, Frederick Pruneau >> > > wrote: >> >>> Hi Anthony, >>> >>> First of all, thanks for your reply! I appreciate your help! >>> >>> I checked conf files that could have changed since the new ip >>> address has been set. I tried to rollback as far as I can but >>> did not find the option that makes incoming calls not working. I >>> checked also my network settings. Everything seems to be good. >>> I'm probably missing something in freeswitch config files. >>> >>> Is there a file (or files) that I need to modify and put my new IP? >>> >>> Thanks again. >>> >>> Fred >>> >>> Le 2013-01-30 20:07, Anthony Minessale a ?crit : >>>> That means when the call was challenged for auth, it never >>>> replied with the new INVITE with www-auth header. >>>> You may have some network settings wrong making the followup >>>> message go to the wrong host. >>>> >>>> >>>> >>>> On Wed, Jan 30, 2013 at 8:18 AM, Frederick Pruneau >>>> >>> > wrote: >>>> >>>> Hi guys! >>>> >>>> I have a problem with my freeswitch server and I need some >>>> help. All >>>> incoming calls are not working since yesterday. All I did >>>> is to change >>>> the IP address. The only error message I have is this one: >>>> >>>> [WARNING] switch_core_state_machine.c:514 >>>> 7b24f8fa-6ae6-11e2-a6be-25229cd2cfbb >>>> sofia/internal/XXX at XXX.XXX.XXX.XXX:5060 >>>> Abandoned >>>> [NOTICE] switch_core_state_machine.c:517 Hangup >>>> sofia/internal/XXX at XXX.XXX.XXX.XXX:5060 >>>> [CS_NEW] >>>> [WRONG_CALL_STATE] >>>> >>>> What does Wrong call state mean and what can I do to fix >>>> this issue? >>>> >>>> Everything else is working. I can make outgoing calls and >>>> call other >>>> extensions. >>>> >>>> Thanks in advance. >>>> >>>> Fred >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>>> >>>> >>>> >>>> >>>> -- >>>> Anthony Minessale II >>>> >>>> FreeSWITCH http://www.freeswitch.org/ >>>> ClueCon http://www.cluecon.com/ >>>> Twitter: http://twitter.com/FreeSWITCH_wire >>>> >>>> AIM: anthm >>>> MSN:anthony_minessale at hotmail.com >>>> >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>> >>>> IRC: irc.freenode.net #freeswitch >>>> >>>> FreeSWITCH Developer Conference >>>> sip:888 at conference.freeswitch.org >>>> >>>> googletalk:conf+888 at conference.freeswitch.org >>>> >>>> pstn:+19193869900 >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > googletalk:conf+888 at conference.freeswitch.org > > pstn:+19193869900 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130201/c7786080/attachment-0001.html -------------- next part -------------- 2013-01-31 16:19:34.292905 [NOTICE] switch_core_session.c:1521 Close Channel sofia/internal/as1000 at XXX.XXX.XXX.10 [CS_DESTROY] 2013-01-31 16:19:34.292905 [DEBUG] switch_core_state_machine.c:556 (sofia/internal/as1000 at XXX.XXX.XXX.10) Callstate Change HANGUP -> DOWN 2013-01-31 16:19:34.292905 [DEBUG] switch_core_state_machine.c:559 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_DESTROY 2013-01-31 16:19:34.292905 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/as1000 at XXX.XXX.XXX.10) State DESTROY 2013-01-31 16:19:34.292905 [DEBUG] mod_sofia.c:396 sofia/internal/as1000 at XXX.XXX.XXX.10 SOFIA DESTROY 2013-01-31 16:19:34.292905 [DEBUG] switch_core_state_machine.c:99 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard DESTROY 2013-01-31 16:19:34.292905 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/as1000 at XXX.XXX.XXX.10) State DESTROY going to sleep 2013-01-31 16:19:34.292905 [NOTICE] mod_logfile.c:213 New log started. 2013-01-31 16:19:36.732903 [NOTICE] switch_channel.c:968 New Channel sofia/internal/as1000 at XXX.XXX.XXX.10 [ea531152-6beb-11e2-a3e1-954bab7d8c64] 2013-01-31 16:19:36.732903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:36.732903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:36.732903 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_NEW 2013-01-31 16:19:36.732903 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/as1000 at XXX.XXX.XXX.10) State NEW 2013-01-31 16:19:36.752902 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:36.752902 [DEBUG] sofia.c:1719 detaching session ea531152-6beb-11e2-a3e1-954bab7d8c64 2013-01-31 16:19:36.892902 [DEBUG] sofia.c:1811 Re-attaching to session ea531152-6beb-11e2-a3e1-954bab7d8c64 2013-01-31 16:19:36.892902 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:36.892902 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:36.912894 [WARNING] sofia_reg.c:2491 Can't find user [as1000 at XXX.XXX.XXX.10] You must define a domain called 'XXX.XXX.XXX.10' in your directory and add a user with the id="as1000" attribute and you must configure your device to use the proper domain in it's authentication credentials. 2013-01-31 16:19:36.912894 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:36.912894 [DEBUG] switch_channel.c:2994 (sofia/internal/as1000 at XXX.XXX.XXX.10) Callstate Change DOWN -> HANGUP 2013-01-31 16:19:36.912894 [NOTICE] sofia.c:1717 Hangup sofia/internal/as1000 at XXX.XXX.XXX.10 [CS_NEW] [CALL_REJECTED] 2013-01-31 16:19:36.912894 [DEBUG] switch_channel.c:3017 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [KILL] 2013-01-31 16:19:36.912894 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:36.932891 [DEBUG] sofia.c:1051 Channel is already hungup. 2013-01-31 16:19:36.932891 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_HANGUP 2013-01-31 16:19:36.932891 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/as1000 at XXX.XXX.XXX.10) State HANGUP 2013-01-31 16:19:36.932891 [DEBUG] mod_sofia.c:503 Channel sofia/internal/as1000 at XXX.XXX.XXX.10 hanging up, cause: CALL_REJECTED 2013-01-31 16:19:36.932891 [DEBUG] switch_core_state_machine.c:48 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard HANGUP, cause: CALL_REJECTED 2013-01-31 16:19:36.932891 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/as1000 at XXX.XXX.XXX.10) State HANGUP going to sleep 2013-01-31 16:19:36.932891 [DEBUG] switch_core_state_machine.c:446 (sofia/internal/as1000 at XXX.XXX.XXX.10) State Change CS_HANGUP -> CS_REPORTING 2013-01-31 16:19:36.932891 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:36.932891 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_REPORTING 2013-01-31 16:19:36.932891 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/as1000 at XXX.XXX.XXX.10) State REPORTING 2013-01-31 16:19:36.932891 [DEBUG] switch_core_state_machine.c:92 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard REPORTING, cause: CALL_REJECTED 2013-01-31 16:19:36.932891 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/as1000 at XXX.XXX.XXX.10) State REPORTING going to sleep 2013-01-31 16:19:36.932891 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/as1000 at XXX.XXX.XXX.10) State Change CS_REPORTING -> CS_DESTROY 2013-01-31 16:19:36.932891 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:36.932891 [DEBUG] switch_core_session.c:1499 Session 38 (sofia/internal/as1000 at XXX.XXX.XXX.10) Locked, Waiting on external entities 2013-01-31 16:19:36.932891 [NOTICE] switch_core_session.c:1517 Session 38 (sofia/internal/as1000 at XXX.XXX.XXX.10) Ended 2013-01-31 16:19:36.932891 [NOTICE] switch_core_session.c:1521 Close Channel sofia/internal/as1000 at XXX.XXX.XXX.10 [CS_DESTROY] 2013-01-31 16:19:36.932891 [DEBUG] switch_core_state_machine.c:556 (sofia/internal/as1000 at XXX.XXX.XXX.10) Callstate Change HANGUP -> DOWN 2013-01-31 16:19:36.932891 [DEBUG] switch_core_state_machine.c:559 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_DESTROY 2013-01-31 16:19:36.932891 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/as1000 at XXX.XXX.XXX.10) State DESTROY 2013-01-31 16:19:36.932891 [DEBUG] mod_sofia.c:396 sofia/internal/as1000 at XXX.XXX.XXX.10 SOFIA DESTROY 2013-01-31 16:19:36.932891 [DEBUG] switch_core_state_machine.c:99 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard DESTROY 2013-01-31 16:19:36.932891 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/as1000 at XXX.XXX.XXX.10) State DESTROY going to sleep 2013-01-31 16:19:38.272903 [NOTICE] switch_channel.c:968 New Channel sofia/internal/as1000 at XXX.XXX.XXX.10 [eb3eed7a-6beb-11e2-a3e3-954bab7d8c64] 2013-01-31 16:19:38.272903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:38.272903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:38.272903 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_NEW 2013-01-31 16:19:38.272903 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/as1000 at XXX.XXX.XXX.10) State NEW 2013-01-31 16:19:38.292900 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:38.292900 [DEBUG] sofia.c:1719 detaching session eb3eed7a-6beb-11e2-a3e3-954bab7d8c64 2013-01-31 16:19:38.692903 [DEBUG] sofia.c:1811 Re-attaching to session eb3eed7a-6beb-11e2-a3e3-954bab7d8c64 2013-01-31 16:19:38.692903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:38.692903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:38.692903 [WARNING] sofia_reg.c:2491 Can't find user [as1000 at XXX.XXX.XXX.10] You must define a domain called 'XXX.XXX.XXX.10' in your directory and add a user with the id="as1000" attribute and you must configure your device to use the proper domain in it's authentication credentials. 2013-01-31 16:19:38.692903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:38.692903 [DEBUG] switch_channel.c:2994 (sofia/internal/as1000 at XXX.XXX.XXX.10) Callstate Change DOWN -> HANGUP 2013-01-31 16:19:38.692903 [NOTICE] sofia.c:1717 Hangup sofia/internal/as1000 at XXX.XXX.XXX.10 [CS_NEW] [CALL_REJECTED] 2013-01-31 16:19:38.692903 [DEBUG] switch_channel.c:3017 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [KILL] 2013-01-31 16:19:38.692903 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:38.712900 [DEBUG] sofia.c:1051 Channel is already hungup. 2013-01-31 16:19:38.712900 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_HANGUP 2013-01-31 16:19:38.712900 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/as1000 at XXX.XXX.XXX.10) State HANGUP 2013-01-31 16:19:38.712900 [DEBUG] mod_sofia.c:503 Channel sofia/internal/as1000 at XXX.XXX.XXX.10 hanging up, cause: CALL_REJECTED 2013-01-31 16:19:38.712900 [DEBUG] switch_core_state_machine.c:48 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard HANGUP, cause: CALL_REJECTED 2013-01-31 16:19:38.712900 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/as1000 at XXX.XXX.XXX.10) State HANGUP going to sleep 2013-01-31 16:19:38.712900 [DEBUG] switch_core_state_machine.c:446 (sofia/internal/as1000 at XXX.XXX.XXX.10) State Change CS_HANGUP -> CS_REPORTING 2013-01-31 16:19:38.712900 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:38.712900 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_REPORTING 2013-01-31 16:19:38.712900 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/as1000 at XXX.XXX.XXX.10) State REPORTING 2013-01-31 16:19:38.712900 [DEBUG] switch_core_state_machine.c:92 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard REPORTING, cause: CALL_REJECTED 2013-01-31 16:19:38.712900 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/as1000 at XXX.XXX.XXX.10) State REPORTING going to sleep 2013-01-31 16:19:38.712900 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/as1000 at XXX.XXX.XXX.10) State Change CS_REPORTING -> CS_DESTROY 2013-01-31 16:19:38.712900 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:38.712900 [DEBUG] switch_core_session.c:1499 Session 39 (sofia/internal/as1000 at XXX.XXX.XXX.10) Locked, Waiting on external entities 2013-01-31 16:19:38.712900 [NOTICE] switch_core_session.c:1517 Session 39 (sofia/internal/as1000 at XXX.XXX.XXX.10) Ended 2013-01-31 16:19:38.712900 [NOTICE] switch_core_session.c:1521 Close Channel sofia/internal/as1000 at XXX.XXX.XXX.10 [CS_DESTROY] 2013-01-31 16:19:38.712900 [DEBUG] switch_core_state_machine.c:556 (sofia/internal/as1000 at XXX.XXX.XXX.10) Callstate Change HANGUP -> DOWN 2013-01-31 16:19:38.712900 [DEBUG] switch_core_state_machine.c:559 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_DESTROY 2013-01-31 16:19:38.712900 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/as1000 at XXX.XXX.XXX.10) State DESTROY 2013-01-31 16:19:38.712900 [DEBUG] mod_sofia.c:396 sofia/internal/as1000 at XXX.XXX.XXX.10 SOFIA DESTROY 2013-01-31 16:19:38.712900 [DEBUG] switch_core_state_machine.c:99 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard DESTROY 2013-01-31 16:19:38.712900 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/as1000 at XXX.XXX.XXX.10) State DESTROY going to sleep 2013-01-31 16:19:40.272903 [NOTICE] switch_channel.c:968 New Channel sofia/internal/as1000 at XXX.XXX.XXX.10 [ec6f4e74-6beb-11e2-a3e5-954bab7d8c64] 2013-01-31 16:19:40.272903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:40.272903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:40.272903 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_NEW 2013-01-31 16:19:40.272903 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/as1000 at XXX.XXX.XXX.10) State NEW 2013-01-31 16:19:40.292895 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:40.292895 [DEBUG] sofia.c:1719 detaching session ec6f4e74-6beb-11e2-a3e5-954bab7d8c64 2013-01-31 16:19:40.602903 [DEBUG] sofia.c:1811 Re-attaching to session ec6f4e74-6beb-11e2-a3e5-954bab7d8c64 2013-01-31 16:19:40.602903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:40.602903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:40.612902 [WARNING] sofia_reg.c:2491 Can't find user [as1000 at XXX.XXX.XXX.10] You must define a domain called 'XXX.XXX.XXX.10' in your directory and add a user with the id="as1000" attribute and you must configure your device to use the proper domain in it's authentication credentials. 2013-01-31 16:19:40.612902 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:40.612902 [DEBUG] switch_channel.c:2994 (sofia/internal/as1000 at XXX.XXX.XXX.10) Callstate Change DOWN -> HANGUP 2013-01-31 16:19:40.612902 [NOTICE] sofia.c:1717 Hangup sofia/internal/as1000 at XXX.XXX.XXX.10 [CS_NEW] [CALL_REJECTED] 2013-01-31 16:19:40.612902 [DEBUG] switch_channel.c:3017 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [KILL] 2013-01-31 16:19:40.612902 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:40.632900 [DEBUG] sofia.c:1051 Channel is already hungup. 2013-01-31 16:19:40.632900 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_HANGUP 2013-01-31 16:19:40.632900 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/as1000 at XXX.XXX.XXX.10) State HANGUP 2013-01-31 16:19:40.632900 [DEBUG] mod_sofia.c:503 Channel sofia/internal/as1000 at XXX.XXX.XXX.10 hanging up, cause: CALL_REJECTED 2013-01-31 16:19:40.632900 [DEBUG] switch_core_state_machine.c:48 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard HANGUP, cause: CALL_REJECTED 2013-01-31 16:19:40.632900 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/as1000 at XXX.XXX.XXX.10) State HANGUP going to sleep 2013-01-31 16:19:40.632900 [DEBUG] switch_core_state_machine.c:446 (sofia/internal/as1000 at XXX.XXX.XXX.10) State Change CS_HANGUP -> CS_REPORTING 2013-01-31 16:19:40.632900 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:40.632900 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_REPORTING 2013-01-31 16:19:40.632900 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/as1000 at XXX.XXX.XXX.10) State REPORTING 2013-01-31 16:19:40.632900 [DEBUG] switch_core_state_machine.c:92 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard REPORTING, cause: CALL_REJECTED 2013-01-31 16:19:40.632900 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/as1000 at XXX.XXX.XXX.10) State REPORTING going to sleep 2013-01-31 16:19:40.632900 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/as1000 at XXX.XXX.XXX.10) State Change CS_REPORTING -> CS_DESTROY 2013-01-31 16:19:40.632900 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:40.632900 [DEBUG] switch_core_session.c:1499 Session 40 (sofia/internal/as1000 at XXX.XXX.XXX.10) Locked, Waiting on external entities 2013-01-31 16:19:40.632900 [NOTICE] switch_core_session.c:1517 Session 40 (sofia/internal/as1000 at XXX.XXX.XXX.10) Ended 2013-01-31 16:19:40.632900 [NOTICE] switch_core_session.c:1521 Close Channel sofia/internal/as1000 at XXX.XXX.XXX.10 [CS_DESTROY] 2013-01-31 16:19:40.632900 [DEBUG] switch_core_state_machine.c:556 (sofia/internal/as1000 at XXX.XXX.XXX.10) Callstate Change HANGUP -> DOWN 2013-01-31 16:19:40.632900 [DEBUG] switch_core_state_machine.c:559 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_DESTROY 2013-01-31 16:19:40.632900 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/as1000 at XXX.XXX.XXX.10) State DESTROY 2013-01-31 16:19:40.632900 [DEBUG] mod_sofia.c:396 sofia/internal/as1000 at XXX.XXX.XXX.10 SOFIA DESTROY 2013-01-31 16:19:40.632900 [DEBUG] switch_core_state_machine.c:99 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard DESTROY 2013-01-31 16:19:40.632900 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/as1000 at XXX.XXX.XXX.10) State DESTROY going to sleep 2013-01-31 16:19:43.702903 [NOTICE] switch_channel.c:968 New Channel sofia/internal/as1000 at XXX.XXX.XXX.10 [ee7b13d8-6beb-11e2-a3e7-954bab7d8c64] 2013-01-31 16:19:43.702903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:43.702903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:43.702903 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_NEW 2013-01-31 16:19:43.702903 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/as1000 at XXX.XXX.XXX.10) State NEW 2013-01-31 16:19:43.722903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:43.722903 [DEBUG] sofia.c:1719 detaching session ee7b13d8-6beb-11e2-a3e7-954bab7d8c64 2013-01-31 16:19:43.952903 [DEBUG] sofia.c:1811 Re-attaching to session ee7b13d8-6beb-11e2-a3e7-954bab7d8c64 2013-01-31 16:19:43.952903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:43.952903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:43.962903 [WARNING] sofia_reg.c:2491 Can't find user [as1000 at XXX.XXX.XXX.10] You must define a domain called 'XXX.XXX.XXX.10' in your directory and add a user with the id="as1000" attribute and you must configure your device to use the proper domain in it's authentication credentials. 2013-01-31 16:19:43.962903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:43.962903 [DEBUG] switch_channel.c:2994 (sofia/internal/as1000 at XXX.XXX.XXX.10) Callstate Change DOWN -> HANGUP 2013-01-31 16:19:43.962903 [NOTICE] sofia.c:1717 Hangup sofia/internal/as1000 at XXX.XXX.XXX.10 [CS_NEW] [CALL_REJECTED] 2013-01-31 16:19:43.962903 [DEBUG] switch_channel.c:3017 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [KILL] 2013-01-31 16:19:43.962903 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:43.982903 [DEBUG] sofia.c:1051 Channel is already hungup. 2013-01-31 16:19:43.982903 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_HANGUP 2013-01-31 16:19:43.982903 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/as1000 at XXX.XXX.XXX.10) State HANGUP 2013-01-31 16:19:43.982903 [DEBUG] mod_sofia.c:503 Channel sofia/internal/as1000 at XXX.XXX.XXX.10 hanging up, cause: CALL_REJECTED 2013-01-31 16:19:43.982903 [DEBUG] switch_core_state_machine.c:48 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard HANGUP, cause: CALL_REJECTED 2013-01-31 16:19:43.982903 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/as1000 at XXX.XXX.XXX.10) State HANGUP going to sleep 2013-01-31 16:19:43.982903 [DEBUG] switch_core_state_machine.c:446 (sofia/internal/as1000 at XXX.XXX.XXX.10) State Change CS_HANGUP -> CS_REPORTING 2013-01-31 16:19:43.982903 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:43.982903 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_REPORTING 2013-01-31 16:19:43.982903 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/as1000 at XXX.XXX.XXX.10) State REPORTING 2013-01-31 16:19:43.982903 [DEBUG] switch_core_state_machine.c:92 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard REPORTING, cause: CALL_REJECTED 2013-01-31 16:19:43.982903 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/as1000 at XXX.XXX.XXX.10) State REPORTING going to sleep 2013-01-31 16:19:43.982903 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/as1000 at XXX.XXX.XXX.10) State Change CS_REPORTING -> CS_DESTROY 2013-01-31 16:19:43.982903 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:43.982903 [DEBUG] switch_core_session.c:1499 Session 41 (sofia/internal/as1000 at XXX.XXX.XXX.10) Locked, Waiting on external entities 2013-01-31 16:19:43.982903 [NOTICE] switch_core_session.c:1517 Session 41 (sofia/internal/as1000 at XXX.XXX.XXX.10) Ended 2013-01-31 16:19:43.982903 [NOTICE] switch_core_session.c:1521 Close Channel sofia/internal/as1000 at XXX.XXX.XXX.10 [CS_DESTROY] 2013-01-31 16:19:43.982903 [DEBUG] switch_core_state_machine.c:556 (sofia/internal/as1000 at XXX.XXX.XXX.10) Callstate Change HANGUP -> DOWN 2013-01-31 16:19:43.982903 [DEBUG] switch_core_state_machine.c:559 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_DESTROY 2013-01-31 16:19:43.982903 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/as1000 at XXX.XXX.XXX.10) State DESTROY 2013-01-31 16:19:43.982903 [DEBUG] mod_sofia.c:396 sofia/internal/as1000 at XXX.XXX.XXX.10 SOFIA DESTROY 2013-01-31 16:19:43.982903 [DEBUG] switch_core_state_machine.c:99 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard DESTROY 2013-01-31 16:19:43.982903 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/as1000 at XXX.XXX.XXX.10) State DESTROY going to sleep 2013-01-31 16:19:46.872895 [NOTICE] switch_channel.c:968 New Channel sofia/internal/as1000 at XXX.XXX.XXX.10 [f05e6646-6beb-11e2-a3e9-954bab7d8c64] 2013-01-31 16:19:46.872895 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:46.872895 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:46.872895 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_NEW 2013-01-31 16:19:46.872895 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/as1000 at XXX.XXX.XXX.10) State NEW 2013-01-31 16:19:46.892900 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:46.892900 [DEBUG] sofia.c:1719 detaching session f05e6646-6beb-11e2-a3e9-954bab7d8c64 2013-01-31 16:19:47.122903 [DEBUG] sofia.c:1811 Re-attaching to session f05e6646-6beb-11e2-a3e9-954bab7d8c64 2013-01-31 16:19:47.122903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:47.122903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:47.132902 [WARNING] sofia_reg.c:2491 Can't find user [as1000 at XXX.XXX.XXX.10] You must define a domain called 'XXX.XXX.XXX.10' in your directory and add a user with the id="as1000" attribute and you must configure your device to use the proper domain in it's authentication credentials. 2013-01-31 16:19:47.132902 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:47.132902 [DEBUG] switch_channel.c:2994 (sofia/internal/as1000 at XXX.XXX.XXX.10) Callstate Change DOWN -> HANGUP 2013-01-31 16:19:47.132902 [NOTICE] sofia.c:1717 Hangup sofia/internal/as1000 at XXX.XXX.XXX.10 [CS_NEW] [CALL_REJECTED] 2013-01-31 16:19:47.132902 [DEBUG] switch_channel.c:3017 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [KILL] 2013-01-31 16:19:47.132902 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:47.152894 [DEBUG] sofia.c:1051 Channel is already hungup. 2013-01-31 16:19:47.152894 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_HANGUP 2013-01-31 16:19:47.152894 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/as1000 at XXX.XXX.XXX.10) State HANGUP 2013-01-31 16:19:47.152894 [DEBUG] mod_sofia.c:503 Channel sofia/internal/as1000 at XXX.XXX.XXX.10 hanging up, cause: CALL_REJECTED 2013-01-31 16:19:47.152894 [DEBUG] switch_core_state_machine.c:48 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard HANGUP, cause: CALL_REJECTED 2013-01-31 16:19:47.152894 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/as1000 at XXX.XXX.XXX.10) State HANGUP going to sleep 2013-01-31 16:19:47.152894 [DEBUG] switch_core_state_machine.c:446 (sofia/internal/as1000 at XXX.XXX.XXX.10) State Change CS_HANGUP -> CS_REPORTING 2013-01-31 16:19:47.152894 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:47.152894 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_REPORTING 2013-01-31 16:19:47.152894 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/as1000 at XXX.XXX.XXX.10) State REPORTING 2013-01-31 16:19:47.152894 [DEBUG] switch_core_state_machine.c:92 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard REPORTING, cause: CALL_REJECTED 2013-01-31 16:19:47.152894 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/as1000 at XXX.XXX.XXX.10) State REPORTING going to sleep 2013-01-31 16:19:47.152894 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/as1000 at XXX.XXX.XXX.10) State Change CS_REPORTING -> CS_DESTROY 2013-01-31 16:19:47.152894 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:47.152894 [DEBUG] switch_core_session.c:1499 Session 42 (sofia/internal/as1000 at XXX.XXX.XXX.10) Locked, Waiting on external entities 2013-01-31 16:19:47.152894 [NOTICE] switch_core_session.c:1517 Session 42 (sofia/internal/as1000 at XXX.XXX.XXX.10) Ended 2013-01-31 16:19:47.152894 [NOTICE] switch_core_session.c:1521 Close Channel sofia/internal/as1000 at XXX.XXX.XXX.10 [CS_DESTROY] 2013-01-31 16:19:47.152894 [DEBUG] switch_core_state_machine.c:556 (sofia/internal/as1000 at XXX.XXX.XXX.10) Callstate Change HANGUP -> DOWN 2013-01-31 16:19:47.152894 [DEBUG] switch_core_state_machine.c:559 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_DESTROY 2013-01-31 16:19:47.152894 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/as1000 at XXX.XXX.XXX.10) State DESTROY 2013-01-31 16:19:47.152894 [DEBUG] mod_sofia.c:396 sofia/internal/as1000 at XXX.XXX.XXX.10 SOFIA DESTROY 2013-01-31 16:19:47.152894 [DEBUG] switch_core_state_machine.c:99 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard DESTROY 2013-01-31 16:19:47.152894 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/as1000 at XXX.XXX.XXX.10) State DESTROY going to sleep 2013-01-31 16:19:48.212902 [NOTICE] switch_channel.c:968 New Channel sofia/internal/as1000 at XXX.XXX.XXX.10 [f12be3b4-6beb-11e2-a3eb-954bab7d8c64] 2013-01-31 16:19:48.212902 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:48.212902 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:48.212902 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_NEW 2013-01-31 16:19:48.212902 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/as1000 at XXX.XXX.XXX.10) State NEW 2013-01-31 16:19:48.232900 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:48.232900 [DEBUG] sofia.c:1719 detaching session f12be3b4-6beb-11e2-a3eb-954bab7d8c64 2013-01-31 16:19:48.382903 [DEBUG] sofia.c:1811 Re-attaching to session f12be3b4-6beb-11e2-a3eb-954bab7d8c64 2013-01-31 16:19:48.382903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:48.382903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:48.392894 [WARNING] sofia_reg.c:2491 Can't find user [as1000 at XXX.XXX.XXX.10] You must define a domain called 'XXX.XXX.XXX.10' in your directory and add a user with the id="as1000" attribute and you must configure your device to use the proper domain in it's authentication credentials. 2013-01-31 16:19:48.392894 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:48.392894 [DEBUG] switch_channel.c:2994 (sofia/internal/as1000 at XXX.XXX.XXX.10) Callstate Change DOWN -> HANGUP 2013-01-31 16:19:48.392894 [NOTICE] sofia.c:1717 Hangup sofia/internal/as1000 at XXX.XXX.XXX.10 [CS_NEW] [CALL_REJECTED] 2013-01-31 16:19:48.392894 [DEBUG] switch_channel.c:3017 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [KILL] 2013-01-31 16:19:48.392894 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:48.412897 [DEBUG] sofia.c:1051 Channel is already hungup. 2013-01-31 16:19:48.412897 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_HANGUP 2013-01-31 16:19:48.412897 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/as1000 at XXX.XXX.XXX.10) State HANGUP 2013-01-31 16:19:48.412897 [DEBUG] mod_sofia.c:503 Channel sofia/internal/as1000 at XXX.XXX.XXX.10 hanging up, cause: CALL_REJECTED 2013-01-31 16:19:48.412897 [DEBUG] switch_core_state_machine.c:48 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard HANGUP, cause: CALL_REJECTED 2013-01-31 16:19:48.412897 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/as1000 at XXX.XXX.XXX.10) State HANGUP going to sleep 2013-01-31 16:19:48.412897 [DEBUG] switch_core_state_machine.c:446 (sofia/internal/as1000 at XXX.XXX.XXX.10) State Change CS_HANGUP -> CS_REPORTING 2013-01-31 16:19:48.412897 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:48.412897 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_REPORTING 2013-01-31 16:19:48.412897 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/as1000 at XXX.XXX.XXX.10) State REPORTING 2013-01-31 16:19:48.412897 [DEBUG] switch_core_state_machine.c:92 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard REPORTING, cause: CALL_REJECTED 2013-01-31 16:19:48.412897 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/as1000 at XXX.XXX.XXX.10) State REPORTING going to sleep 2013-01-31 16:19:48.412897 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/as1000 at XXX.XXX.XXX.10) State Change CS_REPORTING -> CS_DESTROY 2013-01-31 16:19:48.412897 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:48.412897 [DEBUG] switch_core_session.c:1499 Session 43 (sofia/internal/as1000 at XXX.XXX.XXX.10) Locked, Waiting on external entities 2013-01-31 16:19:48.412897 [NOTICE] switch_core_session.c:1517 Session 43 (sofia/internal/as1000 at XXX.XXX.XXX.10) Ended 2013-01-31 16:19:48.412897 [NOTICE] switch_core_session.c:1521 Close Channel sofia/internal/as1000 at XXX.XXX.XXX.10 [CS_DESTROY] 2013-01-31 16:19:48.412897 [DEBUG] switch_core_state_machine.c:556 (sofia/internal/as1000 at XXX.XXX.XXX.10) Callstate Change HANGUP -> DOWN 2013-01-31 16:19:48.412897 [DEBUG] switch_core_state_machine.c:559 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_DESTROY 2013-01-31 16:19:48.412897 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/as1000 at XXX.XXX.XXX.10) State DESTROY 2013-01-31 16:19:48.412897 [DEBUG] mod_sofia.c:396 sofia/internal/as1000 at XXX.XXX.XXX.10 SOFIA DESTROY 2013-01-31 16:19:48.412897 [DEBUG] switch_core_state_machine.c:99 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard DESTROY 2013-01-31 16:19:48.412897 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/as1000 at XXX.XXX.XXX.10) State DESTROY going to sleep 2013-01-31 16:19:51.462895 [NOTICE] switch_channel.c:968 New Channel sofia/internal/as1000 at XXX.XXX.XXX.10 [f31b35da-6beb-11e2-a3ed-954bab7d8c64] 2013-01-31 16:19:51.462895 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:51.462895 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:51.462895 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_NEW 2013-01-31 16:19:51.462895 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/as1000 at XXX.XXX.XXX.10) State NEW 2013-01-31 16:19:51.482900 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:51.482900 [DEBUG] sofia.c:1719 detaching session f31b35da-6beb-11e2-a3ed-954bab7d8c64 2013-01-31 16:19:51.742902 [DEBUG] sofia.c:1811 Re-attaching to session f31b35da-6beb-11e2-a3ed-954bab7d8c64 2013-01-31 16:19:51.742902 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:51.742902 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:51.762900 [WARNING] sofia_reg.c:2491 Can't find user [as1000 at XXX.XXX.XXX.10] You must define a domain called 'XXX.XXX.XXX.10' in your directory and add a user with the id="as1000" attribute and you must configure your device to use the proper domain in it's authentication credentials. 2013-01-31 16:19:51.762900 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:51.762900 [DEBUG] switch_channel.c:2994 (sofia/internal/as1000 at XXX.XXX.XXX.10) Callstate Change DOWN -> HANGUP 2013-01-31 16:19:51.762900 [NOTICE] sofia.c:1717 Hangup sofia/internal/as1000 at XXX.XXX.XXX.10 [CS_NEW] [CALL_REJECTED] 2013-01-31 16:19:51.762900 [DEBUG] switch_channel.c:3017 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [KILL] 2013-01-31 16:19:51.762900 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:51.782895 [DEBUG] sofia.c:1051 Channel is already hungup. 2013-01-31 16:19:51.782895 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_HANGUP 2013-01-31 16:19:51.782895 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/as1000 at XXX.XXX.XXX.10) State HANGUP 2013-01-31 16:19:51.782895 [DEBUG] mod_sofia.c:503 Channel sofia/internal/as1000 at XXX.XXX.XXX.10 hanging up, cause: CALL_REJECTED 2013-01-31 16:19:51.782895 [DEBUG] switch_core_state_machine.c:48 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard HANGUP, cause: CALL_REJECTED 2013-01-31 16:19:51.782895 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/as1000 at XXX.XXX.XXX.10) State HANGUP going to sleep 2013-01-31 16:19:51.782895 [DEBUG] switch_core_state_machine.c:446 (sofia/internal/as1000 at XXX.XXX.XXX.10) State Change CS_HANGUP -> CS_REPORTING 2013-01-31 16:19:51.782895 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:51.782895 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_REPORTING 2013-01-31 16:19:51.782895 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/as1000 at XXX.XXX.XXX.10) State REPORTING 2013-01-31 16:19:51.782895 [DEBUG] switch_core_state_machine.c:92 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard REPORTING, cause: CALL_REJECTED 2013-01-31 16:19:51.782895 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/as1000 at XXX.XXX.XXX.10) State REPORTING going to sleep 2013-01-31 16:19:51.782895 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/as1000 at XXX.XXX.XXX.10) State Change CS_REPORTING -> CS_DESTROY 2013-01-31 16:19:51.782895 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:51.782895 [DEBUG] switch_core_session.c:1499 Session 44 (sofia/internal/as1000 at XXX.XXX.XXX.10) Locked, Waiting on external entities 2013-01-31 16:19:51.782895 [NOTICE] switch_core_session.c:1517 Session 44 (sofia/internal/as1000 at XXX.XXX.XXX.10) Ended 2013-01-31 16:19:51.782895 [NOTICE] switch_core_session.c:1521 Close Channel sofia/internal/as1000 at XXX.XXX.XXX.10 [CS_DESTROY] 2013-01-31 16:19:51.782895 [DEBUG] switch_core_state_machine.c:556 (sofia/internal/as1000 at XXX.XXX.XXX.10) Callstate Change HANGUP -> DOWN 2013-01-31 16:19:51.782895 [DEBUG] switch_core_state_machine.c:559 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_DESTROY 2013-01-31 16:19:51.782895 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/as1000 at XXX.XXX.XXX.10) State DESTROY 2013-01-31 16:19:51.782895 [DEBUG] mod_sofia.c:396 sofia/internal/as1000 at XXX.XXX.XXX.10 SOFIA DESTROY 2013-01-31 16:19:51.782895 [DEBUG] switch_core_state_machine.c:99 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard DESTROY 2013-01-31 16:19:51.782895 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/as1000 at XXX.XXX.XXX.10) State DESTROY going to sleep 2013-01-31 16:19:54.292904 [NOTICE] switch_channel.c:968 New Channel sofia/internal/as1000 at XXX.XXX.XXX.10 [f4cb315a-6beb-11e2-a3ef-954bab7d8c64] 2013-01-31 16:19:54.292904 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:54.292904 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:54.292904 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_NEW 2013-01-31 16:19:54.292904 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/as1000 at XXX.XXX.XXX.10) State NEW 2013-01-31 16:19:54.312896 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:54.312896 [DEBUG] sofia.c:1719 detaching session f4cb315a-6beb-11e2-a3ef-954bab7d8c64 2013-01-31 16:19:54.402894 [DEBUG] sofia.c:1811 Re-attaching to session f4cb315a-6beb-11e2-a3ef-954bab7d8c64 2013-01-31 16:19:54.402894 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:54.402894 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:54.412894 [WARNING] sofia_reg.c:2491 Can't find user [as1000 at XXX.XXX.XXX.10] You must define a domain called 'XXX.XXX.XXX.10' in your directory and add a user with the id="as1000" attribute and you must configure your device to use the proper domain in it's authentication credentials. 2013-01-31 16:19:54.412894 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:54.412894 [DEBUG] switch_channel.c:2994 (sofia/internal/as1000 at XXX.XXX.XXX.10) Callstate Change DOWN -> HANGUP 2013-01-31 16:19:54.412894 [NOTICE] sofia.c:1717 Hangup sofia/internal/as1000 at XXX.XXX.XXX.10 [CS_NEW] [CALL_REJECTED] 2013-01-31 16:19:54.412894 [DEBUG] switch_channel.c:3017 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [KILL] 2013-01-31 16:19:54.412894 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:54.432894 [DEBUG] sofia.c:1051 Channel is already hungup. 2013-01-31 16:19:54.432894 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_HANGUP 2013-01-31 16:19:54.432894 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/as1000 at XXX.XXX.XXX.10) State HANGUP 2013-01-31 16:19:54.432894 [DEBUG] mod_sofia.c:503 Channel sofia/internal/as1000 at XXX.XXX.XXX.10 hanging up, cause: CALL_REJECTED 2013-01-31 16:19:54.432894 [DEBUG] switch_core_state_machine.c:48 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard HANGUP, cause: CALL_REJECTED 2013-01-31 16:19:54.432894 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/as1000 at XXX.XXX.XXX.10) State HANGUP going to sleep 2013-01-31 16:19:54.432894 [DEBUG] switch_core_state_machine.c:446 (sofia/internal/as1000 at XXX.XXX.XXX.10) State Change CS_HANGUP -> CS_REPORTING 2013-01-31 16:19:54.432894 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:54.432894 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_REPORTING 2013-01-31 16:19:54.432894 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/as1000 at XXX.XXX.XXX.10) State REPORTING 2013-01-31 16:19:54.432894 [DEBUG] switch_core_state_machine.c:92 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard REPORTING, cause: CALL_REJECTED 2013-01-31 16:19:54.432894 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/as1000 at XXX.XXX.XXX.10) State REPORTING going to sleep 2013-01-31 16:19:54.432894 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/as1000 at XXX.XXX.XXX.10) State Change CS_REPORTING -> CS_DESTROY 2013-01-31 16:19:54.432894 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:54.432894 [DEBUG] switch_core_session.c:1499 Session 45 (sofia/internal/as1000 at XXX.XXX.XXX.10) Locked, Waiting on external entities 2013-01-31 16:19:54.432894 [NOTICE] switch_core_session.c:1517 Session 45 (sofia/internal/as1000 at XXX.XXX.XXX.10) Ended 2013-01-31 16:19:54.432894 [NOTICE] switch_core_session.c:1521 Close Channel sofia/internal/as1000 at XXX.XXX.XXX.10 [CS_DESTROY] 2013-01-31 16:19:54.432894 [DEBUG] switch_core_state_machine.c:556 (sofia/internal/as1000 at XXX.XXX.XXX.10) Callstate Change HANGUP -> DOWN 2013-01-31 16:19:54.432894 [DEBUG] switch_core_state_machine.c:559 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_DESTROY 2013-01-31 16:19:54.432894 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/as1000 at XXX.XXX.XXX.10) State DESTROY 2013-01-31 16:19:54.432894 [DEBUG] mod_sofia.c:396 sofia/internal/as1000 at XXX.XXX.XXX.10 SOFIA DESTROY 2013-01-31 16:19:54.432894 [DEBUG] switch_core_state_machine.c:99 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard DESTROY 2013-01-31 16:19:54.432894 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/as1000 at XXX.XXX.XXX.10) State DESTROY going to sleep 2013-01-31 16:19:56.362903 [NOTICE] switch_channel.c:968 New Channel sofia/internal/as1000 at XXX.XXX.XXX.10 [f6074b1c-6beb-11e2-a3f1-954bab7d8c64] 2013-01-31 16:19:56.362903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:56.362903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:56.362903 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_NEW 2013-01-31 16:19:56.362903 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/as1000 at XXX.XXX.XXX.10) State NEW 2013-01-31 16:19:56.382903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:56.382903 [DEBUG] sofia.c:1719 detaching session f6074b1c-6beb-11e2-a3f1-954bab7d8c64 2013-01-31 16:19:56.752903 [DEBUG] sofia.c:1811 Re-attaching to session f6074b1c-6beb-11e2-a3f1-954bab7d8c64 2013-01-31 16:19:56.752903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:56.752903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:56.762903 [WARNING] sofia_reg.c:2491 Can't find user [as1000 at XXX.XXX.XXX.10] You must define a domain called 'XXX.XXX.XXX.10' in your directory and add a user with the id="as1000" attribute and you must configure your device to use the proper domain in it's authentication credentials. 2013-01-31 16:19:56.762903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:56.762903 [DEBUG] switch_channel.c:2994 (sofia/internal/as1000 at XXX.XXX.XXX.10) Callstate Change DOWN -> HANGUP 2013-01-31 16:19:56.762903 [NOTICE] sofia.c:1717 Hangup sofia/internal/as1000 at XXX.XXX.XXX.10 [CS_NEW] [CALL_REJECTED] 2013-01-31 16:19:56.762903 [DEBUG] switch_channel.c:3017 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [KILL] 2013-01-31 16:19:56.762903 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:56.782904 [DEBUG] sofia.c:1051 Channel is already hungup. 2013-01-31 16:19:56.782904 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_HANGUP 2013-01-31 16:19:56.782904 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/as1000 at XXX.XXX.XXX.10) State HANGUP 2013-01-31 16:19:56.782904 [DEBUG] mod_sofia.c:503 Channel sofia/internal/as1000 at XXX.XXX.XXX.10 hanging up, cause: CALL_REJECTED 2013-01-31 16:19:56.782904 [DEBUG] switch_core_state_machine.c:48 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard HANGUP, cause: CALL_REJECTED 2013-01-31 16:19:56.782904 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/as1000 at XXX.XXX.XXX.10) State HANGUP going to sleep 2013-01-31 16:19:56.782904 [DEBUG] switch_core_state_machine.c:446 (sofia/internal/as1000 at XXX.XXX.XXX.10) State Change CS_HANGUP -> CS_REPORTING 2013-01-31 16:19:56.782904 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:56.782904 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_REPORTING 2013-01-31 16:19:56.782904 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/as1000 at XXX.XXX.XXX.10) State REPORTING 2013-01-31 16:19:56.782904 [DEBUG] switch_core_state_machine.c:92 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard REPORTING, cause: CALL_REJECTED 2013-01-31 16:19:56.782904 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/as1000 at XXX.XXX.XXX.10) State REPORTING going to sleep 2013-01-31 16:19:56.782904 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/as1000 at XXX.XXX.XXX.10) State Change CS_REPORTING -> CS_DESTROY 2013-01-31 16:19:56.782904 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:56.782904 [DEBUG] switch_core_session.c:1499 Session 46 (sofia/internal/as1000 at XXX.XXX.XXX.10) Locked, Waiting on external entities 2013-01-31 16:19:56.782904 [NOTICE] switch_core_session.c:1517 Session 46 (sofia/internal/as1000 at XXX.XXX.XXX.10) Ended 2013-01-31 16:19:56.782904 [NOTICE] switch_core_session.c:1521 Close Channel sofia/internal/as1000 at XXX.XXX.XXX.10 [CS_DESTROY] 2013-01-31 16:19:56.782904 [DEBUG] switch_core_state_machine.c:556 (sofia/internal/as1000 at XXX.XXX.XXX.10) Callstate Change HANGUP -> DOWN 2013-01-31 16:19:56.782904 [DEBUG] switch_core_state_machine.c:559 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_DESTROY 2013-01-31 16:19:56.782904 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/as1000 at XXX.XXX.XXX.10) State DESTROY 2013-01-31 16:19:56.782904 [DEBUG] mod_sofia.c:396 sofia/internal/as1000 at XXX.XXX.XXX.10 SOFIA DESTROY 2013-01-31 16:19:56.782904 [DEBUG] switch_core_state_machine.c:99 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard DESTROY 2013-01-31 16:19:56.782904 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/as1000 at XXX.XXX.XXX.10) State DESTROY going to sleep 2013-01-31 16:19:57.492903 [NOTICE] switch_channel.c:968 New Channel sofia/internal/as1000 at XXX.XXX.XXX.10 [f6b36ee2-6beb-11e2-a3f3-954bab7d8c64] 2013-01-31 16:19:57.492903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:57.492903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:57.492903 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_NEW 2013-01-31 16:19:57.492903 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/as1000 at XXX.XXX.XXX.10) State NEW 2013-01-31 16:19:57.512900 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:57.512900 [DEBUG] sofia.c:1719 detaching session f6b36ee2-6beb-11e2-a3f3-954bab7d8c64 2013-01-31 16:19:57.622894 [DEBUG] sofia.c:1811 Re-attaching to session f6b36ee2-6beb-11e2-a3f3-954bab7d8c64 2013-01-31 16:19:57.622894 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:57.622894 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:57.632894 [WARNING] sofia_reg.c:2491 Can't find user [as1000 at XXX.XXX.XXX.10] You must define a domain called 'XXX.XXX.XXX.10' in your directory and add a user with the id="as1000" attribute and you must configure your device to use the proper domain in it's authentication credentials. 2013-01-31 16:19:57.632894 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:57.632894 [DEBUG] switch_channel.c:2994 (sofia/internal/as1000 at XXX.XXX.XXX.10) Callstate Change DOWN -> HANGUP 2013-01-31 16:19:57.632894 [NOTICE] sofia.c:1717 Hangup sofia/internal/as1000 at XXX.XXX.XXX.10 [CS_NEW] [CALL_REJECTED] 2013-01-31 16:19:57.632894 [DEBUG] switch_channel.c:3017 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [KILL] 2013-01-31 16:19:57.632894 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:57.652894 [DEBUG] sofia.c:1051 Channel is already hungup. 2013-01-31 16:19:57.652894 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_HANGUP 2013-01-31 16:19:57.652894 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/as1000 at XXX.XXX.XXX.10) State HANGUP 2013-01-31 16:19:57.652894 [DEBUG] mod_sofia.c:503 Channel sofia/internal/as1000 at XXX.XXX.XXX.10 hanging up, cause: CALL_REJECTED 2013-01-31 16:19:57.652894 [DEBUG] switch_core_state_machine.c:48 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard HANGUP, cause: CALL_REJECTED 2013-01-31 16:19:57.652894 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/as1000 at XXX.XXX.XXX.10) State HANGUP going to sleep 2013-01-31 16:19:57.652894 [DEBUG] switch_core_state_machine.c:446 (sofia/internal/as1000 at XXX.XXX.XXX.10) State Change CS_HANGUP -> CS_REPORTING 2013-01-31 16:19:57.652894 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:57.652894 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_REPORTING 2013-01-31 16:19:57.652894 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/as1000 at XXX.XXX.XXX.10) State REPORTING 2013-01-31 16:19:57.652894 [DEBUG] switch_core_state_machine.c:92 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard REPORTING, cause: CALL_REJECTED 2013-01-31 16:19:57.652894 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/as1000 at XXX.XXX.XXX.10) State REPORTING going to sleep 2013-01-31 16:19:57.652894 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/as1000 at XXX.XXX.XXX.10) State Change CS_REPORTING -> CS_DESTROY 2013-01-31 16:19:57.652894 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:57.652894 [DEBUG] switch_core_session.c:1499 Session 47 (sofia/internal/as1000 at XXX.XXX.XXX.10) Locked, Waiting on external entities 2013-01-31 16:19:57.652894 [NOTICE] switch_core_session.c:1517 Session 47 (sofia/internal/as1000 at XXX.XXX.XXX.10) Ended 2013-01-31 16:19:57.652894 [NOTICE] switch_core_session.c:1521 Close Channel sofia/internal/as1000 at XXX.XXX.XXX.10 [CS_DESTROY] 2013-01-31 16:19:57.652894 [DEBUG] switch_core_state_machine.c:556 (sofia/internal/as1000 at XXX.XXX.XXX.10) Callstate Change HANGUP -> DOWN 2013-01-31 16:19:57.652894 [DEBUG] switch_core_state_machine.c:559 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_DESTROY 2013-01-31 16:19:57.652894 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/as1000 at XXX.XXX.XXX.10) State DESTROY 2013-01-31 16:19:57.652894 [DEBUG] mod_sofia.c:396 sofia/internal/as1000 at XXX.XXX.XXX.10 SOFIA DESTROY 2013-01-31 16:19:57.652894 [DEBUG] switch_core_state_machine.c:99 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard DESTROY 2013-01-31 16:19:57.652894 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/as1000 at XXX.XXX.XXX.10) State DESTROY going to sleep 2013-01-31 16:19:59.022903 [NOTICE] switch_channel.c:968 New Channel sofia/internal/as1000 at XXX.XXX.XXX.10 [f79c571a-6beb-11e2-a3f5-954bab7d8c64] 2013-01-31 16:19:59.022903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:59.022903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:59.022903 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_NEW 2013-01-31 16:19:59.022903 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/as1000 at XXX.XXX.XXX.10) State NEW 2013-01-31 16:19:59.042894 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:59.042894 [DEBUG] sofia.c:1719 detaching session f79c571a-6beb-11e2-a3f5-954bab7d8c64 2013-01-31 16:19:59.562898 [DEBUG] sofia.c:1811 Re-attaching to session f79c571a-6beb-11e2-a3f5-954bab7d8c64 2013-01-31 16:19:59.562898 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:59.562898 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:59.582894 [WARNING] sofia_reg.c:2491 Can't find user [as1000 at XXX.XXX.XXX.10] You must define a domain called 'XXX.XXX.XXX.10' in your directory and add a user with the id="as1000" attribute and you must configure your device to use the proper domain in it's authentication credentials. 2013-01-31 16:19:59.582894 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:59.582894 [DEBUG] switch_channel.c:2994 (sofia/internal/as1000 at XXX.XXX.XXX.10) Callstate Change DOWN -> HANGUP 2013-01-31 16:19:59.582894 [NOTICE] sofia.c:1717 Hangup sofia/internal/as1000 at XXX.XXX.XXX.10 [CS_NEW] [CALL_REJECTED] 2013-01-31 16:19:59.582894 [DEBUG] switch_channel.c:3017 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [KILL] 2013-01-31 16:19:59.582894 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:59.602894 [DEBUG] sofia.c:1051 Channel is already hungup. 2013-01-31 16:19:59.602894 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_HANGUP 2013-01-31 16:19:59.602894 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/as1000 at XXX.XXX.XXX.10) State HANGUP 2013-01-31 16:19:59.602894 [DEBUG] mod_sofia.c:503 Channel sofia/internal/as1000 at XXX.XXX.XXX.10 hanging up, cause: CALL_REJECTED 2013-01-31 16:19:59.602894 [DEBUG] switch_core_state_machine.c:48 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard HANGUP, cause: CALL_REJECTED 2013-01-31 16:19:59.602894 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/as1000 at XXX.XXX.XXX.10) State HANGUP going to sleep 2013-01-31 16:19:59.602894 [DEBUG] switch_core_state_machine.c:446 (sofia/internal/as1000 at XXX.XXX.XXX.10) State Change CS_HANGUP -> CS_REPORTING 2013-01-31 16:19:59.602894 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:59.602894 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_REPORTING 2013-01-31 16:19:59.602894 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/as1000 at XXX.XXX.XXX.10) State REPORTING 2013-01-31 16:19:59.602894 [DEBUG] switch_core_state_machine.c:92 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard REPORTING, cause: CALL_REJECTED 2013-01-31 16:19:59.602894 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/as1000 at XXX.XXX.XXX.10) State REPORTING going to sleep 2013-01-31 16:19:59.602894 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/as1000 at XXX.XXX.XXX.10) State Change CS_REPORTING -> CS_DESTROY 2013-01-31 16:19:59.602894 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:59.602894 [DEBUG] switch_core_session.c:1499 Session 48 (sofia/internal/as1000 at XXX.XXX.XXX.10) Locked, Waiting on external entities 2013-01-31 16:19:59.602894 [NOTICE] switch_core_session.c:1517 Session 48 (sofia/internal/as1000 at XXX.XXX.XXX.10) Ended 2013-01-31 16:19:59.602894 [NOTICE] switch_core_session.c:1521 Close Channel sofia/internal/as1000 at XXX.XXX.XXX.10 [CS_DESTROY] 2013-01-31 16:19:59.602894 [DEBUG] switch_core_state_machine.c:556 (sofia/internal/as1000 at XXX.XXX.XXX.10) Callstate Change HANGUP -> DOWN 2013-01-31 16:19:59.602894 [DEBUG] switch_core_state_machine.c:559 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_DESTROY 2013-01-31 16:19:59.602894 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/as1000 at XXX.XXX.XXX.10) State DESTROY 2013-01-31 16:19:59.602894 [DEBUG] mod_sofia.c:396 sofia/internal/as1000 at XXX.XXX.XXX.10 SOFIA DESTROY 2013-01-31 16:19:59.602894 [DEBUG] switch_core_state_machine.c:99 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard DESTROY 2013-01-31 16:19:59.602894 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/as1000 at XXX.XXX.XXX.10) State DESTROY going to sleep 2013-01-31 16:19:59.752893 [NOTICE] switch_channel.c:968 New Channel sofia/internal/as1000 at XXX.XXX.XXX.10 [f80b9a26-6beb-11e2-a3f7-954bab7d8c64] 2013-01-31 16:19:59.752893 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:59.752893 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:59.752893 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_NEW 2013-01-31 16:19:59.752893 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/as1000 at XXX.XXX.XXX.10) State NEW 2013-01-31 16:19:59.772895 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:19:59.772895 [DEBUG] sofia.c:1719 detaching session f80b9a26-6beb-11e2-a3f7-954bab7d8c64 2013-01-31 16:20:01.302894 [NOTICE] switch_channel.c:968 New Channel sofia/internal/as1000 at XXX.XXX.XXX.10 [f8f84c04-6beb-11e2-a3f9-954bab7d8c64] 2013-01-31 16:20:01.302894 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:01.302894 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:01.302894 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_NEW 2013-01-31 16:20:01.302894 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/as1000 at XXX.XXX.XXX.10) State NEW 2013-01-31 16:20:01.322900 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:01.322900 [DEBUG] sofia.c:1719 detaching session f8f84c04-6beb-11e2-a3f9-954bab7d8c64 2013-01-31 16:20:01.522903 [DEBUG] sofia.c:1811 Re-attaching to session f8f84c04-6beb-11e2-a3f9-954bab7d8c64 2013-01-31 16:20:01.522903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:01.522903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:01.522903 [WARNING] sofia_reg.c:2491 Can't find user [as1000 at XXX.XXX.XXX.10] You must define a domain called 'XXX.XXX.XXX.10' in your directory and add a user with the id="as1000" attribute and you must configure your device to use the proper domain in it's authentication credentials. 2013-01-31 16:20:01.522903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:01.522903 [DEBUG] switch_channel.c:2994 (sofia/internal/as1000 at XXX.XXX.XXX.10) Callstate Change DOWN -> HANGUP 2013-01-31 16:20:01.522903 [NOTICE] sofia.c:1717 Hangup sofia/internal/as1000 at XXX.XXX.XXX.10 [CS_NEW] [CALL_REJECTED] 2013-01-31 16:20:01.522903 [DEBUG] switch_channel.c:3017 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [KILL] 2013-01-31 16:20:01.522903 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:01.542900 [DEBUG] sofia.c:1051 Channel is already hungup. 2013-01-31 16:20:01.542900 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_HANGUP 2013-01-31 16:20:01.542900 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/as1000 at XXX.XXX.XXX.10) State HANGUP 2013-01-31 16:20:01.542900 [DEBUG] mod_sofia.c:503 Channel sofia/internal/as1000 at XXX.XXX.XXX.10 hanging up, cause: CALL_REJECTED 2013-01-31 16:20:01.542900 [DEBUG] switch_core_state_machine.c:48 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard HANGUP, cause: CALL_REJECTED 2013-01-31 16:20:01.542900 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/as1000 at XXX.XXX.XXX.10) State HANGUP going to sleep 2013-01-31 16:20:01.542900 [DEBUG] switch_core_state_machine.c:446 (sofia/internal/as1000 at XXX.XXX.XXX.10) State Change CS_HANGUP -> CS_REPORTING 2013-01-31 16:20:01.542900 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:01.542900 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_REPORTING 2013-01-31 16:20:01.542900 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/as1000 at XXX.XXX.XXX.10) State REPORTING 2013-01-31 16:20:01.542900 [DEBUG] switch_core_state_machine.c:92 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard REPORTING, cause: CALL_REJECTED 2013-01-31 16:20:01.542900 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/as1000 at XXX.XXX.XXX.10) State REPORTING going to sleep 2013-01-31 16:20:01.542900 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/as1000 at XXX.XXX.XXX.10) State Change CS_REPORTING -> CS_DESTROY 2013-01-31 16:20:01.542900 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:01.542900 [DEBUG] switch_core_session.c:1499 Session 50 (sofia/internal/as1000 at XXX.XXX.XXX.10) Locked, Waiting on external entities 2013-01-31 16:20:01.542900 [NOTICE] switch_core_session.c:1517 Session 50 (sofia/internal/as1000 at XXX.XXX.XXX.10) Ended 2013-01-31 16:20:01.542900 [NOTICE] switch_core_session.c:1521 Close Channel sofia/internal/as1000 at XXX.XXX.XXX.10 [CS_DESTROY] 2013-01-31 16:20:01.542900 [DEBUG] switch_core_state_machine.c:556 (sofia/internal/as1000 at XXX.XXX.XXX.10) Callstate Change HANGUP -> DOWN 2013-01-31 16:20:01.542900 [DEBUG] switch_core_state_machine.c:559 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_DESTROY 2013-01-31 16:20:01.542900 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/as1000 at XXX.XXX.XXX.10) State DESTROY 2013-01-31 16:20:01.542900 [DEBUG] mod_sofia.c:396 sofia/internal/as1000 at XXX.XXX.XXX.10 SOFIA DESTROY 2013-01-31 16:20:01.542900 [DEBUG] switch_core_state_machine.c:99 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard DESTROY 2013-01-31 16:20:01.542900 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/as1000 at XXX.XXX.XXX.10) State DESTROY going to sleep 2013-01-31 16:20:02.282902 [NOTICE] switch_channel.c:968 New Channel sofia/internal/as1000 at XXX.XXX.XXX.10 [f98e9416-6beb-11e2-a3fb-954bab7d8c64] 2013-01-31 16:20:02.282902 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:02.282902 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:02.282902 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_NEW 2013-01-31 16:20:02.282902 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/as1000 at XXX.XXX.XXX.10) State NEW 2013-01-31 16:20:02.302902 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:02.302902 [DEBUG] sofia.c:1719 detaching session f98e9416-6beb-11e2-a3fb-954bab7d8c64 2013-01-31 16:20:02.772904 [DEBUG] sofia.c:1811 Re-attaching to session f98e9416-6beb-11e2-a3fb-954bab7d8c64 2013-01-31 16:20:02.772904 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:02.772904 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:02.782900 [WARNING] sofia_reg.c:2491 Can't find user [as1000 at XXX.XXX.XXX.10] You must define a domain called 'XXX.XXX.XXX.10' in your directory and add a user with the id="as1000" attribute and you must configure your device to use the proper domain in it's authentication credentials. 2013-01-31 16:20:02.782900 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:02.782900 [DEBUG] switch_channel.c:2994 (sofia/internal/as1000 at XXX.XXX.XXX.10) Callstate Change DOWN -> HANGUP 2013-01-31 16:20:02.782900 [NOTICE] sofia.c:1717 Hangup sofia/internal/as1000 at XXX.XXX.XXX.10 [CS_NEW] [CALL_REJECTED] 2013-01-31 16:20:02.782900 [DEBUG] switch_channel.c:3017 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [KILL] 2013-01-31 16:20:02.782900 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:02.802899 [DEBUG] sofia.c:1051 Channel is already hungup. 2013-01-31 16:20:02.802899 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_HANGUP 2013-01-31 16:20:02.802899 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/as1000 at XXX.XXX.XXX.10) State HANGUP 2013-01-31 16:20:02.802899 [DEBUG] mod_sofia.c:503 Channel sofia/internal/as1000 at XXX.XXX.XXX.10 hanging up, cause: CALL_REJECTED 2013-01-31 16:20:02.802899 [DEBUG] switch_core_state_machine.c:48 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard HANGUP, cause: CALL_REJECTED 2013-01-31 16:20:02.802899 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/as1000 at XXX.XXX.XXX.10) State HANGUP going to sleep 2013-01-31 16:20:02.802899 [DEBUG] switch_core_state_machine.c:446 (sofia/internal/as1000 at XXX.XXX.XXX.10) State Change CS_HANGUP -> CS_REPORTING 2013-01-31 16:20:02.802899 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:02.802899 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_REPORTING 2013-01-31 16:20:02.802899 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/as1000 at XXX.XXX.XXX.10) State REPORTING 2013-01-31 16:20:02.802899 [DEBUG] switch_core_state_machine.c:92 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard REPORTING, cause: CALL_REJECTED 2013-01-31 16:20:02.802899 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/as1000 at XXX.XXX.XXX.10) State REPORTING going to sleep 2013-01-31 16:20:02.802899 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/as1000 at XXX.XXX.XXX.10) State Change CS_REPORTING -> CS_DESTROY 2013-01-31 16:20:02.802899 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:02.802899 [DEBUG] switch_core_session.c:1499 Session 51 (sofia/internal/as1000 at XXX.XXX.XXX.10) Locked, Waiting on external entities 2013-01-31 16:20:02.802899 [NOTICE] switch_core_session.c:1517 Session 51 (sofia/internal/as1000 at XXX.XXX.XXX.10) Ended 2013-01-31 16:20:02.802899 [NOTICE] switch_core_session.c:1521 Close Channel sofia/internal/as1000 at XXX.XXX.XXX.10 [CS_DESTROY] 2013-01-31 16:20:02.802899 [DEBUG] switch_core_state_machine.c:556 (sofia/internal/as1000 at XXX.XXX.XXX.10) Callstate Change HANGUP -> DOWN 2013-01-31 16:20:02.802899 [DEBUG] switch_core_state_machine.c:559 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_DESTROY 2013-01-31 16:20:02.802899 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/as1000 at XXX.XXX.XXX.10) State DESTROY 2013-01-31 16:20:02.802899 [DEBUG] mod_sofia.c:396 sofia/internal/as1000 at XXX.XXX.XXX.10 SOFIA DESTROY 2013-01-31 16:20:02.802899 [DEBUG] switch_core_state_machine.c:99 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard DESTROY 2013-01-31 16:20:02.802899 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/as1000 at XXX.XXX.XXX.10) State DESTROY going to sleep 2013-01-31 16:20:04.082903 [NOTICE] switch_channel.c:968 New Channel sofia/internal/as1000 at XXX.XXX.XXX.10 [faa0545c-6beb-11e2-a3fd-954bab7d8c64] 2013-01-31 16:20:04.082903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:04.082903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:04.082903 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_NEW 2013-01-31 16:20:04.082903 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/as1000 at XXX.XXX.XXX.10) State NEW 2013-01-31 16:20:04.102894 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:04.102894 [DEBUG] sofia.c:1719 detaching session faa0545c-6beb-11e2-a3fd-954bab7d8c64 2013-01-31 16:20:04.512903 [DEBUG] sofia.c:1811 Re-attaching to session faa0545c-6beb-11e2-a3fd-954bab7d8c64 2013-01-31 16:20:04.512903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:04.512903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:04.522903 [WARNING] sofia_reg.c:2491 Can't find user [as1000 at XXX.XXX.XXX.10] You must define a domain called 'XXX.XXX.XXX.10' in your directory and add a user with the id="as1000" attribute and you must configure your device to use the proper domain in it's authentication credentials. 2013-01-31 16:20:04.522903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:04.522903 [DEBUG] switch_channel.c:2994 (sofia/internal/as1000 at XXX.XXX.XXX.10) Callstate Change DOWN -> HANGUP 2013-01-31 16:20:04.522903 [NOTICE] sofia.c:1717 Hangup sofia/internal/as1000 at XXX.XXX.XXX.10 [CS_NEW] [CALL_REJECTED] 2013-01-31 16:20:04.522903 [DEBUG] switch_channel.c:3017 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [KILL] 2013-01-31 16:20:04.522903 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:04.542892 [DEBUG] sofia.c:1051 Channel is already hungup. 2013-01-31 16:20:04.542892 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_HANGUP 2013-01-31 16:20:04.542892 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/as1000 at XXX.XXX.XXX.10) State HANGUP 2013-01-31 16:20:04.542892 [DEBUG] mod_sofia.c:503 Channel sofia/internal/as1000 at XXX.XXX.XXX.10 hanging up, cause: CALL_REJECTED 2013-01-31 16:20:04.542892 [DEBUG] switch_core_state_machine.c:48 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard HANGUP, cause: CALL_REJECTED 2013-01-31 16:20:04.542892 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/as1000 at XXX.XXX.XXX.10) State HANGUP going to sleep 2013-01-31 16:20:04.542892 [DEBUG] switch_core_state_machine.c:446 (sofia/internal/as1000 at XXX.XXX.XXX.10) State Change CS_HANGUP -> CS_REPORTING 2013-01-31 16:20:04.542892 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:04.542892 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_REPORTING 2013-01-31 16:20:04.542892 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/as1000 at XXX.XXX.XXX.10) State REPORTING 2013-01-31 16:20:04.542892 [DEBUG] switch_core_state_machine.c:92 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard REPORTING, cause: CALL_REJECTED 2013-01-31 16:20:04.542892 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/as1000 at XXX.XXX.XXX.10) State REPORTING going to sleep 2013-01-31 16:20:04.542892 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/as1000 at XXX.XXX.XXX.10) State Change CS_REPORTING -> CS_DESTROY 2013-01-31 16:20:04.542892 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:04.542892 [DEBUG] switch_core_session.c:1499 Session 52 (sofia/internal/as1000 at XXX.XXX.XXX.10) Locked, Waiting on external entities 2013-01-31 16:20:04.542892 [NOTICE] switch_core_session.c:1517 Session 52 (sofia/internal/as1000 at XXX.XXX.XXX.10) Ended 2013-01-31 16:20:04.542892 [NOTICE] switch_core_session.c:1521 Close Channel sofia/internal/as1000 at XXX.XXX.XXX.10 [CS_DESTROY] 2013-01-31 16:20:04.542892 [DEBUG] switch_core_state_machine.c:556 (sofia/internal/as1000 at XXX.XXX.XXX.10) Callstate Change HANGUP -> DOWN 2013-01-31 16:20:04.542892 [DEBUG] switch_core_state_machine.c:559 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_DESTROY 2013-01-31 16:20:04.542892 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/as1000 at XXX.XXX.XXX.10) State DESTROY 2013-01-31 16:20:04.542892 [DEBUG] mod_sofia.c:396 sofia/internal/as1000 at XXX.XXX.XXX.10 SOFIA DESTROY 2013-01-31 16:20:04.542892 [DEBUG] switch_core_state_machine.c:99 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard DESTROY 2013-01-31 16:20:04.542892 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/as1000 at XXX.XXX.XXX.10) State DESTROY going to sleep 2013-01-31 16:20:05.412903 [NOTICE] switch_channel.c:968 New Channel sofia/internal/as1000 at XXX.XXX.XXX.10 [fb6cb25e-6beb-11e2-a3ff-954bab7d8c64] 2013-01-31 16:20:05.412903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:05.412903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:05.412903 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_NEW 2013-01-31 16:20:05.422895 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/as1000 at XXX.XXX.XXX.10) State NEW 2013-01-31 16:20:05.442896 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:05.442896 [DEBUG] sofia.c:1719 detaching session fb6cb25e-6beb-11e2-a3ff-954bab7d8c64 2013-01-31 16:20:05.862895 [DEBUG] sofia.c:1811 Re-attaching to session fb6cb25e-6beb-11e2-a3ff-954bab7d8c64 2013-01-31 16:20:05.862895 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:05.862895 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:05.882894 [WARNING] sofia_reg.c:2491 Can't find user [as1000 at XXX.XXX.XXX.10] You must define a domain called 'XXX.XXX.XXX.10' in your directory and add a user with the id="as1000" attribute and you must configure your device to use the proper domain in it's authentication credentials. 2013-01-31 16:20:05.882894 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:05.882894 [DEBUG] switch_channel.c:2994 (sofia/internal/as1000 at XXX.XXX.XXX.10) Callstate Change DOWN -> HANGUP 2013-01-31 16:20:05.882894 [NOTICE] sofia.c:1717 Hangup sofia/internal/as1000 at XXX.XXX.XXX.10 [CS_NEW] [CALL_REJECTED] 2013-01-31 16:20:05.882894 [DEBUG] switch_channel.c:3017 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [KILL] 2013-01-31 16:20:05.882894 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:05.902894 [DEBUG] sofia.c:1051 Channel is already hungup. 2013-01-31 16:20:05.902894 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_HANGUP 2013-01-31 16:20:05.902894 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/as1000 at XXX.XXX.XXX.10) State HANGUP 2013-01-31 16:20:05.902894 [DEBUG] mod_sofia.c:503 Channel sofia/internal/as1000 at XXX.XXX.XXX.10 hanging up, cause: CALL_REJECTED 2013-01-31 16:20:05.902894 [DEBUG] switch_core_state_machine.c:48 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard HANGUP, cause: CALL_REJECTED 2013-01-31 16:20:05.902894 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/as1000 at XXX.XXX.XXX.10) State HANGUP going to sleep 2013-01-31 16:20:05.902894 [DEBUG] switch_core_state_machine.c:446 (sofia/internal/as1000 at XXX.XXX.XXX.10) State Change CS_HANGUP -> CS_REPORTING 2013-01-31 16:20:05.902894 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:05.902894 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_REPORTING 2013-01-31 16:20:05.902894 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/as1000 at XXX.XXX.XXX.10) State REPORTING 2013-01-31 16:20:05.902894 [DEBUG] switch_core_state_machine.c:92 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard REPORTING, cause: CALL_REJECTED 2013-01-31 16:20:05.902894 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/as1000 at XXX.XXX.XXX.10) State REPORTING going to sleep 2013-01-31 16:20:05.902894 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/as1000 at XXX.XXX.XXX.10) State Change CS_REPORTING -> CS_DESTROY 2013-01-31 16:20:05.902894 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:05.902894 [DEBUG] switch_core_session.c:1499 Session 53 (sofia/internal/as1000 at XXX.XXX.XXX.10) Locked, Waiting on external entities 2013-01-31 16:20:05.902894 [NOTICE] switch_core_session.c:1517 Session 53 (sofia/internal/as1000 at XXX.XXX.XXX.10) Ended 2013-01-31 16:20:05.902894 [NOTICE] switch_core_session.c:1521 Close Channel sofia/internal/as1000 at XXX.XXX.XXX.10 [CS_DESTROY] 2013-01-31 16:20:05.902894 [DEBUG] switch_core_state_machine.c:556 (sofia/internal/as1000 at XXX.XXX.XXX.10) Callstate Change HANGUP -> DOWN 2013-01-31 16:20:05.902894 [DEBUG] switch_core_state_machine.c:559 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_DESTROY 2013-01-31 16:20:05.902894 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/as1000 at XXX.XXX.XXX.10) State DESTROY 2013-01-31 16:20:05.902894 [DEBUG] mod_sofia.c:396 sofia/internal/as1000 at XXX.XXX.XXX.10 SOFIA DESTROY 2013-01-31 16:20:05.902894 [DEBUG] switch_core_state_machine.c:99 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard DESTROY 2013-01-31 16:20:05.902894 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/as1000 at XXX.XXX.XXX.10) State DESTROY going to sleep 2013-01-31 16:20:08.102902 [NOTICE] switch_channel.c:968 New Channel sofia/internal/as1000 at XXX.XXX.XXX.10 [fd062d66-6beb-11e2-a401-954bab7d8c64] 2013-01-31 16:20:08.102902 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:08.102902 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:08.102902 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_NEW 2013-01-31 16:20:08.102902 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/as1000 at XXX.XXX.XXX.10) State NEW 2013-01-31 16:20:08.122900 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:08.122900 [DEBUG] sofia.c:1719 detaching session fd062d66-6beb-11e2-a401-954bab7d8c64 2013-01-31 16:20:08.222903 [DEBUG] sofia.c:1811 Re-attaching to session fd062d66-6beb-11e2-a401-954bab7d8c64 2013-01-31 16:20:08.222903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:08.222903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:08.242904 [WARNING] sofia_reg.c:2491 Can't find user [as1000 at XXX.XXX.XXX.10] You must define a domain called 'XXX.XXX.XXX.10' in your directory and add a user with the id="as1000" attribute and you must configure your device to use the proper domain in it's authentication credentials. 2013-01-31 16:20:08.242904 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:08.242904 [DEBUG] switch_channel.c:2994 (sofia/internal/as1000 at XXX.XXX.XXX.10) Callstate Change DOWN -> HANGUP 2013-01-31 16:20:08.242904 [NOTICE] sofia.c:1717 Hangup sofia/internal/as1000 at XXX.XXX.XXX.10 [CS_NEW] [CALL_REJECTED] 2013-01-31 16:20:08.242904 [DEBUG] switch_channel.c:3017 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [KILL] 2013-01-31 16:20:08.242904 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:08.262894 [DEBUG] sofia.c:1051 Channel is already hungup. 2013-01-31 16:20:08.262894 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_HANGUP 2013-01-31 16:20:08.262894 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/as1000 at XXX.XXX.XXX.10) State HANGUP 2013-01-31 16:20:08.262894 [DEBUG] mod_sofia.c:503 Channel sofia/internal/as1000 at XXX.XXX.XXX.10 hanging up, cause: CALL_REJECTED 2013-01-31 16:20:08.262894 [DEBUG] switch_core_state_machine.c:48 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard HANGUP, cause: CALL_REJECTED 2013-01-31 16:20:08.262894 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/as1000 at XXX.XXX.XXX.10) State HANGUP going to sleep 2013-01-31 16:20:08.262894 [DEBUG] switch_core_state_machine.c:446 (sofia/internal/as1000 at XXX.XXX.XXX.10) State Change CS_HANGUP -> CS_REPORTING 2013-01-31 16:20:08.262894 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:08.262894 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_REPORTING 2013-01-31 16:20:08.262894 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/as1000 at XXX.XXX.XXX.10) State REPORTING 2013-01-31 16:20:08.262894 [DEBUG] switch_core_state_machine.c:92 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard REPORTING, cause: CALL_REJECTED 2013-01-31 16:20:08.262894 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/as1000 at XXX.XXX.XXX.10) State REPORTING going to sleep 2013-01-31 16:20:08.262894 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/as1000 at XXX.XXX.XXX.10) State Change CS_REPORTING -> CS_DESTROY 2013-01-31 16:20:08.262894 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:08.262894 [DEBUG] switch_core_session.c:1499 Session 54 (sofia/internal/as1000 at XXX.XXX.XXX.10) Locked, Waiting on external entities 2013-01-31 16:20:08.262894 [NOTICE] switch_core_session.c:1517 Session 54 (sofia/internal/as1000 at XXX.XXX.XXX.10) Ended 2013-01-31 16:20:08.262894 [NOTICE] switch_core_session.c:1521 Close Channel sofia/internal/as1000 at XXX.XXX.XXX.10 [CS_DESTROY] 2013-01-31 16:20:08.262894 [DEBUG] switch_core_state_machine.c:556 (sofia/internal/as1000 at XXX.XXX.XXX.10) Callstate Change HANGUP -> DOWN 2013-01-31 16:20:08.262894 [DEBUG] switch_core_state_machine.c:559 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_DESTROY 2013-01-31 16:20:08.262894 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/as1000 at XXX.XXX.XXX.10) State DESTROY 2013-01-31 16:20:08.262894 [DEBUG] mod_sofia.c:396 sofia/internal/as1000 at XXX.XXX.XXX.10 SOFIA DESTROY 2013-01-31 16:20:08.262894 [DEBUG] switch_core_state_machine.c:99 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard DESTROY 2013-01-31 16:20:08.262894 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/as1000 at XXX.XXX.XXX.10) State DESTROY going to sleep 2013-01-31 16:20:09.592903 [NOTICE] switch_channel.c:968 New Channel sofia/internal/as1000 at XXX.XXX.XXX.10 [fdea7250-6beb-11e2-a403-954bab7d8c64] 2013-01-31 16:20:09.592903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:09.592903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:09.592903 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_NEW 2013-01-31 16:20:09.592903 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/as1000 at XXX.XXX.XXX.10) State NEW 2013-01-31 16:20:09.622900 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:09.622900 [DEBUG] sofia.c:1719 detaching session fdea7250-6beb-11e2-a403-954bab7d8c64 2013-01-31 16:20:09.762903 [WARNING] switch_core_state_machine.c:514 f80b9a26-6beb-11e2-a3f7-954bab7d8c64 sofia/internal/as1000 at XXX.XXX.XXX.10 Abandoned 2013-01-31 16:20:09.762903 [DEBUG] switch_channel.c:2994 (sofia/internal/as1000 at XXX.XXX.XXX.10) Callstate Change DOWN -> HANGUP 2013-01-31 16:20:09.762903 [NOTICE] switch_core_state_machine.c:517 Hangup sofia/internal/as1000 at XXX.XXX.XXX.10 [CS_NEW] [WRONG_CALL_STATE] 2013-01-31 16:20:09.762903 [DEBUG] switch_channel.c:3017 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [KILL] 2013-01-31 16:20:09.762903 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:09.762903 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_HANGUP 2013-01-31 16:20:09.762903 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/as1000 at XXX.XXX.XXX.10) State HANGUP 2013-01-31 16:20:09.762903 [DEBUG] mod_sofia.c:503 Channel sofia/internal/as1000 at XXX.XXX.XXX.10 hanging up, cause: WRONG_CALL_STATE 2013-01-31 16:20:09.762903 [DEBUG] switch_core_state_machine.c:48 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard HANGUP, cause: WRONG_CALL_STATE 2013-01-31 16:20:09.762903 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/as1000 at XXX.XXX.XXX.10) State HANGUP going to sleep 2013-01-31 16:20:09.762903 [DEBUG] switch_core_state_machine.c:446 (sofia/internal/as1000 at XXX.XXX.XXX.10) State Change CS_HANGUP -> CS_REPORTING 2013-01-31 16:20:09.762903 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:09.762903 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_REPORTING 2013-01-31 16:20:09.762903 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/as1000 at XXX.XXX.XXX.10) State REPORTING 2013-01-31 16:20:09.762903 [DEBUG] switch_core_state_machine.c:92 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard REPORTING, cause: WRONG_CALL_STATE 2013-01-31 16:20:09.762903 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/as1000 at XXX.XXX.XXX.10) State REPORTING going to sleep 2013-01-31 16:20:09.762903 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/as1000 at XXX.XXX.XXX.10) State Change CS_REPORTING -> CS_DESTROY 2013-01-31 16:20:09.762903 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:09.762903 [DEBUG] switch_core_session.c:1499 Session 49 (sofia/internal/as1000 at XXX.XXX.XXX.10) Locked, Waiting on external entities 2013-01-31 16:20:09.762903 [NOTICE] switch_core_session.c:1517 Session 49 (sofia/internal/as1000 at XXX.XXX.XXX.10) Ended 2013-01-31 16:20:09.762903 [NOTICE] switch_core_session.c:1521 Close Channel sofia/internal/as1000 at XXX.XXX.XXX.10 [CS_DESTROY] 2013-01-31 16:20:09.762903 [DEBUG] switch_core_state_machine.c:556 (sofia/internal/as1000 at XXX.XXX.XXX.10) Callstate Change HANGUP -> DOWN 2013-01-31 16:20:09.762903 [DEBUG] switch_core_state_machine.c:559 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_DESTROY 2013-01-31 16:20:09.762903 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/as1000 at XXX.XXX.XXX.10) State DESTROY 2013-01-31 16:20:09.762903 [DEBUG] mod_sofia.c:396 sofia/internal/as1000 at XXX.XXX.XXX.10 SOFIA DESTROY 2013-01-31 16:20:09.762903 [DEBUG] switch_core_state_machine.c:99 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard DESTROY 2013-01-31 16:20:09.762903 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/as1000 at XXX.XXX.XXX.10) State DESTROY going to sleep 2013-01-31 16:20:09.842902 [DEBUG] sofia.c:1811 Re-attaching to session fdea7250-6beb-11e2-a403-954bab7d8c64 2013-01-31 16:20:09.842902 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:09.842902 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:09.862895 [WARNING] sofia_reg.c:2491 Can't find user [as1000 at XXX.XXX.XXX.10] You must define a domain called 'XXX.XXX.XXX.10' in your directory and add a user with the id="as1000" attribute and you must configure your device to use the proper domain in it's authentication credentials. 2013-01-31 16:20:09.862895 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:09.862895 [DEBUG] switch_channel.c:2994 (sofia/internal/as1000 at XXX.XXX.XXX.10) Callstate Change DOWN -> HANGUP 2013-01-31 16:20:09.862895 [NOTICE] sofia.c:1717 Hangup sofia/internal/as1000 at XXX.XXX.XXX.10 [CS_NEW] [CALL_REJECTED] 2013-01-31 16:20:09.862895 [DEBUG] switch_channel.c:3017 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [KILL] 2013-01-31 16:20:09.862895 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:09.882904 [DEBUG] sofia.c:1051 Channel is already hungup. 2013-01-31 16:20:09.882904 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_HANGUP 2013-01-31 16:20:09.882904 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/as1000 at XXX.XXX.XXX.10) State HANGUP 2013-01-31 16:20:09.882904 [DEBUG] mod_sofia.c:503 Channel sofia/internal/as1000 at XXX.XXX.XXX.10 hanging up, cause: CALL_REJECTED 2013-01-31 16:20:09.882904 [DEBUG] switch_core_state_machine.c:48 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard HANGUP, cause: CALL_REJECTED 2013-01-31 16:20:09.882904 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/as1000 at XXX.XXX.XXX.10) State HANGUP going to sleep 2013-01-31 16:20:09.882904 [DEBUG] switch_core_state_machine.c:446 (sofia/internal/as1000 at XXX.XXX.XXX.10) State Change CS_HANGUP -> CS_REPORTING 2013-01-31 16:20:09.882904 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:09.882904 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_REPORTING 2013-01-31 16:20:09.882904 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/as1000 at XXX.XXX.XXX.10) State REPORTING 2013-01-31 16:20:09.882904 [DEBUG] switch_core_state_machine.c:92 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard REPORTING, cause: CALL_REJECTED 2013-01-31 16:20:09.882904 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/as1000 at XXX.XXX.XXX.10) State REPORTING going to sleep 2013-01-31 16:20:09.882904 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/as1000 at XXX.XXX.XXX.10) State Change CS_REPORTING -> CS_DESTROY 2013-01-31 16:20:09.882904 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:09.882904 [DEBUG] switch_core_session.c:1499 Session 55 (sofia/internal/as1000 at XXX.XXX.XXX.10) Locked, Waiting on external entities 2013-01-31 16:20:09.882904 [NOTICE] switch_core_session.c:1517 Session 55 (sofia/internal/as1000 at XXX.XXX.XXX.10) Ended 2013-01-31 16:20:09.882904 [NOTICE] switch_core_session.c:1521 Close Channel sofia/internal/as1000 at XXX.XXX.XXX.10 [CS_DESTROY] 2013-01-31 16:20:09.882904 [DEBUG] switch_core_state_machine.c:556 (sofia/internal/as1000 at XXX.XXX.XXX.10) Callstate Change HANGUP -> DOWN 2013-01-31 16:20:09.882904 [DEBUG] switch_core_state_machine.c:559 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_DESTROY 2013-01-31 16:20:09.882904 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/as1000 at XXX.XXX.XXX.10) State DESTROY 2013-01-31 16:20:09.882904 [DEBUG] mod_sofia.c:396 sofia/internal/as1000 at XXX.XXX.XXX.10 SOFIA DESTROY 2013-01-31 16:20:09.882904 [DEBUG] switch_core_state_machine.c:99 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard DESTROY 2013-01-31 16:20:09.882904 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/as1000 at XXX.XXX.XXX.10) State DESTROY going to sleep 2013-01-31 16:20:10.592904 [NOTICE] switch_channel.c:968 New Channel sofia/internal/as1000 at XXX.XXX.XXX.10 [fe82e472-6beb-11e2-a405-954bab7d8c64] 2013-01-31 16:20:10.592904 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:10.592904 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:10.592904 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_NEW 2013-01-31 16:20:10.592904 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/as1000 at XXX.XXX.XXX.10) State NEW 2013-01-31 16:20:10.612903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:10.612903 [DEBUG] sofia.c:1719 detaching session fe82e472-6beb-11e2-a405-954bab7d8c64 2013-01-31 16:20:10.982904 [DEBUG] sofia.c:1811 Re-attaching to session fe82e472-6beb-11e2-a405-954bab7d8c64 2013-01-31 16:20:10.982904 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:10.982904 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:10.992903 [WARNING] sofia_reg.c:2491 Can't find user [as1000 at XXX.XXX.XXX.10] You must define a domain called 'XXX.XXX.XXX.10' in your directory and add a user with the id="as1000" attribute and you must configure your device to use the proper domain in it's authentication credentials. 2013-01-31 16:20:11.002904 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:11.002904 [DEBUG] switch_channel.c:2994 (sofia/internal/as1000 at XXX.XXX.XXX.10) Callstate Change DOWN -> HANGUP 2013-01-31 16:20:11.002904 [NOTICE] sofia.c:1717 Hangup sofia/internal/as1000 at XXX.XXX.XXX.10 [CS_NEW] [CALL_REJECTED] 2013-01-31 16:20:11.002904 [DEBUG] switch_channel.c:3017 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [KILL] 2013-01-31 16:20:11.002904 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:11.022903 [DEBUG] sofia.c:1051 Channel is already hungup. 2013-01-31 16:20:11.022903 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_HANGUP 2013-01-31 16:20:11.022903 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/as1000 at XXX.XXX.XXX.10) State HANGUP 2013-01-31 16:20:11.022903 [DEBUG] mod_sofia.c:503 Channel sofia/internal/as1000 at XXX.XXX.XXX.10 hanging up, cause: CALL_REJECTED 2013-01-31 16:20:11.022903 [DEBUG] switch_core_state_machine.c:48 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard HANGUP, cause: CALL_REJECTED 2013-01-31 16:20:11.022903 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/as1000 at XXX.XXX.XXX.10) State HANGUP going to sleep 2013-01-31 16:20:11.022903 [DEBUG] switch_core_state_machine.c:446 (sofia/internal/as1000 at XXX.XXX.XXX.10) State Change CS_HANGUP -> CS_REPORTING 2013-01-31 16:20:11.022903 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:11.022903 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_REPORTING 2013-01-31 16:20:11.022903 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/as1000 at XXX.XXX.XXX.10) State REPORTING 2013-01-31 16:20:11.022903 [DEBUG] switch_core_state_machine.c:92 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard REPORTING, cause: CALL_REJECTED 2013-01-31 16:20:11.022903 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/as1000 at XXX.XXX.XXX.10) State REPORTING going to sleep 2013-01-31 16:20:11.022903 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/as1000 at XXX.XXX.XXX.10) State Change CS_REPORTING -> CS_DESTROY 2013-01-31 16:20:11.022903 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:11.022903 [DEBUG] switch_core_session.c:1499 Session 56 (sofia/internal/as1000 at XXX.XXX.XXX.10) Locked, Waiting on external entities 2013-01-31 16:20:11.022903 [NOTICE] switch_core_session.c:1517 Session 56 (sofia/internal/as1000 at XXX.XXX.XXX.10) Ended 2013-01-31 16:20:11.022903 [NOTICE] switch_core_session.c:1521 Close Channel sofia/internal/as1000 at XXX.XXX.XXX.10 [CS_DESTROY] 2013-01-31 16:20:11.022903 [DEBUG] switch_core_state_machine.c:556 (sofia/internal/as1000 at XXX.XXX.XXX.10) Callstate Change HANGUP -> DOWN 2013-01-31 16:20:11.022903 [DEBUG] switch_core_state_machine.c:559 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_DESTROY 2013-01-31 16:20:11.022903 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/as1000 at XXX.XXX.XXX.10) State DESTROY 2013-01-31 16:20:11.022903 [DEBUG] mod_sofia.c:396 sofia/internal/as1000 at XXX.XXX.XXX.10 SOFIA DESTROY 2013-01-31 16:20:11.022903 [DEBUG] switch_core_state_machine.c:99 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard DESTROY 2013-01-31 16:20:11.022903 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/as1000 at XXX.XXX.XXX.10) State DESTROY going to sleep 2013-01-31 16:20:12.262903 [NOTICE] switch_channel.c:968 New Channel sofia/internal/as1000 at XXX.XXX.XXX.10 [ff807c90-6beb-11e2-a407-954bab7d8c64] 2013-01-31 16:20:12.262903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:12.262903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:12.262903 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_NEW 2013-01-31 16:20:12.262903 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/as1000 at XXX.XXX.XXX.10) State NEW 2013-01-31 16:20:12.282895 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:12.282895 [DEBUG] sofia.c:1719 detaching session ff807c90-6beb-11e2-a407-954bab7d8c64 2013-01-31 16:20:12.362907 [DEBUG] sofia.c:1811 Re-attaching to session ff807c90-6beb-11e2-a407-954bab7d8c64 2013-01-31 16:20:12.362907 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:12.362907 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:12.382903 [WARNING] sofia_reg.c:2491 Can't find user [as1000 at XXX.XXX.XXX.10] You must define a domain called 'XXX.XXX.XXX.10' in your directory and add a user with the id="as1000" attribute and you must configure your device to use the proper domain in it's authentication credentials. 2013-01-31 16:20:12.382903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:12.382903 [DEBUG] switch_channel.c:2994 (sofia/internal/as1000 at XXX.XXX.XXX.10) Callstate Change DOWN -> HANGUP 2013-01-31 16:20:12.382903 [NOTICE] sofia.c:1717 Hangup sofia/internal/as1000 at XXX.XXX.XXX.10 [CS_NEW] [CALL_REJECTED] 2013-01-31 16:20:12.382903 [DEBUG] switch_channel.c:3017 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [KILL] 2013-01-31 16:20:12.382903 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:12.402904 [DEBUG] sofia.c:1051 Channel is already hungup. 2013-01-31 16:20:12.402904 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_HANGUP 2013-01-31 16:20:12.402904 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/as1000 at XXX.XXX.XXX.10) State HANGUP 2013-01-31 16:20:12.402904 [DEBUG] mod_sofia.c:503 Channel sofia/internal/as1000 at XXX.XXX.XXX.10 hanging up, cause: CALL_REJECTED 2013-01-31 16:20:12.402904 [DEBUG] switch_core_state_machine.c:48 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard HANGUP, cause: CALL_REJECTED 2013-01-31 16:20:12.402904 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/as1000 at XXX.XXX.XXX.10) State HANGUP going to sleep 2013-01-31 16:20:12.402904 [DEBUG] switch_core_state_machine.c:446 (sofia/internal/as1000 at XXX.XXX.XXX.10) State Change CS_HANGUP -> CS_REPORTING 2013-01-31 16:20:12.402904 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:12.402904 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_REPORTING 2013-01-31 16:20:12.402904 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/as1000 at XXX.XXX.XXX.10) State REPORTING 2013-01-31 16:20:12.402904 [DEBUG] switch_core_state_machine.c:92 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard REPORTING, cause: CALL_REJECTED 2013-01-31 16:20:12.402904 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/as1000 at XXX.XXX.XXX.10) State REPORTING going to sleep 2013-01-31 16:20:12.402904 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/as1000 at XXX.XXX.XXX.10) State Change CS_REPORTING -> CS_DESTROY 2013-01-31 16:20:12.402904 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:12.402904 [DEBUG] switch_core_session.c:1499 Session 57 (sofia/internal/as1000 at XXX.XXX.XXX.10) Locked, Waiting on external entities 2013-01-31 16:20:12.402904 [NOTICE] switch_core_session.c:1517 Session 57 (sofia/internal/as1000 at XXX.XXX.XXX.10) Ended 2013-01-31 16:20:12.402904 [NOTICE] switch_core_session.c:1521 Close Channel sofia/internal/as1000 at XXX.XXX.XXX.10 [CS_DESTROY] 2013-01-31 16:20:12.402904 [DEBUG] switch_core_state_machine.c:556 (sofia/internal/as1000 at XXX.XXX.XXX.10) Callstate Change HANGUP -> DOWN 2013-01-31 16:20:12.402904 [DEBUG] switch_core_state_machine.c:559 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_DESTROY 2013-01-31 16:20:12.402904 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/as1000 at XXX.XXX.XXX.10) State DESTROY 2013-01-31 16:20:12.402904 [DEBUG] mod_sofia.c:396 sofia/internal/as1000 at XXX.XXX.XXX.10 SOFIA DESTROY 2013-01-31 16:20:12.402904 [DEBUG] switch_core_state_machine.c:99 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard DESTROY 2013-01-31 16:20:12.402904 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/as1000 at XXX.XXX.XXX.10) State DESTROY going to sleep 2013-01-31 16:20:13.992899 [NOTICE] switch_channel.c:968 New Channel sofia/internal/as1000 at XXX.XXX.XXX.10 [0089bd0e-6bec-11e2-a409-954bab7d8c64] 2013-01-31 16:20:13.992899 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:13.992899 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:13.992899 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_NEW 2013-01-31 16:20:13.992899 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/as1000 at XXX.XXX.XXX.10) State NEW 2013-01-31 16:20:14.012895 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:14.012895 [DEBUG] sofia.c:1719 detaching session 0089bd0e-6bec-11e2-a409-954bab7d8c64 2013-01-31 16:20:14.162903 [DEBUG] sofia.c:1811 Re-attaching to session 0089bd0e-6bec-11e2-a409-954bab7d8c64 2013-01-31 16:20:14.162903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:14.162903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:14.172900 [WARNING] sofia_reg.c:2491 Can't find user [as1000 at XXX.XXX.XXX.10] You must define a domain called 'XXX.XXX.XXX.10' in your directory and add a user with the id="as1000" attribute and you must configure your device to use the proper domain in it's authentication credentials. 2013-01-31 16:20:14.182899 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:14.182899 [DEBUG] switch_channel.c:2994 (sofia/internal/as1000 at XXX.XXX.XXX.10) Callstate Change DOWN -> HANGUP 2013-01-31 16:20:14.182899 [NOTICE] sofia.c:1717 Hangup sofia/internal/as1000 at XXX.XXX.XXX.10 [CS_NEW] [CALL_REJECTED] 2013-01-31 16:20:14.182899 [DEBUG] switch_channel.c:3017 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [KILL] 2013-01-31 16:20:14.182899 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:14.202904 [DEBUG] sofia.c:1051 Channel is already hungup. 2013-01-31 16:20:14.202904 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_HANGUP 2013-01-31 16:20:14.202904 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/as1000 at XXX.XXX.XXX.10) State HANGUP 2013-01-31 16:20:14.202904 [DEBUG] mod_sofia.c:503 Channel sofia/internal/as1000 at XXX.XXX.XXX.10 hanging up, cause: CALL_REJECTED 2013-01-31 16:20:14.202904 [DEBUG] switch_core_state_machine.c:48 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard HANGUP, cause: CALL_REJECTED 2013-01-31 16:20:14.202904 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/as1000 at XXX.XXX.XXX.10) State HANGUP going to sleep 2013-01-31 16:20:14.202904 [DEBUG] switch_core_state_machine.c:446 (sofia/internal/as1000 at XXX.XXX.XXX.10) State Change CS_HANGUP -> CS_REPORTING 2013-01-31 16:20:14.202904 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:14.202904 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_REPORTING 2013-01-31 16:20:14.202904 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/as1000 at XXX.XXX.XXX.10) State REPORTING 2013-01-31 16:20:14.202904 [DEBUG] switch_core_state_machine.c:92 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard REPORTING, cause: CALL_REJECTED 2013-01-31 16:20:14.202904 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/as1000 at XXX.XXX.XXX.10) State REPORTING going to sleep 2013-01-31 16:20:14.202904 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/as1000 at XXX.XXX.XXX.10) State Change CS_REPORTING -> CS_DESTROY 2013-01-31 16:20:14.202904 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:14.202904 [DEBUG] switch_core_session.c:1499 Session 58 (sofia/internal/as1000 at XXX.XXX.XXX.10) Locked, Waiting on external entities 2013-01-31 16:20:14.202904 [NOTICE] switch_core_session.c:1517 Session 58 (sofia/internal/as1000 at XXX.XXX.XXX.10) Ended 2013-01-31 16:20:14.202904 [NOTICE] switch_core_session.c:1521 Close Channel sofia/internal/as1000 at XXX.XXX.XXX.10 [CS_DESTROY] 2013-01-31 16:20:14.202904 [DEBUG] switch_core_state_machine.c:556 (sofia/internal/as1000 at XXX.XXX.XXX.10) Callstate Change HANGUP -> DOWN 2013-01-31 16:20:14.202904 [DEBUG] switch_core_state_machine.c:559 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_DESTROY 2013-01-31 16:20:14.202904 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/as1000 at XXX.XXX.XXX.10) State DESTROY 2013-01-31 16:20:14.202904 [DEBUG] mod_sofia.c:396 sofia/internal/as1000 at XXX.XXX.XXX.10 SOFIA DESTROY 2013-01-31 16:20:14.202904 [DEBUG] switch_core_state_machine.c:99 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard DESTROY 2013-01-31 16:20:14.202904 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/as1000 at XXX.XXX.XXX.10) State DESTROY going to sleep 2013-01-31 16:20:16.032903 [NOTICE] switch_channel.c:968 New Channel sofia/internal/as1000 at XXX.XXX.XXX.10 [01c0cca8-6bec-11e2-a40b-954bab7d8c64] 2013-01-31 16:20:16.032903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:16.032903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:16.032903 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_NEW 2013-01-31 16:20:16.032903 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/as1000 at XXX.XXX.XXX.10) State NEW 2013-01-31 16:20:16.052903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:16.052903 [DEBUG] sofia.c:1719 detaching session 01c0cca8-6bec-11e2-a40b-954bab7d8c64 2013-01-31 16:20:16.452903 [DEBUG] sofia.c:1811 Re-attaching to session 01c0cca8-6bec-11e2-a40b-954bab7d8c64 2013-01-31 16:20:16.452903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:16.452903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:16.452903 [WARNING] sofia_reg.c:2491 Can't find user [as1000 at XXX.XXX.XXX.10] You must define a domain called 'XXX.XXX.XXX.10' in your directory and add a user with the id="as1000" attribute and you must configure your device to use the proper domain in it's authentication credentials. 2013-01-31 16:20:16.452903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:16.452903 [DEBUG] switch_channel.c:2994 (sofia/internal/as1000 at XXX.XXX.XXX.10) Callstate Change DOWN -> HANGUP 2013-01-31 16:20:16.452903 [NOTICE] sofia.c:1717 Hangup sofia/internal/as1000 at XXX.XXX.XXX.10 [CS_NEW] [CALL_REJECTED] 2013-01-31 16:20:16.452903 [DEBUG] switch_channel.c:3017 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [KILL] 2013-01-31 16:20:16.452903 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:16.472903 [DEBUG] sofia.c:1051 Channel is already hungup. 2013-01-31 16:20:16.472903 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_HANGUP 2013-01-31 16:20:16.472903 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/as1000 at XXX.XXX.XXX.10) State HANGUP 2013-01-31 16:20:16.472903 [DEBUG] mod_sofia.c:503 Channel sofia/internal/as1000 at XXX.XXX.XXX.10 hanging up, cause: CALL_REJECTED 2013-01-31 16:20:16.472903 [DEBUG] switch_core_state_machine.c:48 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard HANGUP, cause: CALL_REJECTED 2013-01-31 16:20:16.472903 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/as1000 at XXX.XXX.XXX.10) State HANGUP going to sleep 2013-01-31 16:20:16.472903 [DEBUG] switch_core_state_machine.c:446 (sofia/internal/as1000 at XXX.XXX.XXX.10) State Change CS_HANGUP -> CS_REPORTING 2013-01-31 16:20:16.472903 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:16.472903 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_REPORTING 2013-01-31 16:20:16.472903 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/as1000 at XXX.XXX.XXX.10) State REPORTING 2013-01-31 16:20:16.472903 [DEBUG] switch_core_state_machine.c:92 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard REPORTING, cause: CALL_REJECTED 2013-01-31 16:20:16.472903 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/as1000 at XXX.XXX.XXX.10) State REPORTING going to sleep 2013-01-31 16:20:16.472903 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/as1000 at XXX.XXX.XXX.10) State Change CS_REPORTING -> CS_DESTROY 2013-01-31 16:20:16.472903 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:16.472903 [DEBUG] switch_core_session.c:1499 Session 59 (sofia/internal/as1000 at XXX.XXX.XXX.10) Locked, Waiting on external entities 2013-01-31 16:20:16.472903 [NOTICE] switch_core_session.c:1517 Session 59 (sofia/internal/as1000 at XXX.XXX.XXX.10) Ended 2013-01-31 16:20:16.472903 [NOTICE] switch_core_session.c:1521 Close Channel sofia/internal/as1000 at XXX.XXX.XXX.10 [CS_DESTROY] 2013-01-31 16:20:16.472903 [DEBUG] switch_core_state_machine.c:556 (sofia/internal/as1000 at XXX.XXX.XXX.10) Callstate Change HANGUP -> DOWN 2013-01-31 16:20:16.472903 [DEBUG] switch_core_state_machine.c:559 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_DESTROY 2013-01-31 16:20:16.472903 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/as1000 at XXX.XXX.XXX.10) State DESTROY 2013-01-31 16:20:16.472903 [DEBUG] mod_sofia.c:396 sofia/internal/as1000 at XXX.XXX.XXX.10 SOFIA DESTROY 2013-01-31 16:20:16.472903 [DEBUG] switch_core_state_machine.c:99 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard DESTROY 2013-01-31 16:20:16.472903 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/as1000 at XXX.XXX.XXX.10) State DESTROY going to sleep 2013-01-31 16:20:19.292898 [NOTICE] switch_channel.c:968 New Channel sofia/internal/as1000 at XXX.XXX.XXX.10 [03b1b950-6bec-11e2-a40d-954bab7d8c64] 2013-01-31 16:20:19.292898 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:19.292898 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:19.292898 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_NEW 2013-01-31 16:20:19.292898 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/as1000 at XXX.XXX.XXX.10) State NEW 2013-01-31 16:20:19.312893 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:19.312893 [DEBUG] sofia.c:1719 detaching session 03b1b950-6bec-11e2-a40d-954bab7d8c64 2013-01-31 16:20:19.922902 [DEBUG] sofia.c:1811 Re-attaching to session 03b1b950-6bec-11e2-a40d-954bab7d8c64 2013-01-31 16:20:19.922902 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:19.922902 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:19.932895 [WARNING] sofia_reg.c:2491 Can't find user [as1000 at XXX.XXX.XXX.10] You must define a domain called 'XXX.XXX.XXX.10' in your directory and add a user with the id="as1000" attribute and you must configure your device to use the proper domain in it's authentication credentials. 2013-01-31 16:20:19.932895 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:19.932895 [DEBUG] switch_channel.c:2994 (sofia/internal/as1000 at XXX.XXX.XXX.10) Callstate Change DOWN -> HANGUP 2013-01-31 16:20:19.932895 [NOTICE] sofia.c:1717 Hangup sofia/internal/as1000 at XXX.XXX.XXX.10 [CS_NEW] [CALL_REJECTED] 2013-01-31 16:20:19.932895 [DEBUG] switch_channel.c:3017 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [KILL] 2013-01-31 16:20:19.932895 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:19.952904 [DEBUG] sofia.c:1051 Channel is already hungup. 2013-01-31 16:20:19.952904 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_HANGUP 2013-01-31 16:20:19.952904 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/as1000 at XXX.XXX.XXX.10) State HANGUP 2013-01-31 16:20:19.952904 [DEBUG] mod_sofia.c:503 Channel sofia/internal/as1000 at XXX.XXX.XXX.10 hanging up, cause: CALL_REJECTED 2013-01-31 16:20:19.952904 [DEBUG] switch_core_state_machine.c:48 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard HANGUP, cause: CALL_REJECTED 2013-01-31 16:20:19.952904 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/as1000 at XXX.XXX.XXX.10) State HANGUP going to sleep 2013-01-31 16:20:19.952904 [DEBUG] switch_core_state_machine.c:446 (sofia/internal/as1000 at XXX.XXX.XXX.10) State Change CS_HANGUP -> CS_REPORTING 2013-01-31 16:20:19.952904 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:19.952904 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_REPORTING 2013-01-31 16:20:19.952904 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/as1000 at XXX.XXX.XXX.10) State REPORTING 2013-01-31 16:20:19.952904 [DEBUG] switch_core_state_machine.c:92 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard REPORTING, cause: CALL_REJECTED 2013-01-31 16:20:19.952904 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/as1000 at XXX.XXX.XXX.10) State REPORTING going to sleep 2013-01-31 16:20:19.952904 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/as1000 at XXX.XXX.XXX.10) State Change CS_REPORTING -> CS_DESTROY 2013-01-31 16:20:19.952904 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:19.952904 [DEBUG] switch_core_session.c:1499 Session 60 (sofia/internal/as1000 at XXX.XXX.XXX.10) Locked, Waiting on external entities 2013-01-31 16:20:19.952904 [NOTICE] switch_core_session.c:1517 Session 60 (sofia/internal/as1000 at XXX.XXX.XXX.10) Ended 2013-01-31 16:20:19.952904 [NOTICE] switch_core_session.c:1521 Close Channel sofia/internal/as1000 at XXX.XXX.XXX.10 [CS_DESTROY] 2013-01-31 16:20:19.952904 [DEBUG] switch_core_state_machine.c:556 (sofia/internal/as1000 at XXX.XXX.XXX.10) Callstate Change HANGUP -> DOWN 2013-01-31 16:20:19.952904 [DEBUG] switch_core_state_machine.c:559 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_DESTROY 2013-01-31 16:20:19.952904 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/as1000 at XXX.XXX.XXX.10) State DESTROY 2013-01-31 16:20:19.952904 [DEBUG] mod_sofia.c:396 sofia/internal/as1000 at XXX.XXX.XXX.10 SOFIA DESTROY 2013-01-31 16:20:19.952904 [DEBUG] switch_core_state_machine.c:99 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard DESTROY 2013-01-31 16:20:19.952904 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/as1000 at XXX.XXX.XXX.10) State DESTROY going to sleep 2013-01-31 16:20:20.122903 [NOTICE] switch_channel.c:968 New Channel sofia/internal/as1000 at XXX.XXX.XXX.10 [04300954-6bec-11e2-a40f-954bab7d8c64] 2013-01-31 16:20:20.122903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:20.122903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:20.122903 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_NEW 2013-01-31 16:20:20.122903 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/as1000 at XXX.XXX.XXX.10) State NEW 2013-01-31 16:20:20.142904 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:20.142904 [DEBUG] sofia.c:1719 detaching session 04300954-6bec-11e2-a40f-954bab7d8c64 2013-01-31 16:20:21.842895 [NOTICE] switch_channel.c:968 New Channel sofia/internal/as1000 at XXX.XXX.XXX.10 [05366b5e-6bec-11e2-a411-954bab7d8c64] 2013-01-31 16:20:21.842895 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:21.842895 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:21.842895 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_NEW 2013-01-31 16:20:21.842895 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/as1000 at XXX.XXX.XXX.10) State NEW 2013-01-31 16:20:21.862900 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:21.862900 [DEBUG] sofia.c:1719 detaching session 05366b5e-6bec-11e2-a411-954bab7d8c64 2013-01-31 16:20:22.092903 [DEBUG] sofia.c:1811 Re-attaching to session 05366b5e-6bec-11e2-a411-954bab7d8c64 2013-01-31 16:20:22.092903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:22.092903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:22.102902 [WARNING] sofia_reg.c:2491 Can't find user [as1000 at XXX.XXX.XXX.10] You must define a domain called 'XXX.XXX.XXX.10' in your directory and add a user with the id="as1000" attribute and you must configure your device to use the proper domain in it's authentication credentials. 2013-01-31 16:20:22.102902 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:22.102902 [DEBUG] switch_channel.c:2994 (sofia/internal/as1000 at XXX.XXX.XXX.10) Callstate Change DOWN -> HANGUP 2013-01-31 16:20:22.102902 [NOTICE] sofia.c:1717 Hangup sofia/internal/as1000 at XXX.XXX.XXX.10 [CS_NEW] [CALL_REJECTED] 2013-01-31 16:20:22.102902 [DEBUG] switch_channel.c:3017 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [KILL] 2013-01-31 16:20:22.102902 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:22.122895 [DEBUG] sofia.c:1051 Channel is already hungup. 2013-01-31 16:20:22.122895 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_HANGUP 2013-01-31 16:20:22.122895 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/as1000 at XXX.XXX.XXX.10) State HANGUP 2013-01-31 16:20:22.122895 [DEBUG] mod_sofia.c:503 Channel sofia/internal/as1000 at XXX.XXX.XXX.10 hanging up, cause: CALL_REJECTED 2013-01-31 16:20:22.122895 [DEBUG] switch_core_state_machine.c:48 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard HANGUP, cause: CALL_REJECTED 2013-01-31 16:20:22.122895 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/as1000 at XXX.XXX.XXX.10) State HANGUP going to sleep 2013-01-31 16:20:22.122895 [DEBUG] switch_core_state_machine.c:446 (sofia/internal/as1000 at XXX.XXX.XXX.10) State Change CS_HANGUP -> CS_REPORTING 2013-01-31 16:20:22.122895 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:22.122895 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_REPORTING 2013-01-31 16:20:22.122895 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/as1000 at XXX.XXX.XXX.10) State REPORTING 2013-01-31 16:20:22.122895 [DEBUG] switch_core_state_machine.c:92 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard REPORTING, cause: CALL_REJECTED 2013-01-31 16:20:22.122895 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/as1000 at XXX.XXX.XXX.10) State REPORTING going to sleep 2013-01-31 16:20:22.122895 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/as1000 at XXX.XXX.XXX.10) State Change CS_REPORTING -> CS_DESTROY 2013-01-31 16:20:22.122895 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:22.122895 [DEBUG] switch_core_session.c:1499 Session 62 (sofia/internal/as1000 at XXX.XXX.XXX.10) Locked, Waiting on external entities 2013-01-31 16:20:22.122895 [NOTICE] switch_core_session.c:1517 Session 62 (sofia/internal/as1000 at XXX.XXX.XXX.10) Ended 2013-01-31 16:20:22.122895 [NOTICE] switch_core_session.c:1521 Close Channel sofia/internal/as1000 at XXX.XXX.XXX.10 [CS_DESTROY] 2013-01-31 16:20:22.122895 [DEBUG] switch_core_state_machine.c:556 (sofia/internal/as1000 at XXX.XXX.XXX.10) Callstate Change HANGUP -> DOWN 2013-01-31 16:20:22.122895 [DEBUG] switch_core_state_machine.c:559 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_DESTROY 2013-01-31 16:20:22.122895 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/as1000 at XXX.XXX.XXX.10) State DESTROY 2013-01-31 16:20:22.122895 [DEBUG] mod_sofia.c:396 sofia/internal/as1000 at XXX.XXX.XXX.10 SOFIA DESTROY 2013-01-31 16:20:22.122895 [DEBUG] switch_core_state_machine.c:99 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard DESTROY 2013-01-31 16:20:22.122895 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/as1000 at XXX.XXX.XXX.10) State DESTROY going to sleep 2013-01-31 16:20:24.322903 [NOTICE] switch_channel.c:968 New Channel sofia/internal/as1000 at XXX.XXX.XXX.10 [06b1f318-6bec-11e2-a413-954bab7d8c64] 2013-01-31 16:20:24.322903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:24.322903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:24.322903 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_NEW 2013-01-31 16:20:24.322903 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/as1000 at XXX.XXX.XXX.10) State NEW 2013-01-31 16:20:24.342900 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:24.342900 [DEBUG] sofia.c:1719 detaching session 06b1f318-6bec-11e2-a413-954bab7d8c64 2013-01-31 16:20:24.432894 [DEBUG] sofia.c:1811 Re-attaching to session 06b1f318-6bec-11e2-a413-954bab7d8c64 2013-01-31 16:20:24.432894 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:24.432894 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:24.442894 [WARNING] sofia_reg.c:2491 Can't find user [as1000 at XXX.XXX.XXX.10] You must define a domain called 'XXX.XXX.XXX.10' in your directory and add a user with the id="as1000" attribute and you must configure your device to use the proper domain in it's authentication credentials. 2013-01-31 16:20:24.452892 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:24.452892 [DEBUG] switch_channel.c:2994 (sofia/internal/as1000 at XXX.XXX.XXX.10) Callstate Change DOWN -> HANGUP 2013-01-31 16:20:24.452892 [NOTICE] sofia.c:1717 Hangup sofia/internal/as1000 at XXX.XXX.XXX.10 [CS_NEW] [CALL_REJECTED] 2013-01-31 16:20:24.452892 [DEBUG] switch_channel.c:3017 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [KILL] 2013-01-31 16:20:24.452892 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:24.462892 [DEBUG] sofia.c:1051 Channel is already hungup. 2013-01-31 16:20:24.462892 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_HANGUP 2013-01-31 16:20:24.462892 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/as1000 at XXX.XXX.XXX.10) State HANGUP 2013-01-31 16:20:24.462892 [DEBUG] mod_sofia.c:503 Channel sofia/internal/as1000 at XXX.XXX.XXX.10 hanging up, cause: CALL_REJECTED 2013-01-31 16:20:24.472896 [DEBUG] switch_core_state_machine.c:48 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard HANGUP, cause: CALL_REJECTED 2013-01-31 16:20:24.472896 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/as1000 at XXX.XXX.XXX.10) State HANGUP going to sleep 2013-01-31 16:20:24.472896 [DEBUG] switch_core_state_machine.c:446 (sofia/internal/as1000 at XXX.XXX.XXX.10) State Change CS_HANGUP -> CS_REPORTING 2013-01-31 16:20:24.472896 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:24.472896 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_REPORTING 2013-01-31 16:20:24.472896 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/as1000 at XXX.XXX.XXX.10) State REPORTING 2013-01-31 16:20:24.472896 [DEBUG] switch_core_state_machine.c:92 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard REPORTING, cause: CALL_REJECTED 2013-01-31 16:20:24.472896 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/as1000 at XXX.XXX.XXX.10) State REPORTING going to sleep 2013-01-31 16:20:24.472896 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/as1000 at XXX.XXX.XXX.10) State Change CS_REPORTING -> CS_DESTROY 2013-01-31 16:20:24.472896 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:24.472896 [DEBUG] switch_core_session.c:1499 Session 63 (sofia/internal/as1000 at XXX.XXX.XXX.10) Locked, Waiting on external entities 2013-01-31 16:20:24.472896 [NOTICE] switch_core_session.c:1517 Session 63 (sofia/internal/as1000 at XXX.XXX.XXX.10) Ended 2013-01-31 16:20:24.472896 [NOTICE] switch_core_session.c:1521 Close Channel sofia/internal/as1000 at XXX.XXX.XXX.10 [CS_DESTROY] 2013-01-31 16:20:24.472896 [DEBUG] switch_core_state_machine.c:556 (sofia/internal/as1000 at XXX.XXX.XXX.10) Callstate Change HANGUP -> DOWN 2013-01-31 16:20:24.472896 [DEBUG] switch_core_state_machine.c:559 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_DESTROY 2013-01-31 16:20:24.472896 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/as1000 at XXX.XXX.XXX.10) State DESTROY 2013-01-31 16:20:24.472896 [DEBUG] mod_sofia.c:396 sofia/internal/as1000 at XXX.XXX.XXX.10 SOFIA DESTROY 2013-01-31 16:20:24.472896 [DEBUG] switch_core_state_machine.c:99 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard DESTROY 2013-01-31 16:20:24.472896 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/as1000 at XXX.XXX.XXX.10) State DESTROY going to sleep 2013-01-31 16:20:25.242903 [NOTICE] switch_channel.c:968 New Channel sofia/internal/as1000 at XXX.XXX.XXX.10 [073e50b0-6bec-11e2-a415-954bab7d8c64] 2013-01-31 16:20:25.242903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:25.242903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:25.242903 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_NEW 2013-01-31 16:20:25.242903 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/as1000 at XXX.XXX.XXX.10) State NEW 2013-01-31 16:20:25.262904 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:25.262904 [DEBUG] sofia.c:1719 detaching session 073e50b0-6bec-11e2-a415-954bab7d8c64 2013-01-31 16:20:25.522903 [DEBUG] sofia.c:1811 Re-attaching to session 073e50b0-6bec-11e2-a415-954bab7d8c64 2013-01-31 16:20:25.522903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:25.522903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:25.522903 [WARNING] sofia_reg.c:2491 Can't find user [as1000 at XXX.XXX.XXX.10] You must define a domain called 'XXX.XXX.XXX.10' in your directory and add a user with the id="as1000" attribute and you must configure your device to use the proper domain in it's authentication credentials. 2013-01-31 16:20:25.532899 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:25.532899 [DEBUG] switch_channel.c:2994 (sofia/internal/as1000 at XXX.XXX.XXX.10) Callstate Change DOWN -> HANGUP 2013-01-31 16:20:25.532899 [NOTICE] sofia.c:1717 Hangup sofia/internal/as1000 at XXX.XXX.XXX.10 [CS_NEW] [CALL_REJECTED] 2013-01-31 16:20:25.532899 [DEBUG] switch_channel.c:3017 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [KILL] 2013-01-31 16:20:25.532899 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:25.552903 [DEBUG] sofia.c:1051 Channel is already hungup. 2013-01-31 16:20:25.552903 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_HANGUP 2013-01-31 16:20:25.552903 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/as1000 at XXX.XXX.XXX.10) State HANGUP 2013-01-31 16:20:25.552903 [DEBUG] mod_sofia.c:503 Channel sofia/internal/as1000 at XXX.XXX.XXX.10 hanging up, cause: CALL_REJECTED 2013-01-31 16:20:25.552903 [DEBUG] switch_core_state_machine.c:48 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard HANGUP, cause: CALL_REJECTED 2013-01-31 16:20:25.552903 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/as1000 at XXX.XXX.XXX.10) State HANGUP going to sleep 2013-01-31 16:20:25.552903 [DEBUG] switch_core_state_machine.c:446 (sofia/internal/as1000 at XXX.XXX.XXX.10) State Change CS_HANGUP -> CS_REPORTING 2013-01-31 16:20:25.552903 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:25.552903 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_REPORTING 2013-01-31 16:20:25.552903 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/as1000 at XXX.XXX.XXX.10) State REPORTING 2013-01-31 16:20:25.552903 [DEBUG] switch_core_state_machine.c:92 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard REPORTING, cause: CALL_REJECTED 2013-01-31 16:20:25.552903 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/as1000 at XXX.XXX.XXX.10) State REPORTING going to sleep 2013-01-31 16:20:25.552903 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/as1000 at XXX.XXX.XXX.10) State Change CS_REPORTING -> CS_DESTROY 2013-01-31 16:20:25.552903 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:25.552903 [DEBUG] switch_core_session.c:1499 Session 64 (sofia/internal/as1000 at XXX.XXX.XXX.10) Locked, Waiting on external entities 2013-01-31 16:20:25.552903 [NOTICE] switch_core_session.c:1517 Session 64 (sofia/internal/as1000 at XXX.XXX.XXX.10) Ended 2013-01-31 16:20:25.552903 [NOTICE] switch_core_session.c:1521 Close Channel sofia/internal/as1000 at XXX.XXX.XXX.10 [CS_DESTROY] 2013-01-31 16:20:25.552903 [DEBUG] switch_core_state_machine.c:556 (sofia/internal/as1000 at XXX.XXX.XXX.10) Callstate Change HANGUP -> DOWN 2013-01-31 16:20:25.552903 [DEBUG] switch_core_state_machine.c:559 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_DESTROY 2013-01-31 16:20:25.552903 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/as1000 at XXX.XXX.XXX.10) State DESTROY 2013-01-31 16:20:25.552903 [DEBUG] mod_sofia.c:396 sofia/internal/as1000 at XXX.XXX.XXX.10 SOFIA DESTROY 2013-01-31 16:20:25.552903 [DEBUG] switch_core_state_machine.c:99 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard DESTROY 2013-01-31 16:20:25.552903 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/as1000 at XXX.XXX.XXX.10) State DESTROY going to sleep 2013-01-31 16:20:27.442904 [NOTICE] switch_channel.c:968 New Channel sofia/internal/as1000 at XXX.XXX.XXX.10 [088d829c-6bec-11e2-a417-954bab7d8c64] 2013-01-31 16:20:27.442904 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:27.442904 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:27.442904 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_NEW 2013-01-31 16:20:27.442904 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/as1000 at XXX.XXX.XXX.10) State NEW 2013-01-31 16:20:27.462900 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:27.462900 [DEBUG] sofia.c:1719 detaching session 088d829c-6bec-11e2-a417-954bab7d8c64 2013-01-31 16:20:27.732903 [DEBUG] sofia.c:1811 Re-attaching to session 088d829c-6bec-11e2-a417-954bab7d8c64 2013-01-31 16:20:27.732903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:27.732903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:27.742898 [WARNING] sofia_reg.c:2491 Can't find user [as1000 at XXX.XXX.XXX.10] You must define a domain called 'XXX.XXX.XXX.10' in your directory and add a user with the id="as1000" attribute and you must configure your device to use the proper domain in it's authentication credentials. 2013-01-31 16:20:27.742898 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:27.742898 [DEBUG] switch_channel.c:2994 (sofia/internal/as1000 at XXX.XXX.XXX.10) Callstate Change DOWN -> HANGUP 2013-01-31 16:20:27.742898 [NOTICE] sofia.c:1717 Hangup sofia/internal/as1000 at XXX.XXX.XXX.10 [CS_NEW] [CALL_REJECTED] 2013-01-31 16:20:27.742898 [DEBUG] switch_channel.c:3017 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [KILL] 2013-01-31 16:20:27.742898 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:27.762904 [DEBUG] sofia.c:1051 Channel is already hungup. 2013-01-31 16:20:27.762904 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_HANGUP 2013-01-31 16:20:27.762904 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/as1000 at XXX.XXX.XXX.10) State HANGUP 2013-01-31 16:20:27.762904 [DEBUG] mod_sofia.c:503 Channel sofia/internal/as1000 at XXX.XXX.XXX.10 hanging up, cause: CALL_REJECTED 2013-01-31 16:20:27.762904 [DEBUG] switch_core_state_machine.c:48 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard HANGUP, cause: CALL_REJECTED 2013-01-31 16:20:27.762904 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/as1000 at XXX.XXX.XXX.10) State HANGUP going to sleep 2013-01-31 16:20:27.762904 [DEBUG] switch_core_state_machine.c:446 (sofia/internal/as1000 at XXX.XXX.XXX.10) State Change CS_HANGUP -> CS_REPORTING 2013-01-31 16:20:27.762904 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:27.762904 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_REPORTING 2013-01-31 16:20:27.762904 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/as1000 at XXX.XXX.XXX.10) State REPORTING 2013-01-31 16:20:27.762904 [DEBUG] switch_core_state_machine.c:92 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard REPORTING, cause: CALL_REJECTED 2013-01-31 16:20:27.762904 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/as1000 at XXX.XXX.XXX.10) State REPORTING going to sleep 2013-01-31 16:20:27.762904 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/as1000 at XXX.XXX.XXX.10) State Change CS_REPORTING -> CS_DESTROY 2013-01-31 16:20:27.762904 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:27.762904 [DEBUG] switch_core_session.c:1499 Session 65 (sofia/internal/as1000 at XXX.XXX.XXX.10) Locked, Waiting on external entities 2013-01-31 16:20:27.762904 [NOTICE] switch_core_session.c:1517 Session 65 (sofia/internal/as1000 at XXX.XXX.XXX.10) Ended 2013-01-31 16:20:27.762904 [NOTICE] switch_core_session.c:1521 Close Channel sofia/internal/as1000 at XXX.XXX.XXX.10 [CS_DESTROY] 2013-01-31 16:20:27.762904 [DEBUG] switch_core_state_machine.c:556 (sofia/internal/as1000 at XXX.XXX.XXX.10) Callstate Change HANGUP -> DOWN 2013-01-31 16:20:27.762904 [DEBUG] switch_core_state_machine.c:559 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_DESTROY 2013-01-31 16:20:27.762904 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/as1000 at XXX.XXX.XXX.10) State DESTROY 2013-01-31 16:20:27.762904 [DEBUG] mod_sofia.c:396 sofia/internal/as1000 at XXX.XXX.XXX.10 SOFIA DESTROY 2013-01-31 16:20:27.762904 [DEBUG] switch_core_state_machine.c:99 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard DESTROY 2013-01-31 16:20:27.762904 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/as1000 at XXX.XXX.XXX.10) State DESTROY going to sleep 2013-01-31 16:20:29.142904 [NOTICE] switch_channel.c:968 New Channel sofia/internal/as1000 at XXX.XXX.XXX.10 [0990ed5a-6bec-11e2-a419-954bab7d8c64] 2013-01-31 16:20:29.142904 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:29.142904 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:29.142904 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_NEW 2013-01-31 16:20:29.142904 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/as1000 at XXX.XXX.XXX.10) State NEW 2013-01-31 16:20:29.162903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:29.162903 [DEBUG] sofia.c:1719 detaching session 0990ed5a-6bec-11e2-a419-954bab7d8c64 2013-01-31 16:20:29.452903 [DEBUG] sofia.c:1811 Re-attaching to session 0990ed5a-6bec-11e2-a419-954bab7d8c64 2013-01-31 16:20:29.452903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:29.452903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:29.462903 [WARNING] sofia_reg.c:2491 Can't find user [as1000 at XXX.XXX.XXX.10] You must define a domain called 'XXX.XXX.XXX.10' in your directory and add a user with the id="as1000" attribute and you must configure your device to use the proper domain in it's authentication credentials. 2013-01-31 16:20:29.462903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:29.462903 [DEBUG] switch_channel.c:2994 (sofia/internal/as1000 at XXX.XXX.XXX.10) Callstate Change DOWN -> HANGUP 2013-01-31 16:20:29.462903 [NOTICE] sofia.c:1717 Hangup sofia/internal/as1000 at XXX.XXX.XXX.10 [CS_NEW] [CALL_REJECTED] 2013-01-31 16:20:29.462903 [DEBUG] switch_channel.c:3017 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [KILL] 2013-01-31 16:20:29.462903 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:29.482903 [DEBUG] sofia.c:1051 Channel is already hungup. 2013-01-31 16:20:29.482903 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_HANGUP 2013-01-31 16:20:29.482903 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/as1000 at XXX.XXX.XXX.10) State HANGUP 2013-01-31 16:20:29.482903 [DEBUG] mod_sofia.c:503 Channel sofia/internal/as1000 at XXX.XXX.XXX.10 hanging up, cause: CALL_REJECTED 2013-01-31 16:20:29.482903 [DEBUG] switch_core_state_machine.c:48 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard HANGUP, cause: CALL_REJECTED 2013-01-31 16:20:29.482903 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/as1000 at XXX.XXX.XXX.10) State HANGUP going to sleep 2013-01-31 16:20:29.482903 [DEBUG] switch_core_state_machine.c:446 (sofia/internal/as1000 at XXX.XXX.XXX.10) State Change CS_HANGUP -> CS_REPORTING 2013-01-31 16:20:29.482903 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:29.482903 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_REPORTING 2013-01-31 16:20:29.482903 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/as1000 at XXX.XXX.XXX.10) State REPORTING 2013-01-31 16:20:29.482903 [DEBUG] switch_core_state_machine.c:92 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard REPORTING, cause: CALL_REJECTED 2013-01-31 16:20:29.482903 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/as1000 at XXX.XXX.XXX.10) State REPORTING going to sleep 2013-01-31 16:20:29.482903 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/as1000 at XXX.XXX.XXX.10) State Change CS_REPORTING -> CS_DESTROY 2013-01-31 16:20:29.482903 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:29.482903 [DEBUG] switch_core_session.c:1499 Session 66 (sofia/internal/as1000 at XXX.XXX.XXX.10) Locked, Waiting on external entities 2013-01-31 16:20:29.482903 [NOTICE] switch_core_session.c:1517 Session 66 (sofia/internal/as1000 at XXX.XXX.XXX.10) Ended 2013-01-31 16:20:29.482903 [NOTICE] switch_core_session.c:1521 Close Channel sofia/internal/as1000 at XXX.XXX.XXX.10 [CS_DESTROY] 2013-01-31 16:20:29.482903 [DEBUG] switch_core_state_machine.c:556 (sofia/internal/as1000 at XXX.XXX.XXX.10) Callstate Change HANGUP -> DOWN 2013-01-31 16:20:29.482903 [DEBUG] switch_core_state_machine.c:559 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_DESTROY 2013-01-31 16:20:29.482903 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/as1000 at XXX.XXX.XXX.10) State DESTROY 2013-01-31 16:20:29.482903 [DEBUG] mod_sofia.c:396 sofia/internal/as1000 at XXX.XXX.XXX.10 SOFIA DESTROY 2013-01-31 16:20:29.482903 [DEBUG] switch_core_state_machine.c:99 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard DESTROY 2013-01-31 16:20:29.482903 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/as1000 at XXX.XXX.XXX.10) State DESTROY going to sleep 2013-01-31 16:20:30.132903 [WARNING] switch_core_state_machine.c:514 04300954-6bec-11e2-a40f-954bab7d8c64 sofia/internal/as1000 at XXX.XXX.XXX.10 Abandoned 2013-01-31 16:20:30.132903 [DEBUG] switch_channel.c:2994 (sofia/internal/as1000 at XXX.XXX.XXX.10) Callstate Change DOWN -> HANGUP 2013-01-31 16:20:30.132903 [NOTICE] switch_core_state_machine.c:517 Hangup sofia/internal/as1000 at XXX.XXX.XXX.10 [CS_NEW] [WRONG_CALL_STATE] 2013-01-31 16:20:30.132903 [DEBUG] switch_channel.c:3017 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [KILL] 2013-01-31 16:20:30.132903 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:30.132903 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_HANGUP 2013-01-31 16:20:30.132903 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/as1000 at XXX.XXX.XXX.10) State HANGUP 2013-01-31 16:20:30.132903 [DEBUG] mod_sofia.c:503 Channel sofia/internal/as1000 at XXX.XXX.XXX.10 hanging up, cause: WRONG_CALL_STATE 2013-01-31 16:20:30.132903 [DEBUG] switch_core_state_machine.c:48 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard HANGUP, cause: WRONG_CALL_STATE 2013-01-31 16:20:30.132903 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/as1000 at XXX.XXX.XXX.10) State HANGUP going to sleep 2013-01-31 16:20:30.132903 [DEBUG] switch_core_state_machine.c:446 (sofia/internal/as1000 at XXX.XXX.XXX.10) State Change CS_HANGUP -> CS_REPORTING 2013-01-31 16:20:30.132903 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:30.132903 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_REPORTING 2013-01-31 16:20:30.132903 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/as1000 at XXX.XXX.XXX.10) State REPORTING 2013-01-31 16:20:30.132903 [DEBUG] switch_core_state_machine.c:92 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard REPORTING, cause: WRONG_CALL_STATE 2013-01-31 16:20:30.132903 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/as1000 at XXX.XXX.XXX.10) State REPORTING going to sleep 2013-01-31 16:20:30.132903 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/as1000 at XXX.XXX.XXX.10) State Change CS_REPORTING -> CS_DESTROY 2013-01-31 16:20:30.132903 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:30.132903 [DEBUG] switch_core_session.c:1499 Session 61 (sofia/internal/as1000 at XXX.XXX.XXX.10) Locked, Waiting on external entities 2013-01-31 16:20:30.132903 [NOTICE] switch_core_session.c:1517 Session 61 (sofia/internal/as1000 at XXX.XXX.XXX.10) Ended 2013-01-31 16:20:30.132903 [NOTICE] switch_core_session.c:1521 Close Channel sofia/internal/as1000 at XXX.XXX.XXX.10 [CS_DESTROY] 2013-01-31 16:20:30.132903 [DEBUG] switch_core_state_machine.c:556 (sofia/internal/as1000 at XXX.XXX.XXX.10) Callstate Change HANGUP -> DOWN 2013-01-31 16:20:30.132903 [DEBUG] switch_core_state_machine.c:559 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_DESTROY 2013-01-31 16:20:30.132903 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/as1000 at XXX.XXX.XXX.10) State DESTROY 2013-01-31 16:20:30.132903 [DEBUG] mod_sofia.c:396 sofia/internal/as1000 at XXX.XXX.XXX.10 SOFIA DESTROY 2013-01-31 16:20:30.132903 [DEBUG] switch_core_state_machine.c:99 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard DESTROY 2013-01-31 16:20:30.132903 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/as1000 at XXX.XXX.XXX.10) State DESTROY going to sleep 2013-01-31 16:20:31.142896 [NOTICE] switch_channel.c:968 New Channel sofia/internal/as1000 at XXX.XXX.XXX.10 [0ac18996-6bec-11e2-a41b-954bab7d8c64] 2013-01-31 16:20:31.142896 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:31.142896 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:31.142896 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_NEW 2013-01-31 16:20:31.142896 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/as1000 at XXX.XXX.XXX.10) State NEW 2013-01-31 16:20:31.162895 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:31.162895 [DEBUG] sofia.c:1719 detaching session 0ac18996-6bec-11e2-a41b-954bab7d8c64 2013-01-31 16:20:31.442904 [DEBUG] sofia.c:1811 Re-attaching to session 0ac18996-6bec-11e2-a41b-954bab7d8c64 2013-01-31 16:20:31.442904 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:31.442904 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:31.442904 [WARNING] sofia_reg.c:2491 Can't find user [as1000 at XXX.XXX.XXX.10] You must define a domain called 'XXX.XXX.XXX.10' in your directory and add a user with the id="as1000" attribute and you must configure your device to use the proper domain in it's authentication credentials. 2013-01-31 16:20:31.442904 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:31.442904 [DEBUG] switch_channel.c:2994 (sofia/internal/as1000 at XXX.XXX.XXX.10) Callstate Change DOWN -> HANGUP 2013-01-31 16:20:31.442904 [NOTICE] sofia.c:1717 Hangup sofia/internal/as1000 at XXX.XXX.XXX.10 [CS_NEW] [CALL_REJECTED] 2013-01-31 16:20:31.442904 [DEBUG] switch_channel.c:3017 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [KILL] 2013-01-31 16:20:31.442904 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:31.462900 [DEBUG] sofia.c:1051 Channel is already hungup. 2013-01-31 16:20:31.462900 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_HANGUP 2013-01-31 16:20:31.462900 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/as1000 at XXX.XXX.XXX.10) State HANGUP 2013-01-31 16:20:31.462900 [DEBUG] mod_sofia.c:503 Channel sofia/internal/as1000 at XXX.XXX.XXX.10 hanging up, cause: CALL_REJECTED 2013-01-31 16:20:31.462900 [DEBUG] switch_core_state_machine.c:48 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard HANGUP, cause: CALL_REJECTED 2013-01-31 16:20:31.462900 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/as1000 at XXX.XXX.XXX.10) State HANGUP going to sleep 2013-01-31 16:20:31.462900 [DEBUG] switch_core_state_machine.c:446 (sofia/internal/as1000 at XXX.XXX.XXX.10) State Change CS_HANGUP -> CS_REPORTING 2013-01-31 16:20:31.462900 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:31.462900 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_REPORTING 2013-01-31 16:20:31.462900 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/as1000 at XXX.XXX.XXX.10) State REPORTING 2013-01-31 16:20:31.462900 [DEBUG] switch_core_state_machine.c:92 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard REPORTING, cause: CALL_REJECTED 2013-01-31 16:20:31.462900 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/as1000 at XXX.XXX.XXX.10) State REPORTING going to sleep 2013-01-31 16:20:31.462900 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/as1000 at XXX.XXX.XXX.10) State Change CS_REPORTING -> CS_DESTROY 2013-01-31 16:20:31.462900 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/as1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:20:31.462900 [DEBUG] switch_core_session.c:1499 Session 67 (sofia/internal/as1000 at XXX.XXX.XXX.10) Locked, Waiting on external entities 2013-01-31 16:20:31.462900 [NOTICE] switch_core_session.c:1517 Session 67 (sofia/internal/as1000 at XXX.XXX.XXX.10) Ended 2013-01-31 16:20:31.462900 [NOTICE] switch_core_session.c:1521 Close Channel sofia/internal/as1000 at XXX.XXX.XXX.10 [CS_DESTROY] 2013-01-31 16:20:31.462900 [DEBUG] switch_core_state_machine.c:556 (sofia/internal/as1000 at XXX.XXX.XXX.10) Callstate Change HANGUP -> DOWN 2013-01-31 16:20:31.462900 [DEBUG] switch_core_state_machine.c:559 (sofia/internal/as1000 at XXX.XXX.XXX.10) Running State Change CS_DESTROY 2013-01-31 16:20:31.462900 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/as1000 at XXX.XXX.XXX.10) State DESTROY 2013-01-31 16:20:31.462900 [DEBUG] mod_sofia.c:396 sofia/internal/as1000 at XXX.XXX.XXX.10 SOFIA DESTROY 2013-01-31 16:20:31.462900 [DEBUG] switch_core_state_machine.c:99 sofia/internal/as1000 at XXX.XXX.XXX.10 Standard DESTROY 2013-01-31 16:20:31.462900 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/as1000 at XXX.XXX.XXX.10) State DESTROY going to sleep 2013-01-31 16:53:32.542903 [NOTICE] switch_channel.c:968 New Channel sofia/internal/1000 at XXX.XXX.XXX.10 [a7c4509e-6bf0-11e2-a41d-954bab7d8c64] 2013-01-31 16:53:32.542903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:53:32.542903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:53:32.542903 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/1000 at XXX.XXX.XXX.10) Running State Change CS_NEW 2013-01-31 16:53:32.542903 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/1000 at XXX.XXX.XXX.10) State NEW 2013-01-31 16:53:32.562903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:53:32.562903 [DEBUG] sofia.c:1719 detaching session a7c4509e-6bf0-11e2-a41d-954bab7d8c64 2013-01-31 16:53:32.892903 [DEBUG] sofia.c:1811 Re-attaching to session a7c4509e-6bf0-11e2-a41d-954bab7d8c64 2013-01-31 16:53:32.892903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:53:32.892903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:53:32.912903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:53:32.912903 [DEBUG] switch_channel.c:2994 (sofia/internal/1000 at XXX.XXX.XXX.10) Callstate Change DOWN -> HANGUP 2013-01-31 16:53:32.912903 [NOTICE] sofia.c:1717 Hangup sofia/internal/1000 at XXX.XXX.XXX.10 [CS_NEW] [CALL_REJECTED] 2013-01-31 16:53:32.912903 [DEBUG] switch_channel.c:3017 Send signal sofia/internal/1000 at XXX.XXX.XXX.10 [KILL] 2013-01-31 16:53:32.912903 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:53:32.932894 [DEBUG] sofia.c:1051 Channel is already hungup. 2013-01-31 16:53:32.932894 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/1000 at XXX.XXX.XXX.10) Running State Change CS_HANGUP 2013-01-31 16:53:32.932894 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/1000 at XXX.XXX.XXX.10) State HANGUP 2013-01-31 16:53:32.932894 [DEBUG] mod_sofia.c:503 Channel sofia/internal/1000 at XXX.XXX.XXX.10 hanging up, cause: CALL_REJECTED 2013-01-31 16:53:32.932894 [DEBUG] switch_core_state_machine.c:48 sofia/internal/1000 at XXX.XXX.XXX.10 Standard HANGUP, cause: CALL_REJECTED 2013-01-31 16:53:32.932894 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/1000 at XXX.XXX.XXX.10) State HANGUP going to sleep 2013-01-31 16:53:32.932894 [DEBUG] switch_core_state_machine.c:446 (sofia/internal/1000 at XXX.XXX.XXX.10) State Change CS_HANGUP -> CS_REPORTING 2013-01-31 16:53:32.932894 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:53:32.932894 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/1000 at XXX.XXX.XXX.10) Running State Change CS_REPORTING 2013-01-31 16:53:32.932894 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/1000 at XXX.XXX.XXX.10) State REPORTING 2013-01-31 16:53:32.932894 [DEBUG] switch_core_state_machine.c:92 sofia/internal/1000 at XXX.XXX.XXX.10 Standard REPORTING, cause: CALL_REJECTED 2013-01-31 16:53:32.932894 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/1000 at XXX.XXX.XXX.10) State REPORTING going to sleep 2013-01-31 16:53:32.932894 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/1000 at XXX.XXX.XXX.10) State Change CS_REPORTING -> CS_DESTROY 2013-01-31 16:53:32.932894 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:53:32.932894 [DEBUG] switch_core_session.c:1499 Session 68 (sofia/internal/1000 at XXX.XXX.XXX.10) Locked, Waiting on external entities 2013-01-31 16:53:32.932894 [NOTICE] switch_core_session.c:1517 Session 68 (sofia/internal/1000 at XXX.XXX.XXX.10) Ended 2013-01-31 16:53:32.932894 [NOTICE] switch_core_session.c:1521 Close Channel sofia/internal/1000 at XXX.XXX.XXX.10 [CS_DESTROY] 2013-01-31 16:53:32.932894 [DEBUG] switch_core_state_machine.c:556 (sofia/internal/1000 at XXX.XXX.XXX.10) Callstate Change HANGUP -> DOWN 2013-01-31 16:53:32.932894 [DEBUG] switch_core_state_machine.c:559 (sofia/internal/1000 at XXX.XXX.XXX.10) Running State Change CS_DESTROY 2013-01-31 16:53:32.932894 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/1000 at XXX.XXX.XXX.10) State DESTROY 2013-01-31 16:53:32.932894 [DEBUG] mod_sofia.c:396 sofia/internal/1000 at XXX.XXX.XXX.10 SOFIA DESTROY 2013-01-31 16:53:32.932894 [DEBUG] switch_core_state_machine.c:99 sofia/internal/1000 at XXX.XXX.XXX.10 Standard DESTROY 2013-01-31 16:53:32.932894 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/1000 at XXX.XXX.XXX.10) State DESTROY going to sleep 2013-01-31 16:53:33.382903 [NOTICE] switch_channel.c:968 New Channel sofia/internal/1000 at XXX.XXX.XXX.10 [a84474fe-6bf0-11e2-a41f-954bab7d8c64] 2013-01-31 16:53:33.382903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:53:33.382903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:53:33.382903 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/1000 at XXX.XXX.XXX.10) Running State Change CS_NEW 2013-01-31 16:53:33.382903 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/1000 at XXX.XXX.XXX.10) State NEW 2013-01-31 16:53:33.402894 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:53:33.402894 [DEBUG] sofia.c:1719 detaching session a84474fe-6bf0-11e2-a41f-954bab7d8c64 2013-01-31 16:53:33.762894 [DEBUG] sofia.c:1811 Re-attaching to session a84474fe-6bf0-11e2-a41f-954bab7d8c64 2013-01-31 16:53:33.762894 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:53:33.762894 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:53:33.772896 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:53:33.772896 [DEBUG] switch_channel.c:2994 (sofia/internal/1000 at XXX.XXX.XXX.10) Callstate Change DOWN -> HANGUP 2013-01-31 16:53:33.772896 [NOTICE] sofia.c:1717 Hangup sofia/internal/1000 at XXX.XXX.XXX.10 [CS_NEW] [CALL_REJECTED] 2013-01-31 16:53:33.772896 [DEBUG] switch_channel.c:3017 Send signal sofia/internal/1000 at XXX.XXX.XXX.10 [KILL] 2013-01-31 16:53:33.772896 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:53:33.792893 [DEBUG] sofia.c:1051 Channel is already hungup. 2013-01-31 16:53:33.792893 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/1000 at XXX.XXX.XXX.10) Running State Change CS_HANGUP 2013-01-31 16:53:33.792893 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/1000 at XXX.XXX.XXX.10) State HANGUP 2013-01-31 16:53:33.792893 [DEBUG] mod_sofia.c:503 Channel sofia/internal/1000 at XXX.XXX.XXX.10 hanging up, cause: CALL_REJECTED 2013-01-31 16:53:33.792893 [DEBUG] switch_core_state_machine.c:48 sofia/internal/1000 at XXX.XXX.XXX.10 Standard HANGUP, cause: CALL_REJECTED 2013-01-31 16:53:33.792893 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/1000 at XXX.XXX.XXX.10) State HANGUP going to sleep 2013-01-31 16:53:33.792893 [DEBUG] switch_core_state_machine.c:446 (sofia/internal/1000 at XXX.XXX.XXX.10) State Change CS_HANGUP -> CS_REPORTING 2013-01-31 16:53:33.792893 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:53:33.792893 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/1000 at XXX.XXX.XXX.10) Running State Change CS_REPORTING 2013-01-31 16:53:33.792893 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/1000 at XXX.XXX.XXX.10) State REPORTING 2013-01-31 16:53:33.792893 [DEBUG] switch_core_state_machine.c:92 sofia/internal/1000 at XXX.XXX.XXX.10 Standard REPORTING, cause: CALL_REJECTED 2013-01-31 16:53:33.792893 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/1000 at XXX.XXX.XXX.10) State REPORTING going to sleep 2013-01-31 16:53:33.792893 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/1000 at XXX.XXX.XXX.10) State Change CS_REPORTING -> CS_DESTROY 2013-01-31 16:53:33.792893 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:53:33.792893 [DEBUG] switch_core_session.c:1499 Session 69 (sofia/internal/1000 at XXX.XXX.XXX.10) Locked, Waiting on external entities 2013-01-31 16:53:33.792893 [NOTICE] switch_core_session.c:1517 Session 69 (sofia/internal/1000 at XXX.XXX.XXX.10) Ended 2013-01-31 16:53:33.792893 [NOTICE] switch_core_session.c:1521 Close Channel sofia/internal/1000 at XXX.XXX.XXX.10 [CS_DESTROY] 2013-01-31 16:53:33.792893 [DEBUG] switch_core_state_machine.c:556 (sofia/internal/1000 at XXX.XXX.XXX.10) Callstate Change HANGUP -> DOWN 2013-01-31 16:53:33.792893 [DEBUG] switch_core_state_machine.c:559 (sofia/internal/1000 at XXX.XXX.XXX.10) Running State Change CS_DESTROY 2013-01-31 16:53:33.792893 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/1000 at XXX.XXX.XXX.10) State DESTROY 2013-01-31 16:53:33.792893 [DEBUG] mod_sofia.c:396 sofia/internal/1000 at XXX.XXX.XXX.10 SOFIA DESTROY 2013-01-31 16:53:33.792893 [DEBUG] switch_core_state_machine.c:99 sofia/internal/1000 at XXX.XXX.XXX.10 Standard DESTROY 2013-01-31 16:53:33.792893 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/1000 at XXX.XXX.XXX.10) State DESTROY going to sleep 2013-01-31 16:53:34.262903 [NOTICE] switch_channel.c:968 New Channel sofia/internal/1000 at XXX.XXX.XXX.10 [a8cac324-6bf0-11e2-a421-954bab7d8c64] 2013-01-31 16:53:34.262903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:53:34.262903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:53:34.262903 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/1000 at XXX.XXX.XXX.10) Running State Change CS_NEW 2013-01-31 16:53:34.262903 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/1000 at XXX.XXX.XXX.10) State NEW 2013-01-31 16:53:34.282900 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:53:34.282900 [DEBUG] sofia.c:1719 detaching session a8cac324-6bf0-11e2-a421-954bab7d8c64 2013-01-31 16:53:34.612903 [DEBUG] sofia.c:1811 Re-attaching to session a8cac324-6bf0-11e2-a421-954bab7d8c64 2013-01-31 16:53:34.612903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:53:34.612903 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:53:34.632895 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:53:34.632895 [DEBUG] switch_channel.c:2994 (sofia/internal/1000 at XXX.XXX.XXX.10) Callstate Change DOWN -> HANGUP 2013-01-31 16:53:34.632895 [NOTICE] sofia.c:1717 Hangup sofia/internal/1000 at XXX.XXX.XXX.10 [CS_NEW] [CALL_REJECTED] 2013-01-31 16:53:34.632895 [DEBUG] switch_channel.c:3017 Send signal sofia/internal/1000 at XXX.XXX.XXX.10 [KILL] 2013-01-31 16:53:34.632895 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:53:34.652894 [DEBUG] sofia.c:1051 Channel is already hungup. 2013-01-31 16:53:34.652894 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/1000 at XXX.XXX.XXX.10) Running State Change CS_HANGUP 2013-01-31 16:53:34.652894 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/1000 at XXX.XXX.XXX.10) State HANGUP 2013-01-31 16:53:34.652894 [DEBUG] mod_sofia.c:503 Channel sofia/internal/1000 at XXX.XXX.XXX.10 hanging up, cause: CALL_REJECTED 2013-01-31 16:53:34.652894 [DEBUG] switch_core_state_machine.c:48 sofia/internal/1000 at XXX.XXX.XXX.10 Standard HANGUP, cause: CALL_REJECTED 2013-01-31 16:53:34.652894 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/1000 at XXX.XXX.XXX.10) State HANGUP going to sleep 2013-01-31 16:53:34.652894 [DEBUG] switch_core_state_machine.c:446 (sofia/internal/1000 at XXX.XXX.XXX.10) State Change CS_HANGUP -> CS_REPORTING 2013-01-31 16:53:34.652894 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:53:34.652894 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/1000 at XXX.XXX.XXX.10) Running State Change CS_REPORTING 2013-01-31 16:53:34.652894 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/1000 at XXX.XXX.XXX.10) State REPORTING 2013-01-31 16:53:34.652894 [DEBUG] switch_core_state_machine.c:92 sofia/internal/1000 at XXX.XXX.XXX.10 Standard REPORTING, cause: CALL_REJECTED 2013-01-31 16:53:34.652894 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/1000 at XXX.XXX.XXX.10) State REPORTING going to sleep 2013-01-31 16:53:34.652894 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/1000 at XXX.XXX.XXX.10) State Change CS_REPORTING -> CS_DESTROY 2013-01-31 16:53:34.652894 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:53:34.652894 [DEBUG] switch_core_session.c:1499 Session 70 (sofia/internal/1000 at XXX.XXX.XXX.10) Locked, Waiting on external entities 2013-01-31 16:53:34.652894 [NOTICE] switch_core_session.c:1517 Session 70 (sofia/internal/1000 at XXX.XXX.XXX.10) Ended 2013-01-31 16:53:34.652894 [NOTICE] switch_core_session.c:1521 Close Channel sofia/internal/1000 at XXX.XXX.XXX.10 [CS_DESTROY] 2013-01-31 16:53:34.652894 [DEBUG] switch_core_state_machine.c:556 (sofia/internal/1000 at XXX.XXX.XXX.10) Callstate Change HANGUP -> DOWN 2013-01-31 16:53:34.652894 [DEBUG] switch_core_state_machine.c:559 (sofia/internal/1000 at XXX.XXX.XXX.10) Running State Change CS_DESTROY 2013-01-31 16:53:34.652894 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/1000 at XXX.XXX.XXX.10) State DESTROY 2013-01-31 16:53:34.652894 [DEBUG] mod_sofia.c:396 sofia/internal/1000 at XXX.XXX.XXX.10 SOFIA DESTROY 2013-01-31 16:53:34.652894 [DEBUG] switch_core_state_machine.c:99 sofia/internal/1000 at XXX.XXX.XXX.10 Standard DESTROY 2013-01-31 16:53:34.652894 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/1000 at XXX.XXX.XXX.10) State DESTROY going to sleep 2013-01-31 16:53:35.102902 [NOTICE] switch_channel.c:968 New Channel sofia/internal/1000 at XXX.XXX.XXX.10 [a94b0200-6bf0-11e2-a423-954bab7d8c64] 2013-01-31 16:53:35.102902 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:53:35.102902 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:53:35.102902 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/1000 at XXX.XXX.XXX.10) Running State Change CS_NEW 2013-01-31 16:53:35.102902 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/1000 at XXX.XXX.XXX.10) State NEW 2013-01-31 16:53:35.132896 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:53:35.132896 [DEBUG] sofia.c:1719 detaching session a94b0200-6bf0-11e2-a423-954bab7d8c64 2013-01-31 16:53:35.662904 [NOTICE] switch_channel.c:968 New Channel sofia/internal/1000 at XXX.XXX.XXX.10 [a9a066aa-6bf0-11e2-a425-954bab7d8c64] 2013-01-31 16:53:35.662904 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:53:35.662904 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:53:35.662904 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/1000 at XXX.XXX.XXX.10) Running State Change CS_NEW 2013-01-31 16:53:35.662904 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/1000 at XXX.XXX.XXX.10) State NEW 2013-01-31 16:53:35.692893 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:53:35.692893 [DEBUG] sofia.c:1719 detaching session a9a066aa-6bf0-11e2-a425-954bab7d8c64 2013-01-31 16:53:36.052901 [DEBUG] sofia.c:1811 Re-attaching to session a9a066aa-6bf0-11e2-a425-954bab7d8c64 2013-01-31 16:53:36.052901 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:53:36.052901 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:53:36.052901 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:53:36.052901 [DEBUG] switch_channel.c:2994 (sofia/internal/1000 at XXX.XXX.XXX.10) Callstate Change DOWN -> HANGUP 2013-01-31 16:53:36.052901 [NOTICE] sofia.c:1717 Hangup sofia/internal/1000 at XXX.XXX.XXX.10 [CS_NEW] [CALL_REJECTED] 2013-01-31 16:53:36.052901 [DEBUG] switch_channel.c:3017 Send signal sofia/internal/1000 at XXX.XXX.XXX.10 [KILL] 2013-01-31 16:53:36.052901 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:53:36.072895 [DEBUG] sofia.c:1051 Channel is already hungup. 2013-01-31 16:53:36.072895 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/1000 at XXX.XXX.XXX.10) Running State Change CS_HANGUP 2013-01-31 16:53:36.072895 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/1000 at XXX.XXX.XXX.10) State HANGUP 2013-01-31 16:53:36.072895 [DEBUG] mod_sofia.c:503 Channel sofia/internal/1000 at XXX.XXX.XXX.10 hanging up, cause: CALL_REJECTED 2013-01-31 16:53:36.072895 [DEBUG] switch_core_state_machine.c:48 sofia/internal/1000 at XXX.XXX.XXX.10 Standard HANGUP, cause: CALL_REJECTED 2013-01-31 16:53:36.072895 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/1000 at XXX.XXX.XXX.10) State HANGUP going to sleep 2013-01-31 16:53:36.072895 [DEBUG] switch_core_state_machine.c:446 (sofia/internal/1000 at XXX.XXX.XXX.10) State Change CS_HANGUP -> CS_REPORTING 2013-01-31 16:53:36.072895 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:53:36.072895 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/1000 at XXX.XXX.XXX.10) Running State Change CS_REPORTING 2013-01-31 16:53:36.072895 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/1000 at XXX.XXX.XXX.10) State REPORTING 2013-01-31 16:53:36.072895 [DEBUG] switch_core_state_machine.c:92 sofia/internal/1000 at XXX.XXX.XXX.10 Standard REPORTING, cause: CALL_REJECTED 2013-01-31 16:53:36.072895 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/1000 at XXX.XXX.XXX.10) State REPORTING going to sleep 2013-01-31 16:53:36.072895 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/1000 at XXX.XXX.XXX.10) State Change CS_REPORTING -> CS_DESTROY 2013-01-31 16:53:36.072895 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:53:36.072895 [DEBUG] switch_core_session.c:1499 Session 72 (sofia/internal/1000 at XXX.XXX.XXX.10) Locked, Waiting on external entities 2013-01-31 16:53:36.072895 [NOTICE] switch_core_session.c:1517 Session 72 (sofia/internal/1000 at XXX.XXX.XXX.10) Ended 2013-01-31 16:53:36.072895 [NOTICE] switch_core_session.c:1521 Close Channel sofia/internal/1000 at XXX.XXX.XXX.10 [CS_DESTROY] 2013-01-31 16:53:36.072895 [DEBUG] switch_core_state_machine.c:556 (sofia/internal/1000 at XXX.XXX.XXX.10) Callstate Change HANGUP -> DOWN 2013-01-31 16:53:36.072895 [DEBUG] switch_core_state_machine.c:559 (sofia/internal/1000 at XXX.XXX.XXX.10) Running State Change CS_DESTROY 2013-01-31 16:53:36.072895 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/1000 at XXX.XXX.XXX.10) State DESTROY 2013-01-31 16:53:36.072895 [DEBUG] mod_sofia.c:396 sofia/internal/1000 at XXX.XXX.XXX.10 SOFIA DESTROY 2013-01-31 16:53:36.072895 [DEBUG] switch_core_state_machine.c:99 sofia/internal/1000 at XXX.XXX.XXX.10 Standard DESTROY 2013-01-31 16:53:36.072895 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/1000 at XXX.XXX.XXX.10) State DESTROY going to sleep 2013-01-31 16:53:45.122903 [WARNING] switch_core_state_machine.c:514 a94b0200-6bf0-11e2-a423-954bab7d8c64 sofia/internal/1000 at XXX.XXX.XXX.10 Abandoned 2013-01-31 16:53:45.122903 [DEBUG] switch_channel.c:2994 (sofia/internal/1000 at XXX.XXX.XXX.10) Callstate Change DOWN -> HANGUP 2013-01-31 16:53:45.122903 [NOTICE] switch_core_state_machine.c:517 Hangup sofia/internal/1000 at XXX.XXX.XXX.10 [CS_NEW] [WRONG_CALL_STATE] 2013-01-31 16:53:45.122903 [DEBUG] switch_channel.c:3017 Send signal sofia/internal/1000 at XXX.XXX.XXX.10 [KILL] 2013-01-31 16:53:45.122903 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:53:45.122903 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/1000 at XXX.XXX.XXX.10) Running State Change CS_HANGUP 2013-01-31 16:53:45.122903 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/1000 at XXX.XXX.XXX.10) State HANGUP 2013-01-31 16:53:45.122903 [DEBUG] mod_sofia.c:503 Channel sofia/internal/1000 at XXX.XXX.XXX.10 hanging up, cause: WRONG_CALL_STATE 2013-01-31 16:53:45.122903 [DEBUG] switch_core_state_machine.c:48 sofia/internal/1000 at XXX.XXX.XXX.10 Standard HANGUP, cause: WRONG_CALL_STATE 2013-01-31 16:53:45.122903 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/1000 at XXX.XXX.XXX.10) State HANGUP going to sleep 2013-01-31 16:53:45.122903 [DEBUG] switch_core_state_machine.c:446 (sofia/internal/1000 at XXX.XXX.XXX.10) State Change CS_HANGUP -> CS_REPORTING 2013-01-31 16:53:45.122903 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:53:45.122903 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/1000 at XXX.XXX.XXX.10) Running State Change CS_REPORTING 2013-01-31 16:53:45.122903 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/1000 at XXX.XXX.XXX.10) State REPORTING 2013-01-31 16:53:45.122903 [DEBUG] switch_core_state_machine.c:92 sofia/internal/1000 at XXX.XXX.XXX.10 Standard REPORTING, cause: WRONG_CALL_STATE 2013-01-31 16:53:45.122903 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/1000 at XXX.XXX.XXX.10) State REPORTING going to sleep 2013-01-31 16:53:45.122903 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/1000 at XXX.XXX.XXX.10) State Change CS_REPORTING -> CS_DESTROY 2013-01-31 16:53:45.122903 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/1000 at XXX.XXX.XXX.10 [BREAK] 2013-01-31 16:53:45.122903 [DEBUG] switch_core_session.c:1499 Session 71 (sofia/internal/1000 at XXX.XXX.XXX.10) Locked, Waiting on external entities 2013-01-31 16:53:45.122903 [NOTICE] switch_core_session.c:1517 Session 71 (sofia/internal/1000 at XXX.XXX.XXX.10) Ended 2013-01-31 16:53:45.122903 [NOTICE] switch_core_session.c:1521 Close Channel sofia/internal/1000 at XXX.XXX.XXX.10 [CS_DESTROY] 2013-01-31 16:53:45.122903 [DEBUG] switch_core_state_machine.c:556 (sofia/internal/1000 at XXX.XXX.XXX.10) Callstate Change HANGUP -> DOWN 2013-01-31 16:53:45.122903 [DEBUG] switch_core_state_machine.c:559 (sofia/internal/1000 at XXX.XXX.XXX.10) Running State Change CS_DESTROY 2013-01-31 16:53:45.122903 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/1000 at XXX.XXX.XXX.10) State DESTROY 2013-01-31 16:53:45.122903 [DEBUG] mod_sofia.c:396 sofia/internal/1000 at XXX.XXX.XXX.10 SOFIA DESTROY 2013-01-31 16:53:45.122903 [DEBUG] switch_core_state_machine.c:99 sofia/internal/1000 at XXX.XXX.XXX.10 Standard DESTROY 2013-01-31 16:53:45.122903 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/1000 at XXX.XXX.XXX.10) State DESTROY going to sleep 2013-02-01 08:14:14.722897 [INFO] switch_core.c:2395 Shutting down 2013-02-01 08:14:14.722897 [DEBUG] switch_loadable_module.c:649 Chat Thread Ended 2013-02-01 08:14:14.722897 [DEBUG] switch_loadable_module.c:649 Chat Thread Ended 2013-02-01 08:14:14.722897 [DEBUG] switch_loadable_module.c:1152 Write lock interface 'fr' to wait for existing references. 2013-02-01 08:14:14.722897 [NOTICE] switch_loadable_module.c:1160 Deleting Say interface 'fr' 2013-02-01 08:14:14.722897 [DEBUG] switch_loadable_module.c:1049 Write lock interface 'flite' to wait for existing references. 2013-02-01 08:14:14.722897 [NOTICE] switch_loadable_module.c:1058 Deleting Speech interface 'flite' 2013-02-01 08:14:14.722897 [DEBUG] switch_loadable_module.c:1020 Write lock interface 'mod_native_file' to wait for existing references. 2013-02-01 08:14:14.722897 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'H263' 2013-02-01 08:14:14.722897 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'AMR' 2013-02-01 08:14:14.722897 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'SPEEX' 2013-02-01 08:14:14.722897 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'G729' 2013-02-01 08:14:14.722897 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'G726-16' 2013-02-01 08:14:14.722897 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'AAL2-G726-16' 2013-02-01 08:14:14.722897 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'PCMA' 2013-02-01 08:14:14.722897 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'DVI4' 2013-02-01 08:14:14.722897 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'G726-24' 2013-02-01 08:14:14.722897 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'PCMU' 2013-02-01 08:14:14.722897 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'L16' 2013-02-01 08:14:14.722897 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'PROXY' 2013-02-01 08:14:14.722897 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'AAL2-G726-24' 2013-02-01 08:14:14.722897 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'AAL2-G726-32' 2013-02-01 08:14:14.722897 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'G726-32' 2013-02-01 08:14:14.722897 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'G722' 2013-02-01 08:14:14.722897 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'AAL2-G726-40' 2013-02-01 08:14:14.722897 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'G726-40' 2013-02-01 08:14:14.722897 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'GSM' 2013-02-01 08:14:14.722897 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'H263-1998' 2013-02-01 08:14:14.722897 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'H263-2000' 2013-02-01 08:14:14.722897 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'H264' 2013-02-01 08:14:14.722897 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'H261' 2013-02-01 08:14:14.722897 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'G723' 2013-02-01 08:14:14.722897 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'LPC' 2013-02-01 08:14:14.722897 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'PROXY-VID' 2013-02-01 08:14:14.722897 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'VP8' 2013-02-01 08:14:14.722897 [NOTICE] switch_loadable_module.c:936 Deleting Application 'fifo' 2013-02-01 08:14:14.722897 [DEBUG] switch_loadable_module.c:938 Write lock interface 'fifo' to wait for existing references. 2013-02-01 08:14:14.722897 [NOTICE] switch_loadable_module.c:936 Deleting Application 'fifo_track_call' 2013-02-01 08:14:14.722897 [DEBUG] switch_loadable_module.c:938 Write lock interface 'fifo_track_call' to wait for existing references. 2013-02-01 08:14:14.722897 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'fifo' 2013-02-01 08:14:14.722897 [DEBUG] switch_loadable_module.c:991 Write lock interface 'fifo' to wait for existing references. 2013-02-01 08:14:14.722897 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'fifo_member' 2013-02-01 08:14:14.722897 [DEBUG] switch_loadable_module.c:991 Write lock interface 'fifo_member' to wait for existing references. 2013-02-01 08:14:14.722897 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'fifo_add_outbound' 2013-02-01 08:14:14.722897 [DEBUG] switch_loadable_module.c:991 Write lock interface 'fifo_add_outbound' to wait for existing references. 2013-02-01 08:14:14.722897 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'fifo_check_bridge' 2013-02-01 08:14:14.722897 [DEBUG] switch_loadable_module.c:991 Write lock interface 'fifo_check_bridge' to wait for existing references. 2013-02-01 08:14:14.722897 [INFO] switch_core_sqldb.c:1370 fifo Destroying SQL queue. 2013-02-01 08:14:14.722897 [INFO] switch_core_sqldb.c:1315 fifo Stopping SQL thread. 2013-02-01 08:14:14.922905 [NOTICE] switch_event.c:2010 Event Binding deleted for mod_fifo:PRESENCE_PROBE 2013-02-01 08:14:14.922905 [NOTICE] switch_event.c:403 Subclass reservation deleted for /usr/src/freeswitch/src/mod/applications/mod_fifo/mod_fifo.c:fifo::info 2013-02-01 08:14:15.922904 [NOTICE] switch_loadable_module.c:936 Deleting Application 'db' 2013-02-01 08:14:15.922904 [DEBUG] switch_loadable_module.c:938 Write lock interface 'db' to wait for existing references. 2013-02-01 08:14:15.922904 [NOTICE] switch_loadable_module.c:936 Deleting Application 'group' 2013-02-01 08:14:15.922904 [DEBUG] switch_loadable_module.c:938 Write lock interface 'group' to wait for existing references. 2013-02-01 08:14:15.922904 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'db' 2013-02-01 08:14:15.922904 [DEBUG] switch_loadable_module.c:991 Write lock interface 'db' to wait for existing references. 2013-02-01 08:14:15.922904 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'group' 2013-02-01 08:14:15.922904 [DEBUG] switch_loadable_module.c:991 Write lock interface 'group' to wait for existing references. 2013-02-01 08:14:15.922904 [NOTICE] switch_loadable_module.c:1193 Deleting Limit interface 'db' 2013-02-01 08:14:15.922904 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'expr' 2013-02-01 08:14:15.922904 [DEBUG] switch_loadable_module.c:991 Write lock interface 'expr' to wait for existing references. 2013-02-01 08:14:15.922904 [NOTICE] switch_loadable_module.c:936 Deleting Application 'voicemail' 2013-02-01 08:14:15.922904 [DEBUG] switch_loadable_module.c:938 Write lock interface 'voicemail' to wait for existing references. 2013-02-01 08:14:15.922904 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'voicemail' 2013-02-01 08:14:15.922904 [DEBUG] switch_loadable_module.c:991 Write lock interface 'voicemail' to wait for existing references. 2013-02-01 08:14:15.922904 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'voicemail_inject' 2013-02-01 08:14:15.922904 [DEBUG] switch_loadable_module.c:991 Write lock interface 'voicemail_inject' to wait for existing references. 2013-02-01 08:14:15.922904 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'vm_inject' 2013-02-01 08:14:15.922904 [DEBUG] switch_loadable_module.c:991 Write lock interface 'vm_inject' to wait for existing references. 2013-02-01 08:14:15.922904 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'vm_boxcount' 2013-02-01 08:14:15.922904 [DEBUG] switch_loadable_module.c:991 Write lock interface 'vm_boxcount' to wait for existing references. 2013-02-01 08:14:15.922904 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'vm_prefs' 2013-02-01 08:14:15.922904 [DEBUG] switch_loadable_module.c:991 Write lock interface 'vm_prefs' to wait for existing references. 2013-02-01 08:14:15.922904 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'vm_delete' 2013-02-01 08:14:15.922904 [DEBUG] switch_loadable_module.c:991 Write lock interface 'vm_delete' to wait for existing references. 2013-02-01 08:14:15.922904 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'vm_read' 2013-02-01 08:14:15.922904 [DEBUG] switch_loadable_module.c:991 Write lock interface 'vm_read' to wait for existing references. 2013-02-01 08:14:15.922904 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'vm_list' 2013-02-01 08:14:15.922904 [DEBUG] switch_loadable_module.c:991 Write lock interface 'vm_list' to wait for existing references. 2013-02-01 08:14:15.922904 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'vm_fsdb_auth_login' 2013-02-01 08:14:15.922904 [DEBUG] switch_loadable_module.c:991 Write lock interface 'vm_fsdb_auth_login' to wait for existing references. 2013-02-01 08:14:15.922904 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'vm_fsdb_msg_count' 2013-02-01 08:14:15.922904 [DEBUG] switch_loadable_module.c:991 Write lock interface 'vm_fsdb_msg_count' to wait for existing references. 2013-02-01 08:14:15.922904 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'vm_fsdb_msg_list' 2013-02-01 08:14:15.922904 [DEBUG] switch_loadable_module.c:991 Write lock interface 'vm_fsdb_msg_list' to wait for existing references. 2013-02-01 08:14:15.922904 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'vm_fsdb_msg_get' 2013-02-01 08:14:15.922904 [DEBUG] switch_loadable_module.c:991 Write lock interface 'vm_fsdb_msg_get' to wait for existing references. 2013-02-01 08:14:15.922904 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'vm_fsdb_msg_delete' 2013-02-01 08:14:15.922904 [DEBUG] switch_loadable_module.c:991 Write lock interface 'vm_fsdb_msg_delete' to wait for existing references. 2013-02-01 08:14:15.922904 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'vm_fsdb_msg_undelete' 2013-02-01 08:14:15.922904 [DEBUG] switch_loadable_module.c:991 Write lock interface 'vm_fsdb_msg_undelete' to wait for existing references. 2013-02-01 08:14:15.922904 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'vm_fsdb_msg_email' 2013-02-01 08:14:15.922904 [DEBUG] switch_loadable_module.c:991 Write lock interface 'vm_fsdb_msg_email' to wait for existing references. 2013-02-01 08:14:15.922904 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'vm_fsdb_msg_purge' 2013-02-01 08:14:15.922904 [DEBUG] switch_loadable_module.c:991 Write lock interface 'vm_fsdb_msg_purge' to wait for existing references. 2013-02-01 08:14:15.922904 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'vm_fsdb_msg_save' 2013-02-01 08:14:15.922904 [DEBUG] switch_loadable_module.c:991 Write lock interface 'vm_fsdb_msg_save' to wait for existing references. 2013-02-01 08:14:15.922904 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'vm_fsdb_msg_forward' 2013-02-01 08:14:15.922904 [DEBUG] switch_loadable_module.c:991 Write lock interface 'vm_fsdb_msg_forward' to wait for existing references. 2013-02-01 08:14:15.922904 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'vm_fsdb_pref_greeting_set' 2013-02-01 08:14:15.922904 [DEBUG] switch_loadable_module.c:991 Write lock interface 'vm_fsdb_pref_greeting_set' to wait for existing references. 2013-02-01 08:14:15.922904 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'vm_fsdb_pref_recname_set' 2013-02-01 08:14:15.922904 [DEBUG] switch_loadable_module.c:991 Write lock interface 'vm_fsdb_pref_recname_set' to wait for existing references. 2013-02-01 08:14:15.922904 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'vm_fsdb_pref_password_set' 2013-02-01 08:14:15.922904 [DEBUG] switch_loadable_module.c:991 Write lock interface 'vm_fsdb_pref_password_set' to wait for existing references. 2013-02-01 08:14:15.922904 [NOTICE] switch_event.c:403 Subclass reservation deleted for /usr/src/freeswitch/src/mod/applications/mod_voicemail/mod_voicemail.c:vm::maintenance 2013-02-01 08:14:15.922904 [NOTICE] switch_event.c:1970 Event Binding deleted for mod_voicemail:MESSAGE_QUERY 2013-02-01 08:14:15.922904 [DEBUG] mod_voicemail.c:6003 Waiting for write lock (Profile default) 2013-02-01 08:14:15.922904 [DEBUG] mod_voicemail.c:6006 Destroying Profile default 2013-02-01 08:14:15.922904 [DEBUG] switch_loadable_module.c:1020 Write lock interface 'mod_sndfile' to wait for existing references. 2013-02-01 08:14:15.922904 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'aiff' 2013-02-01 08:14:15.922904 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'au' 2013-02-01 08:14:15.922904 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'avr' 2013-02-01 08:14:15.922904 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'caf' 2013-02-01 08:14:15.922904 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'htk' 2013-02-01 08:14:15.922904 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'iff' 2013-02-01 08:14:15.922904 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'mat' 2013-02-01 08:14:15.922904 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'paf' 2013-02-01 08:14:15.922904 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'pvf' 2013-02-01 08:14:15.922904 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'raw' 2013-02-01 08:14:15.922904 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'sd2' 2013-02-01 08:14:15.922904 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'sds' 2013-02-01 08:14:15.922904 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'sf' 2013-02-01 08:14:15.922904 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'voc' 2013-02-01 08:14:15.922904 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'w64' 2013-02-01 08:14:15.922904 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'wav' 2013-02-01 08:14:15.922904 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'wve' 2013-02-01 08:14:15.922904 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'xi' 2013-02-01 08:14:15.922904 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'r8' 2013-02-01 08:14:15.922904 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'r16' 2013-02-01 08:14:15.922904 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'r24' 2013-02-01 08:14:15.922904 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'r32' 2013-02-01 08:14:15.922904 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'gsm' 2013-02-01 08:14:15.922904 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'ul' 2013-02-01 08:14:15.922904 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'ulaw' 2013-02-01 08:14:15.922904 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'al' 2013-02-01 08:14:15.922904 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'alaw' 2013-02-01 08:14:15.922904 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'adpcm' 2013-02-01 08:14:15.922904 [DEBUG] switch_loadable_module.c:1152 Write lock interface 'en' to wait for existing references. 2013-02-01 08:14:15.922904 [NOTICE] switch_loadable_module.c:1160 Deleting Say interface 'en' 2013-02-01 08:14:15.922904 [DEBUG] switch_loadable_module.c:846 Write lock interface 'loopback' to wait for existing references. 2013-02-01 08:14:15.922904 [NOTICE] switch_loadable_module.c:854 Deleting Endpoint 'loopback' 2013-02-01 08:14:15.922904 [NOTICE] switch_loadable_module.c:936 Deleting Application 'unloop' 2013-02-01 08:14:15.922904 [DEBUG] switch_loadable_module.c:938 Write lock interface 'unloop' to wait for existing references. 2013-02-01 08:14:15.922904 [NOTICE] switch_loadable_module.c:921 Deleting Timer 'soft' 2013-02-01 08:14:15.932942 [NOTICE] switch_loadable_module.c:100 Thread ended for CORE_SOFTTIMER_MODULE 2013-02-01 08:14:15.935348 [NOTICE] switch_event.c:2010 Event Binding deleted for CORE_SOFTTIMER_MODULE:RELOADXML 2013-02-01 08:14:15.935366 [NOTICE] switch_loadable_module.c:936 Deleting Application 'conference' 2013-02-01 08:14:15.935400 [DEBUG] switch_loadable_module.c:938 Write lock interface 'conference' to wait for existing references. 2013-02-01 08:14:15.935427 [NOTICE] switch_loadable_module.c:936 Deleting Application 'conference_set_auto_outcall' 2013-02-01 08:14:15.935438 [DEBUG] switch_loadable_module.c:938 Write lock interface 'conference_set_auto_outcall' to wait for existing references. 2013-02-01 08:14:15.935458 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'conference' 2013-02-01 08:14:15.935463 [DEBUG] switch_loadable_module.c:991 Write lock interface 'conference' to wait for existing references. 2013-02-01 08:14:15.935491 [DEBUG] switch_loadable_module.c:1127 Write lock interface 'conf' to wait for existing references. 2013-02-01 08:14:15.935497 [NOTICE] switch_loadable_module.c:1136 Deleting Chat interface 'conf' 2013-02-01 08:14:15.935532 [NOTICE] switch_event.c:1970 Event Binding deleted for mod_conference:PRESENCE_PROBE 2013-02-01 08:14:15.935540 [NOTICE] switch_event.c:1970 Event Binding deleted for mod_conference:CONFERENCE_DATA_QUERY 2013-02-01 08:14:15.935545 [NOTICE] switch_event.c:1970 Event Binding deleted for mod_conference:CALL_SETUP_REQ 2013-02-01 08:14:15.935550 [NOTICE] switch_event.c:403 Subclass reservation deleted for /usr/src/freeswitch/src/mod/applications/mod_conference/mod_conference.c:conference::maintenance 2013-02-01 08:14:15.935555 [NOTICE] switch_loadable_module.c:880 Deleting Codec PROXY-VID 31 PROXY VIDEO PASS-THROUGH 90000hz 0ms 2013-02-01 08:14:15.935581 [NOTICE] switch_loadable_module.c:880 Deleting Codec PROXY 0 PROXY PASS-THROUGH 8000hz 20ms 2013-02-01 08:14:15.935601 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 11025hz 32ms 2013-02-01 08:14:15.935609 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 11025hz 40ms 2013-02-01 08:14:15.935622 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 22050hz 20ms 2013-02-01 08:14:15.935639 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 48000hz 40ms 2013-02-01 08:14:15.935658 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 48000hz 30ms 2013-02-01 08:14:15.935674 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 48000hz 20ms 2013-02-01 08:14:15.935688 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 32000hz 8ms 2013-02-01 08:14:15.935702 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 32000hz 6ms 2013-02-01 08:14:15.935719 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 32000hz 4ms 2013-02-01 08:14:15.935729 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 32000hz 2ms 2013-02-01 08:14:15.935740 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 16000hz 8ms 2013-02-01 08:14:15.935754 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 16000hz 6ms 2013-02-01 08:14:15.935770 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 16000hz 4ms 2013-02-01 08:14:15.935784 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 16000hz 2ms 2013-02-01 08:14:15.935797 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 8000hz 8ms 2013-02-01 08:14:15.935808 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 8000hz 6ms 2013-02-01 08:14:15.935824 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 8000hz 4ms 2013-02-01 08:14:15.935835 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 8000hz 2ms 2013-02-01 08:14:15.935845 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 48000hz 10ms 2013-02-01 08:14:15.935862 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 48000hz 8ms 2013-02-01 08:14:15.935877 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 48000hz 6ms 2013-02-01 08:14:15.935908 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 48000hz 4ms 2013-02-01 08:14:15.935922 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 48000hz 2ms 2013-02-01 08:14:15.935937 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 24000hz 60ms 2013-02-01 08:14:15.935949 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 24000hz 40ms 2013-02-01 08:14:15.935963 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 24000hz 20ms 2013-02-01 08:14:15.935978 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 12000hz 100ms 2013-02-01 08:14:15.935990 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 12000hz 80ms 2013-02-01 08:14:15.936003 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 12000hz 60ms 2013-02-01 08:14:15.936028 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 12000hz 40ms 2013-02-01 08:14:15.936041 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 12000hz 20ms 2013-02-01 08:14:15.936053 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 32000hz 10ms 2013-02-01 08:14:15.936065 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 32000hz 20ms 2013-02-01 08:14:15.936083 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 32000hz 30ms 2013-02-01 08:14:15.936096 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 32000hz 40ms 2013-02-01 08:14:15.936107 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 32000hz 50ms 2013-02-01 08:14:15.936120 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 32000hz 60ms 2013-02-01 08:14:15.936132 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 16000hz 10ms 2013-02-01 08:14:15.936145 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 16000hz 20ms 2013-02-01 08:14:15.936157 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 16000hz 30ms 2013-02-01 08:14:15.936173 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 16000hz 40ms 2013-02-01 08:14:15.936185 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 16000hz 50ms 2013-02-01 08:14:15.936199 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 16000hz 60ms 2013-02-01 08:14:15.936211 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 8000hz 10ms 2013-02-01 08:14:15.936224 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 8000hz 20ms 2013-02-01 08:14:15.936236 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 8000hz 30ms 2013-02-01 08:14:15.936250 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 8000hz 40ms 2013-02-01 08:14:15.936263 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 8000hz 50ms 2013-02-01 08:14:15.936275 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 8000hz 60ms 2013-02-01 08:14:15.936286 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 8000hz 70ms 2013-02-01 08:14:15.936299 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 8000hz 80ms 2013-02-01 08:14:15.936312 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 8000hz 90ms 2013-02-01 08:14:15.936324 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 8000hz 100ms 2013-02-01 08:14:15.936336 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 8000hz 110ms 2013-02-01 08:14:15.936350 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 8000hz 120ms 2013-02-01 08:14:15.936376 [NOTICE] switch_loadable_module.c:880 Deleting Codec PCMU 0 G.711 ulaw 8000hz 10ms 2013-02-01 08:14:15.936390 [NOTICE] switch_loadable_module.c:880 Deleting Codec PCMU 0 G.711 ulaw 8000hz 20ms 2013-02-01 08:14:15.936402 [NOTICE] switch_loadable_module.c:880 Deleting Codec PCMU 0 G.711 ulaw 8000hz 30ms 2013-02-01 08:14:15.936414 [NOTICE] switch_loadable_module.c:880 Deleting Codec PCMU 0 G.711 ulaw 8000hz 40ms 2013-02-01 08:14:15.936434 [NOTICE] switch_loadable_module.c:880 Deleting Codec PCMU 0 G.711 ulaw 8000hz 50ms 2013-02-01 08:14:15.936446 [NOTICE] switch_loadable_module.c:880 Deleting Codec PCMU 0 G.711 ulaw 8000hz 60ms 2013-02-01 08:14:15.936459 [NOTICE] switch_loadable_module.c:880 Deleting Codec PCMU 0 G.711 ulaw 8000hz 70ms 2013-02-01 08:14:15.936472 [NOTICE] switch_loadable_module.c:880 Deleting Codec PCMU 0 G.711 ulaw 8000hz 80ms 2013-02-01 08:14:15.936483 [NOTICE] switch_loadable_module.c:880 Deleting Codec PCMU 0 G.711 ulaw 8000hz 90ms 2013-02-01 08:14:15.936494 [NOTICE] switch_loadable_module.c:880 Deleting Codec PCMU 0 G.711 ulaw 8000hz 100ms 2013-02-01 08:14:15.936506 [NOTICE] switch_loadable_module.c:880 Deleting Codec PCMU 0 G.711 ulaw 8000hz 110ms 2013-02-01 08:14:15.936518 [NOTICE] switch_loadable_module.c:880 Deleting Codec PCMU 0 G.711 ulaw 8000hz 120ms 2013-02-01 08:14:15.936544 [NOTICE] switch_loadable_module.c:880 Deleting Codec PCMA 8 G.711 alaw 8000hz 10ms 2013-02-01 08:14:15.936559 [NOTICE] switch_loadable_module.c:880 Deleting Codec PCMA 8 G.711 alaw 8000hz 20ms 2013-02-01 08:14:15.936572 [NOTICE] switch_loadable_module.c:880 Deleting Codec PCMA 8 G.711 alaw 8000hz 30ms 2013-02-01 08:14:15.936584 [NOTICE] switch_loadable_module.c:880 Deleting Codec PCMA 8 G.711 alaw 8000hz 40ms 2013-02-01 08:14:15.936604 [NOTICE] switch_loadable_module.c:880 Deleting Codec PCMA 8 G.711 alaw 8000hz 50ms 2013-02-01 08:14:15.936617 [NOTICE] switch_loadable_module.c:880 Deleting Codec PCMA 8 G.711 alaw 8000hz 60ms 2013-02-01 08:14:15.936629 [NOTICE] switch_loadable_module.c:880 Deleting Codec PCMA 8 G.711 alaw 8000hz 70ms 2013-02-01 08:14:15.936640 [NOTICE] switch_loadable_module.c:880 Deleting Codec PCMA 8 G.711 alaw 8000hz 80ms 2013-02-01 08:14:15.936654 [NOTICE] switch_loadable_module.c:880 Deleting Codec PCMA 8 G.711 alaw 8000hz 90ms 2013-02-01 08:14:15.936672 [NOTICE] switch_loadable_module.c:880 Deleting Codec PCMA 8 G.711 alaw 8000hz 100ms 2013-02-01 08:14:15.936688 [NOTICE] switch_loadable_module.c:880 Deleting Codec PCMA 8 G.711 alaw 8000hz 110ms 2013-02-01 08:14:15.936703 [NOTICE] switch_loadable_module.c:880 Deleting Codec PCMA 8 G.711 alaw 8000hz 120ms 2013-02-01 08:14:15.936741 [NOTICE] switch_loadable_module.c:936 Deleting Application 'socket' 2013-02-01 08:14:15.936751 [DEBUG] switch_loadable_module.c:938 Write lock interface 'socket' to wait for existing references. 2013-02-01 08:14:15.936769 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'event_sink' 2013-02-01 08:14:15.936775 [DEBUG] switch_loadable_module.c:991 Write lock interface 'event_sink' to wait for existing references. 2013-02-01 08:14:15.936832 [NOTICE] mod_event_socket.c:2829 Shutting Down 2013-02-01 08:14:15.936844 [NOTICE] switch_loadable_module.c:100 Thread ended for mod_event_socket 2013-02-01 08:14:15.936824 [NOTICE] switch_event.c:2010 Event Binding deleted for mod_event_socket:ALL 2013-02-01 08:14:15.936866 [NOTICE] switch_loadable_module.c:880 Deleting Codec AMR 96 AMR 8000hz 20ms 2013-02-01 08:14:15.936936 [DEBUG] switch_loadable_module.c:846 Write lock interface 'error' to wait for existing references. 2013-02-01 08:14:15.936944 [NOTICE] switch_loadable_module.c:854 Deleting Endpoint 'error' 2013-02-01 08:14:15.936968 [DEBUG] switch_loadable_module.c:846 Write lock interface 'group' to wait for existing references. 2013-02-01 08:14:15.936975 [NOTICE] switch_loadable_module.c:854 Deleting Endpoint 'group' 2013-02-01 08:14:15.937007 [DEBUG] switch_loadable_module.c:846 Write lock interface 'user' to wait for existing references. 2013-02-01 08:14:15.937014 [NOTICE] switch_loadable_module.c:854 Deleting Endpoint 'user' 2013-02-01 08:14:15.937036 [DEBUG] switch_loadable_module.c:846 Write lock interface 'pickup' to wait for existing references. 2013-02-01 08:14:15.937043 [NOTICE] switch_loadable_module.c:854 Deleting Endpoint 'pickup' 2013-02-01 08:14:15.937063 [NOTICE] switch_loadable_module.c:905 Deleting Dialplan 'inline' 2013-02-01 08:14:15.937085 [NOTICE] switch_loadable_module.c:936 Deleting Application 'blind_transfer_ack' 2013-02-01 08:14:15.937104 [DEBUG] switch_loadable_module.c:938 Write lock interface 'blind_transfer_ack' to wait for existing references. 2013-02-01 08:14:15.937125 [NOTICE] switch_loadable_module.c:936 Deleting Application 'bind_digit_action' 2013-02-01 08:14:15.937137 [DEBUG] switch_loadable_module.c:938 Write lock interface 'bind_digit_action' to wait for existing references. 2013-02-01 08:14:15.937159 [NOTICE] switch_loadable_module.c:936 Deleting Application 'capture' 2013-02-01 08:14:15.937178 [DEBUG] switch_loadable_module.c:938 Write lock interface 'capture' to wait for existing references. 2013-02-01 08:14:15.937204 [NOTICE] switch_loadable_module.c:936 Deleting Application 'clear_digit_action' 2013-02-01 08:14:15.937226 [DEBUG] switch_loadable_module.c:938 Write lock interface 'clear_digit_action' to wait for existing references. 2013-02-01 08:14:15.937246 [NOTICE] switch_loadable_module.c:936 Deleting Application 'digit_action_set_realm' 2013-02-01 08:14:15.937255 [DEBUG] switch_loadable_module.c:938 Write lock interface 'digit_action_set_realm' to wait for existing references. 2013-02-01 08:14:15.937275 [NOTICE] switch_loadable_module.c:936 Deleting Application 'privacy' 2013-02-01 08:14:15.937284 [DEBUG] switch_loadable_module.c:938 Write lock interface 'privacy' to wait for existing references. 2013-02-01 08:14:15.937308 [NOTICE] switch_loadable_module.c:936 Deleting Application 'set_audio_level' 2013-02-01 08:14:15.937326 [DEBUG] switch_loadable_module.c:938 Write lock interface 'set_audio_level' to wait for existing references. 2013-02-01 08:14:15.937347 [NOTICE] switch_loadable_module.c:936 Deleting Application 'set_mute' 2013-02-01 08:14:15.937357 [DEBUG] switch_loadable_module.c:938 Write lock interface 'set_mute' to wait for existing references. 2013-02-01 08:14:15.937380 [NOTICE] switch_loadable_module.c:936 Deleting Application 'flush_dtmf' 2013-02-01 08:14:15.937390 [DEBUG] switch_loadable_module.c:938 Write lock interface 'flush_dtmf' to wait for existing references. 2013-02-01 08:14:15.937412 [NOTICE] switch_loadable_module.c:936 Deleting Application 'hold' 2013-02-01 08:14:15.937421 [DEBUG] switch_loadable_module.c:938 Write lock interface 'hold' to wait for existing references. 2013-02-01 08:14:15.937443 [NOTICE] switch_loadable_module.c:936 Deleting Application 'unhold' 2013-02-01 08:14:15.937452 [DEBUG] switch_loadable_module.c:938 Write lock interface 'unhold' to wait for existing references. 2013-02-01 08:14:15.937474 [NOTICE] switch_loadable_module.c:936 Deleting Application 'mutex' 2013-02-01 08:14:15.937485 [DEBUG] switch_loadable_module.c:938 Write lock interface 'mutex' to wait for existing references. 2013-02-01 08:14:15.937507 [NOTICE] switch_loadable_module.c:936 Deleting Application 'page' 2013-02-01 08:14:15.937516 [DEBUG] switch_loadable_module.c:938 Write lock interface 'page' to wait for existing references. 2013-02-01 08:14:15.937534 [NOTICE] switch_loadable_module.c:936 Deleting Application 'transfer' 2013-02-01 08:14:15.937547 [DEBUG] switch_loadable_module.c:938 Write lock interface 'transfer' to wait for existing references. 2013-02-01 08:14:15.937568 [NOTICE] switch_loadable_module.c:936 Deleting Application 'check_acl' 2013-02-01 08:14:15.937578 [DEBUG] switch_loadable_module.c:938 Write lock interface 'check_acl' to wait for existing references. 2013-02-01 08:14:15.937599 [NOTICE] switch_loadable_module.c:936 Deleting Application 'verbose_events' 2013-02-01 08:14:15.937609 [DEBUG] switch_loadable_module.c:938 Write lock interface 'verbose_events' to wait for existing references. 2013-02-01 08:14:15.937631 [NOTICE] switch_loadable_module.c:936 Deleting Application 'cng_plc' 2013-02-01 08:14:15.937640 [DEBUG] switch_loadable_module.c:938 Write lock interface 'cng_plc' to wait for existing references. 2013-02-01 08:14:15.937662 [NOTICE] switch_loadable_module.c:936 Deleting Application 'early_hangup' 2013-02-01 08:14:15.937673 [DEBUG] switch_loadable_module.c:938 Write lock interface 'early_hangup' to wait for existing references. 2013-02-01 08:14:15.937694 [NOTICE] switch_loadable_module.c:936 Deleting Application 'sleep' 2013-02-01 08:14:15.937703 [DEBUG] switch_loadable_module.c:938 Write lock interface 'sleep' to wait for existing references. 2013-02-01 08:14:15.937725 [NOTICE] switch_loadable_module.c:936 Deleting Application 'delay_echo' 2013-02-01 08:14:15.937737 [DEBUG] switch_loadable_module.c:938 Write lock interface 'delay_echo' to wait for existing references. 2013-02-01 08:14:15.937758 [NOTICE] switch_loadable_module.c:936 Deleting Application 'strftime' 2013-02-01 08:14:15.937767 [DEBUG] switch_loadable_module.c:938 Write lock interface 'strftime' to wait for existing references. 2013-02-01 08:14:15.937788 [NOTICE] switch_loadable_module.c:936 Deleting Application 'phrase' 2013-02-01 08:14:15.937798 [DEBUG] switch_loadable_module.c:938 Write lock interface 'phrase' to wait for existing references. 2013-02-01 08:14:15.937820 [NOTICE] switch_loadable_module.c:936 Deleting Application 'eval' 2013-02-01 08:14:15.937829 [DEBUG] switch_loadable_module.c:938 Write lock interface 'eval' to wait for existing references. 2013-02-01 08:14:15.937850 [NOTICE] switch_loadable_module.c:936 Deleting Application 'stop' 2013-02-01 08:14:15.937860 [DEBUG] switch_loadable_module.c:938 Write lock interface 'stop' to wait for existing references. 2013-02-01 08:14:15.937881 [NOTICE] switch_loadable_module.c:936 Deleting Application 'set_zombie_exec' 2013-02-01 08:14:15.937907 [DEBUG] switch_loadable_module.c:938 Write lock interface 'set_zombie_exec' to wait for existing references. 2013-02-01 08:14:15.937929 [NOTICE] switch_loadable_module.c:936 Deleting Application 'pre_answer' 2013-02-01 08:14:15.937940 [DEBUG] switch_loadable_module.c:938 Write lock interface 'pre_answer' to wait for existing references. 2013-02-01 08:14:15.937962 [NOTICE] switch_loadable_module.c:936 Deleting Application 'answer' 2013-02-01 08:14:15.937971 [DEBUG] switch_loadable_module.c:938 Write lock interface 'answer' to wait for existing references. 2013-02-01 08:14:15.937993 [NOTICE] switch_loadable_module.c:936 Deleting Application 'wait_for_answer' 2013-02-01 08:14:15.938003 [DEBUG] switch_loadable_module.c:938 Write lock interface 'wait_for_answer' to wait for existing references. 2013-02-01 08:14:15.938023 [NOTICE] switch_loadable_module.c:936 Deleting Application 'hangup' 2013-02-01 08:14:15.938033 [DEBUG] switch_loadable_module.c:938 Write lock interface 'hangup' to wait for existing references. 2013-02-01 08:14:15.938054 [NOTICE] switch_loadable_module.c:936 Deleting Application 'set_name' 2013-02-01 08:14:15.938064 [DEBUG] switch_loadable_module.c:938 Write lock interface 'set_name' to wait for existing references. 2013-02-01 08:14:15.938085 [NOTICE] switch_loadable_module.c:936 Deleting Application 'presence' 2013-02-01 08:14:15.938094 [DEBUG] switch_loadable_module.c:938 Write lock interface 'presence' to wait for existing references. 2013-02-01 08:14:15.938114 [NOTICE] switch_loadable_module.c:936 Deleting Application 'log' 2013-02-01 08:14:15.938123 [DEBUG] switch_loadable_module.c:938 Write lock interface 'log' to wait for existing references. 2013-02-01 08:14:15.938145 [NOTICE] switch_loadable_module.c:936 Deleting Application 'info' 2013-02-01 08:14:15.938154 [DEBUG] switch_loadable_module.c:938 Write lock interface 'info' to wait for existing references. 2013-02-01 08:14:15.938176 [NOTICE] switch_loadable_module.c:936 Deleting Application 'event' 2013-02-01 08:14:15.938186 [DEBUG] switch_loadable_module.c:938 Write lock interface 'event' to wait for existing references. 2013-02-01 08:14:15.938207 [NOTICE] switch_loadable_module.c:936 Deleting Application 'sound_test' 2013-02-01 08:14:15.938216 [DEBUG] switch_loadable_module.c:938 Write lock interface 'sound_test' to wait for existing references. 2013-02-01 08:14:15.938237 [NOTICE] switch_loadable_module.c:936 Deleting Application 'export' 2013-02-01 08:14:15.938246 [DEBUG] switch_loadable_module.c:938 Write lock interface 'export' to wait for existing references. 2013-02-01 08:14:15.938269 [NOTICE] switch_loadable_module.c:936 Deleting Application 'bridge_export' 2013-02-01 08:14:15.938294 [DEBUG] switch_loadable_module.c:938 Write lock interface 'bridge_export' to wait for existing references. 2013-02-01 08:14:15.938315 [NOTICE] switch_loadable_module.c:936 Deleting Application 'set' 2013-02-01 08:14:15.938324 [DEBUG] switch_loadable_module.c:938 Write lock interface 'set' to wait for existing references. 2013-02-01 08:14:15.938345 [NOTICE] switch_loadable_module.c:936 Deleting Application 'multiset' 2013-02-01 08:14:15.938354 [DEBUG] switch_loadable_module.c:938 Write lock interface 'multiset' to wait for existing references. 2013-02-01 08:14:15.938376 [NOTICE] switch_loadable_module.c:936 Deleting Application 'push' 2013-02-01 08:14:15.938387 [DEBUG] switch_loadable_module.c:938 Write lock interface 'push' to wait for existing references. 2013-02-01 08:14:15.938408 [NOTICE] switch_loadable_module.c:936 Deleting Application 'unshift' 2013-02-01 08:14:15.938418 [DEBUG] switch_loadable_module.c:938 Write lock interface 'unshift' to wait for existing references. 2013-02-01 08:14:15.938439 [NOTICE] switch_loadable_module.c:936 Deleting Application 'set_global' 2013-02-01 08:14:15.938448 [DEBUG] switch_loadable_module.c:938 Write lock interface 'set_global' to wait for existing references. 2013-02-01 08:14:15.938469 [NOTICE] switch_loadable_module.c:936 Deleting Application 'set_profile_var' 2013-02-01 08:14:15.938478 [DEBUG] switch_loadable_module.c:938 Write lock interface 'set_profile_var' to wait for existing references. 2013-02-01 08:14:15.938500 [NOTICE] switch_loadable_module.c:936 Deleting Application 'unset' 2013-02-01 08:14:15.938510 [DEBUG] switch_loadable_module.c:938 Write lock interface 'unset' to wait for existing references. 2013-02-01 08:14:15.938530 [NOTICE] switch_loadable_module.c:936 Deleting Application 'ring_ready' 2013-02-01 08:14:15.938540 [DEBUG] switch_loadable_module.c:938 Write lock interface 'ring_ready' to wait for existing references. 2013-02-01 08:14:15.938560 [NOTICE] switch_loadable_module.c:936 Deleting Application 'remove_bugs' 2013-02-01 08:14:15.938570 [DEBUG] switch_loadable_module.c:938 Write lock interface 'remove_bugs' to wait for existing references. 2013-02-01 08:14:15.938592 [NOTICE] switch_loadable_module.c:936 Deleting Application 'break' 2013-02-01 08:14:15.938601 [DEBUG] switch_loadable_module.c:938 Write lock interface 'break' to wait for existing references. 2013-02-01 08:14:15.938623 [NOTICE] switch_loadable_module.c:936 Deleting Application 'detect_speech' 2013-02-01 08:14:15.938632 [DEBUG] switch_loadable_module.c:938 Write lock interface 'detect_speech' to wait for existing references. 2013-02-01 08:14:15.938655 [NOTICE] switch_loadable_module.c:936 Deleting Application 'play_and_detect_speech' 2013-02-01 08:14:15.938665 [DEBUG] switch_loadable_module.c:938 Write lock interface 'play_and_detect_speech' to wait for existing references. 2013-02-01 08:14:15.938685 [NOTICE] switch_loadable_module.c:936 Deleting Application 'ivr' 2013-02-01 08:14:15.938695 [DEBUG] switch_loadable_module.c:938 Write lock interface 'ivr' to wait for existing references. 2013-02-01 08:14:15.938717 [NOTICE] switch_loadable_module.c:936 Deleting Application 'redirect' 2013-02-01 08:14:15.938726 [DEBUG] switch_loadable_module.c:938 Write lock interface 'redirect' to wait for existing references. 2013-02-01 08:14:15.938747 [NOTICE] switch_loadable_module.c:936 Deleting Application 'video_refresh' 2013-02-01 08:14:15.938757 [DEBUG] switch_loadable_module.c:938 Write lock interface 'video_refresh' to wait for existing references. 2013-02-01 08:14:15.938778 [NOTICE] switch_loadable_module.c:936 Deleting Application 'send_info' 2013-02-01 08:14:15.938787 [DEBUG] switch_loadable_module.c:938 Write lock interface 'send_info' to wait for existing references. 2013-02-01 08:14:15.938809 [NOTICE] switch_loadable_module.c:936 Deleting Application 'jitterbuffer' 2013-02-01 08:14:15.938823 [DEBUG] switch_loadable_module.c:938 Write lock interface 'jitterbuffer' to wait for existing references. 2013-02-01 08:14:15.938844 [NOTICE] switch_loadable_module.c:936 Deleting Application 'send_display' 2013-02-01 08:14:15.938854 [DEBUG] switch_loadable_module.c:938 Write lock interface 'send_display' to wait for existing references. 2013-02-01 08:14:15.938874 [NOTICE] switch_loadable_module.c:936 Deleting Application 'respond' 2013-02-01 08:14:15.938884 [DEBUG] switch_loadable_module.c:938 Write lock interface 'respond' to wait for existing references. 2013-02-01 08:14:15.938916 [NOTICE] switch_loadable_module.c:936 Deleting Application 'deflect' 2013-02-01 08:14:15.938926 [DEBUG] switch_loadable_module.c:938 Write lock interface 'deflect' to wait for existing references. 2013-02-01 08:14:15.938947 [NOTICE] switch_loadable_module.c:936 Deleting Application 'recovery_refresh' 2013-02-01 08:14:15.938957 [DEBUG] switch_loadable_module.c:938 Write lock interface 'recovery_refresh' to wait for existing references. 2013-02-01 08:14:15.938979 [NOTICE] switch_loadable_module.c:936 Deleting Application 'queue_dtmf' 2013-02-01 08:14:15.938989 [DEBUG] switch_loadable_module.c:938 Write lock interface 'queue_dtmf' to wait for existing references. 2013-02-01 08:14:15.939010 [NOTICE] switch_loadable_module.c:936 Deleting Application 'send_dtmf' 2013-02-01 08:14:15.939019 [DEBUG] switch_loadable_module.c:938 Write lock interface 'send_dtmf' to wait for existing references. 2013-02-01 08:14:15.939039 [NOTICE] switch_loadable_module.c:936 Deleting Application 'sched_cancel' 2013-02-01 08:14:15.939048 [DEBUG] switch_loadable_module.c:938 Write lock interface 'sched_cancel' to wait for existing references. 2013-02-01 08:14:15.939071 [NOTICE] switch_loadable_module.c:936 Deleting Application 'sched_hangup' 2013-02-01 08:14:15.939080 [DEBUG] switch_loadable_module.c:938 Write lock interface 'sched_hangup' to wait for existing references. 2013-02-01 08:14:15.939103 [NOTICE] switch_loadable_module.c:936 Deleting Application 'sched_broadcast' 2013-02-01 08:14:15.939113 [DEBUG] switch_loadable_module.c:938 Write lock interface 'sched_broadcast' to wait for existing references. 2013-02-01 08:14:15.939135 [NOTICE] switch_loadable_module.c:936 Deleting Application 'sched_transfer' 2013-02-01 08:14:15.939145 [DEBUG] switch_loadable_module.c:938 Write lock interface 'sched_transfer' to wait for existing references. 2013-02-01 08:14:15.939167 [NOTICE] switch_loadable_module.c:936 Deleting Application 'execute_extension' 2013-02-01 08:14:15.939177 [DEBUG] switch_loadable_module.c:938 Write lock interface 'execute_extension' to wait for existing references. 2013-02-01 08:14:15.939198 [NOTICE] switch_loadable_module.c:936 Deleting Application 'sched_heartbeat' 2013-02-01 08:14:15.939208 [DEBUG] switch_loadable_module.c:938 Write lock interface 'sched_heartbeat' to wait for existing references. 2013-02-01 08:14:15.939230 [NOTICE] switch_loadable_module.c:936 Deleting Application 'enable_heartbeat' 2013-02-01 08:14:15.939240 [DEBUG] switch_loadable_module.c:938 Write lock interface 'enable_heartbeat' to wait for existing references. 2013-02-01 08:14:15.939261 [NOTICE] switch_loadable_module.c:936 Deleting Application 'mkdir' 2013-02-01 08:14:15.939273 [DEBUG] switch_loadable_module.c:938 Write lock interface 'mkdir' to wait for existing references. 2013-02-01 08:14:15.939293 [NOTICE] switch_loadable_module.c:936 Deleting Application 'rename' 2013-02-01 08:14:15.939303 [DEBUG] switch_loadable_module.c:938 Write lock interface 'rename' to wait for existing references. 2013-02-01 08:14:15.939323 [NOTICE] switch_loadable_module.c:936 Deleting Application 'soft_hold' 2013-02-01 08:14:15.939333 [DEBUG] switch_loadable_module.c:938 Write lock interface 'soft_hold' to wait for existing references. 2013-02-01 08:14:15.939355 [NOTICE] switch_loadable_module.c:936 Deleting Application 'bind_meta_app' 2013-02-01 08:14:15.939367 [DEBUG] switch_loadable_module.c:938 Write lock interface 'bind_meta_app' to wait for existing references. 2013-02-01 08:14:15.939388 [NOTICE] switch_loadable_module.c:936 Deleting Application 'unbind_meta_app' 2013-02-01 08:14:15.939397 [DEBUG] switch_loadable_module.c:938 Write lock interface 'unbind_meta_app' to wait for existing references. 2013-02-01 08:14:15.939415 [NOTICE] switch_loadable_module.c:936 Deleting Application 'block_dfmf' 2013-02-01 08:14:15.939424 [DEBUG] switch_loadable_module.c:938 Write lock interface 'block_dfmf' to wait for existing references. 2013-02-01 08:14:15.939447 [NOTICE] switch_loadable_module.c:936 Deleting Application 'unblock_dtmf' 2013-02-01 08:14:15.939457 [DEBUG] switch_loadable_module.c:938 Write lock interface 'unblock_dtmf' to wait for existing references. 2013-02-01 08:14:15.939478 [NOTICE] switch_loadable_module.c:936 Deleting Application 'intercept' 2013-02-01 08:14:15.939487 [DEBUG] switch_loadable_module.c:938 Write lock interface 'intercept' to wait for existing references. 2013-02-01 08:14:15.939509 [NOTICE] switch_loadable_module.c:936 Deleting Application 'eavesdrop' 2013-02-01 08:14:15.939518 [DEBUG] switch_loadable_module.c:938 Write lock interface 'eavesdrop' to wait for existing references. 2013-02-01 08:14:15.939540 [NOTICE] switch_loadable_module.c:936 Deleting Application 'three_way' 2013-02-01 08:14:15.939549 [DEBUG] switch_loadable_module.c:938 Write lock interface 'three_way' to wait for existing references. 2013-02-01 08:14:15.939571 [NOTICE] switch_loadable_module.c:936 Deleting Application 'set_user' 2013-02-01 08:14:15.939580 [DEBUG] switch_loadable_module.c:938 Write lock interface 'set_user' to wait for existing references. 2013-02-01 08:14:15.939601 [NOTICE] switch_loadable_module.c:936 Deleting Application 'stop_dtmf' 2013-02-01 08:14:15.939610 [DEBUG] switch_loadable_module.c:938 Write lock interface 'stop_dtmf' to wait for existing references. 2013-02-01 08:14:15.939629 [NOTICE] switch_loadable_module.c:936 Deleting Application 'start_dtmf' 2013-02-01 08:14:15.939638 [DEBUG] switch_loadable_module.c:938 Write lock interface 'start_dtmf' to wait for existing references. 2013-02-01 08:14:15.939656 [NOTICE] switch_loadable_module.c:936 Deleting Application 'stop_dtmf_generate' 2013-02-01 08:14:15.939666 [DEBUG] switch_loadable_module.c:938 Write lock interface 'stop_dtmf_generate' to wait for existing references. 2013-02-01 08:14:15.939687 [NOTICE] switch_loadable_module.c:936 Deleting Application 'start_dtmf_generate' 2013-02-01 08:14:15.939699 [DEBUG] switch_loadable_module.c:938 Write lock interface 'start_dtmf_generate' to wait for existing references. 2013-02-01 08:14:15.939719 [NOTICE] switch_loadable_module.c:936 Deleting Application 'stop_tone_detect' 2013-02-01 08:14:15.939729 [DEBUG] switch_loadable_module.c:938 Write lock interface 'stop_tone_detect' to wait for existing references. 2013-02-01 08:14:15.939750 [NOTICE] switch_loadable_module.c:936 Deleting Application 'fax_detect' 2013-02-01 08:14:15.939760 [DEBUG] switch_loadable_module.c:938 Write lock interface 'fax_detect' to wait for existing references. 2013-02-01 08:14:15.939779 [NOTICE] switch_loadable_module.c:936 Deleting Application 'tone_detect' 2013-02-01 08:14:15.939789 [DEBUG] switch_loadable_module.c:938 Write lock interface 'tone_detect' to wait for existing references. 2013-02-01 08:14:15.939810 [NOTICE] switch_loadable_module.c:936 Deleting Application 'echo' 2013-02-01 08:14:15.939820 [DEBUG] switch_loadable_module.c:938 Write lock interface 'echo' to wait for existing references. 2013-02-01 08:14:15.939841 [NOTICE] switch_loadable_module.c:936 Deleting Application 'park' 2013-02-01 08:14:15.939850 [DEBUG] switch_loadable_module.c:938 Write lock interface 'park' to wait for existing references. 2013-02-01 08:14:15.939871 [NOTICE] switch_loadable_module.c:936 Deleting Application 'park_state' 2013-02-01 08:14:15.939882 [DEBUG] switch_loadable_module.c:938 Write lock interface 'park_state' to wait for existing references. 2013-02-01 08:14:15.939907 [NOTICE] switch_loadable_module.c:936 Deleting Application 'gentones' 2013-02-01 08:14:15.939917 [DEBUG] switch_loadable_module.c:938 Write lock interface 'gentones' to wait for existing references. 2013-02-01 08:14:15.939935 [NOTICE] switch_loadable_module.c:936 Deleting Application 'playback' 2013-02-01 08:14:15.939945 [DEBUG] switch_loadable_module.c:938 Write lock interface 'playback' to wait for existing references. 2013-02-01 08:14:15.939966 [NOTICE] switch_loadable_module.c:936 Deleting Application 'endless_playback' 2013-02-01 08:14:15.939976 [DEBUG] switch_loadable_module.c:938 Write lock interface 'endless_playback' to wait for existing references. 2013-02-01 08:14:15.939998 [NOTICE] switch_loadable_module.c:936 Deleting Application 'att_xfer' 2013-02-01 08:14:15.940007 [DEBUG] switch_loadable_module.c:938 Write lock interface 'att_xfer' to wait for existing references. 2013-02-01 08:14:15.940030 [NOTICE] switch_loadable_module.c:936 Deleting Application 'read' 2013-02-01 08:14:15.940039 [DEBUG] switch_loadable_module.c:938 Write lock interface 'read' to wait for existing references. 2013-02-01 08:14:15.940062 [NOTICE] switch_loadable_module.c:936 Deleting Application 'play_and_get_digits' 2013-02-01 08:14:15.940072 [DEBUG] switch_loadable_module.c:938 Write lock interface 'play_and_get_digits' to wait for existing references. 2013-02-01 08:14:15.940094 [NOTICE] switch_loadable_module.c:936 Deleting Application 'stop_record_session' 2013-02-01 08:14:15.940104 [DEBUG] switch_loadable_module.c:938 Write lock interface 'stop_record_session' to wait for existing references. 2013-02-01 08:14:15.940125 [NOTICE] switch_loadable_module.c:936 Deleting Application 'record_session' 2013-02-01 08:14:15.940136 [DEBUG] switch_loadable_module.c:938 Write lock interface 'record_session' to wait for existing references. 2013-02-01 08:14:15.940157 [NOTICE] switch_loadable_module.c:936 Deleting Application 'record' 2013-02-01 08:14:15.940166 [DEBUG] switch_loadable_module.c:938 Write lock interface 'record' to wait for existing references. 2013-02-01 08:14:15.940188 [NOTICE] switch_loadable_module.c:936 Deleting Application 'preprocess' 2013-02-01 08:14:15.940197 [DEBUG] switch_loadable_module.c:938 Write lock interface 'preprocess' to wait for existing references. 2013-02-01 08:14:15.940218 [NOTICE] switch_loadable_module.c:936 Deleting Application 'stop_displace_session' 2013-02-01 08:14:15.940228 [DEBUG] switch_loadable_module.c:938 Write lock interface 'stop_displace_session' to wait for existing references. 2013-02-01 08:14:15.940249 [NOTICE] switch_loadable_module.c:936 Deleting Application 'displace_session' 2013-02-01 08:14:15.940259 [DEBUG] switch_loadable_module.c:938 Write lock interface 'displace_session' to wait for existing references. 2013-02-01 08:14:15.940281 [NOTICE] switch_loadable_module.c:936 Deleting Application 'speak' 2013-02-01 08:14:15.940291 [DEBUG] switch_loadable_module.c:938 Write lock interface 'speak' to wait for existing references. 2013-02-01 08:14:15.940312 [NOTICE] switch_loadable_module.c:936 Deleting Application 'clear_speech_cache' 2013-02-01 08:14:15.940322 [DEBUG] switch_loadable_module.c:938 Write lock interface 'clear_speech_cache' to wait for existing references. 2013-02-01 08:14:15.940341 [NOTICE] switch_loadable_module.c:936 Deleting Application 'bridge' 2013-02-01 08:14:15.940353 [DEBUG] switch_loadable_module.c:938 Write lock interface 'bridge' to wait for existing references. 2013-02-01 08:14:15.940373 [NOTICE] switch_loadable_module.c:936 Deleting Application 'system' 2013-02-01 08:14:15.940383 [DEBUG] switch_loadable_module.c:938 Write lock interface 'system' to wait for existing references. 2013-02-01 08:14:15.940401 [NOTICE] switch_loadable_module.c:936 Deleting Application 'say' 2013-02-01 08:14:15.940412 [DEBUG] switch_loadable_module.c:938 Write lock interface 'say' to wait for existing references. 2013-02-01 08:14:15.940433 [NOTICE] switch_loadable_module.c:936 Deleting Application 'wait_for_silence' 2013-02-01 08:14:15.940443 [DEBUG] switch_loadable_module.c:938 Write lock interface 'wait_for_silence' to wait for existing references. 2013-02-01 08:14:15.940465 [NOTICE] switch_loadable_module.c:936 Deleting Application 'session_loglevel' 2013-02-01 08:14:15.940474 [DEBUG] switch_loadable_module.c:938 Write lock interface 'session_loglevel' to wait for existing references. 2013-02-01 08:14:15.940496 [NOTICE] switch_loadable_module.c:936 Deleting Application 'limit' 2013-02-01 08:14:15.940505 [DEBUG] switch_loadable_module.c:938 Write lock interface 'limit' to wait for existing references. 2013-02-01 08:14:15.940526 [NOTICE] switch_loadable_module.c:936 Deleting Application 'limit_hash' 2013-02-01 08:14:15.940535 [DEBUG] switch_loadable_module.c:938 Write lock interface 'limit_hash' to wait for existing references. 2013-02-01 08:14:15.940554 [NOTICE] switch_loadable_module.c:936 Deleting Application 'limit_execute' 2013-02-01 08:14:15.940565 [DEBUG] switch_loadable_module.c:938 Write lock interface 'limit_execute' to wait for existing references. 2013-02-01 08:14:15.940586 [NOTICE] switch_loadable_module.c:936 Deleting Application 'limit_hash_execute' 2013-02-01 08:14:15.940596 [DEBUG] switch_loadable_module.c:938 Write lock interface 'limit_hash_execute' to wait for existing references. 2013-02-01 08:14:15.940618 [NOTICE] switch_loadable_module.c:936 Deleting Application 'pickup' 2013-02-01 08:14:15.940627 [DEBUG] switch_loadable_module.c:938 Write lock interface 'pickup' to wait for existing references. 2013-02-01 08:14:15.940648 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'strepoch' 2013-02-01 08:14:15.940653 [DEBUG] switch_loadable_module.c:991 Write lock interface 'strepoch' to wait for existing references. 2013-02-01 08:14:15.940676 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'page' 2013-02-01 08:14:15.940681 [DEBUG] switch_loadable_module.c:991 Write lock interface 'page' to wait for existing references. 2013-02-01 08:14:15.940705 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'strmicroepoch' 2013-02-01 08:14:15.940710 [DEBUG] switch_loadable_module.c:991 Write lock interface 'strmicroepoch' to wait for existing references. 2013-02-01 08:14:15.940732 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'chat' 2013-02-01 08:14:15.940737 [DEBUG] switch_loadable_module.c:991 Write lock interface 'chat' to wait for existing references. 2013-02-01 08:14:15.940761 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'strftime' 2013-02-01 08:14:15.940766 [DEBUG] switch_loadable_module.c:991 Write lock interface 'strftime' to wait for existing references. 2013-02-01 08:14:15.940789 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'presence' 2013-02-01 08:14:15.940794 [DEBUG] switch_loadable_module.c:991 Write lock interface 'presence' to wait for existing references. 2013-02-01 08:14:15.940817 [DEBUG] switch_loadable_module.c:1020 Write lock interface 'mod_dptools' to wait for existing references. 2013-02-01 08:14:15.940824 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'file_string' 2013-02-01 08:14:15.940843 [DEBUG] switch_loadable_module.c:1127 Write lock interface 'event' to wait for existing references. 2013-02-01 08:14:15.940849 [NOTICE] switch_loadable_module.c:1136 Deleting Chat interface 'event' 2013-02-01 08:14:15.940870 [DEBUG] switch_loadable_module.c:1127 Write lock interface 'api' to wait for existing references. 2013-02-01 08:14:15.940875 [NOTICE] switch_loadable_module.c:1136 Deleting Chat interface 'api' 2013-02-01 08:14:15.940911 [NOTICE] switch_event.c:1970 Event Binding deleted for mod_dptools:PRESENCE_PROBE 2013-02-01 08:14:15.940920 [NOTICE] switch_loadable_module.c:880 Deleting Codec G723 4 G.723.1 6.3k 8000hz 120ms 2013-02-01 08:14:15.940942 [NOTICE] switch_loadable_module.c:880 Deleting Codec G723 4 G.723.1 6.3k 8000hz 90ms 2013-02-01 08:14:15.940955 [NOTICE] switch_loadable_module.c:880 Deleting Codec G723 4 G.723.1 6.3k 8000hz 60ms 2013-02-01 08:14:15.940967 [NOTICE] switch_loadable_module.c:880 Deleting Codec G723 4 G.723.1 6.3k 8000hz 30ms 2013-02-01 08:14:15.941000 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'console' 2013-02-01 08:14:15.941007 [DEBUG] switch_loadable_module.c:991 Write lock interface 'console' to wait for existing references. 2013-02-01 08:39:39.324100 [NOTICE] switch_loadable_module.c:214 Adding Dialplan 'enum' 2013-02-01 08:39:39.324126 [NOTICE] switch_loadable_module.c:254 Adding Application 'enum' 2013-02-01 08:39:39.324147 [NOTICE] switch_loadable_module.c:298 Adding API Function 'enum' 2013-02-01 08:39:39.324165 [NOTICE] switch_loadable_module.c:298 Adding API Function 'enum_auto' 2013-02-01 08:39:39.324367 [DEBUG] mod_cdr_csv.c:339 Adding default template. 2013-02-01 08:39:39.324386 [DEBUG] mod_cdr_csv.c:386 Adding template sql. 2013-02-01 08:39:39.324391 [DEBUG] mod_cdr_csv.c:386 Adding template example. 2013-02-01 08:39:39.324395 [DEBUG] mod_cdr_csv.c:386 Adding template snom. 2013-02-01 08:39:39.324399 [DEBUG] mod_cdr_csv.c:386 Adding template linksys. 2013-02-01 08:39:39.324403 [DEBUG] mod_cdr_csv.c:386 Adding template asterisk. 2013-02-01 08:39:39.324407 [DEBUG] mod_cdr_csv.c:386 Adding template opencdrrate. 2013-02-01 08:39:39.324441 [NOTICE] switch_loadable_module.c:298 Adding API Function 'cdr_csv' 2013-02-01 08:39:39.324703 [NOTICE] switch_loadable_module.c:254 Adding Application 'socket' 2013-02-01 08:39:39.324727 [NOTICE] switch_loadable_module.c:298 Adding API Function 'event_sink' 2013-02-01 08:39:39.325622 [INFO] mod_sofia.c:5728 Starting initial message thread. 2013-02-01 08:39:39.325703 [WARNING] sofia.c:1592 MSG Thread 0 Started 2013-02-01 08:39:39.325770 [DEBUG] sofia.c:3581 debug [0] 2013-02-01 08:39:39.325782 [DEBUG] sofia.c:3581 sip-trace [no] 2013-02-01 08:39:39.325809 [DEBUG] sofia.c:3581 sip-capture [no] 2013-02-01 08:39:39.325822 [DEBUG] sofia.c:3581 rfc2833-pt [101] 2013-02-01 08:39:39.325833 [DEBUG] sofia.c:3581 sip-port [5080] 2013-02-01 08:39:39.325839 [DEBUG] sofia.c:3581 dialplan [XML] 2013-02-01 08:39:39.325853 [DEBUG] sofia.c:3581 context [public] 2013-02-01 08:39:39.325865 [DEBUG] sofia.c:3581 dtmf-duration [2000] 2013-02-01 08:39:39.325874 [DEBUG] sofia.c:3581 inbound-codec-prefs [G722,PCMU,PCMA,GSM] 2013-02-01 08:39:39.325897 [DEBUG] sofia.c:3581 outbound-codec-prefs [PCMU,PCMA,GSM] 2013-02-01 08:39:39.325912 [DEBUG] sofia.c:3581 hold-music [%{bong-ring}] 2013-02-01 08:39:39.325922 [DEBUG] sofia.c:3581 rtp-timer-name [soft] 2013-02-01 08:39:39.325931 [DEBUG] sofia.c:3581 local-network-acl [localnet.auto] 2013-02-01 08:39:39.325941 [DEBUG] sofia.c:3581 manage-presence [false] 2013-02-01 08:39:39.325953 [DEBUG] sofia.c:3581 inbound-codec-negotiation [generous] 2013-02-01 08:39:39.325962 [DEBUG] sofia.c:3581 nonce-ttl [60] 2013-02-01 08:39:39.325972 [DEBUG] sofia.c:3581 auth-calls [true] 2013-02-01 08:39:39.325983 [DEBUG] sofia.c:3581 inbound-late-negotiation [true] 2013-02-01 08:39:39.325990 [DEBUG] sofia.c:3581 inbound-zrtp-passthru [true] 2013-02-01 08:39:39.325995 [DEBUG] sofia.c:3581 rtp-ip [XXX.XXX.XXX.10] 2013-02-01 08:39:39.326001 [DEBUG] sofia.c:3581 sip-ip [XXX.XXX.XXX.10] 2013-02-01 08:39:39.326008 [DEBUG] sofia.c:3581 ext-rtp-ip [auto-nat] 2013-02-01 08:39:39.326013 [DEBUG] sofia.c:3581 ext-sip-ip [auto-nat] 2013-02-01 08:39:39.326018 [DEBUG] sofia.c:3581 rtp-timeout-sec [300] 2013-02-01 08:39:39.326023 [DEBUG] sofia.c:3581 rtp-hold-timeout-sec [1800] 2013-02-01 08:39:39.326032 [DEBUG] sofia.c:3581 tls [false] 2013-02-01 08:39:39.326040 [DEBUG] sofia.c:3581 tls-only [false] 2013-02-01 08:39:39.326048 [DEBUG] sofia.c:3581 tls-bind-params [transport=tls] 2013-02-01 08:39:39.326059 [DEBUG] sofia.c:3581 tls-sip-port [5081] 2013-02-01 08:39:39.326068 [DEBUG] sofia.c:3581 tls-cert-dir [/usr/local/freeswitch/conf/ssl] 2013-02-01 08:39:39.326081 [DEBUG] sofia.c:3581 tls-passphrase [] 2013-02-01 08:39:39.326092 [DEBUG] sofia.c:3581 tls-verify-date [true] 2013-02-01 08:39:39.326103 [DEBUG] sofia.c:3581 tls-verify-policy [none] 2013-02-01 08:39:39.326115 [DEBUG] sofia.c:3581 tls-verify-depth [2] 2013-02-01 08:39:39.326125 [DEBUG] sofia.c:3581 tls-verify-in-subjects [] 2013-02-01 08:39:39.326137 [DEBUG] sofia.c:3581 tls-version [tlsv1] 2013-02-01 08:39:39.326228 [NOTICE] sofia.c:4643 Started Profile external [sofia_reg_external] 2013-02-01 08:39:39.326252 [DEBUG] sofia.c:2311 Creating agent for external 2013-02-01 08:39:39.326270 [DEBUG] sofia.c:3581 debug [0] 2013-02-01 08:39:39.326281 [DEBUG] sofia.c:3581 sip-trace [no] 2013-02-01 08:39:39.326301 [DEBUG] sofia.c:3581 context [public] 2013-02-01 08:39:39.326315 [DEBUG] sofia.c:3581 rfc2833-pt [101] 2013-02-01 08:39:39.326323 [DEBUG] sofia.c:3581 sip-port [5060] 2013-02-01 08:39:39.326329 [DEBUG] sofia.c:3581 dialplan [XML] 2013-02-01 08:39:39.326340 [DEBUG] sofia.c:3581 dtmf-duration [2000] 2013-02-01 08:39:39.326355 [DEBUG] sofia.c:3581 inbound-codec-prefs [G722,PCMU,PCMA,GSM] 2013-02-01 08:39:39.326366 [DEBUG] sofia.c:3581 outbound-codec-prefs [G722,PCMU,PCMA,GSM] 2013-02-01 08:39:39.326376 [DEBUG] sofia.c:3581 use-rtp-timer [true] 2013-02-01 08:39:39.326387 [DEBUG] sofia.c:3581 rtp-timer-name [soft] 2013-02-01 08:39:39.326395 [DEBUG] sofia.c:3581 rtp-ip [::1] 2013-02-01 08:39:39.326401 [DEBUG] sofia.c:3581 sip-ip [::1] 2013-02-01 08:39:39.326412 [DEBUG] sofia.c:3581 hold-music [%{bong-ring}] 2013-02-01 08:39:39.326420 [DEBUG] sofia.c:3581 apply-inbound-acl [domains] 2013-02-01 08:39:39.326516 [DEBUG] sofia.c:3581 record-template [/usr/local/freeswitch/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav] 2013-02-01 08:39:39.326536 [DEBUG] sofia.c:3581 manage-presence [true] 2013-02-01 08:39:39.326545 [DEBUG] sofia.c:3581 inbound-codec-negotiation [generous] 2013-02-01 08:39:39.326556 [DEBUG] sofia.c:3581 tls [false] 2013-02-01 08:39:39.326570 [DEBUG] sofia.c:3581 tls-bind-params [transport=tls] 2013-02-01 08:39:39.326579 [DEBUG] sofia.c:3581 tls-sip-port [5061] 2013-02-01 08:39:39.326588 [DEBUG] sofia.c:3581 tls-cert-dir [/usr/local/freeswitch/conf/ssl] 2013-02-01 08:39:39.326600 [DEBUG] sofia.c:3581 tls-version [tlsv1] 2013-02-01 08:39:39.326608 [DEBUG] sofia.c:3581 inbound-late-negotiation [true] 2013-02-01 08:39:39.326614 [DEBUG] sofia.c:3581 inbound-zrtp-passthru [true] 2013-02-01 08:39:39.326619 [DEBUG] sofia.c:3581 nonce-ttl [60] 2013-02-01 08:39:39.326628 [DEBUG] sofia.c:3581 auth-calls [true] 2013-02-01 08:39:39.326635 [DEBUG] sofia.c:3581 auth-all-packets [false] 2013-02-01 08:39:39.326642 [DEBUG] sofia.c:3581 rtp-timeout-sec [300] 2013-02-01 08:39:39.326654 [DEBUG] sofia.c:3581 rtp-hold-timeout-sec [1800] 2013-02-01 08:39:39.326665 [DEBUG] sofia.c:3581 force-register-domain [XXX.XXX.XXX.10] 2013-02-01 08:39:39.326670 [DEBUG] sofia.c:3581 force-register-db-domain [XXX.XXX.XXX.10] 2013-02-01 08:39:39.326777 [NOTICE] sofia.c:4643 Started Profile internal-ipv6 [sofia_reg_internal-ipv6] 2013-02-01 08:39:39.326796 [DEBUG] sofia.c:2311 Creating agent for internal-ipv6 2013-02-01 08:39:39.326817 [DEBUG] sofia.c:3581 debug [0] 2013-02-01 08:39:39.326825 [DEBUG] sofia.c:3581 sip-trace [yes] 2013-02-01 08:39:39.326829 [DEBUG] sofia.c:3581 sip-capture [no] 2013-02-01 08:39:39.326848 [DEBUG] sofia.c:3581 watchdog-enabled [no] 2013-02-01 08:39:39.326854 [DEBUG] sofia.c:3581 watchdog-step-timeout [30000] 2013-02-01 08:39:39.326859 [DEBUG] sofia.c:3581 watchdog-event-timeout [30000] 2013-02-01 08:39:39.326863 [DEBUG] sofia.c:3581 log-auth-failures [false] 2013-02-01 08:39:39.326867 [DEBUG] sofia.c:3581 forward-unsolicited-mwi-notify [false] 2013-02-01 08:39:39.326871 [DEBUG] sofia.c:3581 context [default] 2013-02-01 08:39:39.326879 [DEBUG] sofia.c:3581 rfc2833-pt [101] 2013-02-01 08:39:39.326887 [DEBUG] sofia.c:3581 sip-port [5060] 2013-02-01 08:39:39.326893 [DEBUG] sofia.c:3581 dialplan [XML] 2013-02-01 08:39:39.326905 [DEBUG] sofia.c:3581 dtmf-duration [2000] 2013-02-01 08:39:39.326913 [DEBUG] sofia.c:3581 inbound-codec-prefs [G722,PCMU,PCMA,GSM] 2013-02-01 08:39:39.326929 [DEBUG] sofia.c:3581 outbound-codec-prefs [G722,PCMU,PCMA,GSM] 2013-02-01 08:39:39.326941 [DEBUG] sofia.c:3581 rtp-timer-name [soft] 2013-02-01 08:39:39.326949 [DEBUG] sofia.c:3581 rtp-ip [XXX.XXX.XXX.10] 2013-02-01 08:39:39.326955 [DEBUG] sofia.c:3581 sip-ip [XXX.XXX.XXX.10] 2013-02-01 08:39:39.326963 [DEBUG] sofia.c:3581 hold-music [%{bong-ring}] 2013-02-01 08:39:39.326969 [DEBUG] sofia.c:3581 apply-nat-acl [nat.auto] 2013-02-01 08:39:39.326982 [DEBUG] sofia.c:3581 local-network-acl [localnet.auto] 2013-02-01 08:39:39.326990 [DEBUG] sofia.c:3581 apply-register-acl [domains] 2013-02-01 08:39:39.326998 [DEBUG] sofia.c:3581 record-path [/usr/local/freeswitch/recordings] 2013-02-01 08:39:39.327005 [DEBUG] sofia.c:3581 record-template [${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav] 2013-02-01 08:39:39.327012 [DEBUG] sofia.c:3581 manage-presence [true] 2013-02-01 08:39:39.327021 [DEBUG] sofia.c:3581 presence-hosts [XXX.XXX.XXX.10,XXX.XXX.XXX.10] 2013-02-01 08:39:39.327027 [DEBUG] sofia.c:3581 presence-privacy [false] 2013-02-01 08:39:39.327034 [DEBUG] sofia.c:3581 inbound-codec-negotiation [generous] 2013-02-01 08:39:39.327041 [DEBUG] sofia.c:3581 tls [false] 2013-02-01 08:39:39.327051 [DEBUG] sofia.c:3581 tls-only [false] 2013-02-01 08:39:39.327059 [DEBUG] sofia.c:3581 tls-bind-params [transport=tls] 2013-02-01 08:39:39.327070 [DEBUG] sofia.c:3581 tls-sip-port [5061] 2013-02-01 08:39:39.327082 [DEBUG] sofia.c:3581 tls-cert-dir [/usr/local/freeswitch/conf/ssl] 2013-02-01 08:39:39.327093 [DEBUG] sofia.c:3581 tls-passphrase [] 2013-02-01 08:39:39.327106 [DEBUG] sofia.c:3581 tls-verify-date [true] 2013-02-01 08:39:39.327115 [DEBUG] sofia.c:3581 tls-verify-policy [none] 2013-02-01 08:39:39.327125 [DEBUG] sofia.c:3581 tls-verify-depth [2] 2013-02-01 08:39:39.327150 [DEBUG] sofia.c:3581 tls-verify-in-subjects [] 2013-02-01 08:39:39.327161 [DEBUG] sofia.c:3581 tls-version [tlsv1] 2013-02-01 08:39:39.327170 [DEBUG] sofia.c:3581 inbound-late-negotiation [true] 2013-02-01 08:39:39.327175 [DEBUG] sofia.c:3581 inbound-zrtp-passthru [true] 2013-02-01 08:39:39.327180 [DEBUG] sofia.c:3581 nonce-ttl [60] 2013-02-01 08:39:39.327187 [DEBUG] sofia.c:3581 auth-calls [true] 2013-02-01 08:39:39.327194 [DEBUG] sofia.c:3581 inbound-reg-force-matching-username [true] 2013-02-01 08:39:39.327205 [DEBUG] sofia.c:3581 auth-all-packets [false] 2013-02-01 08:39:39.327215 [DEBUG] sofia.c:3581 ext-rtp-ip [auto-nat] 2013-02-01 08:39:39.327220 [DEBUG] sofia.c:3581 ext-sip-ip [auto-nat] 2013-02-01 08:39:39.327228 [DEBUG] sofia.c:3581 rtp-timeout-sec [300] 2013-02-01 08:39:39.327234 [DEBUG] sofia.c:3581 rtp-hold-timeout-sec [1800] 2013-02-01 08:39:39.327242 [DEBUG] sofia.c:3581 force-register-domain [XXX.XXX.XXX.10] 2013-02-01 08:39:39.327249 [DEBUG] sofia.c:3581 force-subscription-domain [XXX.XXX.XXX.10] 2013-02-01 08:39:39.327255 [DEBUG] sofia.c:3581 force-register-db-domain [XXX.XXX.XXX.10] 2013-02-01 08:39:39.327260 [DEBUG] sofia.c:3581 challenge-realm [auto_from] 2013-02-01 08:39:39.327370 [NOTICE] sofia.c:4643 Started Profile internal [sofia_reg_internal] 2013-02-01 08:39:39.327379 [DEBUG] sofia.c:2311 Creating agent for internal 2013-02-01 08:39:39.327392 [DEBUG] mod_sofia.c:5741 Waiting for profiles to start 2013-02-01 08:39:39.331092 [DEBUG] sofia.c:2401 Created agent for external 2013-02-01 08:39:39.331092 [DEBUG] sofia.c:2401 Created agent for internal 2013-02-01 08:39:39.331092 [DEBUG] sofia.c:2401 Created agent for internal-ipv6 2013-02-01 08:39:39.331110 [DEBUG] sofia.c:2444 Set params for external 2013-02-01 08:39:39.331110 [DEBUG] sofia.c:2444 Set params for internal 2013-02-01 08:39:39.331112 [DEBUG] sofia.c:2488 Activated db for external 2013-02-01 08:39:39.331113 [DEBUG] sofia.c:2488 Activated db for internal 2013-02-01 08:39:39.331112 [DEBUG] sofia.c:2444 Set params for internal-ipv6 2013-02-01 08:39:39.331115 [DEBUG] sofia.c:2488 Activated db for internal-ipv6 2013-02-01 08:39:39.331728 [INFO] switch_core_sqldb.c:1329 sofia:external Starting SQL thread. 2013-02-01 08:39:39.331733 [INFO] switch_core_sqldb.c:1329 sofia:internal Starting SQL thread. 2013-02-01 08:39:39.331853 [DEBUG] sofia.c:2525 Starting thread for external 2013-02-01 08:39:39.331855 [DEBUG] sofia.c:2525 Starting thread for internal 2013-02-01 08:39:39.331875 [DEBUG] sofia.c:2211 Launching worker thread for internal 2013-02-01 08:39:39.331875 [DEBUG] sofia.c:2211 Launching worker thread for external 2013-02-01 08:39:39.331939 [NOTICE] sofia.c:3237 Adding Alias [XXX.XXX.XXX.10] for profile [internal] 2013-02-01 08:39:39.331935 [INFO] switch_core_sqldb.c:1329 sofia:internal-ipv6 Starting SQL thread. 2013-02-01 08:39:39.331947 [NOTICE] sofia_reg.c:2992 Added gateway 'thinktel' to profile 'external' 2013-02-01 08:39:39.332008 [DEBUG] sofia.c:2525 Starting thread for internal-ipv6 2013-02-01 08:39:39.332014 [DEBUG] sofia.c:2211 Launching worker thread for internal-ipv6 2013-02-01 08:39:40.827758 [NOTICE] switch_loadable_module.c:146 Adding Endpoint 'sofia' 2013-02-01 08:39:40.827785 [NOTICE] switch_loadable_module.c:146 Adding Endpoint 'rtp' 2013-02-01 08:39:40.827804 [NOTICE] switch_loadable_module.c:254 Adding Application 'sofia_sla' 2013-02-01 08:39:40.827824 [NOTICE] switch_loadable_module.c:298 Adding API Function 'sofia' 2013-02-01 08:39:40.827845 [NOTICE] switch_loadable_module.c:298 Adding API Function 'sofia_gateway_data' 2013-02-01 08:39:40.827875 [NOTICE] switch_loadable_module.c:298 Adding API Function 'sofia_username_of' 2013-02-01 08:39:40.827896 [NOTICE] switch_loadable_module.c:298 Adding API Function 'sofia_contact' 2013-02-01 08:39:40.827920 [NOTICE] switch_loadable_module.c:298 Adding API Function 'sofia_count_reg' 2013-02-01 08:39:40.827947 [NOTICE] switch_loadable_module.c:298 Adding API Function 'sofia_dig' 2013-02-01 08:39:40.827972 [NOTICE] switch_loadable_module.c:403 Adding Chat interface 'sip' 2013-02-01 08:39:40.827995 [NOTICE] switch_loadable_module.c:447 Adding Management interface 'mod_sofia' OID[.1.3.6.1.4.1.27880.1001] 2013-02-01 08:39:40.828488 [NOTICE] switch_loadable_module.c:146 Adding Endpoint 'loopback' 2013-02-01 08:39:40.828511 [NOTICE] switch_loadable_module.c:254 Adding Application 'unloop' 2013-02-01 08:39:40.829818 [NOTICE] switch_loadable_module.c:298 Adding API Function 'acl' 2013-02-01 08:39:40.829844 [NOTICE] switch_loadable_module.c:298 Adding API Function 'alias' 2013-02-01 08:39:40.829862 [NOTICE] switch_loadable_module.c:298 Adding API Function 'banner' 2013-02-01 08:39:40.829896 [NOTICE] switch_loadable_module.c:298 Adding API Function 'bgapi' 2013-02-01 08:39:40.829920 [NOTICE] switch_loadable_module.c:298 Adding API Function 'bg_system' 2013-02-01 08:39:40.829941 [NOTICE] switch_loadable_module.c:298 Adding API Function 'break' 2013-02-01 08:39:40.829966 [NOTICE] switch_loadable_module.c:298 Adding API Function 'complete' 2013-02-01 08:39:40.829989 [NOTICE] switch_loadable_module.c:298 Adding API Function 'cond' 2013-02-01 08:39:40.830010 [NOTICE] switch_loadable_module.c:298 Adding API Function 'console_complete' 2013-02-01 08:39:40.830027 [NOTICE] switch_loadable_module.c:298 Adding API Function 'console_complete_xml' 2013-02-01 08:39:40.830054 [NOTICE] switch_loadable_module.c:298 Adding API Function 'create_uuid' 2013-02-01 08:39:40.830078 [NOTICE] switch_loadable_module.c:298 Adding API Function 'db_cache' 2013-02-01 08:39:40.830102 [NOTICE] switch_loadable_module.c:298 Adding API Function 'domain_exists' 2013-02-01 08:39:40.830118 [NOTICE] switch_loadable_module.c:298 Adding API Function 'echo' 2013-02-01 08:39:40.830133 [NOTICE] switch_loadable_module.c:298 Adding API Function 'escape' 2013-02-01 08:39:40.830152 [NOTICE] switch_loadable_module.c:298 Adding API Function 'eval' 2013-02-01 08:39:40.830175 [NOTICE] switch_loadable_module.c:298 Adding API Function 'expand' 2013-02-01 08:39:40.830198 [NOTICE] switch_loadable_module.c:298 Adding API Function 'find_user_xml' 2013-02-01 08:39:40.830222 [NOTICE] switch_loadable_module.c:298 Adding API Function 'fsctl' 2013-02-01 08:39:40.830247 [NOTICE] switch_loadable_module.c:298 Adding API Function '...' 2013-02-01 08:39:40.830271 [NOTICE] switch_loadable_module.c:298 Adding API Function 'shutdown' 2013-02-01 08:39:40.830295 [NOTICE] switch_loadable_module.c:298 Adding API Function 'version' 2013-02-01 08:39:40.830325 [NOTICE] switch_loadable_module.c:298 Adding API Function 'global_getvar' 2013-02-01 08:39:40.830346 [NOTICE] switch_loadable_module.c:298 Adding API Function 'global_setvar' 2013-02-01 08:39:40.830366 [NOTICE] switch_loadable_module.c:298 Adding API Function 'group_call' 2013-02-01 08:39:40.830390 [NOTICE] switch_loadable_module.c:298 Adding API Function 'help' 2013-02-01 08:39:40.830412 [NOTICE] switch_loadable_module.c:298 Adding API Function 'host_lookup' 2013-02-01 08:39:40.830438 [NOTICE] switch_loadable_module.c:298 Adding API Function 'hostname' 2013-02-01 08:39:40.830461 [NOTICE] switch_loadable_module.c:298 Adding API Function 'switchname' 2013-02-01 08:39:40.830478 [NOTICE] switch_loadable_module.c:298 Adding API Function 'hupall' 2013-02-01 08:39:40.830503 [NOTICE] switch_loadable_module.c:298 Adding API Function 'in_group' 2013-02-01 08:39:40.830524 [NOTICE] switch_loadable_module.c:298 Adding API Function 'is_lan_addr' 2013-02-01 08:39:40.830546 [NOTICE] switch_loadable_module.c:298 Adding API Function 'limit_usage' 2013-02-01 08:39:40.830571 [NOTICE] switch_loadable_module.c:298 Adding API Function 'limit_hash_usage' 2013-02-01 08:39:40.830597 [NOTICE] switch_loadable_module.c:298 Adding API Function 'limit_status' 2013-02-01 08:39:40.830622 [NOTICE] switch_loadable_module.c:298 Adding API Function 'limit_reset' 2013-02-01 08:39:40.830646 [NOTICE] switch_loadable_module.c:298 Adding API Function 'limit_interval_reset' 2013-02-01 08:39:40.830669 [NOTICE] switch_loadable_module.c:298 Adding API Function 'list_users' 2013-02-01 08:39:40.830688 [NOTICE] switch_loadable_module.c:298 Adding API Function 'load' 2013-02-01 08:39:40.830714 [NOTICE] switch_loadable_module.c:298 Adding API Function 'log' 2013-02-01 08:39:40.830737 [NOTICE] switch_loadable_module.c:298 Adding API Function 'md5' 2013-02-01 08:39:40.830761 [NOTICE] switch_loadable_module.c:298 Adding API Function 'module_exists' 2013-02-01 08:39:40.830785 [NOTICE] switch_loadable_module.c:298 Adding API Function 'msleep' 2013-02-01 08:39:40.830808 [NOTICE] switch_loadable_module.c:298 Adding API Function 'nat_map' 2013-02-01 08:39:40.830831 [NOTICE] switch_loadable_module.c:298 Adding API Function 'originate' 2013-02-01 08:39:40.830856 [NOTICE] switch_loadable_module.c:298 Adding API Function 'pause' 2013-02-01 08:39:40.830882 [NOTICE] switch_loadable_module.c:298 Adding API Function 'regex' 2013-02-01 08:39:40.830906 [NOTICE] switch_loadable_module.c:298 Adding API Function 'reloadacl' 2013-02-01 08:39:40.830931 [NOTICE] switch_loadable_module.c:298 Adding API Function 'reload' 2013-02-01 08:39:40.830953 [NOTICE] switch_loadable_module.c:298 Adding API Function 'reloadxml' 2013-02-01 08:39:40.830979 [NOTICE] switch_loadable_module.c:298 Adding API Function 'replace' 2013-02-01 08:39:40.831002 [NOTICE] switch_loadable_module.c:298 Adding API Function 'say_string' 2013-02-01 08:39:40.831025 [NOTICE] switch_loadable_module.c:298 Adding API Function 'sched_api' 2013-02-01 08:39:40.831047 [NOTICE] switch_loadable_module.c:298 Adding API Function 'sched_broadcast' 2013-02-01 08:39:40.831079 [NOTICE] switch_loadable_module.c:298 Adding API Function 'sched_del' 2013-02-01 08:39:40.831101 [NOTICE] switch_loadable_module.c:298 Adding API Function 'sched_hangup' 2013-02-01 08:39:40.831128 [NOTICE] switch_loadable_module.c:298 Adding API Function 'sched_transfer' 2013-02-01 08:39:40.831152 [NOTICE] switch_loadable_module.c:298 Adding API Function 'show' 2013-02-01 08:39:40.831178 [NOTICE] switch_loadable_module.c:298 Adding API Function 'sql_escape' 2013-02-01 08:39:40.831202 [NOTICE] switch_loadable_module.c:298 Adding API Function 'status' 2013-02-01 08:39:40.831225 [NOTICE] switch_loadable_module.c:298 Adding API Function 'strftime_tz' 2013-02-01 08:39:40.831249 [NOTICE] switch_loadable_module.c:298 Adding API Function 'stun' 2013-02-01 08:39:40.831274 [NOTICE] switch_loadable_module.c:298 Adding API Function 'system' 2013-02-01 08:39:40.831299 [NOTICE] switch_loadable_module.c:298 Adding API Function 'time_test' 2013-02-01 08:39:40.831322 [NOTICE] switch_loadable_module.c:298 Adding API Function 'timer_test' 2013-02-01 08:39:40.831346 [NOTICE] switch_loadable_module.c:298 Adding API Function 'tone_detect' 2013-02-01 08:39:40.831368 [NOTICE] switch_loadable_module.c:298 Adding API Function 'unload' 2013-02-01 08:39:40.831394 [NOTICE] switch_loadable_module.c:298 Adding API Function 'unsched_api' 2013-02-01 08:39:40.831416 [NOTICE] switch_loadable_module.c:298 Adding API Function 'reg_url' 2013-02-01 08:39:40.831441 [NOTICE] switch_loadable_module.c:298 Adding API Function 'url_decode' 2013-02-01 08:39:40.831462 [NOTICE] switch_loadable_module.c:298 Adding API Function 'url_encode' 2013-02-01 08:39:40.831486 [NOTICE] switch_loadable_module.c:298 Adding API Function 'user_data' 2013-02-01 08:39:40.831509 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_early_ok' 2013-02-01 08:39:40.831534 [NOTICE] switch_loadable_module.c:298 Adding API Function 'user_exists' 2013-02-01 08:39:40.831556 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_answer' 2013-02-01 08:39:40.831582 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_audio' 2013-02-01 08:39:40.831605 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_break' 2013-02-01 08:39:40.831629 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_bridge' 2013-02-01 08:39:40.831651 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_broadcast' 2013-02-01 08:39:40.831676 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_buglist' 2013-02-01 08:39:40.831699 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_chat' 2013-02-01 08:39:40.831722 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_debug_audio' 2013-02-01 08:39:40.831747 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_deflect' 2013-02-01 08:39:40.831772 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_displace' 2013-02-01 08:39:40.831795 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_display' 2013-02-01 08:39:40.831820 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_dump' 2013-02-01 08:39:40.831844 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_exists' 2013-02-01 08:39:40.831868 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_fileman' 2013-02-01 08:39:40.831891 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_flush_dtmf' 2013-02-01 08:39:40.831916 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_getvar' 2013-02-01 08:39:40.831939 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_hold' 2013-02-01 08:39:40.831963 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_kill' 2013-02-01 08:39:40.831986 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_send_message' 2013-02-01 08:39:40.832009 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_send_info' 2013-02-01 08:39:40.832036 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_video_refresh' 2013-02-01 08:39:40.832061 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_outgoing_answer' 2013-02-01 08:39:40.832085 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_limit' 2013-02-01 08:39:40.832110 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_limit_release' 2013-02-01 08:39:40.832134 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_limit_release' 2013-02-01 08:39:40.832159 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_loglevel' 2013-02-01 08:39:40.832184 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_media' 2013-02-01 08:39:40.832209 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_media_reneg' 2013-02-01 08:39:40.832232 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_park' 2013-02-01 08:39:40.832258 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_pause' 2013-02-01 08:39:40.832285 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_phone_event' 2013-02-01 08:39:40.832310 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_pre_answer' 2013-02-01 08:39:40.832336 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_preprocess' 2013-02-01 08:39:40.832358 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_record' 2013-02-01 08:39:40.832383 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_recovery_refresh' 2013-02-01 08:39:40.832404 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_recv_dtmf' 2013-02-01 08:39:40.832428 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_send_dtmf' 2013-02-01 08:39:40.832452 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_session_heartbeat' 2013-02-01 08:39:40.832476 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_setvar_multi' 2013-02-01 08:39:40.832500 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_setvar' 2013-02-01 08:39:40.832521 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_transfer' 2013-02-01 08:39:40.832547 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_dual_transfer' 2013-02-01 08:39:40.832571 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_simplify' 2013-02-01 08:39:40.832596 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_jitterbuffer' 2013-02-01 08:39:40.832619 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_zombie_exec' 2013-02-01 08:39:40.832658 [NOTICE] switch_loadable_module.c:298 Adding API Function 'xml_flush_cache' 2013-02-01 08:39:40.832678 [NOTICE] switch_loadable_module.c:298 Adding API Function 'xml_locate' 2013-02-01 08:39:40.832705 [NOTICE] switch_loadable_module.c:298 Adding API Function 'xml_wrap' 2013-02-01 08:39:40.832728 [NOTICE] switch_loadable_module.c:298 Adding API Function 'file_exists' 2013-02-01 08:39:40.833425 [NOTICE] switch_loadable_module.c:254 Adding Application 'conference' 2013-02-01 08:39:40.833445 [NOTICE] switch_loadable_module.c:254 Adding Application 'conference_set_auto_outcall' 2013-02-01 08:39:40.833469 [NOTICE] switch_loadable_module.c:298 Adding API Function 'conference' 2013-02-01 08:39:40.833493 [NOTICE] switch_loadable_module.c:403 Adding Chat interface 'conf' 2013-02-01 08:39:40.834566 [NOTICE] switch_loadable_module.c:254 Adding Application 'db' 2013-02-01 08:39:40.834588 [NOTICE] switch_loadable_module.c:254 Adding Application 'group' 2013-02-01 08:39:40.834606 [NOTICE] switch_loadable_module.c:298 Adding API Function 'db' 2013-02-01 08:39:40.834624 [NOTICE] switch_loadable_module.c:298 Adding API Function 'group' 2013-02-01 08:39:40.834644 [NOTICE] switch_loadable_module.c:473 Adding Limit interface 'db' 2013-02-01 08:39:41.333659 [NOTICE] switch_loadable_module.c:146 Adding Endpoint 'error' 2013-02-01 08:39:41.333688 [NOTICE] switch_loadable_module.c:146 Adding Endpoint 'group' 2013-02-01 08:39:41.333708 [NOTICE] switch_loadable_module.c:146 Adding Endpoint 'user' 2013-02-01 08:39:41.333728 [NOTICE] switch_loadable_module.c:146 Adding Endpoint 'pickup' 2013-02-01 08:39:41.333746 [NOTICE] switch_loadable_module.c:214 Adding Dialplan 'inline' 2013-02-01 08:39:41.333770 [NOTICE] switch_loadable_module.c:254 Adding Application 'blind_transfer_ack' 2013-02-01 08:39:41.333797 [NOTICE] switch_loadable_module.c:254 Adding Application 'bind_digit_action' 2013-02-01 08:39:41.333823 [NOTICE] switch_loadable_module.c:254 Adding Application 'capture' 2013-02-01 08:39:41.333848 [NOTICE] switch_loadable_module.c:254 Adding Application 'clear_digit_action' 2013-02-01 08:39:41.333870 [NOTICE] switch_loadable_module.c:254 Adding Application 'digit_action_set_realm' 2013-02-01 08:39:41.333894 [NOTICE] switch_loadable_module.c:254 Adding Application 'privacy' 2013-02-01 08:39:41.333914 [NOTICE] switch_loadable_module.c:254 Adding Application 'set_audio_level' 2013-02-01 08:39:41.333934 [NOTICE] switch_loadable_module.c:254 Adding Application 'set_mute' 2013-02-01 08:39:41.333958 [NOTICE] switch_loadable_module.c:254 Adding Application 'flush_dtmf' 2013-02-01 08:39:41.333984 [NOTICE] switch_loadable_module.c:254 Adding Application 'hold' 2013-02-01 08:39:41.334007 [NOTICE] switch_loadable_module.c:254 Adding Application 'unhold' 2013-02-01 08:39:41.334022 [NOTICE] switch_loadable_module.c:254 Adding Application 'mutex' 2013-02-01 08:39:41.334045 [NOTICE] switch_loadable_module.c:254 Adding Application 'page' 2013-02-01 08:39:41.334067 [NOTICE] switch_loadable_module.c:254 Adding Application 'transfer' 2013-02-01 08:39:41.334091 [NOTICE] switch_loadable_module.c:254 Adding Application 'check_acl' 2013-02-01 08:39:41.334120 [NOTICE] switch_loadable_module.c:254 Adding Application 'verbose_events' 2013-02-01 08:39:41.334140 [NOTICE] switch_loadable_module.c:254 Adding Application 'cng_plc' 2013-02-01 08:39:41.334164 [NOTICE] switch_loadable_module.c:254 Adding Application 'early_hangup' 2013-02-01 08:39:41.334182 [NOTICE] switch_loadable_module.c:254 Adding Application 'sleep' 2013-02-01 08:39:41.334208 [NOTICE] switch_loadable_module.c:254 Adding Application 'delay_echo' 2013-02-01 08:39:41.334233 [NOTICE] switch_loadable_module.c:254 Adding Application 'strftime' 2013-02-01 08:39:41.334251 [NOTICE] switch_loadable_module.c:254 Adding Application 'phrase' 2013-02-01 08:39:41.334284 [NOTICE] switch_loadable_module.c:254 Adding Application 'eval' 2013-02-01 08:39:41.334309 [NOTICE] switch_loadable_module.c:254 Adding Application 'stop' 2013-02-01 08:39:41.334333 [NOTICE] switch_loadable_module.c:254 Adding Application 'set_zombie_exec' 2013-02-01 08:39:41.334365 [NOTICE] switch_loadable_module.c:254 Adding Application 'pre_answer' 2013-02-01 08:39:41.334388 [NOTICE] switch_loadable_module.c:254 Adding Application 'answer' 2013-02-01 08:39:41.334414 [NOTICE] switch_loadable_module.c:254 Adding Application 'wait_for_answer' 2013-02-01 08:39:41.334438 [NOTICE] switch_loadable_module.c:254 Adding Application 'hangup' 2013-02-01 08:39:41.334463 [NOTICE] switch_loadable_module.c:254 Adding Application 'set_name' 2013-02-01 08:39:41.334487 [NOTICE] switch_loadable_module.c:254 Adding Application 'presence' 2013-02-01 08:39:41.334513 [NOTICE] switch_loadable_module.c:254 Adding Application 'log' 2013-02-01 08:39:41.334538 [NOTICE] switch_loadable_module.c:254 Adding Application 'info' 2013-02-01 08:39:41.334562 [NOTICE] switch_loadable_module.c:254 Adding Application 'event' 2013-02-01 08:39:41.334587 [NOTICE] switch_loadable_module.c:254 Adding Application 'sound_test' 2013-02-01 08:39:41.334610 [NOTICE] switch_loadable_module.c:254 Adding Application 'export' 2013-02-01 08:39:41.334635 [NOTICE] switch_loadable_module.c:254 Adding Application 'bridge_export' 2013-02-01 08:39:41.334660 [NOTICE] switch_loadable_module.c:254 Adding Application 'set' 2013-02-01 08:39:41.334684 [NOTICE] switch_loadable_module.c:254 Adding Application 'multiset' 2013-02-01 08:39:41.334709 [NOTICE] switch_loadable_module.c:254 Adding Application 'push' 2013-02-01 08:39:41.334731 [NOTICE] switch_loadable_module.c:254 Adding Application 'unshift' 2013-02-01 08:39:41.334756 [NOTICE] switch_loadable_module.c:254 Adding Application 'set_global' 2013-02-01 08:39:41.334780 [NOTICE] switch_loadable_module.c:254 Adding Application 'set_profile_var' 2013-02-01 08:39:41.334805 [NOTICE] switch_loadable_module.c:254 Adding Application 'unset' 2013-02-01 08:39:41.334829 [NOTICE] switch_loadable_module.c:254 Adding Application 'ring_ready' 2013-02-01 08:39:41.334855 [NOTICE] switch_loadable_module.c:254 Adding Application 'remove_bugs' 2013-02-01 08:39:41.334879 [NOTICE] switch_loadable_module.c:254 Adding Application 'break' 2013-02-01 08:39:41.334903 [NOTICE] switch_loadable_module.c:254 Adding Application 'detect_speech' 2013-02-01 08:39:41.334927 [NOTICE] switch_loadable_module.c:254 Adding Application 'play_and_detect_speech' 2013-02-01 08:39:41.334952 [NOTICE] switch_loadable_module.c:254 Adding Application 'ivr' 2013-02-01 08:39:41.334977 [NOTICE] switch_loadable_module.c:254 Adding Application 'redirect' 2013-02-01 08:39:41.335003 [NOTICE] switch_loadable_module.c:254 Adding Application 'video_refresh' 2013-02-01 08:39:41.335026 [NOTICE] switch_loadable_module.c:254 Adding Application 'send_info' 2013-02-01 08:39:41.335049 [NOTICE] switch_loadable_module.c:254 Adding Application 'jitterbuffer' 2013-02-01 08:39:41.335075 [NOTICE] switch_loadable_module.c:254 Adding Application 'send_display' 2013-02-01 08:39:41.335100 [NOTICE] switch_loadable_module.c:254 Adding Application 'respond' 2013-02-01 08:39:41.335124 [NOTICE] switch_loadable_module.c:254 Adding Application 'deflect' 2013-02-01 08:39:41.335172 [NOTICE] switch_loadable_module.c:254 Adding Application 'recovery_refresh' 2013-02-01 08:39:41.335195 [NOTICE] switch_loadable_module.c:254 Adding Application 'queue_dtmf' 2013-02-01 08:39:41.335220 [NOTICE] switch_loadable_module.c:254 Adding Application 'send_dtmf' 2013-02-01 08:39:41.335247 [NOTICE] switch_loadable_module.c:254 Adding Application 'sched_cancel' 2013-02-01 08:39:41.335275 [NOTICE] switch_loadable_module.c:254 Adding Application 'sched_hangup' 2013-02-01 08:39:41.335301 [NOTICE] switch_loadable_module.c:254 Adding Application 'sched_broadcast' 2013-02-01 08:39:41.335326 [NOTICE] switch_loadable_module.c:254 Adding Application 'sched_transfer' 2013-02-01 08:39:41.335353 [NOTICE] switch_loadable_module.c:254 Adding Application 'execute_extension' 2013-02-01 08:39:41.335378 [NOTICE] switch_loadable_module.c:254 Adding Application 'sched_heartbeat' 2013-02-01 08:39:41.335404 [NOTICE] switch_loadable_module.c:254 Adding Application 'enable_heartbeat' 2013-02-01 08:39:41.335428 [NOTICE] switch_loadable_module.c:254 Adding Application 'mkdir' 2013-02-01 08:39:41.335455 [NOTICE] switch_loadable_module.c:254 Adding Application 'rename' 2013-02-01 08:39:41.335479 [NOTICE] switch_loadable_module.c:254 Adding Application 'soft_hold' 2013-02-01 08:39:41.335504 [NOTICE] switch_loadable_module.c:254 Adding Application 'bind_meta_app' 2013-02-01 08:39:41.335528 [NOTICE] switch_loadable_module.c:254 Adding Application 'unbind_meta_app' 2013-02-01 08:39:41.335555 [NOTICE] switch_loadable_module.c:254 Adding Application 'block_dfmf' 2013-02-01 08:39:41.335579 [NOTICE] switch_loadable_module.c:254 Adding Application 'unblock_dtmf' 2013-02-01 08:39:41.335604 [NOTICE] switch_loadable_module.c:254 Adding Application 'intercept' 2013-02-01 08:39:41.335627 [NOTICE] switch_loadable_module.c:254 Adding Application 'eavesdrop' 2013-02-01 08:39:41.335652 [NOTICE] switch_loadable_module.c:254 Adding Application 'three_way' 2013-02-01 08:39:41.335677 [NOTICE] switch_loadable_module.c:254 Adding Application 'set_user' 2013-02-01 08:39:41.335702 [NOTICE] switch_loadable_module.c:254 Adding Application 'stop_dtmf' 2013-02-01 08:39:41.335726 [NOTICE] switch_loadable_module.c:254 Adding Application 'start_dtmf' 2013-02-01 08:39:41.335751 [NOTICE] switch_loadable_module.c:254 Adding Application 'stop_dtmf_generate' 2013-02-01 08:39:41.335775 [NOTICE] switch_loadable_module.c:254 Adding Application 'start_dtmf_generate' 2013-02-01 08:39:41.335802 [NOTICE] switch_loadable_module.c:254 Adding Application 'stop_tone_detect' 2013-02-01 08:39:41.335826 [NOTICE] switch_loadable_module.c:254 Adding Application 'fax_detect' 2013-02-01 08:39:41.335851 [NOTICE] switch_loadable_module.c:254 Adding Application 'tone_detect' 2013-02-01 08:39:41.335873 [NOTICE] switch_loadable_module.c:254 Adding Application 'echo' 2013-02-01 08:39:41.335898 [NOTICE] switch_loadable_module.c:254 Adding Application 'park' 2013-02-01 08:39:41.335925 [NOTICE] switch_loadable_module.c:254 Adding Application 'park_state' 2013-02-01 08:39:41.335949 [NOTICE] switch_loadable_module.c:254 Adding Application 'gentones' 2013-02-01 08:39:41.335974 [NOTICE] switch_loadable_module.c:254 Adding Application 'playback' 2013-02-01 08:39:41.336001 [NOTICE] switch_loadable_module.c:254 Adding Application 'endless_playback' 2013-02-01 08:39:41.336026 [NOTICE] switch_loadable_module.c:254 Adding Application 'att_xfer' 2013-02-01 08:39:41.336050 [NOTICE] switch_loadable_module.c:254 Adding Application 'read' 2013-02-01 08:39:41.336076 [NOTICE] switch_loadable_module.c:254 Adding Application 'play_and_get_digits' 2013-02-01 08:39:41.336101 [NOTICE] switch_loadable_module.c:254 Adding Application 'stop_record_session' 2013-02-01 08:39:41.336126 [NOTICE] switch_loadable_module.c:254 Adding Application 'record_session' 2013-02-01 08:39:41.336150 [NOTICE] switch_loadable_module.c:254 Adding Application 'record' 2013-02-01 08:39:41.336176 [NOTICE] switch_loadable_module.c:254 Adding Application 'preprocess' 2013-02-01 08:39:41.336201 [NOTICE] switch_loadable_module.c:254 Adding Application 'stop_displace_session' 2013-02-01 08:39:41.336225 [NOTICE] switch_loadable_module.c:254 Adding Application 'displace_session' 2013-02-01 08:39:41.336250 [NOTICE] switch_loadable_module.c:254 Adding Application 'speak' 2013-02-01 08:39:41.336277 [NOTICE] switch_loadable_module.c:254 Adding Application 'clear_speech_cache' 2013-02-01 08:39:41.336302 [NOTICE] switch_loadable_module.c:254 Adding Application 'bridge' 2013-02-01 08:39:41.336326 [NOTICE] switch_loadable_module.c:254 Adding Application 'system' 2013-02-01 08:39:41.336351 [NOTICE] switch_loadable_module.c:254 Adding Application 'say' 2013-02-01 08:39:41.336375 [NOTICE] switch_loadable_module.c:254 Adding Application 'wait_for_silence' 2013-02-01 08:39:41.336401 [NOTICE] switch_loadable_module.c:254 Adding Application 'session_loglevel' 2013-02-01 08:39:41.336424 [NOTICE] switch_loadable_module.c:254 Adding Application 'limit' 2013-02-01 08:39:41.336450 [NOTICE] switch_loadable_module.c:254 Adding Application 'limit_hash' 2013-02-01 08:39:41.336471 [NOTICE] switch_loadable_module.c:254 Adding Application 'limit_execute' 2013-02-01 08:39:41.336497 [NOTICE] switch_loadable_module.c:254 Adding Application 'limit_hash_execute' 2013-02-01 08:39:41.336522 [NOTICE] switch_loadable_module.c:254 Adding Application 'pickup' 2013-02-01 08:39:41.336547 [NOTICE] switch_loadable_module.c:298 Adding API Function 'strepoch' 2013-02-01 08:39:41.336570 [NOTICE] switch_loadable_module.c:298 Adding API Function 'page' 2013-02-01 08:39:41.336597 [NOTICE] switch_loadable_module.c:298 Adding API Function 'strmicroepoch' 2013-02-01 08:39:41.336620 [NOTICE] switch_loadable_module.c:298 Adding API Function 'chat' 2013-02-01 08:39:41.336645 [NOTICE] switch_loadable_module.c:298 Adding API Function 'strftime' 2013-02-01 08:39:41.336669 [NOTICE] switch_loadable_module.c:298 Adding API Function 'presence' 2013-02-01 08:39:41.336695 [NOTICE] switch_loadable_module.c:322 Adding File Format 'file_string' 2013-02-01 08:39:41.336719 [NOTICE] switch_loadable_module.c:403 Adding Chat interface 'event' 2013-02-01 08:39:41.336741 [NOTICE] switch_loadable_module.c:403 Adding Chat interface 'api' 2013-02-01 08:39:41.337009 [NOTICE] switch_loadable_module.c:298 Adding API Function 'expr' 2013-02-01 08:39:41.338416 [INFO] switch_core_sqldb.c:1329 fifo Starting SQL thread. 2013-02-01 08:39:41.339731 [INFO] mod_fifo.c:4260 cool_fifo at XXX.XXX.XXX.10 configured 2013-02-01 08:39:41.339855 [NOTICE] switch_loadable_module.c:254 Adding Application 'fifo' 2013-02-01 08:39:41.339883 [NOTICE] switch_loadable_module.c:254 Adding Application 'fifo_track_call' 2013-02-01 08:39:41.339921 [NOTICE] switch_loadable_module.c:298 Adding API Function 'fifo' 2013-02-01 08:39:41.339947 [NOTICE] switch_loadable_module.c:298 Adding API Function 'fifo_member' 2013-02-01 08:39:41.339971 [NOTICE] switch_loadable_module.c:298 Adding API Function 'fifo_add_outbound' 2013-02-01 08:39:41.339994 [NOTICE] switch_loadable_module.c:298 Adding API Function 'fifo_check_bridge' 2013-02-01 08:39:41.340341 [DEBUG] switch_scheduler.c:214 Added task 3 limit_hash_cleanup (mod_hash) to run at 1359726881 2013-02-01 08:39:41.340443 [NOTICE] switch_loadable_module.c:254 Adding Application 'hash' 2013-02-01 08:39:41.340462 [NOTICE] switch_loadable_module.c:298 Adding API Function 'hash' 2013-02-01 08:39:41.340482 [NOTICE] switch_loadable_module.c:298 Adding API Function 'hash_dump' 2013-02-01 08:39:41.340501 [NOTICE] switch_loadable_module.c:298 Adding API Function 'hash_remote' 2013-02-01 08:39:41.340532 [NOTICE] switch_loadable_module.c:473 Adding Limit interface 'hash' 2013-02-01 08:39:41.342166 [INFO] mod_voicemail.c:738 Added Profile default 2013-02-01 08:39:41.342215 [NOTICE] switch_loadable_module.c:254 Adding Application 'voicemail' 2013-02-01 08:39:41.342241 [NOTICE] switch_loadable_module.c:298 Adding API Function 'voicemail' 2013-02-01 08:39:41.342261 [NOTICE] switch_loadable_module.c:298 Adding API Function 'voicemail_inject' 2013-02-01 08:39:41.342282 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_inject' 2013-02-01 08:39:41.342307 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_boxcount' 2013-02-01 08:39:41.342330 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_prefs' 2013-02-01 08:39:41.342351 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_delete' 2013-02-01 08:39:41.342377 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_read' 2013-02-01 08:39:41.342402 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_list' 2013-02-01 08:39:41.342423 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_fsdb_auth_login' 2013-02-01 08:39:41.342449 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_fsdb_msg_count' 2013-02-01 08:39:41.342474 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_fsdb_msg_list' 2013-02-01 08:39:41.342499 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_fsdb_msg_get' 2013-02-01 08:39:41.342523 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_fsdb_msg_delete' 2013-02-01 08:39:41.342542 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_fsdb_msg_undelete' 2013-02-01 08:39:41.342571 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_fsdb_msg_email' 2013-02-01 08:39:41.342595 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_fsdb_msg_purge' 2013-02-01 08:39:41.342616 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_fsdb_msg_save' 2013-02-01 08:39:41.342641 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_fsdb_msg_forward' 2013-02-01 08:39:41.342666 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_fsdb_pref_greeting_set' 2013-02-01 08:39:41.342692 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_fsdb_pref_recname_set' 2013-02-01 08:39:41.342717 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_fsdb_pref_password_set' 2013-02-01 08:39:41.342957 [NOTICE] switch_loadable_module.c:254 Adding Application 'esf_page_group' 2013-02-01 08:39:41.343191 [NOTICE] switch_loadable_module.c:254 Adding Application 'play_fsv' 2013-02-01 08:39:41.343213 [NOTICE] switch_loadable_module.c:254 Adding Application 'record_fsv' 2013-02-01 08:39:41.343235 [NOTICE] switch_loadable_module.c:322 Adding File Format 'fsv' 2013-02-01 08:39:41.343455 [NOTICE] mod_cluechoo.c:83 Hello World! 2013-02-01 08:39:41.343477 [NOTICE] switch_loadable_module.c:254 Adding Application 'cluechoo' 2013-02-01 08:39:41.343502 [NOTICE] switch_loadable_module.c:298 Adding API Function 'cluechoo' 2013-02-01 08:39:41.343730 [NOTICE] switch_loadable_module.c:254 Adding Application 'valet_park' 2013-02-01 08:39:41.343768 [NOTICE] switch_loadable_module.c:298 Adding API Function 'valet_info' 2013-02-01 08:39:41.344066 [NOTICE] mod_httapi.c:2068 Profile [default] JSON Function [http://www.freeswitch.org/api/index.cgi] 2013-02-01 08:39:41.344112 [NOTICE] switch_loadable_module.c:254 Adding Application 'httapi' 2013-02-01 08:39:41.344137 [NOTICE] switch_loadable_module.c:298 Adding API Function 'httapi' 2013-02-01 08:39:41.344160 [NOTICE] switch_loadable_module.c:322 Adding File Format 'http' 2013-02-01 08:39:41.344381 [NOTICE] switch_loadable_module.c:214 Adding Dialplan 'XML' 2013-02-01 08:39:41.344594 [NOTICE] switch_loadable_module.c:146 Adding Endpoint 'SIP' 2013-02-01 08:39:41.344616 [NOTICE] switch_loadable_module.c:146 Adding Endpoint 'IAX2' 2013-02-01 08:39:41.344641 [NOTICE] switch_loadable_module.c:214 Adding Dialplan 'asterisk' 2013-02-01 08:39:41.344661 [NOTICE] switch_loadable_module.c:254 Adding Application 'Dial' 2013-02-01 08:39:41.344693 [NOTICE] switch_loadable_module.c:254 Adding Application 'Goto' 2013-02-01 08:39:41.344728 [NOTICE] switch_loadable_module.c:254 Adding Application 'AvoidingDeadlock' 2013-02-01 08:39:41.346415 [DEBUG] mod_spandsp_fax.c:192 FAX timer thread started. 2013-02-01 08:39:41.367006 [DEBUG] mod_spandsp.c:777 mod_spandsp loaded, using spandsp library version [20120902 163333] 2013-02-01 08:39:41.367035 [NOTICE] switch_loadable_module.c:146 Adding Endpoint 'modem' 2013-02-01 08:39:41.367062 [NOTICE] switch_loadable_module.c:187 Adding Codec DVI4 6 ADPCM (IMA) 16000hz 10ms 64000bps 2013-02-01 08:39:41.367069 [NOTICE] switch_loadable_module.c:187 Adding Codec DVI4 6 ADPCM (IMA) 16000hz 20ms 64000bps 2013-02-01 08:39:41.367074 [NOTICE] switch_loadable_module.c:187 Adding Codec DVI4 6 ADPCM (IMA) 16000hz 30ms 64000bps 2013-02-01 08:39:41.367079 [NOTICE] switch_loadable_module.c:187 Adding Codec DVI4 6 ADPCM (IMA) 16000hz 40ms 64000bps 2013-02-01 08:39:41.367083 [NOTICE] switch_loadable_module.c:187 Adding Codec DVI4 6 ADPCM (IMA) 16000hz 50ms 64000bps 2013-02-01 08:39:41.367092 [NOTICE] switch_loadable_module.c:187 Adding Codec DVI4 6 ADPCM (IMA) 16000hz 60ms 64000bps 2013-02-01 08:39:41.367097 [NOTICE] switch_loadable_module.c:187 Adding Codec DVI4 5 ADPCM (IMA) 8000hz 10ms 32000bps 2013-02-01 08:39:41.367101 [NOTICE] switch_loadable_module.c:187 Adding Codec DVI4 5 ADPCM (IMA) 8000hz 20ms 32000bps 2013-02-01 08:39:41.367105 [NOTICE] switch_loadable_module.c:187 Adding Codec DVI4 5 ADPCM (IMA) 8000hz 30ms 32000bps 2013-02-01 08:39:41.367110 [NOTICE] switch_loadable_module.c:187 Adding Codec DVI4 5 ADPCM (IMA) 8000hz 40ms 32000bps 2013-02-01 08:39:41.367115 [NOTICE] switch_loadable_module.c:187 Adding Codec DVI4 5 ADPCM (IMA) 8000hz 50ms 32000bps 2013-02-01 08:39:41.367119 [NOTICE] switch_loadable_module.c:187 Adding Codec DVI4 5 ADPCM (IMA) 8000hz 60ms 32000bps 2013-02-01 08:39:41.367123 [NOTICE] switch_loadable_module.c:187 Adding Codec DVI4 5 ADPCM (IMA) 8000hz 70ms 32000bps 2013-02-01 08:39:41.367135 [NOTICE] switch_loadable_module.c:187 Adding Codec DVI4 5 ADPCM (IMA) 8000hz 80ms 32000bps 2013-02-01 08:39:41.367137 [NOTICE] switch_loadable_module.c:187 Adding Codec DVI4 5 ADPCM (IMA) 8000hz 90ms 32000bps 2013-02-01 08:39:41.367139 [NOTICE] switch_loadable_module.c:187 Adding Codec DVI4 5 ADPCM (IMA) 8000hz 100ms 32000bps 2013-02-01 08:39:41.367141 [NOTICE] switch_loadable_module.c:187 Adding Codec DVI4 5 ADPCM (IMA) 8000hz 110ms 32000bps 2013-02-01 08:39:41.367142 [NOTICE] switch_loadable_module.c:187 Adding Codec DVI4 5 ADPCM (IMA) 8000hz 120ms 32000bps 2013-02-01 08:39:41.367152 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-16 124 G.726 16k (AAL2) 8000hz 10ms 16000bps 2013-02-01 08:39:41.367164 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-16 124 G.726 16k (AAL2) 8000hz 20ms 16000bps 2013-02-01 08:39:41.367166 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-16 124 G.726 16k (AAL2) 8000hz 30ms 16000bps 2013-02-01 08:39:41.367168 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-16 124 G.726 16k (AAL2) 8000hz 40ms 16000bps 2013-02-01 08:39:41.367171 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-16 124 G.726 16k (AAL2) 8000hz 50ms 16000bps 2013-02-01 08:39:41.367173 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-16 124 G.726 16k (AAL2) 8000hz 60ms 16000bps 2013-02-01 08:39:41.367175 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-16 124 G.726 16k (AAL2) 8000hz 70ms 16000bps 2013-02-01 08:39:41.367177 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-16 124 G.726 16k (AAL2) 8000hz 80ms 16000bps 2013-02-01 08:39:41.367179 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-16 124 G.726 16k (AAL2) 8000hz 90ms 16000bps 2013-02-01 08:39:41.367180 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-16 124 G.726 16k (AAL2) 8000hz 100ms 16000bps 2013-02-01 08:39:41.367183 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-16 124 G.726 16k (AAL2) 8000hz 110ms 16000bps 2013-02-01 08:39:41.367185 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-16 124 G.726 16k (AAL2) 8000hz 120ms 16000bps 2013-02-01 08:39:41.367194 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-16 124 G.726 16k 8000hz 10ms 16000bps 2013-02-01 08:39:41.367198 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-16 124 G.726 16k 8000hz 20ms 16000bps 2013-02-01 08:39:41.367200 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-16 124 G.726 16k 8000hz 30ms 16000bps 2013-02-01 08:39:41.367202 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-16 124 G.726 16k 8000hz 40ms 16000bps 2013-02-01 08:39:41.367205 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-16 124 G.726 16k 8000hz 50ms 16000bps 2013-02-01 08:39:41.367207 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-16 124 G.726 16k 8000hz 60ms 16000bps 2013-02-01 08:39:41.367209 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-16 124 G.726 16k 8000hz 70ms 16000bps 2013-02-01 08:39:41.367211 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-16 124 G.726 16k 8000hz 80ms 16000bps 2013-02-01 08:39:41.367213 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-16 124 G.726 16k 8000hz 90ms 16000bps 2013-02-01 08:39:41.367216 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-16 124 G.726 16k 8000hz 100ms 16000bps 2013-02-01 08:39:41.367227 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-16 124 G.726 16k 8000hz 110ms 16000bps 2013-02-01 08:39:41.367229 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-16 124 G.726 16k 8000hz 120ms 16000bps 2013-02-01 08:39:41.367237 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-24 123 G.726 24k (AAL2) 8000hz 10ms 24000bps 2013-02-01 08:39:41.367253 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-24 123 G.726 24k (AAL2) 8000hz 20ms 24000bps 2013-02-01 08:39:41.367255 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-24 123 G.726 24k (AAL2) 8000hz 30ms 24000bps 2013-02-01 08:39:41.367257 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-24 123 G.726 24k (AAL2) 8000hz 40ms 24000bps 2013-02-01 08:39:41.367259 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-24 123 G.726 24k (AAL2) 8000hz 50ms 24000bps 2013-02-01 08:39:41.367260 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-24 123 G.726 24k (AAL2) 8000hz 60ms 24000bps 2013-02-01 08:39:41.367262 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-24 123 G.726 24k (AAL2) 8000hz 70ms 24000bps 2013-02-01 08:39:41.367264 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-24 123 G.726 24k (AAL2) 8000hz 80ms 24000bps 2013-02-01 08:39:41.367265 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-24 123 G.726 24k (AAL2) 8000hz 90ms 24000bps 2013-02-01 08:39:41.367267 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-24 123 G.726 24k (AAL2) 8000hz 100ms 24000bps 2013-02-01 08:39:41.367268 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-24 123 G.726 24k (AAL2) 8000hz 110ms 24000bps 2013-02-01 08:39:41.367271 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-24 123 G.726 24k (AAL2) 8000hz 120ms 24000bps 2013-02-01 08:39:41.367280 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-24 123 G.726 24k 8000hz 10ms 24000bps 2013-02-01 08:39:41.367284 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-24 123 G.726 24k 8000hz 20ms 24000bps 2013-02-01 08:39:41.367286 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-24 123 G.726 24k 8000hz 30ms 24000bps 2013-02-01 08:39:41.367288 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-24 123 G.726 24k 8000hz 40ms 24000bps 2013-02-01 08:39:41.367291 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-24 123 G.726 24k 8000hz 50ms 24000bps 2013-02-01 08:39:41.367293 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-24 123 G.726 24k 8000hz 60ms 24000bps 2013-02-01 08:39:41.367295 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-24 123 G.726 24k 8000hz 70ms 24000bps 2013-02-01 08:39:41.367297 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-24 123 G.726 24k 8000hz 80ms 24000bps 2013-02-01 08:39:41.367299 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-24 123 G.726 24k 8000hz 90ms 24000bps 2013-02-01 08:39:41.367300 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-24 123 G.726 24k 8000hz 100ms 24000bps 2013-02-01 08:39:41.367310 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-24 123 G.726 24k 8000hz 110ms 24000bps 2013-02-01 08:39:41.367313 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-24 123 G.726 24k 8000hz 120ms 24000bps 2013-02-01 08:39:41.367320 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-32 122 G.726 32k (AAL2) 8000hz 10ms 32000bps 2013-02-01 08:39:41.367323 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-32 122 G.726 32k (AAL2) 8000hz 20ms 32000bps 2013-02-01 08:39:41.367334 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-32 122 G.726 32k (AAL2) 8000hz 30ms 32000bps 2013-02-01 08:39:41.367336 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-32 122 G.726 32k (AAL2) 8000hz 40ms 32000bps 2013-02-01 08:39:41.367338 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-32 122 G.726 32k (AAL2) 8000hz 50ms 32000bps 2013-02-01 08:39:41.367340 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-32 122 G.726 32k (AAL2) 8000hz 60ms 32000bps 2013-02-01 08:39:41.367343 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-32 122 G.726 32k (AAL2) 8000hz 70ms 32000bps 2013-02-01 08:39:41.367345 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-32 122 G.726 32k (AAL2) 8000hz 80ms 32000bps 2013-02-01 08:39:41.367347 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-32 122 G.726 32k (AAL2) 8000hz 90ms 32000bps 2013-02-01 08:39:41.367350 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-32 122 G.726 32k (AAL2) 8000hz 100ms 32000bps 2013-02-01 08:39:41.367355 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-32 122 G.726 32k (AAL2) 8000hz 110ms 32000bps 2013-02-01 08:39:41.367356 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-32 122 G.726 32k (AAL2) 8000hz 120ms 32000bps 2013-02-01 08:39:41.367365 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-32 122 G.726 32k 8000hz 10ms 32000bps 2013-02-01 08:39:41.367377 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-32 122 G.726 32k 8000hz 20ms 32000bps 2013-02-01 08:39:41.367379 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-32 122 G.726 32k 8000hz 30ms 32000bps 2013-02-01 08:39:41.367389 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-32 122 G.726 32k 8000hz 40ms 32000bps 2013-02-01 08:39:41.367391 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-32 122 G.726 32k 8000hz 50ms 32000bps 2013-02-01 08:39:41.367394 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-32 122 G.726 32k 8000hz 60ms 32000bps 2013-02-01 08:39:41.367397 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-32 122 G.726 32k 8000hz 70ms 32000bps 2013-02-01 08:39:41.367408 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-32 122 G.726 32k 8000hz 80ms 32000bps 2013-02-01 08:39:41.367412 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-32 122 G.726 32k 8000hz 90ms 32000bps 2013-02-01 08:39:41.367416 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-32 122 G.726 32k 8000hz 100ms 32000bps 2013-02-01 08:39:41.367420 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-32 122 G.726 32k 8000hz 110ms 32000bps 2013-02-01 08:39:41.367426 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-32 122 G.726 32k 8000hz 120ms 32000bps 2013-02-01 08:39:41.367444 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-40 121 G.726 40k (AAL2) 8000hz 10ms 40000bps 2013-02-01 08:39:41.367451 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-40 121 G.726 40k (AAL2) 8000hz 20ms 40000bps 2013-02-01 08:39:41.367456 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-40 121 G.726 40k (AAL2) 8000hz 30ms 40000bps 2013-02-01 08:39:41.367460 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-40 121 G.726 40k (AAL2) 8000hz 40ms 40000bps 2013-02-01 08:39:41.367465 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-40 121 G.726 40k (AAL2) 8000hz 50ms 40000bps 2013-02-01 08:39:41.367470 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-40 121 G.726 40k (AAL2) 8000hz 60ms 40000bps 2013-02-01 08:39:41.367474 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-40 121 G.726 40k (AAL2) 8000hz 70ms 40000bps 2013-02-01 08:39:41.367478 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-40 121 G.726 40k (AAL2) 8000hz 80ms 40000bps 2013-02-01 08:39:41.367483 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-40 121 G.726 40k (AAL2) 8000hz 90ms 40000bps 2013-02-01 08:39:41.367489 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-40 121 G.726 40k (AAL2) 8000hz 100ms 40000bps 2013-02-01 08:39:41.367493 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-40 121 G.726 40k (AAL2) 8000hz 110ms 40000bps 2013-02-01 08:39:41.367497 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-40 121 G.726 40k (AAL2) 8000hz 120ms 40000bps 2013-02-01 08:39:41.367514 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-40 121 G.726 40k 8000hz 10ms 40000bps 2013-02-01 08:39:41.367520 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-40 121 G.726 40k 8000hz 20ms 40000bps 2013-02-01 08:39:41.367525 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-40 121 G.726 40k 8000hz 30ms 40000bps 2013-02-01 08:39:41.367528 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-40 121 G.726 40k 8000hz 40ms 40000bps 2013-02-01 08:39:41.367533 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-40 121 G.726 40k 8000hz 50ms 40000bps 2013-02-01 08:39:41.367536 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-40 121 G.726 40k 8000hz 60ms 40000bps 2013-02-01 08:39:41.367540 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-40 121 G.726 40k 8000hz 70ms 40000bps 2013-02-01 08:39:41.367545 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-40 121 G.726 40k 8000hz 80ms 40000bps 2013-02-01 08:39:41.367549 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-40 121 G.726 40k 8000hz 90ms 40000bps 2013-02-01 08:39:41.367553 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-40 121 G.726 40k 8000hz 100ms 40000bps 2013-02-01 08:39:41.367558 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-40 121 G.726 40k 8000hz 110ms 40000bps 2013-02-01 08:39:41.367563 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-40 121 G.726 40k 8000hz 120ms 40000bps 2013-02-01 08:39:41.367580 [NOTICE] switch_loadable_module.c:187 Adding Codec G722 9 G.722 16000hz 10ms 64000bps 2013-02-01 08:39:41.367587 [NOTICE] switch_loadable_module.c:187 Adding Codec G722 9 G.722 16000hz 20ms 64000bps 2013-02-01 08:39:41.367592 [NOTICE] switch_loadable_module.c:187 Adding Codec G722 9 G.722 16000hz 30ms 64000bps 2013-02-01 08:39:41.367596 [NOTICE] switch_loadable_module.c:187 Adding Codec G722 9 G.722 16000hz 40ms 64000bps 2013-02-01 08:39:41.367603 [NOTICE] switch_loadable_module.c:187 Adding Codec G722 9 G.722 16000hz 50ms 64000bps 2013-02-01 08:39:41.367607 [NOTICE] switch_loadable_module.c:187 Adding Codec G722 9 G.722 16000hz 60ms 64000bps 2013-02-01 08:39:41.367626 [NOTICE] switch_loadable_module.c:187 Adding Codec GSM 3 GSM 8000hz 20ms 13200bps 2013-02-01 08:39:41.367632 [NOTICE] switch_loadable_module.c:187 Adding Codec GSM 3 GSM 8000hz 40ms 13200bps 2013-02-01 08:39:41.367637 [NOTICE] switch_loadable_module.c:187 Adding Codec GSM 3 GSM 8000hz 60ms 13200bps 2013-02-01 08:39:41.367640 [NOTICE] switch_loadable_module.c:187 Adding Codec GSM 3 GSM 8000hz 80ms 13200bps 2013-02-01 08:39:41.367644 [NOTICE] switch_loadable_module.c:187 Adding Codec GSM 3 GSM 8000hz 100ms 13200bps 2013-02-01 08:39:41.367648 [NOTICE] switch_loadable_module.c:187 Adding Codec GSM 3 GSM 8000hz 120ms 13200bps 2013-02-01 08:39:41.367667 [NOTICE] switch_loadable_module.c:187 Adding Codec LPC 7 LPC-10 8000hz 90ms 2400bps 2013-02-01 08:39:41.367688 [NOTICE] switch_loadable_module.c:254 Adding Application 't38_gateway' 2013-02-01 08:39:41.367715 [NOTICE] switch_loadable_module.c:254 Adding Application 'rxfax' 2013-02-01 08:39:41.367739 [NOTICE] switch_loadable_module.c:254 Adding Application 'txfax' 2013-02-01 08:39:41.367756 [NOTICE] switch_loadable_module.c:254 Adding Application 'spandsp_stop_dtmf' 2013-02-01 08:39:41.367783 [NOTICE] switch_loadable_module.c:254 Adding Application 'spandsp_start_dtmf' 2013-02-01 08:39:41.367799 [NOTICE] switch_loadable_module.c:254 Adding Application 'spandsp_stop_inject_tdd' 2013-02-01 08:39:41.367827 [NOTICE] switch_loadable_module.c:254 Adding Application 'spandsp_inject_tdd' 2013-02-01 08:39:41.367850 [NOTICE] switch_loadable_module.c:254 Adding Application 'spandsp_stop_detect_tdd' 2013-02-01 08:39:41.367871 [NOTICE] switch_loadable_module.c:254 Adding Application 'spandsp_detect_tdd' 2013-02-01 08:39:41.367899 [NOTICE] switch_loadable_module.c:254 Adding Application 'spandsp_send_tdd' 2013-02-01 08:39:41.367917 [NOTICE] switch_loadable_module.c:254 Adding Application 'spandsp_start_fax_detect' 2013-02-01 08:39:41.367941 [NOTICE] switch_loadable_module.c:254 Adding Application 'spandsp_stop_fax_detect' 2013-02-01 08:39:41.367968 [NOTICE] switch_loadable_module.c:254 Adding Application 'start_tone_detect' 2013-02-01 08:39:41.367993 [NOTICE] switch_loadable_module.c:254 Adding Application 'stop_tone_detect' 2013-02-01 08:39:41.368015 [NOTICE] switch_loadable_module.c:298 Adding API Function 'start_tone_detect' 2013-02-01 08:39:41.368036 [NOTICE] switch_loadable_module.c:298 Adding API Function 'stop_tone_detect' 2013-02-01 08:39:41.368062 [NOTICE] switch_loadable_module.c:298 Adding API Function 'start_tdd_detect' 2013-02-01 08:39:41.368085 [NOTICE] switch_loadable_module.c:298 Adding API Function 'stop_tdd_detect' 2013-02-01 08:39:41.368107 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_send_tdd' 2013-02-01 08:39:41.368330 [NOTICE] switch_loadable_module.c:187 Adding Codec G723 4 G.723.1 6.3k 8000hz 120ms 6300bps 2013-02-01 08:39:41.368336 [NOTICE] switch_loadable_module.c:187 Adding Codec G723 4 G.723.1 6.3k 8000hz 90ms 6300bps 2013-02-01 08:39:41.368340 [NOTICE] switch_loadable_module.c:187 Adding Codec G723 4 G.723.1 6.3k 8000hz 60ms 6300bps 2013-02-01 08:39:41.368344 [NOTICE] switch_loadable_module.c:187 Adding Codec G723 4 G.723.1 6.3k 8000hz 30ms 6300bps 2013-02-01 08:39:41.368552 [NOTICE] switch_loadable_module.c:187 Adding Codec G729 18 G.729 8000hz 10ms 8000bps 2013-02-01 08:39:41.368558 [NOTICE] switch_loadable_module.c:187 Adding Codec G729 18 G.729 8000hz 20ms 8000bps 2013-02-01 08:39:41.368562 [NOTICE] switch_loadable_module.c:187 Adding Codec G729 18 G.729 8000hz 30ms 8000bps 2013-02-01 08:39:41.368565 [NOTICE] switch_loadable_module.c:187 Adding Codec G729 18 G.729 8000hz 40ms 8000bps 2013-02-01 08:39:41.368569 [NOTICE] switch_loadable_module.c:187 Adding Codec G729 18 G.729 8000hz 50ms 8000bps 2013-02-01 08:39:41.368573 [NOTICE] switch_loadable_module.c:187 Adding Codec G729 18 G.729 8000hz 60ms 8000bps 2013-02-01 08:39:41.368576 [NOTICE] switch_loadable_module.c:187 Adding Codec G729 18 G.729 8000hz 70ms 8000bps 2013-02-01 08:39:41.368584 [NOTICE] switch_loadable_module.c:187 Adding Codec G729 18 G.729 8000hz 80ms 8000bps 2013-02-01 08:39:41.368588 [NOTICE] switch_loadable_module.c:187 Adding Codec G729 18 G.729 8000hz 90ms 8000bps 2013-02-01 08:39:41.368591 [NOTICE] switch_loadable_module.c:187 Adding Codec G729 18 G.729 8000hz 100ms 8000bps 2013-02-01 08:39:41.368595 [NOTICE] switch_loadable_module.c:187 Adding Codec G729 18 G.729 8000hz 110ms 8000bps 2013-02-01 08:39:41.368599 [NOTICE] switch_loadable_module.c:187 Adding Codec G729 18 G.729 8000hz 120ms 8000bps 2013-02-01 08:39:41.368793 [NOTICE] switch_loadable_module.c:187 Adding Codec AMR 96 AMR 8000hz 20ms 12200bps 2013-02-01 08:39:41.369068 [NOTICE] switch_loadable_module.c:187 Adding Codec SPEEX 99 Speex 32000hz 20ms 44000bps 2013-02-01 08:39:41.369074 [NOTICE] switch_loadable_module.c:187 Adding Codec SPEEX 99 Speex 16000hz 20ms 42200bps 2013-02-01 08:39:41.369078 [NOTICE] switch_loadable_module.c:187 Adding Codec SPEEX 99 Speex 8000hz 20ms 24600bps 2013-02-01 08:39:41.369272 [NOTICE] switch_loadable_module.c:187 Adding Codec H264 97 H.264 Video (passthru) 90000hz 0ms 0bps 2013-02-01 08:39:41.369292 [NOTICE] switch_loadable_module.c:187 Adding Codec H263 34 H.263 Video (passthru) 90000hz 0ms 0bps 2013-02-01 08:39:41.369312 [NOTICE] switch_loadable_module.c:187 Adding Codec H263-1998 115 H.263+ Video (passthru) 90000hz 0ms 0bps 2013-02-01 08:39:41.369336 [NOTICE] switch_loadable_module.c:187 Adding Codec H263-2000 121 H.263++ Video (passthru) 90000hz 0ms 0bps 2013-02-01 08:39:41.369368 [NOTICE] switch_loadable_module.c:187 Adding Codec H261 31 H.261 Video (passthru) 90000hz 0ms 0bps 2013-02-01 08:39:41.369571 [NOTICE] switch_loadable_module.c:187 Adding Codec VP8 99 VP8 Video (passthru) 90000hz 0ms 0bps 2013-02-01 08:39:41.369976 [INFO] mod_sndfile.c:363 LibSndFile Version : libsndfile-1.0.19 Supported Formats ================================================================================ AIFF (Apple/SGI) (extension "aiff") AU (Sun/NeXT) (extension "au") AVR (Audio Visual Research) (extension "avr") CAF (Apple Core Audio File) (extension "caf") HTK (HMM Tool Kit) (extension "htk") IFF (Amiga IFF/SVX8/SV16) (extension "iff") MAT4 (GNU Octave 2.0 / Matlab 4.2) (extension "mat") MAT5 (GNU Octave 2.1 / Matlab 5.0) (extension "mat") PAF (Ensoniq PARIS) (extension "paf") PVF (Portable Voice Format) (extension "pvf") RAW (header-less) (extension "raw") SD2 (Sound Designer II) (extension "sd2") SDS (Midi Sample Dump Standard) (extension "sds") SF (Berkeley/IRCAM/CARL) (extension "sf") VOC (Creative Labs) (extension "voc") W64 (SoundFoundry WAVE 64) (extension "w64") WAV (Microsoft) (extension "wav") WAV (NIST Sphere) (extension "wav") WAVEX (Microsoft) (extension "wav") WVE (Psion Series 3) (extension "wve") XI (FastTracker 2) (extension "xi") ================================================================================ 2013-02-01 08:39:41.370080 [NOTICE] switch_loadable_module.c:322 Adding File Format 'aiff' 2013-02-01 08:39:41.370108 [NOTICE] switch_loadable_module.c:322 Adding File Format 'au' 2013-02-01 08:39:41.370137 [NOTICE] switch_loadable_module.c:322 Adding File Format 'avr' 2013-02-01 08:39:41.370157 [NOTICE] switch_loadable_module.c:322 Adding File Format 'caf' 2013-02-01 08:39:41.370178 [NOTICE] switch_loadable_module.c:322 Adding File Format 'htk' 2013-02-01 08:39:41.370199 [NOTICE] switch_loadable_module.c:322 Adding File Format 'iff' 2013-02-01 08:39:41.370235 [NOTICE] switch_loadable_module.c:322 Adding File Format 'mat' 2013-02-01 08:39:41.370259 [NOTICE] switch_loadable_module.c:322 Adding File Format 'paf' 2013-02-01 08:39:41.370283 [NOTICE] switch_loadable_module.c:322 Adding File Format 'pvf' 2013-02-01 08:39:41.370304 [NOTICE] switch_loadable_module.c:322 Adding File Format 'raw' 2013-02-01 08:39:41.370325 [NOTICE] switch_loadable_module.c:322 Adding File Format 'sd2' 2013-02-01 08:39:41.370348 [NOTICE] switch_loadable_module.c:322 Adding File Format 'sds' 2013-02-01 08:39:41.370370 [NOTICE] switch_loadable_module.c:322 Adding File Format 'sf' 2013-02-01 08:39:41.370396 [NOTICE] switch_loadable_module.c:322 Adding File Format 'voc' 2013-02-01 08:39:41.370420 [NOTICE] switch_loadable_module.c:322 Adding File Format 'w64' 2013-02-01 08:39:41.370443 [NOTICE] switch_loadable_module.c:322 Adding File Format 'wav' 2013-02-01 08:39:41.370466 [NOTICE] switch_loadable_module.c:322 Adding File Format 'wve' 2013-02-01 08:39:41.370491 [NOTICE] switch_loadable_module.c:322 Adding File Format 'xi' 2013-02-01 08:39:41.370514 [NOTICE] switch_loadable_module.c:322 Adding File Format 'r8' 2013-02-01 08:39:41.370537 [NOTICE] switch_loadable_module.c:322 Adding File Format 'r16' 2013-02-01 08:39:41.370558 [NOTICE] switch_loadable_module.c:322 Adding File Format 'r24' 2013-02-01 08:39:41.370582 [NOTICE] switch_loadable_module.c:322 Adding File Format 'r32' 2013-02-01 08:39:41.370603 [NOTICE] switch_loadable_module.c:322 Adding File Format 'gsm' 2013-02-01 08:39:41.370627 [NOTICE] switch_loadable_module.c:322 Adding File Format 'ul' 2013-02-01 08:39:41.370649 [NOTICE] switch_loadable_module.c:322 Adding File Format 'ulaw' 2013-02-01 08:39:41.370672 [NOTICE] switch_loadable_module.c:322 Adding File Format 'al' 2013-02-01 08:39:41.370694 [NOTICE] switch_loadable_module.c:322 Adding File Format 'alaw' 2013-02-01 08:39:41.370727 [NOTICE] switch_loadable_module.c:322 Adding File Format 'adpcm' 2013-02-01 08:39:41.370970 [NOTICE] switch_loadable_module.c:322 Adding File Format 'H263' 2013-02-01 08:39:41.370994 [NOTICE] switch_loadable_module.c:322 Adding File Format 'AMR' 2013-02-01 08:39:41.371020 [NOTICE] switch_loadable_module.c:322 Adding File Format 'SPEEX' 2013-02-01 08:39:41.371044 [NOTICE] switch_loadable_module.c:322 Adding File Format 'G729' 2013-02-01 08:39:41.371068 [NOTICE] switch_loadable_module.c:322 Adding File Format 'G726-16' 2013-02-01 08:39:41.371090 [NOTICE] switch_loadable_module.c:322 Adding File Format 'AAL2-G726-16' 2013-02-01 08:39:41.371115 [NOTICE] switch_loadable_module.c:322 Adding File Format 'PCMA' 2013-02-01 08:39:41.371137 [NOTICE] switch_loadable_module.c:322 Adding File Format 'DVI4' 2013-02-01 08:39:41.371161 [NOTICE] switch_loadable_module.c:322 Adding File Format 'G726-24' 2013-02-01 08:39:41.371183 [NOTICE] switch_loadable_module.c:322 Adding File Format 'PCMU' 2013-02-01 08:39:41.371207 [NOTICE] switch_loadable_module.c:322 Adding File Format 'L16' 2013-02-01 08:39:41.371242 [NOTICE] switch_loadable_module.c:322 Adding File Format 'PROXY' 2013-02-01 08:39:41.371275 [NOTICE] switch_loadable_module.c:322 Adding File Format 'AAL2-G726-24' 2013-02-01 08:39:41.371299 [NOTICE] switch_loadable_module.c:322 Adding File Format 'AAL2-G726-32' 2013-02-01 08:39:41.371332 [NOTICE] switch_loadable_module.c:322 Adding File Format 'G726-32' 2013-02-01 08:39:41.371367 [NOTICE] switch_loadable_module.c:322 Adding File Format 'G722' 2013-02-01 08:39:41.371390 [NOTICE] switch_loadable_module.c:322 Adding File Format 'AAL2-G726-40' 2013-02-01 08:39:41.371413 [NOTICE] switch_loadable_module.c:322 Adding File Format 'G726-40' 2013-02-01 08:39:41.371448 [NOTICE] switch_loadable_module.c:322 Adding File Format 'GSM' 2013-02-01 08:39:41.371485 [NOTICE] switch_loadable_module.c:322 Adding File Format 'H263-1998' 2013-02-01 08:39:41.371509 [NOTICE] switch_loadable_module.c:322 Adding File Format 'H263-2000' 2013-02-01 08:39:41.371533 [NOTICE] switch_loadable_module.c:322 Adding File Format 'H264' 2013-02-01 08:39:41.371569 [NOTICE] switch_loadable_module.c:322 Adding File Format 'H261' 2013-02-01 08:39:41.371604 [NOTICE] switch_loadable_module.c:322 Adding File Format 'G723' 2013-02-01 08:39:41.371627 [NOTICE] switch_loadable_module.c:322 Adding File Format 'LPC' 2013-02-01 08:39:41.371651 [NOTICE] switch_loadable_module.c:322 Adding File Format 'PROXY-VID' 2013-02-01 08:39:41.371687 [NOTICE] switch_loadable_module.c:322 Adding File Format 'VP8' 2013-02-01 08:39:41.372101 [NOTICE] switch_loadable_module.c:298 Adding API Function 'stop_local_stream' 2013-02-01 08:39:41.372131 [NOTICE] switch_loadable_module.c:298 Adding API Function 'start_local_stream' 2013-02-01 08:39:41.372158 [NOTICE] switch_loadable_module.c:298 Adding API Function 'show_local_stream' 2013-02-01 08:39:41.372203 [NOTICE] switch_loadable_module.c:322 Adding File Format 'local_stream' 2013-02-01 08:39:41.372571 [NOTICE] switch_loadable_module.c:322 Adding File Format 'tone_stream' 2013-02-01 08:39:41.372598 [NOTICE] switch_loadable_module.c:322 Adding File Format 'silence_stream' 2013-02-01 08:39:41.376731 [DEBUG] switch_loadable_module.c:1281 Loading module with global namespace at request of module 2013-02-01 08:39:41.376990 [DEBUG] mod_spidermonkey.c:980 Successfully Loaded [/usr/local/freeswitch/mod/mod_spidermonkey_teletone.so] 2013-02-01 08:39:41.377202 [DEBUG] mod_spidermonkey.c:980 Successfully Loaded [/usr/local/freeswitch/mod/mod_spidermonkey_core_db.so] 2013-02-01 08:39:41.377396 [DEBUG] mod_spidermonkey.c:980 Successfully Loaded [/usr/local/freeswitch/mod/mod_spidermonkey_socket.so] 2013-02-01 08:39:41.377425 [NOTICE] switch_loadable_module.c:254 Adding Application 'javascript' 2013-02-01 08:39:41.377452 [NOTICE] switch_loadable_module.c:276 Adding Chat Application 'javascript' 2013-02-01 08:39:41.377485 [NOTICE] switch_loadable_module.c:298 Adding API Function 'jsrun' 2013-02-01 08:39:41.377508 [NOTICE] switch_loadable_module.c:298 Adding API Function 'jsapi' 2013-02-01 08:39:41.378011 [DEBUG] switch_loadable_module.c:1281 Loading module with global namespace at request of module 2013-02-01 08:39:41.378033 [NOTICE] switch_loadable_module.c:214 Adding Dialplan 'LUA' 2013-02-01 08:39:41.378044 [NOTICE] switch_loadable_module.c:254 Adding Application 'lua' 2013-02-01 08:39:41.378062 [NOTICE] switch_loadable_module.c:276 Adding Chat Application 'lua' 2013-02-01 08:39:41.378072 [NOTICE] switch_loadable_module.c:298 Adding API Function 'luarun' 2013-02-01 08:39:41.378099 [NOTICE] switch_loadable_module.c:298 Adding API Function 'lua' 2013-02-01 08:39:41.380693 [NOTICE] switch_loadable_module.c:343 Adding Speech interface 'flite' 2013-02-01 08:39:41.380825 [NOTICE] switch_loadable_module.c:423 Adding Say interface 'en' 2013-02-01 08:39:41.380936 [NOTICE] switch_loadable_module.c:423 Adding Say interface 'fr' 2013-02-01 08:39:41.381153 [DEBUG] mod_event_socket.c:2805 Socket up listening on 127.0.0.1:8021 2013-02-01 08:39:41.381151 [DEBUG] switch_loadable_module.c:640 Chat Thread Started 2013-02-01 08:39:41.381180 [DEBUG] switch_loadable_module.c:640 Chat Thread Started 2013-02-01 08:39:41.381209 [NOTICE] switch_core.c:1211 Created ip list rfc1918.auto default (deny) 2013-02-01 08:39:41.381217 [NOTICE] switch_utils.c:295 Adding 10.0.0.0/8 (allow) [] to list rfc1918.auto 2013-02-01 08:39:41.381222 [NOTICE] switch_utils.c:295 Adding 172.16.0.0/12 (allow) [] to list rfc1918.auto 2013-02-01 08:39:41.381225 [NOTICE] switch_utils.c:295 Adding 192.168.0.0/16 (allow) [] to list rfc1918.auto 2013-02-01 08:39:41.381227 [NOTICE] switch_core.c:1219 Created ip list wan.auto default (allow) 2013-02-01 08:39:41.381231 [NOTICE] switch_utils.c:295 Adding 10.0.0.0/8 (deny) [] to list wan.auto 2013-02-01 08:39:41.381233 [NOTICE] switch_utils.c:295 Adding 172.16.0.0/12 (deny) [] to list wan.auto 2013-02-01 08:39:41.381235 [NOTICE] switch_utils.c:295 Adding 192.168.0.0/16 (deny) [] to list wan.auto 2013-02-01 08:39:41.381237 [NOTICE] switch_core.c:1227 Created ip list nat.auto default (deny) 2013-02-01 08:39:41.381239 [NOTICE] switch_core.c:1229 Adding XXX.XXX.XXX.10/255.255.255.240 (deny) to list nat.auto 2013-02-01 08:39:41.381243 [NOTICE] switch_utils.c:295 Adding 10.0.0.0/8 (allow) [] to list nat.auto 2013-02-01 08:39:41.381245 [NOTICE] switch_utils.c:295 Adding 172.16.0.0/12 (allow) [] to list nat.auto 2013-02-01 08:39:41.381247 [NOTICE] switch_utils.c:295 Adding 192.168.0.0/16 (allow) [] to list nat.auto 2013-02-01 08:39:41.381249 [NOTICE] switch_core.c:1238 Created ip list loopback.auto default (deny) 2013-02-01 08:39:41.381251 [NOTICE] switch_utils.c:295 Adding 127.0.0.0/8 (allow) [] to list loopback.auto 2013-02-01 08:39:41.381254 [NOTICE] switch_core.c:1244 Created ip list localnet.auto default (deny) 2013-02-01 08:39:41.381257 [NOTICE] switch_core.c:1247 Adding XXX.XXX.XXX.10/255.255.255.240 (allow) to list localnet.auto 2013-02-01 08:39:41.381271 [NOTICE] switch_utils.c:295 Adding 192.168.42.0/24 (deny) [] to list lan 2013-02-01 08:39:41.381274 [NOTICE] switch_core.c:1343 Adding 192.168.42.0/24 (deny) to list lan 2013-02-01 08:39:41.381277 [NOTICE] switch_utils.c:295 Adding 192.168.42.42/32 (allow) [] to list lan 2013-02-01 08:39:41.381279 [NOTICE] switch_core.c:1343 Adding 192.168.42.42/32 (allow) to list lan 2013-02-01 08:39:41.381299 [NOTICE] switch_utils.c:295 Adding 216.113.106.182/32 (allow) [0000000000 at XXX.XXX.XXX.10] to list domains 2013-02-01 08:39:41.381313 [NOTICE] switch_utils.c:295 Adding YYY.YYY.YYY.YYY/32 (allow) [0000000000 at XXX.XXX.XXX.10] to list domains 2013-02-01 08:39:41.381319 [INFO] switch_time.c:476 Clock synchronized to system time. 2013-02-01 08:39:41.381319 [NOTICE] switch_utils.c:295 Adding 10.5.3.154/32 (allow) [4388999130 at XXX.XXX.XXX.10] to list domains 2013-02-01 08:39:41.381343 [NOTICE] switch_utils.c:295 Adding 192.0.2.0/24 (allow) [brian at XXX.XXX.XXX.10] to list domains 2013-02-01 08:39:41.381348 [NOTICE] switch_utils.c:295 Adding 159.18.161.101/32 (allow) [] to list domains 2013-02-01 08:39:41.381352 [NOTICE] switch_core.c:1343 Adding 159.18.161.101/32 (allow) to list domains 2013-02-01 08:40:10.252905 [WARNING] sofia.c:4719 Ping succeeded thinktel with code 200 - count 1/1/1, state UP 2013-02-01 08:45:37.492902 [INFO] switch_core.c:2395 Shutting down 2013-02-01 08:45:37.492902 [DEBUG] switch_loadable_module.c:649 Chat Thread Ended 2013-02-01 08:45:37.492902 [DEBUG] switch_loadable_module.c:649 Chat Thread Ended 2013-02-01 08:45:37.492902 [DEBUG] switch_loadable_module.c:1152 Write lock interface 'fr' to wait for existing references. 2013-02-01 08:45:37.492902 [NOTICE] switch_loadable_module.c:1160 Deleting Say interface 'fr' 2013-02-01 08:45:37.492902 [DEBUG] switch_loadable_module.c:1049 Write lock interface 'flite' to wait for existing references. 2013-02-01 08:45:37.492902 [NOTICE] switch_loadable_module.c:1058 Deleting Speech interface 'flite' 2013-02-01 08:45:37.492902 [DEBUG] switch_loadable_module.c:1020 Write lock interface 'mod_native_file' to wait for existing references. 2013-02-01 08:45:37.492902 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'H263' 2013-02-01 08:45:37.492902 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'AMR' 2013-02-01 08:45:37.492902 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'SPEEX' 2013-02-01 08:45:37.492902 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'G729' 2013-02-01 08:45:37.492902 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'G726-16' 2013-02-01 08:45:37.492902 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'AAL2-G726-16' 2013-02-01 08:45:37.492902 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'PCMA' 2013-02-01 08:45:37.492902 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'DVI4' 2013-02-01 08:45:37.492902 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'G726-24' 2013-02-01 08:45:37.492902 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'PCMU' 2013-02-01 08:45:37.492902 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'L16' 2013-02-01 08:45:37.492902 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'PROXY' 2013-02-01 08:45:37.492902 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'AAL2-G726-24' 2013-02-01 08:45:37.492902 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'AAL2-G726-32' 2013-02-01 08:45:37.492902 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'G726-32' 2013-02-01 08:45:37.492902 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'G722' 2013-02-01 08:45:37.492902 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'AAL2-G726-40' 2013-02-01 08:45:37.492902 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'G726-40' 2013-02-01 08:45:37.492902 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'GSM' 2013-02-01 08:45:37.492902 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'H263-1998' 2013-02-01 08:45:37.492902 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'H263-2000' 2013-02-01 08:45:37.492902 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'H264' 2013-02-01 08:45:37.492902 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'H261' 2013-02-01 08:45:37.492902 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'G723' 2013-02-01 08:45:37.492902 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'LPC' 2013-02-01 08:45:37.492902 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'PROXY-VID' 2013-02-01 08:45:37.492902 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'VP8' 2013-02-01 08:45:37.492902 [NOTICE] switch_loadable_module.c:936 Deleting Application 'fifo' 2013-02-01 08:45:37.492902 [DEBUG] switch_loadable_module.c:938 Write lock interface 'fifo' to wait for existing references. 2013-02-01 08:45:37.492902 [NOTICE] switch_loadable_module.c:936 Deleting Application 'fifo_track_call' 2013-02-01 08:45:37.492902 [DEBUG] switch_loadable_module.c:938 Write lock interface 'fifo_track_call' to wait for existing references. 2013-02-01 08:45:37.492902 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'fifo' 2013-02-01 08:45:37.492902 [DEBUG] switch_loadable_module.c:991 Write lock interface 'fifo' to wait for existing references. 2013-02-01 08:45:37.492902 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'fifo_member' 2013-02-01 08:45:37.492902 [DEBUG] switch_loadable_module.c:991 Write lock interface 'fifo_member' to wait for existing references. 2013-02-01 08:45:37.492902 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'fifo_add_outbound' 2013-02-01 08:45:37.492902 [DEBUG] switch_loadable_module.c:991 Write lock interface 'fifo_add_outbound' to wait for existing references. 2013-02-01 08:45:37.492902 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'fifo_check_bridge' 2013-02-01 08:45:37.492902 [DEBUG] switch_loadable_module.c:991 Write lock interface 'fifo_check_bridge' to wait for existing references. 2013-02-01 08:45:37.492902 [INFO] switch_core_sqldb.c:1370 fifo Destroying SQL queue. 2013-02-01 08:45:37.492902 [INFO] switch_core_sqldb.c:1315 fifo Stopping SQL thread. 2013-02-01 08:45:37.692904 [NOTICE] switch_event.c:2010 Event Binding deleted for mod_fifo:PRESENCE_PROBE 2013-02-01 08:45:37.692904 [NOTICE] switch_event.c:403 Subclass reservation deleted for /usr/src/freeswitch/src/mod/applications/mod_fifo/mod_fifo.c:fifo::info 2013-02-01 08:45:38.332904 [NOTICE] switch_loadable_module.c:936 Deleting Application 'db' 2013-02-01 08:45:38.332904 [DEBUG] switch_loadable_module.c:938 Write lock interface 'db' to wait for existing references. 2013-02-01 08:45:38.332904 [NOTICE] switch_loadable_module.c:936 Deleting Application 'group' 2013-02-01 08:45:38.332904 [DEBUG] switch_loadable_module.c:938 Write lock interface 'group' to wait for existing references. 2013-02-01 08:45:38.332904 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'db' 2013-02-01 08:45:38.332904 [DEBUG] switch_loadable_module.c:991 Write lock interface 'db' to wait for existing references. 2013-02-01 08:45:38.332904 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'group' 2013-02-01 08:45:38.332904 [DEBUG] switch_loadable_module.c:991 Write lock interface 'group' to wait for existing references. 2013-02-01 08:45:38.332904 [NOTICE] switch_loadable_module.c:1193 Deleting Limit interface 'db' 2013-02-01 08:45:38.332904 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'expr' 2013-02-01 08:45:38.332904 [DEBUG] switch_loadable_module.c:991 Write lock interface 'expr' to wait for existing references. 2013-02-01 08:45:38.332904 [NOTICE] switch_loadable_module.c:936 Deleting Application 'voicemail' 2013-02-01 08:45:38.332904 [DEBUG] switch_loadable_module.c:938 Write lock interface 'voicemail' to wait for existing references. 2013-02-01 08:45:38.332904 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'voicemail' 2013-02-01 08:45:38.332904 [DEBUG] switch_loadable_module.c:991 Write lock interface 'voicemail' to wait for existing references. 2013-02-01 08:45:38.332904 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'voicemail_inject' 2013-02-01 08:45:38.332904 [DEBUG] switch_loadable_module.c:991 Write lock interface 'voicemail_inject' to wait for existing references. 2013-02-01 08:45:38.332904 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'vm_inject' 2013-02-01 08:45:38.332904 [DEBUG] switch_loadable_module.c:991 Write lock interface 'vm_inject' to wait for existing references. 2013-02-01 08:45:38.332904 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'vm_boxcount' 2013-02-01 08:45:38.332904 [DEBUG] switch_loadable_module.c:991 Write lock interface 'vm_boxcount' to wait for existing references. 2013-02-01 08:45:38.332904 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'vm_prefs' 2013-02-01 08:45:38.332904 [DEBUG] switch_loadable_module.c:991 Write lock interface 'vm_prefs' to wait for existing references. 2013-02-01 08:45:38.332904 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'vm_delete' 2013-02-01 08:45:38.332904 [DEBUG] switch_loadable_module.c:991 Write lock interface 'vm_delete' to wait for existing references. 2013-02-01 08:45:38.332904 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'vm_read' 2013-02-01 08:45:38.332904 [DEBUG] switch_loadable_module.c:991 Write lock interface 'vm_read' to wait for existing references. 2013-02-01 08:45:38.332904 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'vm_list' 2013-02-01 08:45:38.332904 [DEBUG] switch_loadable_module.c:991 Write lock interface 'vm_list' to wait for existing references. 2013-02-01 08:45:38.332904 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'vm_fsdb_auth_login' 2013-02-01 08:45:38.332904 [DEBUG] switch_loadable_module.c:991 Write lock interface 'vm_fsdb_auth_login' to wait for existing references. 2013-02-01 08:45:38.332904 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'vm_fsdb_msg_count' 2013-02-01 08:45:38.332904 [DEBUG] switch_loadable_module.c:991 Write lock interface 'vm_fsdb_msg_count' to wait for existing references. 2013-02-01 08:45:38.332904 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'vm_fsdb_msg_list' 2013-02-01 08:45:38.332904 [DEBUG] switch_loadable_module.c:991 Write lock interface 'vm_fsdb_msg_list' to wait for existing references. 2013-02-01 08:45:38.332904 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'vm_fsdb_msg_get' 2013-02-01 08:45:38.332904 [DEBUG] switch_loadable_module.c:991 Write lock interface 'vm_fsdb_msg_get' to wait for existing references. 2013-02-01 08:45:38.332904 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'vm_fsdb_msg_delete' 2013-02-01 08:45:38.332904 [DEBUG] switch_loadable_module.c:991 Write lock interface 'vm_fsdb_msg_delete' to wait for existing references. 2013-02-01 08:45:38.332904 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'vm_fsdb_msg_undelete' 2013-02-01 08:45:38.332904 [DEBUG] switch_loadable_module.c:991 Write lock interface 'vm_fsdb_msg_undelete' to wait for existing references. 2013-02-01 08:45:38.332904 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'vm_fsdb_msg_email' 2013-02-01 08:45:38.332904 [DEBUG] switch_loadable_module.c:991 Write lock interface 'vm_fsdb_msg_email' to wait for existing references. 2013-02-01 08:45:38.332904 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'vm_fsdb_msg_purge' 2013-02-01 08:45:38.332904 [DEBUG] switch_loadable_module.c:991 Write lock interface 'vm_fsdb_msg_purge' to wait for existing references. 2013-02-01 08:45:38.332904 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'vm_fsdb_msg_save' 2013-02-01 08:45:38.332904 [DEBUG] switch_loadable_module.c:991 Write lock interface 'vm_fsdb_msg_save' to wait for existing references. 2013-02-01 08:45:38.332904 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'vm_fsdb_msg_forward' 2013-02-01 08:45:38.332904 [DEBUG] switch_loadable_module.c:991 Write lock interface 'vm_fsdb_msg_forward' to wait for existing references. 2013-02-01 08:45:38.332904 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'vm_fsdb_pref_greeting_set' 2013-02-01 08:45:38.332904 [DEBUG] switch_loadable_module.c:991 Write lock interface 'vm_fsdb_pref_greeting_set' to wait for existing references. 2013-02-01 08:45:38.332904 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'vm_fsdb_pref_recname_set' 2013-02-01 08:45:38.332904 [DEBUG] switch_loadable_module.c:991 Write lock interface 'vm_fsdb_pref_recname_set' to wait for existing references. 2013-02-01 08:45:38.332904 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'vm_fsdb_pref_password_set' 2013-02-01 08:45:38.332904 [DEBUG] switch_loadable_module.c:991 Write lock interface 'vm_fsdb_pref_password_set' to wait for existing references. 2013-02-01 08:45:38.332904 [NOTICE] switch_event.c:403 Subclass reservation deleted for /usr/src/freeswitch/src/mod/applications/mod_voicemail/mod_voicemail.c:vm::maintenance 2013-02-01 08:45:38.332904 [NOTICE] switch_event.c:1970 Event Binding deleted for mod_voicemail:MESSAGE_QUERY 2013-02-01 08:45:38.332904 [DEBUG] mod_voicemail.c:6003 Waiting for write lock (Profile default) 2013-02-01 08:45:38.332904 [DEBUG] mod_voicemail.c:6006 Destroying Profile default 2013-02-01 08:45:38.332904 [DEBUG] switch_loadable_module.c:1020 Write lock interface 'mod_sndfile' to wait for existing references. 2013-02-01 08:45:38.332904 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'aiff' 2013-02-01 08:45:38.332904 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'au' 2013-02-01 08:45:38.332904 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'avr' 2013-02-01 08:45:38.332904 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'caf' 2013-02-01 08:45:38.332904 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'htk' 2013-02-01 08:45:38.332904 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'iff' 2013-02-01 08:45:38.332904 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'mat' 2013-02-01 08:45:38.332904 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'paf' 2013-02-01 08:45:38.332904 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'pvf' 2013-02-01 08:45:38.332904 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'raw' 2013-02-01 08:45:38.332904 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'sd2' 2013-02-01 08:45:38.332904 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'sds' 2013-02-01 08:45:38.332904 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'sf' 2013-02-01 08:45:38.332904 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'voc' 2013-02-01 08:45:38.332904 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'w64' 2013-02-01 08:45:38.332904 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'wav' 2013-02-01 08:45:38.332904 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'wve' 2013-02-01 08:45:38.332904 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'xi' 2013-02-01 08:45:38.332904 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'r8' 2013-02-01 08:45:38.332904 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'r16' 2013-02-01 08:45:38.332904 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'r24' 2013-02-01 08:45:38.332904 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'r32' 2013-02-01 08:45:38.332904 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'gsm' 2013-02-01 08:45:38.332904 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'ul' 2013-02-01 08:45:38.332904 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'ulaw' 2013-02-01 08:45:38.332904 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'al' 2013-02-01 08:45:38.332904 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'alaw' 2013-02-01 08:45:38.332904 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'adpcm' 2013-02-01 08:45:38.332904 [DEBUG] switch_loadable_module.c:1152 Write lock interface 'en' to wait for existing references. 2013-02-01 08:45:38.332904 [NOTICE] switch_loadable_module.c:1160 Deleting Say interface 'en' 2013-02-01 08:45:38.332904 [DEBUG] switch_loadable_module.c:846 Write lock interface 'loopback' to wait for existing references. 2013-02-01 08:45:38.332904 [NOTICE] switch_loadable_module.c:854 Deleting Endpoint 'loopback' 2013-02-01 08:45:38.332904 [NOTICE] switch_loadable_module.c:936 Deleting Application 'unloop' 2013-02-01 08:45:38.332904 [DEBUG] switch_loadable_module.c:938 Write lock interface 'unloop' to wait for existing references. 2013-02-01 08:45:38.332904 [NOTICE] switch_loadable_module.c:921 Deleting Timer 'soft' 2013-02-01 08:45:38.352936 [NOTICE] switch_loadable_module.c:100 Thread ended for CORE_SOFTTIMER_MODULE 2013-02-01 08:45:38.358483 [NOTICE] switch_event.c:2010 Event Binding deleted for CORE_SOFTTIMER_MODULE:RELOADXML 2013-02-01 08:45:38.358501 [NOTICE] switch_loadable_module.c:936 Deleting Application 'conference' 2013-02-01 08:45:38.358534 [DEBUG] switch_loadable_module.c:938 Write lock interface 'conference' to wait for existing references. 2013-02-01 08:45:38.358559 [NOTICE] switch_loadable_module.c:936 Deleting Application 'conference_set_auto_outcall' 2013-02-01 08:45:38.358569 [DEBUG] switch_loadable_module.c:938 Write lock interface 'conference_set_auto_outcall' to wait for existing references. 2013-02-01 08:45:38.358586 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'conference' 2013-02-01 08:45:38.358592 [DEBUG] switch_loadable_module.c:991 Write lock interface 'conference' to wait for existing references. 2013-02-01 08:45:38.358620 [DEBUG] switch_loadable_module.c:1127 Write lock interface 'conf' to wait for existing references. 2013-02-01 08:45:38.358627 [NOTICE] switch_loadable_module.c:1136 Deleting Chat interface 'conf' 2013-02-01 08:45:38.358666 [NOTICE] switch_event.c:1970 Event Binding deleted for mod_conference:PRESENCE_PROBE 2013-02-01 08:45:38.358673 [NOTICE] switch_event.c:1970 Event Binding deleted for mod_conference:CONFERENCE_DATA_QUERY 2013-02-01 08:45:38.358677 [NOTICE] switch_event.c:1970 Event Binding deleted for mod_conference:CALL_SETUP_REQ 2013-02-01 08:45:38.358683 [NOTICE] switch_event.c:403 Subclass reservation deleted for /usr/src/freeswitch/src/mod/applications/mod_conference/mod_conference.c:conference::maintenance 2013-02-01 08:45:38.358689 [NOTICE] switch_loadable_module.c:880 Deleting Codec PROXY-VID 31 PROXY VIDEO PASS-THROUGH 90000hz 0ms 2013-02-01 08:45:38.358721 [NOTICE] switch_loadable_module.c:880 Deleting Codec PROXY 0 PROXY PASS-THROUGH 8000hz 20ms 2013-02-01 08:45:38.358741 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 11025hz 32ms 2013-02-01 08:45:38.358749 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 11025hz 40ms 2013-02-01 08:45:38.358755 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 22050hz 20ms 2013-02-01 08:45:38.358761 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 48000hz 40ms 2013-02-01 08:45:38.358784 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 48000hz 30ms 2013-02-01 08:45:38.358806 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 48000hz 20ms 2013-02-01 08:45:38.358817 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 32000hz 8ms 2013-02-01 08:45:38.358830 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 32000hz 6ms 2013-02-01 08:45:38.358850 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 32000hz 4ms 2013-02-01 08:45:38.358861 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 32000hz 2ms 2013-02-01 08:45:38.358874 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 16000hz 8ms 2013-02-01 08:45:38.358900 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 16000hz 6ms 2013-02-01 08:45:38.358916 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 16000hz 4ms 2013-02-01 08:45:38.358927 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 16000hz 2ms 2013-02-01 08:45:38.358939 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 8000hz 8ms 2013-02-01 08:45:38.358951 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 8000hz 6ms 2013-02-01 08:45:38.358970 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 8000hz 4ms 2013-02-01 08:45:38.358983 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 8000hz 2ms 2013-02-01 08:45:38.358995 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 48000hz 10ms 2013-02-01 08:45:38.359007 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 48000hz 8ms 2013-02-01 08:45:38.359022 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 48000hz 6ms 2013-02-01 08:45:38.359034 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 48000hz 4ms 2013-02-01 08:45:38.359048 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 48000hz 2ms 2013-02-01 08:45:38.359060 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 24000hz 60ms 2013-02-01 08:45:38.359073 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 24000hz 40ms 2013-02-01 08:45:38.359086 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 24000hz 20ms 2013-02-01 08:45:38.359098 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 12000hz 100ms 2013-02-01 08:45:38.359112 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 12000hz 80ms 2013-02-01 08:45:38.359124 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 12000hz 60ms 2013-02-01 08:45:38.359141 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 12000hz 40ms 2013-02-01 08:45:38.359152 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 12000hz 20ms 2013-02-01 08:45:38.359166 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 32000hz 10ms 2013-02-01 08:45:38.359178 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 32000hz 20ms 2013-02-01 08:45:38.359194 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 32000hz 30ms 2013-02-01 08:45:38.359206 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 32000hz 40ms 2013-02-01 08:45:38.359219 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 32000hz 50ms 2013-02-01 08:45:38.359234 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 32000hz 60ms 2013-02-01 08:45:38.359246 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 16000hz 10ms 2013-02-01 08:45:38.359258 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 16000hz 20ms 2013-02-01 08:45:38.359270 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 16000hz 30ms 2013-02-01 08:45:38.359282 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 16000hz 40ms 2013-02-01 08:45:38.359294 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 16000hz 50ms 2013-02-01 08:45:38.359308 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 16000hz 60ms 2013-02-01 08:45:38.359322 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 8000hz 10ms 2013-02-01 08:45:38.359334 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 8000hz 20ms 2013-02-01 08:45:38.359346 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 8000hz 30ms 2013-02-01 08:45:38.359358 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 8000hz 40ms 2013-02-01 08:45:38.359373 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 8000hz 50ms 2013-02-01 08:45:38.359388 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 8000hz 60ms 2013-02-01 08:45:38.359404 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 8000hz 70ms 2013-02-01 08:45:38.359422 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 8000hz 80ms 2013-02-01 08:45:38.359437 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 8000hz 90ms 2013-02-01 08:45:38.359449 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 8000hz 100ms 2013-02-01 08:45:38.359461 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 8000hz 110ms 2013-02-01 08:45:38.359472 [NOTICE] switch_loadable_module.c:880 Deleting Codec L16 70 RAW Signed Linear (16 bit) 8000hz 120ms 2013-02-01 08:45:38.359499 [NOTICE] switch_loadable_module.c:880 Deleting Codec PCMU 0 G.711 ulaw 8000hz 10ms 2013-02-01 08:45:38.359512 [NOTICE] switch_loadable_module.c:880 Deleting Codec PCMU 0 G.711 ulaw 8000hz 20ms 2013-02-01 08:45:38.359527 [NOTICE] switch_loadable_module.c:880 Deleting Codec PCMU 0 G.711 ulaw 8000hz 30ms 2013-02-01 08:45:38.359539 [NOTICE] switch_loadable_module.c:880 Deleting Codec PCMU 0 G.711 ulaw 8000hz 40ms 2013-02-01 08:45:38.359567 [NOTICE] switch_loadable_module.c:880 Deleting Codec PCMU 0 G.711 ulaw 8000hz 50ms 2013-02-01 08:45:38.359579 [NOTICE] switch_loadable_module.c:880 Deleting Codec PCMU 0 G.711 ulaw 8000hz 60ms 2013-02-01 08:45:38.359592 [NOTICE] switch_loadable_module.c:880 Deleting Codec PCMU 0 G.711 ulaw 8000hz 70ms 2013-02-01 08:45:38.359604 [NOTICE] switch_loadable_module.c:880 Deleting Codec PCMU 0 G.711 ulaw 8000hz 80ms 2013-02-01 08:45:38.359616 [NOTICE] switch_loadable_module.c:880 Deleting Codec PCMU 0 G.711 ulaw 8000hz 90ms 2013-02-01 08:45:38.359629 [NOTICE] switch_loadable_module.c:880 Deleting Codec PCMU 0 G.711 ulaw 8000hz 100ms 2013-02-01 08:45:38.359641 [NOTICE] switch_loadable_module.c:880 Deleting Codec PCMU 0 G.711 ulaw 8000hz 110ms 2013-02-01 08:45:38.359653 [NOTICE] switch_loadable_module.c:880 Deleting Codec PCMU 0 G.711 ulaw 8000hz 120ms 2013-02-01 08:45:38.359685 [NOTICE] switch_loadable_module.c:880 Deleting Codec PCMA 8 G.711 alaw 8000hz 10ms 2013-02-01 08:45:38.359700 [NOTICE] switch_loadable_module.c:880 Deleting Codec PCMA 8 G.711 alaw 8000hz 20ms 2013-02-01 08:45:38.359720 [NOTICE] switch_loadable_module.c:880 Deleting Codec PCMA 8 G.711 alaw 8000hz 30ms 2013-02-01 08:45:38.359742 [NOTICE] switch_loadable_module.c:880 Deleting Codec PCMA 8 G.711 alaw 8000hz 40ms 2013-02-01 08:45:38.359761 [NOTICE] switch_loadable_module.c:880 Deleting Codec PCMA 8 G.711 alaw 8000hz 50ms 2013-02-01 08:45:38.359776 [NOTICE] switch_loadable_module.c:880 Deleting Codec PCMA 8 G.711 alaw 8000hz 60ms 2013-02-01 08:45:38.359795 [NOTICE] switch_loadable_module.c:880 Deleting Codec PCMA 8 G.711 alaw 8000hz 70ms 2013-02-01 08:45:38.359811 [NOTICE] switch_loadable_module.c:880 Deleting Codec PCMA 8 G.711 alaw 8000hz 80ms 2013-02-01 08:45:38.359832 [NOTICE] switch_loadable_module.c:880 Deleting Codec PCMA 8 G.711 alaw 8000hz 90ms 2013-02-01 08:45:38.359850 [NOTICE] switch_loadable_module.c:880 Deleting Codec PCMA 8 G.711 alaw 8000hz 100ms 2013-02-01 08:45:38.359862 [NOTICE] switch_loadable_module.c:880 Deleting Codec PCMA 8 G.711 alaw 8000hz 110ms 2013-02-01 08:45:38.359873 [NOTICE] switch_loadable_module.c:880 Deleting Codec PCMA 8 G.711 alaw 8000hz 120ms 2013-02-01 08:45:38.359914 [NOTICE] switch_loadable_module.c:936 Deleting Application 'socket' 2013-02-01 08:45:38.359923 [DEBUG] switch_loadable_module.c:938 Write lock interface 'socket' to wait for existing references. 2013-02-01 08:45:38.359941 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'event_sink' 2013-02-01 08:45:38.359947 [DEBUG] switch_loadable_module.c:991 Write lock interface 'event_sink' to wait for existing references. 2013-02-01 08:45:38.360005 [NOTICE] mod_event_socket.c:2829 Shutting Down 2013-02-01 08:45:38.360031 [NOTICE] switch_event.c:2010 Event Binding deleted for mod_event_socket:ALL 2013-02-01 08:45:38.360042 [NOTICE] switch_loadable_module.c:100 Thread ended for mod_event_socket 2013-02-01 08:45:38.360051 [NOTICE] switch_loadable_module.c:880 Deleting Codec AMR 96 AMR 8000hz 20ms 2013-02-01 08:45:38.360135 [DEBUG] switch_loadable_module.c:846 Write lock interface 'error' to wait for existing references. 2013-02-01 08:45:38.360142 [NOTICE] switch_loadable_module.c:854 Deleting Endpoint 'error' 2013-02-01 08:45:38.360166 [DEBUG] switch_loadable_module.c:846 Write lock interface 'group' to wait for existing references. 2013-02-01 08:45:38.360172 [NOTICE] switch_loadable_module.c:854 Deleting Endpoint 'group' 2013-02-01 08:45:38.360191 [DEBUG] switch_loadable_module.c:846 Write lock interface 'user' to wait for existing references. 2013-02-01 08:45:38.360197 [NOTICE] switch_loadable_module.c:854 Deleting Endpoint 'user' 2013-02-01 08:45:38.360223 [DEBUG] switch_loadable_module.c:846 Write lock interface 'pickup' to wait for existing references. 2013-02-01 08:45:38.360230 [NOTICE] switch_loadable_module.c:854 Deleting Endpoint 'pickup' 2013-02-01 08:45:38.360247 [NOTICE] switch_loadable_module.c:905 Deleting Dialplan 'inline' 2013-02-01 08:45:38.360281 [NOTICE] switch_loadable_module.c:936 Deleting Application 'blind_transfer_ack' 2013-02-01 08:45:38.360292 [DEBUG] switch_loadable_module.c:938 Write lock interface 'blind_transfer_ack' to wait for existing references. 2013-02-01 08:45:38.360318 [NOTICE] switch_loadable_module.c:936 Deleting Application 'bind_digit_action' 2013-02-01 08:45:38.360342 [DEBUG] switch_loadable_module.c:938 Write lock interface 'bind_digit_action' to wait for existing references. 2013-02-01 08:45:38.360362 [NOTICE] switch_loadable_module.c:936 Deleting Application 'capture' 2013-02-01 08:45:38.360371 [DEBUG] switch_loadable_module.c:938 Write lock interface 'capture' to wait for existing references. 2013-02-01 08:45:38.360387 [NOTICE] switch_loadable_module.c:936 Deleting Application 'clear_digit_action' 2013-02-01 08:45:38.360396 [DEBUG] switch_loadable_module.c:938 Write lock interface 'clear_digit_action' to wait for existing references. 2013-02-01 08:45:38.360420 [NOTICE] switch_loadable_module.c:936 Deleting Application 'digit_action_set_realm' 2013-02-01 08:45:38.360438 [DEBUG] switch_loadable_module.c:938 Write lock interface 'digit_action_set_realm' to wait for existing references. 2013-02-01 08:45:38.360460 [NOTICE] switch_loadable_module.c:936 Deleting Application 'privacy' 2013-02-01 08:45:38.360474 [DEBUG] switch_loadable_module.c:938 Write lock interface 'privacy' to wait for existing references. 2013-02-01 08:45:38.360498 [NOTICE] switch_loadable_module.c:936 Deleting Application 'set_audio_level' 2013-02-01 08:45:38.360508 [DEBUG] switch_loadable_module.c:938 Write lock interface 'set_audio_level' to wait for existing references. 2013-02-01 08:45:38.360529 [NOTICE] switch_loadable_module.c:936 Deleting Application 'set_mute' 2013-02-01 08:45:38.360539 [DEBUG] switch_loadable_module.c:938 Write lock interface 'set_mute' to wait for existing references. 2013-02-01 08:45:38.360556 [NOTICE] switch_loadable_module.c:936 Deleting Application 'flush_dtmf' 2013-02-01 08:45:38.360565 [DEBUG] switch_loadable_module.c:938 Write lock interface 'flush_dtmf' to wait for existing references. 2013-02-01 08:45:38.360587 [NOTICE] switch_loadable_module.c:936 Deleting Application 'hold' 2013-02-01 08:45:38.360608 [DEBUG] switch_loadable_module.c:938 Write lock interface 'hold' to wait for existing references. 2013-02-01 08:45:38.360628 [NOTICE] switch_loadable_module.c:936 Deleting Application 'unhold' 2013-02-01 08:45:38.360637 [DEBUG] switch_loadable_module.c:938 Write lock interface 'unhold' to wait for existing references. 2013-02-01 08:45:38.360653 [NOTICE] switch_loadable_module.c:936 Deleting Application 'mutex' 2013-02-01 08:45:38.360668 [DEBUG] switch_loadable_module.c:938 Write lock interface 'mutex' to wait for existing references. 2013-02-01 08:45:38.360686 [NOTICE] switch_loadable_module.c:936 Deleting Application 'page' 2013-02-01 08:45:38.360696 [DEBUG] switch_loadable_module.c:938 Write lock interface 'page' to wait for existing references. 2013-02-01 08:45:38.360715 [NOTICE] switch_loadable_module.c:936 Deleting Application 'transfer' 2013-02-01 08:45:38.360726 [DEBUG] switch_loadable_module.c:938 Write lock interface 'transfer' to wait for existing references. 2013-02-01 08:45:38.360747 [NOTICE] switch_loadable_module.c:936 Deleting Application 'check_acl' 2013-02-01 08:45:38.360759 [DEBUG] switch_loadable_module.c:938 Write lock interface 'check_acl' to wait for existing references. 2013-02-01 08:45:38.360779 [NOTICE] switch_loadable_module.c:936 Deleting Application 'verbose_events' 2013-02-01 08:45:38.360789 [DEBUG] switch_loadable_module.c:938 Write lock interface 'verbose_events' to wait for existing references. 2013-02-01 08:45:38.360812 [NOTICE] switch_loadable_module.c:936 Deleting Application 'cng_plc' 2013-02-01 08:45:38.360822 [DEBUG] switch_loadable_module.c:938 Write lock interface 'cng_plc' to wait for existing references. 2013-02-01 08:45:38.360843 [NOTICE] switch_loadable_module.c:936 Deleting Application 'early_hangup' 2013-02-01 08:45:38.360854 [DEBUG] switch_loadable_module.c:938 Write lock interface 'early_hangup' to wait for existing references. 2013-02-01 08:45:38.360874 [NOTICE] switch_loadable_module.c:936 Deleting Application 'sleep' 2013-02-01 08:45:38.360904 [DEBUG] switch_loadable_module.c:938 Write lock interface 'sleep' to wait for existing references. 2013-02-01 08:45:38.360925 [NOTICE] switch_loadable_module.c:936 Deleting Application 'delay_echo' 2013-02-01 08:45:38.360935 [DEBUG] switch_loadable_module.c:938 Write lock interface 'delay_echo' to wait for existing references. 2013-02-01 08:45:38.360957 [NOTICE] switch_loadable_module.c:936 Deleting Application 'strftime' 2013-02-01 08:45:38.360966 [DEBUG] switch_loadable_module.c:938 Write lock interface 'strftime' to wait for existing references. 2013-02-01 08:45:38.360984 [NOTICE] switch_loadable_module.c:936 Deleting Application 'phrase' 2013-02-01 08:45:38.360994 [DEBUG] switch_loadable_module.c:938 Write lock interface 'phrase' to wait for existing references. 2013-02-01 08:45:38.361016 [NOTICE] switch_loadable_module.c:936 Deleting Application 'eval' 2013-02-01 08:45:38.361025 [DEBUG] switch_loadable_module.c:938 Write lock interface 'eval' to wait for existing references. 2013-02-01 08:45:38.361048 [NOTICE] switch_loadable_module.c:936 Deleting Application 'stop' 2013-02-01 08:45:38.361061 [DEBUG] switch_loadable_module.c:938 Write lock interface 'stop' to wait for existing references. 2013-02-01 08:45:38.361081 [NOTICE] switch_loadable_module.c:936 Deleting Application 'set_zombie_exec' 2013-02-01 08:45:38.361092 [DEBUG] switch_loadable_module.c:938 Write lock interface 'set_zombie_exec' to wait for existing references. 2013-02-01 08:45:38.361111 [NOTICE] switch_loadable_module.c:936 Deleting Application 'pre_answer' 2013-02-01 08:45:38.361121 [DEBUG] switch_loadable_module.c:938 Write lock interface 'pre_answer' to wait for existing references. 2013-02-01 08:45:38.361143 [NOTICE] switch_loadable_module.c:936 Deleting Application 'answer' 2013-02-01 08:45:38.361152 [DEBUG] switch_loadable_module.c:938 Write lock interface 'answer' to wait for existing references. 2013-02-01 08:45:38.361171 [NOTICE] switch_loadable_module.c:936 Deleting Application 'wait_for_answer' 2013-02-01 08:45:38.361183 [DEBUG] switch_loadable_module.c:938 Write lock interface 'wait_for_answer' to wait for existing references. 2013-02-01 08:45:38.361203 [NOTICE] switch_loadable_module.c:936 Deleting Application 'hangup' 2013-02-01 08:45:38.361212 [DEBUG] switch_loadable_module.c:938 Write lock interface 'hangup' to wait for existing references. 2013-02-01 08:45:38.361230 [NOTICE] switch_loadable_module.c:936 Deleting Application 'set_name' 2013-02-01 08:45:38.361240 [DEBUG] switch_loadable_module.c:938 Write lock interface 'set_name' to wait for existing references. 2013-02-01 08:45:38.361259 [NOTICE] switch_loadable_module.c:936 Deleting Application 'presence' 2013-02-01 08:45:38.361269 [DEBUG] switch_loadable_module.c:938 Write lock interface 'presence' to wait for existing references. 2013-02-01 08:45:38.361288 [NOTICE] switch_loadable_module.c:936 Deleting Application 'log' 2013-02-01 08:45:38.361298 [DEBUG] switch_loadable_module.c:938 Write lock interface 'log' to wait for existing references. 2013-02-01 08:45:38.361318 [NOTICE] switch_loadable_module.c:936 Deleting Application 'info' 2013-02-01 08:45:38.361328 [DEBUG] switch_loadable_module.c:938 Write lock interface 'info' to wait for existing references. 2013-02-01 08:45:38.361345 [NOTICE] switch_loadable_module.c:936 Deleting Application 'event' 2013-02-01 08:45:38.361356 [DEBUG] switch_loadable_module.c:938 Write lock interface 'event' to wait for existing references. 2013-02-01 08:45:38.361377 [NOTICE] switch_loadable_module.c:936 Deleting Application 'sound_test' 2013-02-01 08:45:38.361386 [DEBUG] switch_loadable_module.c:938 Write lock interface 'sound_test' to wait for existing references. 2013-02-01 08:45:38.361405 [NOTICE] switch_loadable_module.c:936 Deleting Application 'export' 2013-02-01 08:45:38.361416 [DEBUG] switch_loadable_module.c:938 Write lock interface 'export' to wait for existing references. 2013-02-01 08:45:38.361437 [NOTICE] switch_loadable_module.c:936 Deleting Application 'bridge_export' 2013-02-01 08:45:38.361448 [DEBUG] switch_loadable_module.c:938 Write lock interface 'bridge_export' to wait for existing references. 2013-02-01 08:45:38.361469 [NOTICE] switch_loadable_module.c:936 Deleting Application 'set' 2013-02-01 08:45:38.361480 [DEBUG] switch_loadable_module.c:938 Write lock interface 'set' to wait for existing references. 2013-02-01 08:45:38.361501 [NOTICE] switch_loadable_module.c:936 Deleting Application 'multiset' 2013-02-01 08:45:38.361510 [DEBUG] switch_loadable_module.c:938 Write lock interface 'multiset' to wait for existing references. 2013-02-01 08:45:38.361530 [NOTICE] switch_loadable_module.c:936 Deleting Application 'push' 2013-02-01 08:45:38.361539 [DEBUG] switch_loadable_module.c:938 Write lock interface 'push' to wait for existing references. 2013-02-01 08:45:38.361561 [NOTICE] switch_loadable_module.c:936 Deleting Application 'unshift' 2013-02-01 08:45:38.361570 [DEBUG] switch_loadable_module.c:938 Write lock interface 'unshift' to wait for existing references. 2013-02-01 08:45:38.361593 [NOTICE] switch_loadable_module.c:936 Deleting Application 'set_global' 2013-02-01 08:45:38.361602 [DEBUG] switch_loadable_module.c:938 Write lock interface 'set_global' to wait for existing references. 2013-02-01 08:45:38.361622 [NOTICE] switch_loadable_module.c:936 Deleting Application 'set_profile_var' 2013-02-01 08:45:38.361633 [DEBUG] switch_loadable_module.c:938 Write lock interface 'set_profile_var' to wait for existing references. 2013-02-01 08:45:38.361654 [NOTICE] switch_loadable_module.c:936 Deleting Application 'unset' 2013-02-01 08:45:38.361663 [DEBUG] switch_loadable_module.c:938 Write lock interface 'unset' to wait for existing references. 2013-02-01 08:45:38.361683 [NOTICE] switch_loadable_module.c:936 Deleting Application 'ring_ready' 2013-02-01 08:45:38.361692 [DEBUG] switch_loadable_module.c:938 Write lock interface 'ring_ready' to wait for existing references. 2013-02-01 08:45:38.361714 [NOTICE] switch_loadable_module.c:936 Deleting Application 'remove_bugs' 2013-02-01 08:45:38.361727 [DEBUG] switch_loadable_module.c:938 Write lock interface 'remove_bugs' to wait for existing references. 2013-02-01 08:45:38.361747 [NOTICE] switch_loadable_module.c:936 Deleting Application 'break' 2013-02-01 08:45:38.361756 [DEBUG] switch_loadable_module.c:938 Write lock interface 'break' to wait for existing references. 2013-02-01 08:45:38.361774 [NOTICE] switch_loadable_module.c:936 Deleting Application 'detect_speech' 2013-02-01 08:45:38.361786 [DEBUG] switch_loadable_module.c:938 Write lock interface 'detect_speech' to wait for existing references. 2013-02-01 08:45:38.361806 [NOTICE] switch_loadable_module.c:936 Deleting Application 'play_and_detect_speech' 2013-02-01 08:45:38.361817 [DEBUG] switch_loadable_module.c:938 Write lock interface 'play_and_detect_speech' to wait for existing references. 2013-02-01 08:45:38.361838 [NOTICE] switch_loadable_module.c:936 Deleting Application 'ivr' 2013-02-01 08:45:38.361847 [DEBUG] switch_loadable_module.c:938 Write lock interface 'ivr' to wait for existing references. 2013-02-01 08:45:38.361866 [NOTICE] switch_loadable_module.c:936 Deleting Application 'redirect' 2013-02-01 08:45:38.361876 [DEBUG] switch_loadable_module.c:938 Write lock interface 'redirect' to wait for existing references. 2013-02-01 08:45:38.361906 [NOTICE] switch_loadable_module.c:936 Deleting Application 'video_refresh' 2013-02-01 08:45:38.361919 [DEBUG] switch_loadable_module.c:938 Write lock interface 'video_refresh' to wait for existing references. 2013-02-01 08:45:38.361940 [NOTICE] switch_loadable_module.c:936 Deleting Application 'send_info' 2013-02-01 08:45:38.361951 [DEBUG] switch_loadable_module.c:938 Write lock interface 'send_info' to wait for existing references. 2013-02-01 08:45:38.361972 [NOTICE] switch_loadable_module.c:936 Deleting Application 'jitterbuffer' 2013-02-01 08:45:38.361983 [DEBUG] switch_loadable_module.c:938 Write lock interface 'jitterbuffer' to wait for existing references. 2013-02-01 08:45:38.362004 [NOTICE] switch_loadable_module.c:936 Deleting Application 'send_display' 2013-02-01 08:45:38.362014 [DEBUG] switch_loadable_module.c:938 Write lock interface 'send_display' to wait for existing references. 2013-02-01 08:45:38.362035 [NOTICE] switch_loadable_module.c:936 Deleting Application 'respond' 2013-02-01 08:45:38.362044 [DEBUG] switch_loadable_module.c:938 Write lock interface 'respond' to wait for existing references. 2013-02-01 08:45:38.362064 [NOTICE] switch_loadable_module.c:936 Deleting Application 'deflect' 2013-02-01 08:45:38.362073 [DEBUG] switch_loadable_module.c:938 Write lock interface 'deflect' to wait for existing references. 2013-02-01 08:45:38.362093 [NOTICE] switch_loadable_module.c:936 Deleting Application 'recovery_refresh' 2013-02-01 08:45:38.362103 [DEBUG] switch_loadable_module.c:938 Write lock interface 'recovery_refresh' to wait for existing references. 2013-02-01 08:45:38.362125 [NOTICE] switch_loadable_module.c:936 Deleting Application 'queue_dtmf' 2013-02-01 08:45:38.362134 [DEBUG] switch_loadable_module.c:938 Write lock interface 'queue_dtmf' to wait for existing references. 2013-02-01 08:45:38.362152 [NOTICE] switch_loadable_module.c:936 Deleting Application 'send_dtmf' 2013-02-01 08:45:38.362162 [DEBUG] switch_loadable_module.c:938 Write lock interface 'send_dtmf' to wait for existing references. 2013-02-01 08:45:38.362183 [NOTICE] switch_loadable_module.c:936 Deleting Application 'sched_cancel' 2013-02-01 08:45:38.362193 [DEBUG] switch_loadable_module.c:938 Write lock interface 'sched_cancel' to wait for existing references. 2013-02-01 08:45:38.362212 [NOTICE] switch_loadable_module.c:936 Deleting Application 'sched_hangup' 2013-02-01 08:45:38.362222 [DEBUG] switch_loadable_module.c:938 Write lock interface 'sched_hangup' to wait for existing references. 2013-02-01 08:45:38.362243 [NOTICE] switch_loadable_module.c:936 Deleting Application 'sched_broadcast' 2013-02-01 08:45:38.362254 [DEBUG] switch_loadable_module.c:938 Write lock interface 'sched_broadcast' to wait for existing references. 2013-02-01 08:45:38.362274 [NOTICE] switch_loadable_module.c:936 Deleting Application 'sched_transfer' 2013-02-01 08:45:38.362284 [DEBUG] switch_loadable_module.c:938 Write lock interface 'sched_transfer' to wait for existing references. 2013-02-01 08:45:38.362306 [NOTICE] switch_loadable_module.c:936 Deleting Application 'execute_extension' 2013-02-01 08:45:38.362316 [DEBUG] switch_loadable_module.c:938 Write lock interface 'execute_extension' to wait for existing references. 2013-02-01 08:45:38.362338 [NOTICE] switch_loadable_module.c:936 Deleting Application 'sched_heartbeat' 2013-02-01 08:45:38.362350 [DEBUG] switch_loadable_module.c:938 Write lock interface 'sched_heartbeat' to wait for existing references. 2013-02-01 08:45:38.362370 [NOTICE] switch_loadable_module.c:936 Deleting Application 'enable_heartbeat' 2013-02-01 08:45:38.362380 [DEBUG] switch_loadable_module.c:938 Write lock interface 'enable_heartbeat' to wait for existing references. 2013-02-01 08:45:38.362399 [NOTICE] switch_loadable_module.c:936 Deleting Application 'mkdir' 2013-02-01 08:45:38.362408 [DEBUG] switch_loadable_module.c:938 Write lock interface 'mkdir' to wait for existing references. 2013-02-01 08:45:38.362430 [NOTICE] switch_loadable_module.c:936 Deleting Application 'rename' 2013-02-01 08:45:38.362439 [DEBUG] switch_loadable_module.c:938 Write lock interface 'rename' to wait for existing references. 2013-02-01 08:45:38.362458 [NOTICE] switch_loadable_module.c:936 Deleting Application 'soft_hold' 2013-02-01 08:45:38.362484 [DEBUG] switch_loadable_module.c:938 Write lock interface 'soft_hold' to wait for existing references. 2013-02-01 08:45:38.362505 [NOTICE] switch_loadable_module.c:936 Deleting Application 'bind_meta_app' 2013-02-01 08:45:38.362514 [DEBUG] switch_loadable_module.c:938 Write lock interface 'bind_meta_app' to wait for existing references. 2013-02-01 08:45:38.362534 [NOTICE] switch_loadable_module.c:936 Deleting Application 'unbind_meta_app' 2013-02-01 08:45:38.362544 [DEBUG] switch_loadable_module.c:938 Write lock interface 'unbind_meta_app' to wait for existing references. 2013-02-01 08:45:38.362562 [NOTICE] switch_loadable_module.c:936 Deleting Application 'block_dfmf' 2013-02-01 08:45:38.362572 [DEBUG] switch_loadable_module.c:938 Write lock interface 'block_dfmf' to wait for existing references. 2013-02-01 08:45:38.362595 [NOTICE] switch_loadable_module.c:936 Deleting Application 'unblock_dtmf' 2013-02-01 08:45:38.362605 [DEBUG] switch_loadable_module.c:938 Write lock interface 'unblock_dtmf' to wait for existing references. 2013-02-01 08:45:38.362626 [NOTICE] switch_loadable_module.c:936 Deleting Application 'intercept' 2013-02-01 08:45:38.362636 [DEBUG] switch_loadable_module.c:938 Write lock interface 'intercept' to wait for existing references. 2013-02-01 08:45:38.362654 [NOTICE] switch_loadable_module.c:936 Deleting Application 'eavesdrop' 2013-02-01 08:45:38.362664 [DEBUG] switch_loadable_module.c:938 Write lock interface 'eavesdrop' to wait for existing references. 2013-02-01 08:45:38.362685 [NOTICE] switch_loadable_module.c:936 Deleting Application 'three_way' 2013-02-01 08:45:38.362697 [DEBUG] switch_loadable_module.c:938 Write lock interface 'three_way' to wait for existing references. 2013-02-01 08:45:38.362717 [NOTICE] switch_loadable_module.c:936 Deleting Application 'set_user' 2013-02-01 08:45:38.362727 [DEBUG] switch_loadable_module.c:938 Write lock interface 'set_user' to wait for existing references. 2013-02-01 08:45:38.362744 [NOTICE] switch_loadable_module.c:936 Deleting Application 'stop_dtmf' 2013-02-01 08:45:38.362753 [DEBUG] switch_loadable_module.c:938 Write lock interface 'stop_dtmf' to wait for existing references. 2013-02-01 08:45:38.362771 [NOTICE] switch_loadable_module.c:936 Deleting Application 'start_dtmf' 2013-02-01 08:45:38.362782 [DEBUG] switch_loadable_module.c:938 Write lock interface 'start_dtmf' to wait for existing references. 2013-02-01 08:45:38.362802 [NOTICE] switch_loadable_module.c:936 Deleting Application 'stop_dtmf_generate' 2013-02-01 08:45:38.362812 [DEBUG] switch_loadable_module.c:938 Write lock interface 'stop_dtmf_generate' to wait for existing references. 2013-02-01 08:45:38.362834 [NOTICE] switch_loadable_module.c:936 Deleting Application 'start_dtmf_generate' 2013-02-01 08:45:38.362844 [DEBUG] switch_loadable_module.c:938 Write lock interface 'start_dtmf_generate' to wait for existing references. 2013-02-01 08:45:38.362866 [NOTICE] switch_loadable_module.c:936 Deleting Application 'stop_tone_detect' 2013-02-01 08:45:38.362877 [DEBUG] switch_loadable_module.c:938 Write lock interface 'stop_tone_detect' to wait for existing references. 2013-02-01 08:45:38.362903 [NOTICE] switch_loadable_module.c:936 Deleting Application 'fax_detect' 2013-02-01 08:45:38.362914 [DEBUG] switch_loadable_module.c:938 Write lock interface 'fax_detect' to wait for existing references. 2013-02-01 08:45:38.362935 [NOTICE] switch_loadable_module.c:936 Deleting Application 'tone_detect' 2013-02-01 08:45:38.362945 [DEBUG] switch_loadable_module.c:938 Write lock interface 'tone_detect' to wait for existing references. 2013-02-01 08:45:38.362964 [NOTICE] switch_loadable_module.c:936 Deleting Application 'echo' 2013-02-01 08:45:38.362973 [DEBUG] switch_loadable_module.c:938 Write lock interface 'echo' to wait for existing references. 2013-02-01 08:45:38.362991 [NOTICE] switch_loadable_module.c:936 Deleting Application 'park' 2013-02-01 08:45:38.363016 [DEBUG] switch_loadable_module.c:938 Write lock interface 'park' to wait for existing references. 2013-02-01 08:45:38.363036 [NOTICE] switch_loadable_module.c:936 Deleting Application 'park_state' 2013-02-01 08:45:38.363048 [DEBUG] switch_loadable_module.c:938 Write lock interface 'park_state' to wait for existing references. 2013-02-01 08:45:38.363068 [NOTICE] switch_loadable_module.c:936 Deleting Application 'gentones' 2013-02-01 08:45:38.363077 [DEBUG] switch_loadable_module.c:938 Write lock interface 'gentones' to wait for existing references. 2013-02-01 08:45:38.363096 [NOTICE] switch_loadable_module.c:936 Deleting Application 'playback' 2013-02-01 08:45:38.363106 [DEBUG] switch_loadable_module.c:938 Write lock interface 'playback' to wait for existing references. 2013-02-01 08:45:38.363125 [NOTICE] switch_loadable_module.c:936 Deleting Application 'endless_playback' 2013-02-01 08:45:38.363135 [DEBUG] switch_loadable_module.c:938 Write lock interface 'endless_playback' to wait for existing references. 2013-02-01 08:45:38.363157 [NOTICE] switch_loadable_module.c:936 Deleting Application 'att_xfer' 2013-02-01 08:45:38.363166 [DEBUG] switch_loadable_module.c:938 Write lock interface 'att_xfer' to wait for existing references. 2013-02-01 08:45:38.363184 [NOTICE] switch_loadable_module.c:936 Deleting Application 'read' 2013-02-01 08:45:38.363193 [DEBUG] switch_loadable_module.c:938 Write lock interface 'read' to wait for existing references. 2013-02-01 08:45:38.363213 [NOTICE] switch_loadable_module.c:936 Deleting Application 'play_and_get_digits' 2013-02-01 08:45:38.363225 [DEBUG] switch_loadable_module.c:938 Write lock interface 'play_and_get_digits' to wait for existing references. 2013-02-01 08:45:38.363247 [NOTICE] switch_loadable_module.c:936 Deleting Application 'stop_record_session' 2013-02-01 08:45:38.363256 [DEBUG] switch_loadable_module.c:938 Write lock interface 'stop_record_session' to wait for existing references. 2013-02-01 08:45:38.363278 [NOTICE] switch_loadable_module.c:936 Deleting Application 'record_session' 2013-02-01 08:45:38.363288 [DEBUG] switch_loadable_module.c:938 Write lock interface 'record_session' to wait for existing references. 2013-02-01 08:45:38.363306 [NOTICE] switch_loadable_module.c:936 Deleting Application 'record' 2013-02-01 08:45:38.363315 [DEBUG] switch_loadable_module.c:938 Write lock interface 'record' to wait for existing references. 2013-02-01 08:45:38.363335 [NOTICE] switch_loadable_module.c:936 Deleting Application 'preprocess' 2013-02-01 08:45:38.363345 [DEBUG] switch_loadable_module.c:938 Write lock interface 'preprocess' to wait for existing references. 2013-02-01 08:45:38.363364 [NOTICE] switch_loadable_module.c:936 Deleting Application 'stop_displace_session' 2013-02-01 08:45:38.363376 [DEBUG] switch_loadable_module.c:938 Write lock interface 'stop_displace_session' to wait for existing references. 2013-02-01 08:45:38.363397 [NOTICE] switch_loadable_module.c:936 Deleting Application 'displace_session' 2013-02-01 08:45:38.363406 [DEBUG] switch_loadable_module.c:938 Write lock interface 'displace_session' to wait for existing references. 2013-02-01 08:45:38.363425 [NOTICE] switch_loadable_module.c:936 Deleting Application 'speak' 2013-02-01 08:45:38.363434 [DEBUG] switch_loadable_module.c:938 Write lock interface 'speak' to wait for existing references. 2013-02-01 08:45:38.363453 [NOTICE] switch_loadable_module.c:936 Deleting Application 'clear_speech_cache' 2013-02-01 08:45:38.363465 [DEBUG] switch_loadable_module.c:938 Write lock interface 'clear_speech_cache' to wait for existing references. 2013-02-01 08:45:38.363485 [NOTICE] switch_loadable_module.c:936 Deleting Application 'bridge' 2013-02-01 08:45:38.363495 [DEBUG] switch_loadable_module.c:938 Write lock interface 'bridge' to wait for existing references. 2013-02-01 08:45:38.363518 [NOTICE] switch_loadable_module.c:936 Deleting Application 'system' 2013-02-01 08:45:38.363544 [DEBUG] switch_loadable_module.c:938 Write lock interface 'system' to wait for existing references. 2013-02-01 08:45:38.363564 [NOTICE] switch_loadable_module.c:936 Deleting Application 'say' 2013-02-01 08:45:38.363573 [DEBUG] switch_loadable_module.c:938 Write lock interface 'say' to wait for existing references. 2013-02-01 08:45:38.363593 [NOTICE] switch_loadable_module.c:936 Deleting Application 'wait_for_silence' 2013-02-01 08:45:38.363605 [DEBUG] switch_loadable_module.c:938 Write lock interface 'wait_for_silence' to wait for existing references. 2013-02-01 08:45:38.363625 [NOTICE] switch_loadable_module.c:936 Deleting Application 'session_loglevel' 2013-02-01 08:45:38.363635 [DEBUG] switch_loadable_module.c:938 Write lock interface 'session_loglevel' to wait for existing references. 2013-02-01 08:45:38.363656 [NOTICE] switch_loadable_module.c:936 Deleting Application 'limit' 2013-02-01 08:45:38.363671 [DEBUG] switch_loadable_module.c:938 Write lock interface 'limit' to wait for existing references. 2013-02-01 08:45:38.363691 [NOTICE] switch_loadable_module.c:936 Deleting Application 'limit_hash' 2013-02-01 08:45:38.363701 [DEBUG] switch_loadable_module.c:938 Write lock interface 'limit_hash' to wait for existing references. 2013-02-01 08:45:38.363721 [NOTICE] switch_loadable_module.c:936 Deleting Application 'limit_execute' 2013-02-01 08:45:38.363731 [DEBUG] switch_loadable_module.c:938 Write lock interface 'limit_execute' to wait for existing references. 2013-02-01 08:45:38.363752 [NOTICE] switch_loadable_module.c:936 Deleting Application 'limit_hash_execute' 2013-02-01 08:45:38.363762 [DEBUG] switch_loadable_module.c:938 Write lock interface 'limit_hash_execute' to wait for existing references. 2013-02-01 08:45:38.363783 [NOTICE] switch_loadable_module.c:936 Deleting Application 'pickup' 2013-02-01 08:45:38.363793 [DEBUG] switch_loadable_module.c:938 Write lock interface 'pickup' to wait for existing references. 2013-02-01 08:45:38.363811 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'strepoch' 2013-02-01 08:45:38.363816 [DEBUG] switch_loadable_module.c:991 Write lock interface 'strepoch' to wait for existing references. 2013-02-01 08:45:38.363838 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'page' 2013-02-01 08:45:38.363843 [DEBUG] switch_loadable_module.c:991 Write lock interface 'page' to wait for existing references. 2013-02-01 08:45:38.363864 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'strmicroepoch' 2013-02-01 08:45:38.363870 [DEBUG] switch_loadable_module.c:991 Write lock interface 'strmicroepoch' to wait for existing references. 2013-02-01 08:45:38.363898 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'chat' 2013-02-01 08:45:38.363903 [DEBUG] switch_loadable_module.c:991 Write lock interface 'chat' to wait for existing references. 2013-02-01 08:45:38.363924 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'strftime' 2013-02-01 08:45:38.363929 [DEBUG] switch_loadable_module.c:991 Write lock interface 'strftime' to wait for existing references. 2013-02-01 08:45:38.363952 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'presence' 2013-02-01 08:45:38.363956 [DEBUG] switch_loadable_module.c:991 Write lock interface 'presence' to wait for existing references. 2013-02-01 08:45:38.363979 [DEBUG] switch_loadable_module.c:1020 Write lock interface 'mod_dptools' to wait for existing references. 2013-02-01 08:45:38.363985 [NOTICE] switch_loadable_module.c:1030 Deleting File Format 'file_string' 2013-02-01 08:45:38.364005 [DEBUG] switch_loadable_module.c:1127 Write lock interface 'event' to wait for existing references. 2013-02-01 08:45:38.364011 [NOTICE] switch_loadable_module.c:1136 Deleting Chat interface 'event' 2013-02-01 08:45:38.364031 [DEBUG] switch_loadable_module.c:1127 Write lock interface 'api' to wait for existing references. 2013-02-01 08:45:38.364037 [NOTICE] switch_loadable_module.c:1136 Deleting Chat interface 'api' 2013-02-01 08:45:38.364068 [NOTICE] switch_event.c:1970 Event Binding deleted for mod_dptools:PRESENCE_PROBE 2013-02-01 08:45:38.364076 [NOTICE] switch_loadable_module.c:880 Deleting Codec G723 4 G.723.1 6.3k 8000hz 120ms 2013-02-01 08:45:38.364098 [NOTICE] switch_loadable_module.c:880 Deleting Codec G723 4 G.723.1 6.3k 8000hz 90ms 2013-02-01 08:45:38.364113 [NOTICE] switch_loadable_module.c:880 Deleting Codec G723 4 G.723.1 6.3k 8000hz 60ms 2013-02-01 08:45:38.364129 [NOTICE] switch_loadable_module.c:880 Deleting Codec G723 4 G.723.1 6.3k 8000hz 30ms 2013-02-01 08:45:38.364167 [NOTICE] switch_loadable_module.c:989 Deleting API Function 'console' 2013-02-01 08:45:38.364171 [DEBUG] switch_loadable_module.c:991 Write lock interface 'console' to wait for existing references. 2013-02-01 08:45:54.615804 [NOTICE] switch_loadable_module.c:214 Adding Dialplan 'enum' 2013-02-01 08:45:54.615829 [NOTICE] switch_loadable_module.c:254 Adding Application 'enum' 2013-02-01 08:45:54.615850 [NOTICE] switch_loadable_module.c:298 Adding API Function 'enum' 2013-02-01 08:45:54.615872 [NOTICE] switch_loadable_module.c:298 Adding API Function 'enum_auto' 2013-02-01 08:45:54.616086 [DEBUG] mod_cdr_csv.c:339 Adding default template. 2013-02-01 08:45:54.616104 [DEBUG] mod_cdr_csv.c:386 Adding template sql. 2013-02-01 08:45:54.616110 [DEBUG] mod_cdr_csv.c:386 Adding template example. 2013-02-01 08:45:54.616115 [DEBUG] mod_cdr_csv.c:386 Adding template snom. 2013-02-01 08:45:54.616119 [DEBUG] mod_cdr_csv.c:386 Adding template linksys. 2013-02-01 08:45:54.616126 [DEBUG] mod_cdr_csv.c:386 Adding template asterisk. 2013-02-01 08:45:54.616130 [DEBUG] mod_cdr_csv.c:386 Adding template opencdrrate. 2013-02-01 08:45:54.616163 [NOTICE] switch_loadable_module.c:298 Adding API Function 'cdr_csv' 2013-02-01 08:45:54.616423 [NOTICE] switch_loadable_module.c:254 Adding Application 'socket' 2013-02-01 08:45:54.616445 [NOTICE] switch_loadable_module.c:298 Adding API Function 'event_sink' 2013-02-01 08:45:54.617375 [INFO] mod_sofia.c:5728 Starting initial message thread. 2013-02-01 08:45:54.617457 [WARNING] sofia.c:1592 MSG Thread 0 Started 2013-02-01 08:45:54.617516 [DEBUG] sofia.c:3581 debug [0] 2013-02-01 08:45:54.617526 [DEBUG] sofia.c:3581 sip-trace [no] 2013-02-01 08:45:54.617553 [DEBUG] sofia.c:3581 sip-capture [no] 2013-02-01 08:45:54.617568 [DEBUG] sofia.c:3581 rfc2833-pt [101] 2013-02-01 08:45:54.617577 [DEBUG] sofia.c:3581 sip-port [5080] 2013-02-01 08:45:54.617584 [DEBUG] sofia.c:3581 dialplan [XML] 2013-02-01 08:45:54.617598 [DEBUG] sofia.c:3581 context [public] 2013-02-01 08:45:54.617606 [DEBUG] sofia.c:3581 dtmf-duration [2000] 2013-02-01 08:45:54.617615 [DEBUG] sofia.c:3581 inbound-codec-prefs [G722,PCMU,PCMA,GSM] 2013-02-01 08:45:54.617629 [DEBUG] sofia.c:3581 outbound-codec-prefs [PCMU,PCMA,GSM] 2013-02-01 08:45:54.617641 [DEBUG] sofia.c:3581 hold-music [%{bong-ring}] 2013-02-01 08:45:54.617649 [DEBUG] sofia.c:3581 rtp-timer-name [soft] 2013-02-01 08:45:54.617660 [DEBUG] sofia.c:3581 local-network-acl [localnet.auto] 2013-02-01 08:45:54.617670 [DEBUG] sofia.c:3581 manage-presence [false] 2013-02-01 08:45:54.617683 [DEBUG] sofia.c:3581 inbound-codec-negotiation [generous] 2013-02-01 08:45:54.617693 [DEBUG] sofia.c:3581 nonce-ttl [60] 2013-02-01 08:45:54.617700 [DEBUG] sofia.c:3581 auth-calls [true] 2013-02-01 08:45:54.617712 [DEBUG] sofia.c:3581 inbound-late-negotiation [true] 2013-02-01 08:45:54.617719 [DEBUG] sofia.c:3581 inbound-zrtp-passthru [true] 2013-02-01 08:45:54.617725 [DEBUG] sofia.c:3581 rtp-ip [XXX.XXX.XXX.10] 2013-02-01 08:45:54.617731 [DEBUG] sofia.c:3581 sip-ip [XXX.XXX.XXX.10] 2013-02-01 08:45:54.617738 [DEBUG] sofia.c:3581 ext-rtp-ip [auto-nat] 2013-02-01 08:45:54.617744 [DEBUG] sofia.c:3581 ext-sip-ip [auto-nat] 2013-02-01 08:45:54.617749 [DEBUG] sofia.c:3581 rtp-timeout-sec [300] 2013-02-01 08:45:54.617755 [DEBUG] sofia.c:3581 rtp-hold-timeout-sec [1800] 2013-02-01 08:45:54.617767 [DEBUG] sofia.c:3581 tls [false] 2013-02-01 08:45:54.617778 [DEBUG] sofia.c:3581 tls-only [false] 2013-02-01 08:45:54.617787 [DEBUG] sofia.c:3581 tls-bind-params [transport=tls] 2013-02-01 08:45:54.617801 [DEBUG] sofia.c:3581 tls-sip-port [5081] 2013-02-01 08:45:54.617812 [DEBUG] sofia.c:3581 tls-cert-dir [/usr/local/freeswitch/conf/ssl] 2013-02-01 08:45:54.617821 [DEBUG] sofia.c:3581 tls-passphrase [] 2013-02-01 08:45:54.617829 [DEBUG] sofia.c:3581 tls-verify-date [true] 2013-02-01 08:45:54.617839 [DEBUG] sofia.c:3581 tls-verify-policy [none] 2013-02-01 08:45:54.617852 [DEBUG] sofia.c:3581 tls-verify-depth [2] 2013-02-01 08:45:54.617862 [DEBUG] sofia.c:3581 tls-verify-in-subjects [] 2013-02-01 08:45:54.617871 [DEBUG] sofia.c:3581 tls-version [tlsv1] 2013-02-01 08:45:54.617947 [NOTICE] sofia.c:4643 Started Profile external [sofia_reg_external] 2013-02-01 08:45:54.617986 [DEBUG] sofia.c:3581 debug [0] 2013-02-01 08:45:54.617995 [DEBUG] sofia.c:3581 sip-trace [no] 2013-02-01 08:45:54.618003 [DEBUG] sofia.c:2311 Creating agent for external 2013-02-01 08:45:54.618026 [DEBUG] sofia.c:3581 context [public] 2013-02-01 08:45:54.618037 [DEBUG] sofia.c:3581 rfc2833-pt [101] 2013-02-01 08:45:54.618044 [DEBUG] sofia.c:3581 sip-port [5060] 2013-02-01 08:45:54.618049 [DEBUG] sofia.c:3581 dialplan [XML] 2013-02-01 08:45:54.618057 [DEBUG] sofia.c:3581 dtmf-duration [2000] 2013-02-01 08:45:54.618069 [DEBUG] sofia.c:3581 inbound-codec-prefs [G722,PCMU,PCMA,GSM] 2013-02-01 08:45:54.618079 [DEBUG] sofia.c:3581 outbound-codec-prefs [G722,PCMU,PCMA,GSM] 2013-02-01 08:45:54.618087 [DEBUG] sofia.c:3581 use-rtp-timer [true] 2013-02-01 08:45:54.618099 [DEBUG] sofia.c:3581 rtp-timer-name [soft] 2013-02-01 08:45:54.618105 [DEBUG] sofia.c:3581 rtp-ip [::1] 2013-02-01 08:45:54.618111 [DEBUG] sofia.c:3581 sip-ip [::1] 2013-02-01 08:45:54.618119 [DEBUG] sofia.c:3581 hold-music [%{bong-ring}] 2013-02-01 08:45:54.618125 [DEBUG] sofia.c:3581 apply-inbound-acl [domains] 2013-02-01 08:45:54.618132 [DEBUG] sofia.c:3581 record-template [/usr/local/freeswitch/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav] 2013-02-01 08:45:54.618138 [DEBUG] sofia.c:3581 manage-presence [true] 2013-02-01 08:45:54.618144 [DEBUG] sofia.c:3581 inbound-codec-negotiation [generous] 2013-02-01 08:45:54.618151 [DEBUG] sofia.c:3581 tls [false] 2013-02-01 08:45:54.618163 [DEBUG] sofia.c:3581 tls-bind-params [transport=tls] 2013-02-01 08:45:54.618171 [DEBUG] sofia.c:3581 tls-sip-port [5061] 2013-02-01 08:45:54.618180 [DEBUG] sofia.c:3581 tls-cert-dir [/usr/local/freeswitch/conf/ssl] 2013-02-01 08:45:54.618190 [DEBUG] sofia.c:3581 tls-version [tlsv1] 2013-02-01 08:45:54.618198 [DEBUG] sofia.c:3581 inbound-late-negotiation [true] 2013-02-01 08:45:54.618203 [DEBUG] sofia.c:3581 inbound-zrtp-passthru [true] 2013-02-01 08:45:54.618210 [DEBUG] sofia.c:3581 nonce-ttl [60] 2013-02-01 08:45:54.618217 [DEBUG] sofia.c:3581 auth-calls [true] 2013-02-01 08:45:54.618223 [DEBUG] sofia.c:3581 auth-all-packets [false] 2013-02-01 08:45:54.618230 [DEBUG] sofia.c:3581 rtp-timeout-sec [300] 2013-02-01 08:45:54.618238 [DEBUG] sofia.c:3581 rtp-hold-timeout-sec [1800] 2013-02-01 08:45:54.618244 [DEBUG] sofia.c:3581 force-register-domain [XXX.XXX.XXX.10] 2013-02-01 08:45:54.618250 [DEBUG] sofia.c:3581 force-register-db-domain [XXX.XXX.XXX.10] 2013-02-01 08:45:54.618323 [NOTICE] sofia.c:4643 Started Profile internal-ipv6 [sofia_reg_internal-ipv6] 2013-02-01 08:45:54.618348 [DEBUG] sofia.c:3581 debug [0] 2013-02-01 08:45:54.618352 [DEBUG] sofia.c:3581 sip-trace [yes] 2013-02-01 08:45:54.618355 [DEBUG] sofia.c:3581 sip-capture [no] 2013-02-01 08:45:54.618372 [DEBUG] sofia.c:3581 watchdog-enabled [no] 2013-02-01 08:45:54.618380 [DEBUG] sofia.c:3581 watchdog-step-timeout [30000] 2013-02-01 08:45:54.618385 [DEBUG] sofia.c:3581 watchdog-event-timeout [30000] 2013-02-01 08:45:54.618389 [DEBUG] sofia.c:3581 log-auth-failures [false] 2013-02-01 08:45:54.618393 [DEBUG] sofia.c:3581 forward-unsolicited-mwi-notify [false] 2013-02-01 08:45:54.618396 [DEBUG] sofia.c:3581 context [default] 2013-02-01 08:45:54.618404 [DEBUG] sofia.c:3581 rfc2833-pt [101] 2013-02-01 08:45:54.618413 [DEBUG] sofia.c:3581 sip-port [5060] 2013-02-01 08:45:54.618404 [DEBUG] sofia.c:2311 Creating agent for internal-ipv6 2013-02-01 08:45:54.618426 [DEBUG] sofia.c:3581 dialplan [XML] 2013-02-01 08:45:54.618432 [DEBUG] sofia.c:3581 dtmf-duration [2000] 2013-02-01 08:45:54.618436 [DEBUG] sofia.c:3581 inbound-codec-prefs [G722,PCMU,PCMA,GSM] 2013-02-01 08:45:54.618440 [DEBUG] sofia.c:3581 outbound-codec-prefs [G722,PCMU,PCMA,GSM] 2013-02-01 08:45:54.618444 [DEBUG] sofia.c:3581 rtp-timer-name [soft] 2013-02-01 08:45:54.618447 [DEBUG] sofia.c:3581 rtp-ip [XXX.XXX.XXX.10] 2013-02-01 08:45:54.618450 [DEBUG] sofia.c:3581 sip-ip [XXX.XXX.XXX.10] 2013-02-01 08:45:54.618452 [DEBUG] sofia.c:3581 hold-music [%{bong-ring}] 2013-02-01 08:45:54.618455 [DEBUG] sofia.c:3581 apply-nat-acl [nat.auto] 2013-02-01 08:45:54.618461 [DEBUG] sofia.c:3581 local-network-acl [localnet.auto] 2013-02-01 08:45:54.618464 [DEBUG] sofia.c:3581 apply-register-acl [domains] 2013-02-01 08:45:54.618468 [DEBUG] sofia.c:3581 record-path [/usr/local/freeswitch/recordings] 2013-02-01 08:45:54.618471 [DEBUG] sofia.c:3581 record-template [${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav] 2013-02-01 08:45:54.618474 [DEBUG] sofia.c:3581 manage-presence [true] 2013-02-01 08:45:54.618478 [DEBUG] sofia.c:3581 presence-hosts [XXX.XXX.XXX.10,XXX.XXX.XXX.10] 2013-02-01 08:45:54.618482 [DEBUG] sofia.c:3581 presence-privacy [false] 2013-02-01 08:45:54.618485 [DEBUG] sofia.c:3581 inbound-codec-negotiation [generous] 2013-02-01 08:45:54.618488 [DEBUG] sofia.c:3581 tls [false] 2013-02-01 08:45:54.618492 [DEBUG] sofia.c:3581 tls-only [false] 2013-02-01 08:45:54.618497 [DEBUG] sofia.c:3581 tls-bind-params [transport=tls] 2013-02-01 08:45:54.618501 [DEBUG] sofia.c:3581 tls-sip-port [5061] 2013-02-01 08:45:54.618507 [DEBUG] sofia.c:3581 tls-cert-dir [/usr/local/freeswitch/conf/ssl] 2013-02-01 08:45:54.618511 [DEBUG] sofia.c:3581 tls-passphrase [] 2013-02-01 08:45:54.618518 [DEBUG] sofia.c:3581 tls-verify-date [true] 2013-02-01 08:45:54.618525 [DEBUG] sofia.c:3581 tls-verify-policy [none] 2013-02-01 08:45:54.618529 [DEBUG] sofia.c:3581 tls-verify-depth [2] 2013-02-01 08:45:54.618536 [DEBUG] sofia.c:3581 tls-verify-in-subjects [] 2013-02-01 08:45:54.618542 [DEBUG] sofia.c:3581 tls-version [tlsv1] 2013-02-01 08:45:54.618546 [DEBUG] sofia.c:3581 inbound-late-negotiation [true] 2013-02-01 08:45:54.618549 [DEBUG] sofia.c:3581 inbound-zrtp-passthru [true] 2013-02-01 08:45:54.618552 [DEBUG] sofia.c:3581 nonce-ttl [60] 2013-02-01 08:45:54.618555 [DEBUG] sofia.c:3581 auth-calls [true] 2013-02-01 08:45:54.618559 [DEBUG] sofia.c:3581 inbound-reg-force-matching-username [true] 2013-02-01 08:45:54.618565 [DEBUG] sofia.c:3581 auth-all-packets [false] 2013-02-01 08:45:54.618570 [DEBUG] sofia.c:3581 ext-rtp-ip [auto-nat] 2013-02-01 08:45:54.618573 [DEBUG] sofia.c:3581 ext-sip-ip [auto-nat] 2013-02-01 08:45:54.618577 [DEBUG] sofia.c:3581 rtp-timeout-sec [300] 2013-02-01 08:45:54.618581 [DEBUG] sofia.c:3581 rtp-hold-timeout-sec [1800] 2013-02-01 08:45:54.618585 [DEBUG] sofia.c:3581 force-register-domain [XXX.XXX.XXX.10] 2013-02-01 08:45:54.618590 [DEBUG] sofia.c:3581 force-subscription-domain [XXX.XXX.XXX.10] 2013-02-01 08:45:54.618593 [DEBUG] sofia.c:3581 force-register-db-domain [XXX.XXX.XXX.10] 2013-02-01 08:45:54.618597 [DEBUG] sofia.c:3581 challenge-realm [auto_from] 2013-02-01 08:45:54.618694 [DEBUG] sofia.c:2311 Creating agent for internal 2013-02-01 08:45:54.618725 [NOTICE] sofia.c:4643 Started Profile internal [sofia_reg_internal] 2013-02-01 08:45:54.618742 [DEBUG] mod_sofia.c:5741 Waiting for profiles to start 2013-02-01 08:45:54.621827 [DEBUG] sofia.c:2401 Created agent for internal 2013-02-01 08:45:54.621827 [DEBUG] sofia.c:2401 Created agent for internal-ipv6 2013-02-01 08:45:54.621853 [DEBUG] sofia.c:2444 Set params for internal 2013-02-01 08:45:54.621854 [DEBUG] sofia.c:2444 Set params for internal-ipv6 2013-02-01 08:45:54.621856 [DEBUG] sofia.c:2488 Activated db for internal 2013-02-01 08:45:54.621856 [DEBUG] sofia.c:2488 Activated db for internal-ipv6 2013-02-01 08:45:54.622130 [DEBUG] sofia.c:2401 Created agent for external 2013-02-01 08:45:54.622151 [DEBUG] sofia.c:2444 Set params for external 2013-02-01 08:45:54.622154 [DEBUG] sofia.c:2488 Activated db for external 2013-02-01 08:45:54.622741 [INFO] switch_core_sqldb.c:1329 sofia:internal-ipv6 Starting SQL thread. 2013-02-01 08:45:54.622903 [DEBUG] sofia.c:2525 Starting thread for internal-ipv6 2013-02-01 08:45:54.622926 [DEBUG] sofia.c:2211 Launching worker thread for internal-ipv6 2013-02-01 08:45:54.623392 [INFO] switch_core_sqldb.c:1329 sofia:internal Starting SQL thread. 2013-02-01 08:45:54.623445 [DEBUG] sofia.c:2525 Starting thread for internal 2013-02-01 08:45:54.623449 [DEBUG] sofia.c:2211 Launching worker thread for internal 2013-02-01 08:45:54.623473 [NOTICE] sofia.c:3237 Adding Alias [XXX.XXX.XXX.10] for profile [internal] 2013-02-01 08:45:54.623619 [INFO] switch_core_sqldb.c:1329 sofia:external Starting SQL thread. 2013-02-01 08:45:54.623661 [DEBUG] sofia.c:2525 Starting thread for external 2013-02-01 08:45:54.623665 [DEBUG] sofia.c:2211 Launching worker thread for external 2013-02-01 08:45:54.623697 [NOTICE] sofia_reg.c:2992 Added gateway 'thinktel' to profile 'external' 2013-02-01 08:45:56.119151 [NOTICE] switch_loadable_module.c:146 Adding Endpoint 'sofia' 2013-02-01 08:45:56.119179 [NOTICE] switch_loadable_module.c:146 Adding Endpoint 'rtp' 2013-02-01 08:45:56.119201 [NOTICE] switch_loadable_module.c:254 Adding Application 'sofia_sla' 2013-02-01 08:45:56.119225 [NOTICE] switch_loadable_module.c:298 Adding API Function 'sofia' 2013-02-01 08:45:56.119252 [NOTICE] switch_loadable_module.c:298 Adding API Function 'sofia_gateway_data' 2013-02-01 08:45:56.119293 [NOTICE] switch_loadable_module.c:298 Adding API Function 'sofia_username_of' 2013-02-01 08:45:56.119314 [NOTICE] switch_loadable_module.c:298 Adding API Function 'sofia_contact' 2013-02-01 08:45:56.119350 [NOTICE] switch_loadable_module.c:298 Adding API Function 'sofia_count_reg' 2013-02-01 08:45:56.119374 [NOTICE] switch_loadable_module.c:298 Adding API Function 'sofia_dig' 2013-02-01 08:45:56.119397 [NOTICE] switch_loadable_module.c:403 Adding Chat interface 'sip' 2013-02-01 08:45:56.119432 [NOTICE] switch_loadable_module.c:447 Adding Management interface 'mod_sofia' OID[.1.3.6.1.4.1.27880.1001] 2013-02-01 08:45:56.119714 [NOTICE] switch_loadable_module.c:146 Adding Endpoint 'loopback' 2013-02-01 08:45:56.119736 [NOTICE] switch_loadable_module.c:254 Adding Application 'unloop' 2013-02-01 08:45:56.121043 [NOTICE] switch_loadable_module.c:298 Adding API Function 'acl' 2013-02-01 08:45:56.121067 [NOTICE] switch_loadable_module.c:298 Adding API Function 'alias' 2013-02-01 08:45:56.121089 [NOTICE] switch_loadable_module.c:298 Adding API Function 'banner' 2013-02-01 08:45:56.121109 [NOTICE] switch_loadable_module.c:298 Adding API Function 'bgapi' 2013-02-01 08:45:56.121133 [NOTICE] switch_loadable_module.c:298 Adding API Function 'bg_system' 2013-02-01 08:45:56.121156 [NOTICE] switch_loadable_module.c:298 Adding API Function 'break' 2013-02-01 08:45:56.121180 [NOTICE] switch_loadable_module.c:298 Adding API Function 'complete' 2013-02-01 08:45:56.121202 [NOTICE] switch_loadable_module.c:298 Adding API Function 'cond' 2013-02-01 08:45:56.121225 [NOTICE] switch_loadable_module.c:298 Adding API Function 'console_complete' 2013-02-01 08:45:56.121247 [NOTICE] switch_loadable_module.c:298 Adding API Function 'console_complete_xml' 2013-02-01 08:45:56.121271 [NOTICE] switch_loadable_module.c:298 Adding API Function 'create_uuid' 2013-02-01 08:45:56.121296 [NOTICE] switch_loadable_module.c:298 Adding API Function 'db_cache' 2013-02-01 08:45:56.121320 [NOTICE] switch_loadable_module.c:298 Adding API Function 'domain_exists' 2013-02-01 08:45:56.121346 [NOTICE] switch_loadable_module.c:298 Adding API Function 'echo' 2013-02-01 08:45:56.121362 [NOTICE] switch_loadable_module.c:298 Adding API Function 'escape' 2013-02-01 08:45:56.121386 [NOTICE] switch_loadable_module.c:298 Adding API Function 'eval' 2013-02-01 08:45:56.121409 [NOTICE] switch_loadable_module.c:298 Adding API Function 'expand' 2013-02-01 08:45:56.121432 [NOTICE] switch_loadable_module.c:298 Adding API Function 'find_user_xml' 2013-02-01 08:45:56.121454 [NOTICE] switch_loadable_module.c:298 Adding API Function 'fsctl' 2013-02-01 08:45:56.121471 [NOTICE] switch_loadable_module.c:298 Adding API Function '...' 2013-02-01 08:45:56.121494 [NOTICE] switch_loadable_module.c:298 Adding API Function 'shutdown' 2013-02-01 08:45:56.121518 [NOTICE] switch_loadable_module.c:298 Adding API Function 'version' 2013-02-01 08:45:56.121550 [NOTICE] switch_loadable_module.c:298 Adding API Function 'global_getvar' 2013-02-01 08:45:56.121570 [NOTICE] switch_loadable_module.c:298 Adding API Function 'global_setvar' 2013-02-01 08:45:56.121589 [NOTICE] switch_loadable_module.c:298 Adding API Function 'group_call' 2013-02-01 08:45:56.121611 [NOTICE] switch_loadable_module.c:298 Adding API Function 'help' 2013-02-01 08:45:56.121631 [NOTICE] switch_loadable_module.c:298 Adding API Function 'host_lookup' 2013-02-01 08:45:56.121653 [NOTICE] switch_loadable_module.c:298 Adding API Function 'hostname' 2013-02-01 08:45:56.121676 [NOTICE] switch_loadable_module.c:298 Adding API Function 'switchname' 2013-02-01 08:45:56.121697 [NOTICE] switch_loadable_module.c:298 Adding API Function 'hupall' 2013-02-01 08:45:56.121721 [NOTICE] switch_loadable_module.c:298 Adding API Function 'in_group' 2013-02-01 08:45:56.121744 [NOTICE] switch_loadable_module.c:298 Adding API Function 'is_lan_addr' 2013-02-01 08:45:56.121761 [NOTICE] switch_loadable_module.c:298 Adding API Function 'limit_usage' 2013-02-01 08:45:56.121787 [NOTICE] switch_loadable_module.c:298 Adding API Function 'limit_hash_usage' 2013-02-01 08:45:56.121810 [NOTICE] switch_loadable_module.c:298 Adding API Function 'limit_status' 2013-02-01 08:45:56.121836 [NOTICE] switch_loadable_module.c:298 Adding API Function 'limit_reset' 2013-02-01 08:45:56.121859 [NOTICE] switch_loadable_module.c:298 Adding API Function 'limit_interval_reset' 2013-02-01 08:45:56.121881 [NOTICE] switch_loadable_module.c:298 Adding API Function 'list_users' 2013-02-01 08:45:56.121910 [NOTICE] switch_loadable_module.c:298 Adding API Function 'load' 2013-02-01 08:45:56.121927 [NOTICE] switch_loadable_module.c:298 Adding API Function 'log' 2013-02-01 08:45:56.121949 [NOTICE] switch_loadable_module.c:298 Adding API Function 'md5' 2013-02-01 08:45:56.121973 [NOTICE] switch_loadable_module.c:298 Adding API Function 'module_exists' 2013-02-01 08:45:56.121997 [NOTICE] switch_loadable_module.c:298 Adding API Function 'msleep' 2013-02-01 08:45:56.122015 [NOTICE] switch_loadable_module.c:298 Adding API Function 'nat_map' 2013-02-01 08:45:56.122035 [NOTICE] switch_loadable_module.c:298 Adding API Function 'originate' 2013-02-01 08:45:56.122062 [NOTICE] switch_loadable_module.c:298 Adding API Function 'pause' 2013-02-01 08:45:56.122087 [NOTICE] switch_loadable_module.c:298 Adding API Function 'regex' 2013-02-01 08:45:56.122110 [NOTICE] switch_loadable_module.c:298 Adding API Function 'reloadacl' 2013-02-01 08:45:56.122135 [NOTICE] switch_loadable_module.c:298 Adding API Function 'reload' 2013-02-01 08:45:56.122158 [NOTICE] switch_loadable_module.c:298 Adding API Function 'reloadxml' 2013-02-01 08:45:56.122183 [NOTICE] switch_loadable_module.c:298 Adding API Function 'replace' 2013-02-01 08:45:56.122206 [NOTICE] switch_loadable_module.c:298 Adding API Function 'say_string' 2013-02-01 08:45:56.122231 [NOTICE] switch_loadable_module.c:298 Adding API Function 'sched_api' 2013-02-01 08:45:56.122258 [NOTICE] switch_loadable_module.c:298 Adding API Function 'sched_broadcast' 2013-02-01 08:45:56.122287 [NOTICE] switch_loadable_module.c:298 Adding API Function 'sched_del' 2013-02-01 08:45:56.122313 [NOTICE] switch_loadable_module.c:298 Adding API Function 'sched_hangup' 2013-02-01 08:45:56.122339 [NOTICE] switch_loadable_module.c:298 Adding API Function 'sched_transfer' 2013-02-01 08:45:56.122364 [NOTICE] switch_loadable_module.c:298 Adding API Function 'show' 2013-02-01 08:45:56.122391 [NOTICE] switch_loadable_module.c:298 Adding API Function 'sql_escape' 2013-02-01 08:45:56.122417 [NOTICE] switch_loadable_module.c:298 Adding API Function 'status' 2013-02-01 08:45:56.122441 [NOTICE] switch_loadable_module.c:298 Adding API Function 'strftime_tz' 2013-02-01 08:45:56.122466 [NOTICE] switch_loadable_module.c:298 Adding API Function 'stun' 2013-02-01 08:45:56.122490 [NOTICE] switch_loadable_module.c:298 Adding API Function 'system' 2013-02-01 08:45:56.122516 [NOTICE] switch_loadable_module.c:298 Adding API Function 'time_test' 2013-02-01 08:45:56.122539 [NOTICE] switch_loadable_module.c:298 Adding API Function 'timer_test' 2013-02-01 08:45:56.122565 [NOTICE] switch_loadable_module.c:298 Adding API Function 'tone_detect' 2013-02-01 08:45:56.122591 [NOTICE] switch_loadable_module.c:298 Adding API Function 'unload' 2013-02-01 08:45:56.122616 [NOTICE] switch_loadable_module.c:298 Adding API Function 'unsched_api' 2013-02-01 08:45:56.122640 [NOTICE] switch_loadable_module.c:298 Adding API Function 'reg_url' 2013-02-01 08:45:56.122664 [NOTICE] switch_loadable_module.c:298 Adding API Function 'url_decode' 2013-02-01 08:45:56.122687 [NOTICE] switch_loadable_module.c:298 Adding API Function 'url_encode' 2013-02-01 08:45:56.122713 [NOTICE] switch_loadable_module.c:298 Adding API Function 'user_data' 2013-02-01 08:45:56.122739 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_early_ok' 2013-02-01 08:45:56.122763 [NOTICE] switch_loadable_module.c:298 Adding API Function 'user_exists' 2013-02-01 08:45:56.122788 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_answer' 2013-02-01 08:45:56.122813 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_audio' 2013-02-01 08:45:56.122837 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_break' 2013-02-01 08:45:56.122861 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_bridge' 2013-02-01 08:45:56.122885 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_broadcast' 2013-02-01 08:45:56.122915 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_buglist' 2013-02-01 08:45:56.122943 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_chat' 2013-02-01 08:45:56.122965 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_debug_audio' 2013-02-01 08:45:56.122991 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_deflect' 2013-02-01 08:45:56.123014 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_displace' 2013-02-01 08:45:56.123039 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_display' 2013-02-01 08:45:56.123066 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_dump' 2013-02-01 08:45:56.123090 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_exists' 2013-02-01 08:45:56.123115 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_fileman' 2013-02-01 08:45:56.123138 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_flush_dtmf' 2013-02-01 08:45:56.123163 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_getvar' 2013-02-01 08:45:56.123187 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_hold' 2013-02-01 08:45:56.123213 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_kill' 2013-02-01 08:45:56.123237 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_send_message' 2013-02-01 08:45:56.123262 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_send_info' 2013-02-01 08:45:56.123288 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_video_refresh' 2013-02-01 08:45:56.123312 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_outgoing_answer' 2013-02-01 08:45:56.123337 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_limit' 2013-02-01 08:45:56.123363 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_limit_release' 2013-02-01 08:45:56.123389 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_limit_release' 2013-02-01 08:45:56.123413 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_loglevel' 2013-02-01 08:45:56.123437 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_media' 2013-02-01 08:45:56.123462 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_media_reneg' 2013-02-01 08:45:56.123486 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_park' 2013-02-01 08:45:56.123511 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_pause' 2013-02-01 08:45:56.123535 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_phone_event' 2013-02-01 08:45:56.123560 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_pre_answer' 2013-02-01 08:45:56.123583 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_preprocess' 2013-02-01 08:45:56.123609 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_record' 2013-02-01 08:45:56.123632 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_recovery_refresh' 2013-02-01 08:45:56.123656 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_recv_dtmf' 2013-02-01 08:45:56.123679 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_send_dtmf' 2013-02-01 08:45:56.123705 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_session_heartbeat' 2013-02-01 08:45:56.123729 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_setvar_multi' 2013-02-01 08:45:56.123754 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_setvar' 2013-02-01 08:45:56.123778 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_transfer' 2013-02-01 08:45:56.123803 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_dual_transfer' 2013-02-01 08:45:56.123830 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_simplify' 2013-02-01 08:45:56.123855 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_jitterbuffer' 2013-02-01 08:45:56.123881 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_zombie_exec' 2013-02-01 08:45:56.123920 [NOTICE] switch_loadable_module.c:298 Adding API Function 'xml_flush_cache' 2013-02-01 08:45:56.123939 [NOTICE] switch_loadable_module.c:298 Adding API Function 'xml_locate' 2013-02-01 08:45:56.123965 [NOTICE] switch_loadable_module.c:298 Adding API Function 'xml_wrap' 2013-02-01 08:45:56.123989 [NOTICE] switch_loadable_module.c:298 Adding API Function 'file_exists' 2013-02-01 08:45:56.124679 [NOTICE] switch_loadable_module.c:254 Adding Application 'conference' 2013-02-01 08:45:56.124697 [NOTICE] switch_loadable_module.c:254 Adding Application 'conference_set_auto_outcall' 2013-02-01 08:45:56.124721 [NOTICE] switch_loadable_module.c:298 Adding API Function 'conference' 2013-02-01 08:45:56.124745 [NOTICE] switch_loadable_module.c:403 Adding Chat interface 'conf' 2013-02-01 08:45:56.125819 [NOTICE] switch_loadable_module.c:254 Adding Application 'db' 2013-02-01 08:45:56.125841 [NOTICE] switch_loadable_module.c:254 Adding Application 'group' 2013-02-01 08:45:56.125857 [NOTICE] switch_loadable_module.c:298 Adding API Function 'db' 2013-02-01 08:45:56.125876 [NOTICE] switch_loadable_module.c:298 Adding API Function 'group' 2013-02-01 08:45:56.125904 [NOTICE] switch_loadable_module.c:473 Adding Limit interface 'db' 2013-02-01 08:45:56.624950 [NOTICE] switch_loadable_module.c:146 Adding Endpoint 'error' 2013-02-01 08:45:56.624979 [NOTICE] switch_loadable_module.c:146 Adding Endpoint 'group' 2013-02-01 08:45:56.624999 [NOTICE] switch_loadable_module.c:146 Adding Endpoint 'user' 2013-02-01 08:45:56.625016 [NOTICE] switch_loadable_module.c:146 Adding Endpoint 'pickup' 2013-02-01 08:45:56.625031 [NOTICE] switch_loadable_module.c:214 Adding Dialplan 'inline' 2013-02-01 08:45:56.625046 [NOTICE] switch_loadable_module.c:254 Adding Application 'blind_transfer_ack' 2013-02-01 08:45:56.625069 [NOTICE] switch_loadable_module.c:254 Adding Application 'bind_digit_action' 2013-02-01 08:45:56.625093 [NOTICE] switch_loadable_module.c:254 Adding Application 'capture' 2013-02-01 08:45:56.625115 [NOTICE] switch_loadable_module.c:254 Adding Application 'clear_digit_action' 2013-02-01 08:45:56.625137 [NOTICE] switch_loadable_module.c:254 Adding Application 'digit_action_set_realm' 2013-02-01 08:45:56.625163 [NOTICE] switch_loadable_module.c:254 Adding Application 'privacy' 2013-02-01 08:45:56.625187 [NOTICE] switch_loadable_module.c:254 Adding Application 'set_audio_level' 2013-02-01 08:45:56.625212 [NOTICE] switch_loadable_module.c:254 Adding Application 'set_mute' 2013-02-01 08:45:56.625234 [NOTICE] switch_loadable_module.c:254 Adding Application 'flush_dtmf' 2013-02-01 08:45:56.625260 [NOTICE] switch_loadable_module.c:254 Adding Application 'hold' 2013-02-01 08:45:56.625299 [NOTICE] switch_loadable_module.c:254 Adding Application 'unhold' 2013-02-01 08:45:56.625313 [NOTICE] switch_loadable_module.c:254 Adding Application 'mutex' 2013-02-01 08:45:56.625335 [NOTICE] switch_loadable_module.c:254 Adding Application 'page' 2013-02-01 08:45:56.625357 [NOTICE] switch_loadable_module.c:254 Adding Application 'transfer' 2013-02-01 08:45:56.625379 [NOTICE] switch_loadable_module.c:254 Adding Application 'check_acl' 2013-02-01 08:45:56.625403 [NOTICE] switch_loadable_module.c:254 Adding Application 'verbose_events' 2013-02-01 08:45:56.625427 [NOTICE] switch_loadable_module.c:254 Adding Application 'cng_plc' 2013-02-01 08:45:56.625451 [NOTICE] switch_loadable_module.c:254 Adding Application 'early_hangup' 2013-02-01 08:45:56.625474 [NOTICE] switch_loadable_module.c:254 Adding Application 'sleep' 2013-02-01 08:45:56.625490 [NOTICE] switch_loadable_module.c:254 Adding Application 'delay_echo' 2013-02-01 08:45:56.625513 [NOTICE] switch_loadable_module.c:254 Adding Application 'strftime' 2013-02-01 08:45:56.625537 [NOTICE] switch_loadable_module.c:254 Adding Application 'phrase' 2013-02-01 08:45:56.625561 [NOTICE] switch_loadable_module.c:254 Adding Application 'eval' 2013-02-01 08:45:56.625584 [NOTICE] switch_loadable_module.c:254 Adding Application 'stop' 2013-02-01 08:45:56.625607 [NOTICE] switch_loadable_module.c:254 Adding Application 'set_zombie_exec' 2013-02-01 08:45:56.625640 [NOTICE] switch_loadable_module.c:254 Adding Application 'pre_answer' 2013-02-01 08:45:56.625661 [NOTICE] switch_loadable_module.c:254 Adding Application 'answer' 2013-02-01 08:45:56.625680 [NOTICE] switch_loadable_module.c:254 Adding Application 'wait_for_answer' 2013-02-01 08:45:56.625703 [NOTICE] switch_loadable_module.c:254 Adding Application 'hangup' 2013-02-01 08:45:56.625728 [NOTICE] switch_loadable_module.c:254 Adding Application 'set_name' 2013-02-01 08:45:56.625752 [NOTICE] switch_loadable_module.c:254 Adding Application 'presence' 2013-02-01 08:45:56.625775 [NOTICE] switch_loadable_module.c:254 Adding Application 'log' 2013-02-01 08:45:56.625798 [NOTICE] switch_loadable_module.c:254 Adding Application 'info' 2013-02-01 08:45:56.625817 [NOTICE] switch_loadable_module.c:254 Adding Application 'event' 2013-02-01 08:45:56.625842 [NOTICE] switch_loadable_module.c:254 Adding Application 'sound_test' 2013-02-01 08:45:56.625865 [NOTICE] switch_loadable_module.c:254 Adding Application 'export' 2013-02-01 08:45:56.625889 [NOTICE] switch_loadable_module.c:254 Adding Application 'bridge_export' 2013-02-01 08:45:56.625913 [NOTICE] switch_loadable_module.c:254 Adding Application 'set' 2013-02-01 08:45:56.625937 [NOTICE] switch_loadable_module.c:254 Adding Application 'multiset' 2013-02-01 08:45:56.625962 [NOTICE] switch_loadable_module.c:254 Adding Application 'push' 2013-02-01 08:45:56.625985 [NOTICE] switch_loadable_module.c:254 Adding Application 'unshift' 2013-02-01 08:45:56.626009 [NOTICE] switch_loadable_module.c:254 Adding Application 'set_global' 2013-02-01 08:45:56.626028 [NOTICE] switch_loadable_module.c:254 Adding Application 'set_profile_var' 2013-02-01 08:45:56.626057 [NOTICE] switch_loadable_module.c:254 Adding Application 'unset' 2013-02-01 08:45:56.626080 [NOTICE] switch_loadable_module.c:254 Adding Application 'ring_ready' 2013-02-01 08:45:56.626105 [NOTICE] switch_loadable_module.c:254 Adding Application 'remove_bugs' 2013-02-01 08:45:56.626129 [NOTICE] switch_loadable_module.c:254 Adding Application 'break' 2013-02-01 08:45:56.626154 [NOTICE] switch_loadable_module.c:254 Adding Application 'detect_speech' 2013-02-01 08:45:56.626179 [NOTICE] switch_loadable_module.c:254 Adding Application 'play_and_detect_speech' 2013-02-01 08:45:56.626203 [NOTICE] switch_loadable_module.c:254 Adding Application 'ivr' 2013-02-01 08:45:56.626229 [NOTICE] switch_loadable_module.c:254 Adding Application 'redirect' 2013-02-01 08:45:56.626253 [NOTICE] switch_loadable_module.c:254 Adding Application 'video_refresh' 2013-02-01 08:45:56.626278 [NOTICE] switch_loadable_module.c:254 Adding Application 'send_info' 2013-02-01 08:45:56.626304 [NOTICE] switch_loadable_module.c:254 Adding Application 'jitterbuffer' 2013-02-01 08:45:56.626328 [NOTICE] switch_loadable_module.c:254 Adding Application 'send_display' 2013-02-01 08:45:56.626355 [NOTICE] switch_loadable_module.c:254 Adding Application 'respond' 2013-02-01 08:45:56.626378 [NOTICE] switch_loadable_module.c:254 Adding Application 'deflect' 2013-02-01 08:45:56.626412 [NOTICE] switch_loadable_module.c:254 Adding Application 'recovery_refresh' 2013-02-01 08:45:56.626435 [NOTICE] switch_loadable_module.c:254 Adding Application 'queue_dtmf' 2013-02-01 08:45:56.626459 [NOTICE] switch_loadable_module.c:254 Adding Application 'send_dtmf' 2013-02-01 08:45:56.626484 [NOTICE] switch_loadable_module.c:254 Adding Application 'sched_cancel' 2013-02-01 08:45:56.626509 [NOTICE] switch_loadable_module.c:254 Adding Application 'sched_hangup' 2013-02-01 08:45:56.626533 [NOTICE] switch_loadable_module.c:254 Adding Application 'sched_broadcast' 2013-02-01 08:45:56.626559 [NOTICE] switch_loadable_module.c:254 Adding Application 'sched_transfer' 2013-02-01 08:45:56.626583 [NOTICE] switch_loadable_module.c:254 Adding Application 'execute_extension' 2013-02-01 08:45:56.626608 [NOTICE] switch_loadable_module.c:254 Adding Application 'sched_heartbeat' 2013-02-01 08:45:56.626632 [NOTICE] switch_loadable_module.c:254 Adding Application 'enable_heartbeat' 2013-02-01 08:45:56.626658 [NOTICE] switch_loadable_module.c:254 Adding Application 'mkdir' 2013-02-01 08:45:56.626682 [NOTICE] switch_loadable_module.c:254 Adding Application 'rename' 2013-02-01 08:45:56.626708 [NOTICE] switch_loadable_module.c:254 Adding Application 'soft_hold' 2013-02-01 08:45:56.626732 [NOTICE] switch_loadable_module.c:254 Adding Application 'bind_meta_app' 2013-02-01 08:45:56.626759 [NOTICE] switch_loadable_module.c:254 Adding Application 'unbind_meta_app' 2013-02-01 08:45:56.626783 [NOTICE] switch_loadable_module.c:254 Adding Application 'block_dfmf' 2013-02-01 08:45:56.626809 [NOTICE] switch_loadable_module.c:254 Adding Application 'unblock_dtmf' 2013-02-01 08:45:56.626833 [NOTICE] switch_loadable_module.c:254 Adding Application 'intercept' 2013-02-01 08:45:56.626859 [NOTICE] switch_loadable_module.c:254 Adding Application 'eavesdrop' 2013-02-01 08:45:56.626882 [NOTICE] switch_loadable_module.c:254 Adding Application 'three_way' 2013-02-01 08:45:56.626908 [NOTICE] switch_loadable_module.c:254 Adding Application 'set_user' 2013-02-01 08:45:56.626932 [NOTICE] switch_loadable_module.c:254 Adding Application 'stop_dtmf' 2013-02-01 08:45:56.626956 [NOTICE] switch_loadable_module.c:254 Adding Application 'start_dtmf' 2013-02-01 08:45:56.626979 [NOTICE] switch_loadable_module.c:254 Adding Application 'stop_dtmf_generate' 2013-02-01 08:45:56.627006 [NOTICE] switch_loadable_module.c:254 Adding Application 'start_dtmf_generate' 2013-02-01 08:45:56.627030 [NOTICE] switch_loadable_module.c:254 Adding Application 'stop_tone_detect' 2013-02-01 08:45:56.627054 [NOTICE] switch_loadable_module.c:254 Adding Application 'fax_detect' 2013-02-01 08:45:56.627079 [NOTICE] switch_loadable_module.c:254 Adding Application 'tone_detect' 2013-02-01 08:45:56.627102 [NOTICE] switch_loadable_module.c:254 Adding Application 'echo' 2013-02-01 08:45:56.627128 [NOTICE] switch_loadable_module.c:254 Adding Application 'park' 2013-02-01 08:45:56.627150 [NOTICE] switch_loadable_module.c:254 Adding Application 'park_state' 2013-02-01 08:45:56.627176 [NOTICE] switch_loadable_module.c:254 Adding Application 'gentones' 2013-02-01 08:45:56.627201 [NOTICE] switch_loadable_module.c:254 Adding Application 'playback' 2013-02-01 08:45:56.627225 [NOTICE] switch_loadable_module.c:254 Adding Application 'endless_playback' 2013-02-01 08:45:56.627251 [NOTICE] switch_loadable_module.c:254 Adding Application 'att_xfer' 2013-02-01 08:45:56.627275 [NOTICE] switch_loadable_module.c:254 Adding Application 'read' 2013-02-01 08:45:56.627301 [NOTICE] switch_loadable_module.c:254 Adding Application 'play_and_get_digits' 2013-02-01 08:45:56.627327 [NOTICE] switch_loadable_module.c:254 Adding Application 'stop_record_session' 2013-02-01 08:45:56.627349 [NOTICE] switch_loadable_module.c:254 Adding Application 'record_session' 2013-02-01 08:45:56.627373 [NOTICE] switch_loadable_module.c:254 Adding Application 'record' 2013-02-01 08:45:56.627400 [NOTICE] switch_loadable_module.c:254 Adding Application 'preprocess' 2013-02-01 08:45:56.627424 [NOTICE] switch_loadable_module.c:254 Adding Application 'stop_displace_session' 2013-02-01 08:45:56.627449 [NOTICE] switch_loadable_module.c:254 Adding Application 'displace_session' 2013-02-01 08:45:56.627476 [NOTICE] switch_loadable_module.c:254 Adding Application 'speak' 2013-02-01 08:45:56.627500 [NOTICE] switch_loadable_module.c:254 Adding Application 'clear_speech_cache' 2013-02-01 08:45:56.627525 [NOTICE] switch_loadable_module.c:254 Adding Application 'bridge' 2013-02-01 08:45:56.627549 [NOTICE] switch_loadable_module.c:254 Adding Application 'system' 2013-02-01 08:45:56.627576 [NOTICE] switch_loadable_module.c:254 Adding Application 'say' 2013-02-01 08:45:56.627600 [NOTICE] switch_loadable_module.c:254 Adding Application 'wait_for_silence' 2013-02-01 08:45:56.627626 [NOTICE] switch_loadable_module.c:254 Adding Application 'session_loglevel' 2013-02-01 08:45:56.627651 [NOTICE] switch_loadable_module.c:254 Adding Application 'limit' 2013-02-01 08:45:56.627677 [NOTICE] switch_loadable_module.c:254 Adding Application 'limit_hash' 2013-02-01 08:45:56.627700 [NOTICE] switch_loadable_module.c:254 Adding Application 'limit_execute' 2013-02-01 08:45:56.627727 [NOTICE] switch_loadable_module.c:254 Adding Application 'limit_hash_execute' 2013-02-01 08:45:56.627752 [NOTICE] switch_loadable_module.c:254 Adding Application 'pickup' 2013-02-01 08:45:56.627778 [NOTICE] switch_loadable_module.c:298 Adding API Function 'strepoch' 2013-02-01 08:45:56.627802 [NOTICE] switch_loadable_module.c:298 Adding API Function 'page' 2013-02-01 08:45:56.627829 [NOTICE] switch_loadable_module.c:298 Adding API Function 'strmicroepoch' 2013-02-01 08:45:56.627853 [NOTICE] switch_loadable_module.c:298 Adding API Function 'chat' 2013-02-01 08:45:56.627880 [NOTICE] switch_loadable_module.c:298 Adding API Function 'strftime' 2013-02-01 08:45:56.627904 [NOTICE] switch_loadable_module.c:298 Adding API Function 'presence' 2013-02-01 08:45:56.627931 [NOTICE] switch_loadable_module.c:322 Adding File Format 'file_string' 2013-02-01 08:45:56.627955 [NOTICE] switch_loadable_module.c:403 Adding Chat interface 'event' 2013-02-01 08:45:56.627981 [NOTICE] switch_loadable_module.c:403 Adding Chat interface 'api' 2013-02-01 08:45:56.628245 [NOTICE] switch_loadable_module.c:298 Adding API Function 'expr' 2013-02-01 08:45:56.629665 [INFO] switch_core_sqldb.c:1329 fifo Starting SQL thread. 2013-02-01 08:45:56.631006 [INFO] mod_fifo.c:4260 cool_fifo at XXX.XXX.XXX.10 configured 2013-02-01 08:45:56.631133 [NOTICE] switch_loadable_module.c:254 Adding Application 'fifo' 2013-02-01 08:45:56.631162 [NOTICE] switch_loadable_module.c:254 Adding Application 'fifo_track_call' 2013-02-01 08:45:56.631188 [NOTICE] switch_loadable_module.c:298 Adding API Function 'fifo' 2013-02-01 08:45:56.631212 [NOTICE] switch_loadable_module.c:298 Adding API Function 'fifo_member' 2013-02-01 08:45:56.631238 [NOTICE] switch_loadable_module.c:298 Adding API Function 'fifo_add_outbound' 2013-02-01 08:45:56.631265 [NOTICE] switch_loadable_module.c:298 Adding API Function 'fifo_check_bridge' 2013-02-01 08:45:56.631616 [DEBUG] switch_scheduler.c:214 Added task 3 limit_hash_cleanup (mod_hash) to run at 1359727256 2013-02-01 08:45:56.631712 [NOTICE] switch_loadable_module.c:254 Adding Application 'hash' 2013-02-01 08:45:56.631731 [NOTICE] switch_loadable_module.c:298 Adding API Function 'hash' 2013-02-01 08:45:56.631751 [NOTICE] switch_loadable_module.c:298 Adding API Function 'hash_dump' 2013-02-01 08:45:56.631770 [NOTICE] switch_loadable_module.c:298 Adding API Function 'hash_remote' 2013-02-01 08:45:56.631799 [NOTICE] switch_loadable_module.c:473 Adding Limit interface 'hash' 2013-02-01 08:45:56.633420 [INFO] mod_voicemail.c:738 Added Profile default 2013-02-01 08:45:56.633467 [NOTICE] switch_loadable_module.c:254 Adding Application 'voicemail' 2013-02-01 08:45:56.633493 [NOTICE] switch_loadable_module.c:298 Adding API Function 'voicemail' 2013-02-01 08:45:56.633514 [NOTICE] switch_loadable_module.c:298 Adding API Function 'voicemail_inject' 2013-02-01 08:45:56.633534 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_inject' 2013-02-01 08:45:56.633557 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_boxcount' 2013-02-01 08:45:56.633579 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_prefs' 2013-02-01 08:45:56.633599 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_delete' 2013-02-01 08:45:56.633626 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_read' 2013-02-01 08:45:56.633648 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_list' 2013-02-01 08:45:56.633674 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_fsdb_auth_login' 2013-02-01 08:45:56.633699 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_fsdb_msg_count' 2013-02-01 08:45:56.633727 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_fsdb_msg_list' 2013-02-01 08:45:56.633753 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_fsdb_msg_get' 2013-02-01 08:45:56.633777 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_fsdb_msg_delete' 2013-02-01 08:45:56.633801 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_fsdb_msg_undelete' 2013-02-01 08:45:56.633825 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_fsdb_msg_email' 2013-02-01 08:45:56.633849 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_fsdb_msg_purge' 2013-02-01 08:45:56.633872 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_fsdb_msg_save' 2013-02-01 08:45:56.633903 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_fsdb_msg_forward' 2013-02-01 08:45:56.633922 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_fsdb_pref_greeting_set' 2013-02-01 08:45:56.633944 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_fsdb_pref_recname_set' 2013-02-01 08:45:56.633969 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_fsdb_pref_password_set' 2013-02-01 08:45:56.634201 [NOTICE] switch_loadable_module.c:254 Adding Application 'esf_page_group' 2013-02-01 08:45:56.634421 [NOTICE] switch_loadable_module.c:254 Adding Application 'play_fsv' 2013-02-01 08:45:56.634445 [NOTICE] switch_loadable_module.c:254 Adding Application 'record_fsv' 2013-02-01 08:45:56.634467 [NOTICE] switch_loadable_module.c:322 Adding File Format 'fsv' 2013-02-01 08:45:56.634679 [NOTICE] mod_cluechoo.c:83 Hello World! 2013-02-01 08:45:56.634701 [NOTICE] switch_loadable_module.c:254 Adding Application 'cluechoo' 2013-02-01 08:45:56.634727 [NOTICE] switch_loadable_module.c:298 Adding API Function 'cluechoo' 2013-02-01 08:45:56.634965 [NOTICE] switch_loadable_module.c:254 Adding Application 'valet_park' 2013-02-01 08:45:56.635005 [NOTICE] switch_loadable_module.c:298 Adding API Function 'valet_info' 2013-02-01 08:45:56.635310 [NOTICE] mod_httapi.c:2068 Profile [default] JSON Function [http://www.freeswitch.org/api/index.cgi] 2013-02-01 08:45:56.635356 [NOTICE] switch_loadable_module.c:254 Adding Application 'httapi' 2013-02-01 08:45:56.635382 [NOTICE] switch_loadable_module.c:298 Adding API Function 'httapi' 2013-02-01 08:45:56.635407 [NOTICE] switch_loadable_module.c:322 Adding File Format 'http' 2013-02-01 08:45:56.635637 [NOTICE] switch_loadable_module.c:214 Adding Dialplan 'XML' 2013-02-01 08:45:56.635855 [NOTICE] switch_loadable_module.c:146 Adding Endpoint 'SIP' 2013-02-01 08:45:56.635877 [NOTICE] switch_loadable_module.c:146 Adding Endpoint 'IAX2' 2013-02-01 08:45:56.635907 [NOTICE] switch_loadable_module.c:214 Adding Dialplan 'asterisk' 2013-02-01 08:45:56.635932 [NOTICE] switch_loadable_module.c:254 Adding Application 'Dial' 2013-02-01 08:45:56.635954 [NOTICE] switch_loadable_module.c:254 Adding Application 'Goto' 2013-02-01 08:45:56.635977 [NOTICE] switch_loadable_module.c:254 Adding Application 'AvoidingDeadlock' 2013-02-01 08:45:56.637293 [DEBUG] mod_spandsp_fax.c:192 FAX timer thread started. 2013-02-01 08:45:56.657879 [DEBUG] mod_spandsp.c:777 mod_spandsp loaded, using spandsp library version [20120902 163333] 2013-02-01 08:45:56.657925 [NOTICE] switch_loadable_module.c:146 Adding Endpoint 'modem' 2013-02-01 08:45:56.657953 [NOTICE] switch_loadable_module.c:187 Adding Codec DVI4 6 ADPCM (IMA) 16000hz 10ms 64000bps 2013-02-01 08:45:56.657960 [NOTICE] switch_loadable_module.c:187 Adding Codec DVI4 6 ADPCM (IMA) 16000hz 20ms 64000bps 2013-02-01 08:45:56.657969 [NOTICE] switch_loadable_module.c:187 Adding Codec DVI4 6 ADPCM (IMA) 16000hz 30ms 64000bps 2013-02-01 08:45:56.657974 [NOTICE] switch_loadable_module.c:187 Adding Codec DVI4 6 ADPCM (IMA) 16000hz 40ms 64000bps 2013-02-01 08:45:56.657985 [NOTICE] switch_loadable_module.c:187 Adding Codec DVI4 6 ADPCM (IMA) 16000hz 50ms 64000bps 2013-02-01 08:45:56.657990 [NOTICE] switch_loadable_module.c:187 Adding Codec DVI4 6 ADPCM (IMA) 16000hz 60ms 64000bps 2013-02-01 08:45:56.657994 [NOTICE] switch_loadable_module.c:187 Adding Codec DVI4 5 ADPCM (IMA) 8000hz 10ms 32000bps 2013-02-01 08:45:56.658000 [NOTICE] switch_loadable_module.c:187 Adding Codec DVI4 5 ADPCM (IMA) 8000hz 20ms 32000bps 2013-02-01 08:45:56.658003 [NOTICE] switch_loadable_module.c:187 Adding Codec DVI4 5 ADPCM (IMA) 8000hz 30ms 32000bps 2013-02-01 08:45:56.658008 [NOTICE] switch_loadable_module.c:187 Adding Codec DVI4 5 ADPCM (IMA) 8000hz 40ms 32000bps 2013-02-01 08:45:56.658013 [NOTICE] switch_loadable_module.c:187 Adding Codec DVI4 5 ADPCM (IMA) 8000hz 50ms 32000bps 2013-02-01 08:45:56.658017 [NOTICE] switch_loadable_module.c:187 Adding Codec DVI4 5 ADPCM (IMA) 8000hz 60ms 32000bps 2013-02-01 08:45:56.658020 [NOTICE] switch_loadable_module.c:187 Adding Codec DVI4 5 ADPCM (IMA) 8000hz 70ms 32000bps 2013-02-01 08:45:56.658025 [NOTICE] switch_loadable_module.c:187 Adding Codec DVI4 5 ADPCM (IMA) 8000hz 80ms 32000bps 2013-02-01 08:45:56.658029 [NOTICE] switch_loadable_module.c:187 Adding Codec DVI4 5 ADPCM (IMA) 8000hz 90ms 32000bps 2013-02-01 08:45:56.658033 [NOTICE] switch_loadable_module.c:187 Adding Codec DVI4 5 ADPCM (IMA) 8000hz 100ms 32000bps 2013-02-01 08:45:56.658037 [NOTICE] switch_loadable_module.c:187 Adding Codec DVI4 5 ADPCM (IMA) 8000hz 110ms 32000bps 2013-02-01 08:45:56.658042 [NOTICE] switch_loadable_module.c:187 Adding Codec DVI4 5 ADPCM (IMA) 8000hz 120ms 32000bps 2013-02-01 08:45:56.658064 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-16 124 G.726 16k (AAL2) 8000hz 10ms 16000bps 2013-02-01 08:45:56.658071 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-16 124 G.726 16k (AAL2) 8000hz 20ms 16000bps 2013-02-01 08:45:56.658076 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-16 124 G.726 16k (AAL2) 8000hz 30ms 16000bps 2013-02-01 08:45:56.658080 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-16 124 G.726 16k (AAL2) 8000hz 40ms 16000bps 2013-02-01 08:45:56.658083 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-16 124 G.726 16k (AAL2) 8000hz 50ms 16000bps 2013-02-01 08:45:56.658087 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-16 124 G.726 16k (AAL2) 8000hz 60ms 16000bps 2013-02-01 08:45:56.658092 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-16 124 G.726 16k (AAL2) 8000hz 70ms 16000bps 2013-02-01 08:45:56.658096 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-16 124 G.726 16k (AAL2) 8000hz 80ms 16000bps 2013-02-01 08:45:56.658100 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-16 124 G.726 16k (AAL2) 8000hz 90ms 16000bps 2013-02-01 08:45:56.658105 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-16 124 G.726 16k (AAL2) 8000hz 100ms 16000bps 2013-02-01 08:45:56.658110 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-16 124 G.726 16k (AAL2) 8000hz 110ms 16000bps 2013-02-01 08:45:56.658117 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-16 124 G.726 16k (AAL2) 8000hz 120ms 16000bps 2013-02-01 08:45:56.658138 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-16 124 G.726 16k 8000hz 10ms 16000bps 2013-02-01 08:45:56.658145 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-16 124 G.726 16k 8000hz 20ms 16000bps 2013-02-01 08:45:56.658150 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-16 124 G.726 16k 8000hz 30ms 16000bps 2013-02-01 08:45:56.658155 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-16 124 G.726 16k 8000hz 40ms 16000bps 2013-02-01 08:45:56.658161 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-16 124 G.726 16k 8000hz 50ms 16000bps 2013-02-01 08:45:56.658165 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-16 124 G.726 16k 8000hz 60ms 16000bps 2013-02-01 08:45:56.658170 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-16 124 G.726 16k 8000hz 70ms 16000bps 2013-02-01 08:45:56.658175 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-16 124 G.726 16k 8000hz 80ms 16000bps 2013-02-01 08:45:56.658181 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-16 124 G.726 16k 8000hz 90ms 16000bps 2013-02-01 08:45:56.658185 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-16 124 G.726 16k 8000hz 100ms 16000bps 2013-02-01 08:45:56.658191 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-16 124 G.726 16k 8000hz 110ms 16000bps 2013-02-01 08:45:56.658195 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-16 124 G.726 16k 8000hz 120ms 16000bps 2013-02-01 08:45:56.658212 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-24 123 G.726 24k (AAL2) 8000hz 10ms 24000bps 2013-02-01 08:45:56.658220 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-24 123 G.726 24k (AAL2) 8000hz 20ms 24000bps 2013-02-01 08:45:56.658225 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-24 123 G.726 24k (AAL2) 8000hz 30ms 24000bps 2013-02-01 08:45:56.658229 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-24 123 G.726 24k (AAL2) 8000hz 40ms 24000bps 2013-02-01 08:45:56.658232 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-24 123 G.726 24k (AAL2) 8000hz 50ms 24000bps 2013-02-01 08:45:56.658236 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-24 123 G.726 24k (AAL2) 8000hz 60ms 24000bps 2013-02-01 08:45:56.658239 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-24 123 G.726 24k (AAL2) 8000hz 70ms 24000bps 2013-02-01 08:45:56.658244 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-24 123 G.726 24k (AAL2) 8000hz 80ms 24000bps 2013-02-01 08:45:56.658249 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-24 123 G.726 24k (AAL2) 8000hz 90ms 24000bps 2013-02-01 08:45:56.658253 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-24 123 G.726 24k (AAL2) 8000hz 100ms 24000bps 2013-02-01 08:45:56.658257 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-24 123 G.726 24k (AAL2) 8000hz 110ms 24000bps 2013-02-01 08:45:56.658263 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-24 123 G.726 24k (AAL2) 8000hz 120ms 24000bps 2013-02-01 08:45:56.658279 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-24 123 G.726 24k 8000hz 10ms 24000bps 2013-02-01 08:45:56.658285 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-24 123 G.726 24k 8000hz 20ms 24000bps 2013-02-01 08:45:56.658289 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-24 123 G.726 24k 8000hz 30ms 24000bps 2013-02-01 08:45:56.658296 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-24 123 G.726 24k 8000hz 40ms 24000bps 2013-02-01 08:45:56.658301 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-24 123 G.726 24k 8000hz 50ms 24000bps 2013-02-01 08:45:56.658305 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-24 123 G.726 24k 8000hz 60ms 24000bps 2013-02-01 08:45:56.658309 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-24 123 G.726 24k 8000hz 70ms 24000bps 2013-02-01 08:45:56.658314 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-24 123 G.726 24k 8000hz 80ms 24000bps 2013-02-01 08:45:56.658319 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-24 123 G.726 24k 8000hz 90ms 24000bps 2013-02-01 08:45:56.658323 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-24 123 G.726 24k 8000hz 100ms 24000bps 2013-02-01 08:45:56.658328 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-24 123 G.726 24k 8000hz 110ms 24000bps 2013-02-01 08:45:56.658332 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-24 123 G.726 24k 8000hz 120ms 24000bps 2013-02-01 08:45:56.658351 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-32 122 G.726 32k (AAL2) 8000hz 10ms 32000bps 2013-02-01 08:45:56.658356 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-32 122 G.726 32k (AAL2) 8000hz 20ms 32000bps 2013-02-01 08:45:56.658362 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-32 122 G.726 32k (AAL2) 8000hz 30ms 32000bps 2013-02-01 08:45:56.658372 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-32 122 G.726 32k (AAL2) 8000hz 40ms 32000bps 2013-02-01 08:45:56.658376 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-32 122 G.726 32k (AAL2) 8000hz 50ms 32000bps 2013-02-01 08:45:56.658380 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-32 122 G.726 32k (AAL2) 8000hz 60ms 32000bps 2013-02-01 08:45:56.658384 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-32 122 G.726 32k (AAL2) 8000hz 70ms 32000bps 2013-02-01 08:45:56.658388 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-32 122 G.726 32k (AAL2) 8000hz 80ms 32000bps 2013-02-01 08:45:56.658394 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-32 122 G.726 32k (AAL2) 8000hz 90ms 32000bps 2013-02-01 08:45:56.658399 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-32 122 G.726 32k (AAL2) 8000hz 100ms 32000bps 2013-02-01 08:45:56.658404 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-32 122 G.726 32k (AAL2) 8000hz 110ms 32000bps 2013-02-01 08:45:56.658408 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-32 122 G.726 32k (AAL2) 8000hz 120ms 32000bps 2013-02-01 08:45:56.658423 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-32 122 G.726 32k 8000hz 10ms 32000bps 2013-02-01 08:45:56.658430 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-32 122 G.726 32k 8000hz 20ms 32000bps 2013-02-01 08:45:56.658435 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-32 122 G.726 32k 8000hz 30ms 32000bps 2013-02-01 08:45:56.658439 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-32 122 G.726 32k 8000hz 40ms 32000bps 2013-02-01 08:45:56.658443 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-32 122 G.726 32k 8000hz 50ms 32000bps 2013-02-01 08:45:56.658446 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-32 122 G.726 32k 8000hz 60ms 32000bps 2013-02-01 08:45:56.658450 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-32 122 G.726 32k 8000hz 70ms 32000bps 2013-02-01 08:45:56.658453 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-32 122 G.726 32k 8000hz 80ms 32000bps 2013-02-01 08:45:56.658457 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-32 122 G.726 32k 8000hz 90ms 32000bps 2013-02-01 08:45:56.658461 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-32 122 G.726 32k 8000hz 100ms 32000bps 2013-02-01 08:45:56.658466 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-32 122 G.726 32k 8000hz 110ms 32000bps 2013-02-01 08:45:56.658470 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-32 122 G.726 32k 8000hz 120ms 32000bps 2013-02-01 08:45:56.658487 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-40 121 G.726 40k (AAL2) 8000hz 10ms 40000bps 2013-02-01 08:45:56.658493 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-40 121 G.726 40k (AAL2) 8000hz 20ms 40000bps 2013-02-01 08:45:56.658497 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-40 121 G.726 40k (AAL2) 8000hz 30ms 40000bps 2013-02-01 08:45:56.658501 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-40 121 G.726 40k (AAL2) 8000hz 40ms 40000bps 2013-02-01 08:45:56.658506 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-40 121 G.726 40k (AAL2) 8000hz 50ms 40000bps 2013-02-01 08:45:56.658511 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-40 121 G.726 40k (AAL2) 8000hz 60ms 40000bps 2013-02-01 08:45:56.658516 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-40 121 G.726 40k (AAL2) 8000hz 70ms 40000bps 2013-02-01 08:45:56.658521 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-40 121 G.726 40k (AAL2) 8000hz 80ms 40000bps 2013-02-01 08:45:56.658525 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-40 121 G.726 40k (AAL2) 8000hz 90ms 40000bps 2013-02-01 08:45:56.658530 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-40 121 G.726 40k (AAL2) 8000hz 100ms 40000bps 2013-02-01 08:45:56.658536 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-40 121 G.726 40k (AAL2) 8000hz 110ms 40000bps 2013-02-01 08:45:56.658541 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-40 121 G.726 40k (AAL2) 8000hz 120ms 40000bps 2013-02-01 08:45:56.658560 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-40 121 G.726 40k 8000hz 10ms 40000bps 2013-02-01 08:45:56.658569 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-40 121 G.726 40k 8000hz 20ms 40000bps 2013-02-01 08:45:56.658573 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-40 121 G.726 40k 8000hz 30ms 40000bps 2013-02-01 08:45:56.658578 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-40 121 G.726 40k 8000hz 40ms 40000bps 2013-02-01 08:45:56.658582 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-40 121 G.726 40k 8000hz 50ms 40000bps 2013-02-01 08:45:56.658587 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-40 121 G.726 40k 8000hz 60ms 40000bps 2013-02-01 08:45:56.658590 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-40 121 G.726 40k 8000hz 70ms 40000bps 2013-02-01 08:45:56.658594 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-40 121 G.726 40k 8000hz 80ms 40000bps 2013-02-01 08:45:56.658599 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-40 121 G.726 40k 8000hz 90ms 40000bps 2013-02-01 08:45:56.658604 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-40 121 G.726 40k 8000hz 100ms 40000bps 2013-02-01 08:45:56.658609 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-40 121 G.726 40k 8000hz 110ms 40000bps 2013-02-01 08:45:56.658617 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-40 121 G.726 40k 8000hz 120ms 40000bps 2013-02-01 08:45:56.658634 [NOTICE] switch_loadable_module.c:187 Adding Codec G722 9 G.722 16000hz 10ms 64000bps 2013-02-01 08:45:56.658641 [NOTICE] switch_loadable_module.c:187 Adding Codec G722 9 G.722 16000hz 20ms 64000bps 2013-02-01 08:45:56.658645 [NOTICE] switch_loadable_module.c:187 Adding Codec G722 9 G.722 16000hz 30ms 64000bps 2013-02-01 08:45:56.658648 [NOTICE] switch_loadable_module.c:187 Adding Codec G722 9 G.722 16000hz 40ms 64000bps 2013-02-01 08:45:56.658652 [NOTICE] switch_loadable_module.c:187 Adding Codec G722 9 G.722 16000hz 50ms 64000bps 2013-02-01 08:45:56.658655 [NOTICE] switch_loadable_module.c:187 Adding Codec G722 9 G.722 16000hz 60ms 64000bps 2013-02-01 08:45:56.658672 [NOTICE] switch_loadable_module.c:187 Adding Codec GSM 3 GSM 8000hz 20ms 13200bps 2013-02-01 08:45:56.658677 [NOTICE] switch_loadable_module.c:187 Adding Codec GSM 3 GSM 8000hz 40ms 13200bps 2013-02-01 08:45:56.658682 [NOTICE] switch_loadable_module.c:187 Adding Codec GSM 3 GSM 8000hz 60ms 13200bps 2013-02-01 08:45:56.658686 [NOTICE] switch_loadable_module.c:187 Adding Codec GSM 3 GSM 8000hz 80ms 13200bps 2013-02-01 08:45:56.658691 [NOTICE] switch_loadable_module.c:187 Adding Codec GSM 3 GSM 8000hz 100ms 13200bps 2013-02-01 08:45:56.658696 [NOTICE] switch_loadable_module.c:187 Adding Codec GSM 3 GSM 8000hz 120ms 13200bps 2013-02-01 08:45:56.658723 [NOTICE] switch_loadable_module.c:187 Adding Codec LPC 7 LPC-10 8000hz 90ms 2400bps 2013-02-01 08:45:56.658749 [NOTICE] switch_loadable_module.c:254 Adding Application 't38_gateway' 2013-02-01 08:45:56.658774 [NOTICE] switch_loadable_module.c:254 Adding Application 'rxfax' 2013-02-01 08:45:56.658791 [NOTICE] switch_loadable_module.c:254 Adding Application 'txfax' 2013-02-01 08:45:56.658814 [NOTICE] switch_loadable_module.c:254 Adding Application 'spandsp_stop_dtmf' 2013-02-01 08:45:56.658837 [NOTICE] switch_loadable_module.c:254 Adding Application 'spandsp_start_dtmf' 2013-02-01 08:45:56.658865 [NOTICE] switch_loadable_module.c:254 Adding Application 'spandsp_stop_inject_tdd' 2013-02-01 08:45:56.658896 [NOTICE] switch_loadable_module.c:254 Adding Application 'spandsp_inject_tdd' 2013-02-01 08:45:56.658922 [NOTICE] switch_loadable_module.c:254 Adding Application 'spandsp_stop_detect_tdd' 2013-02-01 08:45:56.658944 [NOTICE] switch_loadable_module.c:254 Adding Application 'spandsp_detect_tdd' 2013-02-01 08:45:56.658967 [NOTICE] switch_loadable_module.c:254 Adding Application 'spandsp_send_tdd' 2013-02-01 08:45:56.658989 [NOTICE] switch_loadable_module.c:254 Adding Application 'spandsp_start_fax_detect' 2013-02-01 08:45:56.659013 [NOTICE] switch_loadable_module.c:254 Adding Application 'spandsp_stop_fax_detect' 2013-02-01 08:45:56.659031 [NOTICE] switch_loadable_module.c:254 Adding Application 'start_tone_detect' 2013-02-01 08:45:56.659057 [NOTICE] switch_loadable_module.c:254 Adding Application 'stop_tone_detect' 2013-02-01 08:45:56.659080 [NOTICE] switch_loadable_module.c:298 Adding API Function 'start_tone_detect' 2013-02-01 08:45:56.659104 [NOTICE] switch_loadable_module.c:298 Adding API Function 'stop_tone_detect' 2013-02-01 08:45:56.659130 [NOTICE] switch_loadable_module.c:298 Adding API Function 'start_tdd_detect' 2013-02-01 08:45:56.659154 [NOTICE] switch_loadable_module.c:298 Adding API Function 'stop_tdd_detect' 2013-02-01 08:45:56.659179 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_send_tdd' 2013-02-01 08:45:56.659415 [NOTICE] switch_loadable_module.c:187 Adding Codec G723 4 G.723.1 6.3k 8000hz 120ms 6300bps 2013-02-01 08:45:56.659421 [NOTICE] switch_loadable_module.c:187 Adding Codec G723 4 G.723.1 6.3k 8000hz 90ms 6300bps 2013-02-01 08:45:56.659425 [NOTICE] switch_loadable_module.c:187 Adding Codec G723 4 G.723.1 6.3k 8000hz 60ms 6300bps 2013-02-01 08:45:56.659429 [NOTICE] switch_loadable_module.c:187 Adding Codec G723 4 G.723.1 6.3k 8000hz 30ms 6300bps 2013-02-01 08:45:56.659634 [NOTICE] switch_loadable_module.c:187 Adding Codec G729 18 G.729 8000hz 10ms 8000bps 2013-02-01 08:45:56.659640 [NOTICE] switch_loadable_module.c:187 Adding Codec G729 18 G.729 8000hz 20ms 8000bps 2013-02-01 08:45:56.659644 [NOTICE] switch_loadable_module.c:187 Adding Codec G729 18 G.729 8000hz 30ms 8000bps 2013-02-01 08:45:56.659648 [NOTICE] switch_loadable_module.c:187 Adding Codec G729 18 G.729 8000hz 40ms 8000bps 2013-02-01 08:45:56.659652 [NOTICE] switch_loadable_module.c:187 Adding Codec G729 18 G.729 8000hz 50ms 8000bps 2013-02-01 08:45:56.659656 [NOTICE] switch_loadable_module.c:187 Adding Codec G729 18 G.729 8000hz 60ms 8000bps 2013-02-01 08:45:56.659659 [NOTICE] switch_loadable_module.c:187 Adding Codec G729 18 G.729 8000hz 70ms 8000bps 2013-02-01 08:45:56.659665 [NOTICE] switch_loadable_module.c:187 Adding Codec G729 18 G.729 8000hz 80ms 8000bps 2013-02-01 08:45:56.659669 [NOTICE] switch_loadable_module.c:187 Adding Codec G729 18 G.729 8000hz 90ms 8000bps 2013-02-01 08:45:56.659673 [NOTICE] switch_loadable_module.c:187 Adding Codec G729 18 G.729 8000hz 100ms 8000bps 2013-02-01 08:45:56.659676 [NOTICE] switch_loadable_module.c:187 Adding Codec G729 18 G.729 8000hz 110ms 8000bps 2013-02-01 08:45:56.659681 [NOTICE] switch_loadable_module.c:187 Adding Codec G729 18 G.729 8000hz 120ms 8000bps 2013-02-01 08:45:56.659882 [NOTICE] switch_loadable_module.c:187 Adding Codec AMR 96 AMR 8000hz 20ms 12200bps 2013-02-01 08:45:56.660178 [NOTICE] switch_loadable_module.c:187 Adding Codec SPEEX 99 Speex 32000hz 20ms 44000bps 2013-02-01 08:45:56.660184 [NOTICE] switch_loadable_module.c:187 Adding Codec SPEEX 99 Speex 16000hz 20ms 42200bps 2013-02-01 08:45:56.660188 [NOTICE] switch_loadable_module.c:187 Adding Codec SPEEX 99 Speex 8000hz 20ms 24600bps 2013-02-01 08:45:56.660405 [NOTICE] switch_loadable_module.c:187 Adding Codec H264 97 H.264 Video (passthru) 90000hz 0ms 0bps 2013-02-01 08:45:56.660429 [NOTICE] switch_loadable_module.c:187 Adding Codec H263 34 H.263 Video (passthru) 90000hz 0ms 0bps 2013-02-01 08:45:56.660450 [NOTICE] switch_loadable_module.c:187 Adding Codec H263-1998 115 H.263+ Video (passthru) 90000hz 0ms 0bps 2013-02-01 08:45:56.660490 [NOTICE] switch_loadable_module.c:187 Adding Codec H263-2000 121 H.263++ Video (passthru) 90000hz 0ms 0bps 2013-02-01 08:45:56.660516 [NOTICE] switch_loadable_module.c:187 Adding Codec H261 31 H.261 Video (passthru) 90000hz 0ms 0bps 2013-02-01 08:45:56.660719 [NOTICE] switch_loadable_module.c:187 Adding Codec VP8 99 VP8 Video (passthru) 90000hz 0ms 0bps 2013-02-01 08:45:56.661130 [INFO] mod_sndfile.c:363 LibSndFile Version : libsndfile-1.0.19 Supported Formats ================================================================================ AIFF (Apple/SGI) (extension "aiff") AU (Sun/NeXT) (extension "au") AVR (Audio Visual Research) (extension "avr") CAF (Apple Core Audio File) (extension "caf") HTK (HMM Tool Kit) (extension "htk") IFF (Amiga IFF/SVX8/SV16) (extension "iff") MAT4 (GNU Octave 2.0 / Matlab 4.2) (extension "mat") MAT5 (GNU Octave 2.1 / Matlab 5.0) (extension "mat") PAF (Ensoniq PARIS) (extension "paf") PVF (Portable Voice Format) (extension "pvf") RAW (header-less) (extension "raw") SD2 (Sound Designer II) (extension "sd2") SDS (Midi Sample Dump Standard) (extension "sds") SF (Berkeley/IRCAM/CARL) (extension "sf") VOC (Creative Labs) (extension "voc") W64 (SoundFoundry WAVE 64) (extension "w64") WAV (Microsoft) (extension "wav") WAV (NIST Sphere) (extension "wav") WAVEX (Microsoft) (extension "wav") WVE (Psion Series 3) (extension "wve") XI (FastTracker 2) (extension "xi") ================================================================================ 2013-02-01 08:45:56.661233 [NOTICE] switch_loadable_module.c:322 Adding File Format 'aiff' 2013-02-01 08:45:56.661261 [NOTICE] switch_loadable_module.c:322 Adding File Format 'au' 2013-02-01 08:45:56.661285 [NOTICE] switch_loadable_module.c:322 Adding File Format 'avr' 2013-02-01 08:45:56.661312 [NOTICE] switch_loadable_module.c:322 Adding File Format 'caf' 2013-02-01 08:45:56.661337 [NOTICE] switch_loadable_module.c:322 Adding File Format 'htk' 2013-02-01 08:45:56.661360 [NOTICE] switch_loadable_module.c:322 Adding File Format 'iff' 2013-02-01 08:45:56.661383 [NOTICE] switch_loadable_module.c:322 Adding File Format 'mat' 2013-02-01 08:45:56.661406 [NOTICE] switch_loadable_module.c:322 Adding File Format 'paf' 2013-02-01 08:45:56.661429 [NOTICE] switch_loadable_module.c:322 Adding File Format 'pvf' 2013-02-01 08:45:56.661450 [NOTICE] switch_loadable_module.c:322 Adding File Format 'raw' 2013-02-01 08:45:56.661472 [NOTICE] switch_loadable_module.c:322 Adding File Format 'sd2' 2013-02-01 08:45:56.661495 [NOTICE] switch_loadable_module.c:322 Adding File Format 'sds' 2013-02-01 08:45:56.661518 [NOTICE] switch_loadable_module.c:322 Adding File Format 'sf' 2013-02-01 08:45:56.661539 [NOTICE] switch_loadable_module.c:322 Adding File Format 'voc' 2013-02-01 08:45:56.661564 [NOTICE] switch_loadable_module.c:322 Adding File Format 'w64' 2013-02-01 08:45:56.661585 [NOTICE] switch_loadable_module.c:322 Adding File Format 'wav' 2013-02-01 08:45:56.661606 [NOTICE] switch_loadable_module.c:322 Adding File Format 'wve' 2013-02-01 08:45:56.661626 [NOTICE] switch_loadable_module.c:322 Adding File Format 'xi' 2013-02-01 08:45:56.661649 [NOTICE] switch_loadable_module.c:322 Adding File Format 'r8' 2013-02-01 08:45:56.661671 [NOTICE] switch_loadable_module.c:322 Adding File Format 'r16' 2013-02-01 08:45:56.661695 [NOTICE] switch_loadable_module.c:322 Adding File Format 'r24' 2013-02-01 08:45:56.661717 [NOTICE] switch_loadable_module.c:322 Adding File Format 'r32' 2013-02-01 08:45:56.661740 [NOTICE] switch_loadable_module.c:322 Adding File Format 'gsm' 2013-02-01 08:45:56.661767 [NOTICE] switch_loadable_module.c:322 Adding File Format 'ul' 2013-02-01 08:45:56.661790 [NOTICE] switch_loadable_module.c:322 Adding File Format 'ulaw' 2013-02-01 08:45:56.661812 [NOTICE] switch_loadable_module.c:322 Adding File Format 'al' 2013-02-01 08:45:56.661835 [NOTICE] switch_loadable_module.c:322 Adding File Format 'alaw' 2013-02-01 08:45:56.661858 [NOTICE] switch_loadable_module.c:322 Adding File Format 'adpcm' 2013-02-01 08:45:56.662100 [NOTICE] switch_loadable_module.c:322 Adding File Format 'H263' 2013-02-01 08:45:56.662125 [NOTICE] switch_loadable_module.c:322 Adding File Format 'AMR' 2013-02-01 08:45:56.662153 [NOTICE] switch_loadable_module.c:322 Adding File Format 'SPEEX' 2013-02-01 08:45:56.662176 [NOTICE] switch_loadable_module.c:322 Adding File Format 'G729' 2013-02-01 08:45:56.662198 [NOTICE] switch_loadable_module.c:322 Adding File Format 'G726-16' 2013-02-01 08:45:56.662221 [NOTICE] switch_loadable_module.c:322 Adding File Format 'AAL2-G726-16' 2013-02-01 08:45:56.662244 [NOTICE] switch_loadable_module.c:322 Adding File Format 'PCMA' 2013-02-01 08:45:56.662268 [NOTICE] switch_loadable_module.c:322 Adding File Format 'DVI4' 2013-02-01 08:45:56.662290 [NOTICE] switch_loadable_module.c:322 Adding File Format 'G726-24' 2013-02-01 08:45:56.662313 [NOTICE] switch_loadable_module.c:322 Adding File Format 'PCMU' 2013-02-01 08:45:56.662336 [NOTICE] switch_loadable_module.c:322 Adding File Format 'L16' 2013-02-01 08:45:56.662359 [NOTICE] switch_loadable_module.c:322 Adding File Format 'PROXY' 2013-02-01 08:45:56.662382 [NOTICE] switch_loadable_module.c:322 Adding File Format 'AAL2-G726-24' 2013-02-01 08:45:56.662405 [NOTICE] switch_loadable_module.c:322 Adding File Format 'AAL2-G726-32' 2013-02-01 08:45:56.662428 [NOTICE] switch_loadable_module.c:322 Adding File Format 'G726-32' 2013-02-01 08:45:56.662451 [NOTICE] switch_loadable_module.c:322 Adding File Format 'G722' 2013-02-01 08:45:56.662475 [NOTICE] switch_loadable_module.c:322 Adding File Format 'AAL2-G726-40' 2013-02-01 08:45:56.662498 [NOTICE] switch_loadable_module.c:322 Adding File Format 'G726-40' 2013-02-01 08:45:56.662521 [NOTICE] switch_loadable_module.c:322 Adding File Format 'GSM' 2013-02-01 08:45:56.662543 [NOTICE] switch_loadable_module.c:322 Adding File Format 'H263-1998' 2013-02-01 08:45:56.662567 [NOTICE] switch_loadable_module.c:322 Adding File Format 'H263-2000' 2013-02-01 08:45:56.662605 [NOTICE] switch_loadable_module.c:322 Adding File Format 'H264' 2013-02-01 08:45:56.662629 [NOTICE] switch_loadable_module.c:322 Adding File Format 'H261' 2013-02-01 08:45:56.662652 [NOTICE] switch_loadable_module.c:322 Adding File Format 'G723' 2013-02-01 08:45:56.662682 [NOTICE] switch_loadable_module.c:322 Adding File Format 'LPC' 2013-02-01 08:45:56.662715 [NOTICE] switch_loadable_module.c:322 Adding File Format 'PROXY-VID' 2013-02-01 08:45:56.662738 [NOTICE] switch_loadable_module.c:322 Adding File Format 'VP8' 2013-02-01 08:45:56.663149 [NOTICE] switch_loadable_module.c:298 Adding API Function 'stop_local_stream' 2013-02-01 08:45:56.663178 [NOTICE] switch_loadable_module.c:298 Adding API Function 'start_local_stream' 2013-02-01 08:45:56.663205 [NOTICE] switch_loadable_module.c:298 Adding API Function 'show_local_stream' 2013-02-01 08:45:56.663329 [NOTICE] switch_loadable_module.c:322 Adding File Format 'local_stream' 2013-02-01 08:45:56.663903 [NOTICE] switch_loadable_module.c:322 Adding File Format 'tone_stream' 2013-02-01 08:45:56.663941 [NOTICE] switch_loadable_module.c:322 Adding File Format 'silence_stream' 2013-02-01 08:45:56.667854 [DEBUG] switch_loadable_module.c:1281 Loading module with global namespace at request of module 2013-02-01 08:45:56.668108 [DEBUG] mod_spidermonkey.c:980 Successfully Loaded [/usr/local/freeswitch/mod/mod_spidermonkey_teletone.so] 2013-02-01 08:45:56.668321 [DEBUG] mod_spidermonkey.c:980 Successfully Loaded [/usr/local/freeswitch/mod/mod_spidermonkey_core_db.so] 2013-02-01 08:45:56.668516 [DEBUG] mod_spidermonkey.c:980 Successfully Loaded [/usr/local/freeswitch/mod/mod_spidermonkey_socket.so] 2013-02-01 08:45:56.668541 [NOTICE] switch_loadable_module.c:254 Adding Application 'javascript' 2013-02-01 08:45:56.668570 [NOTICE] switch_loadable_module.c:276 Adding Chat Application 'javascript' 2013-02-01 08:45:56.668593 [NOTICE] switch_loadable_module.c:298 Adding API Function 'jsrun' 2013-02-01 08:45:56.668614 [NOTICE] switch_loadable_module.c:298 Adding API Function 'jsapi' 2013-02-01 08:45:56.669543 [DEBUG] switch_loadable_module.c:1281 Loading module with global namespace at request of module 2013-02-01 08:45:56.669583 [NOTICE] switch_loadable_module.c:214 Adding Dialplan 'LUA' 2013-02-01 08:45:56.669607 [NOTICE] switch_loadable_module.c:254 Adding Application 'lua' 2013-02-01 08:45:56.669629 [NOTICE] switch_loadable_module.c:276 Adding Chat Application 'lua' 2013-02-01 08:45:56.669651 [NOTICE] switch_loadable_module.c:298 Adding API Function 'luarun' 2013-02-01 08:45:56.669681 [NOTICE] switch_loadable_module.c:298 Adding API Function 'lua' 2013-02-01 08:45:56.674615 [NOTICE] switch_loadable_module.c:343 Adding Speech interface 'flite' 2013-02-01 08:45:56.674877 [NOTICE] switch_loadable_module.c:423 Adding Say interface 'en' 2013-02-01 08:45:56.675106 [NOTICE] switch_loadable_module.c:423 Adding Say interface 'fr' 2013-02-01 08:45:56.675338 [INFO] switch_time.c:476 Clock synchronized to system time. 2013-02-01 08:45:56.675497 [DEBUG] mod_event_socket.c:2805 Socket up listening on 127.0.0.1:8021 2013-02-01 08:45:56.675522 [DEBUG] switch_loadable_module.c:640 Chat Thread Started 2013-02-01 08:45:56.675528 [DEBUG] switch_loadable_module.c:640 Chat Thread Started 2013-02-01 08:45:56.675616 [NOTICE] switch_core.c:1211 Created ip list rfc1918.auto default (deny) 2013-02-01 08:45:56.675648 [NOTICE] switch_utils.c:295 Adding 10.0.0.0/8 (allow) [] to list rfc1918.auto 2013-02-01 08:45:56.675656 [NOTICE] switch_utils.c:295 Adding 172.16.0.0/12 (allow) [] to list rfc1918.auto 2013-02-01 08:45:56.675662 [NOTICE] switch_utils.c:295 Adding 192.168.0.0/16 (allow) [] to list rfc1918.auto 2013-02-01 08:45:56.675666 [NOTICE] switch_core.c:1219 Created ip list wan.auto default (allow) 2013-02-01 08:45:56.675671 [NOTICE] switch_utils.c:295 Adding 10.0.0.0/8 (deny) [] to list wan.auto 2013-02-01 08:45:56.675675 [NOTICE] switch_utils.c:295 Adding 172.16.0.0/12 (deny) [] to list wan.auto 2013-02-01 08:45:56.675679 [NOTICE] switch_utils.c:295 Adding 192.168.0.0/16 (deny) [] to list wan.auto 2013-02-01 08:45:56.675683 [NOTICE] switch_core.c:1227 Created ip list nat.auto default (deny) 2013-02-01 08:45:56.675687 [NOTICE] switch_core.c:1229 Adding XXX.XXX.XXX.10/255.255.255.240 (deny) to list nat.auto 2013-02-01 08:45:56.675691 [NOTICE] switch_utils.c:295 Adding 10.0.0.0/8 (allow) [] to list nat.auto 2013-02-01 08:45:56.675698 [NOTICE] switch_utils.c:295 Adding 172.16.0.0/12 (allow) [] to list nat.auto 2013-02-01 08:45:56.675702 [NOTICE] switch_utils.c:295 Adding 192.168.0.0/16 (allow) [] to list nat.auto 2013-02-01 08:45:56.675706 [NOTICE] switch_core.c:1238 Created ip list loopback.auto default (deny) 2013-02-01 08:45:56.675710 [NOTICE] switch_utils.c:295 Adding 127.0.0.0/8 (allow) [] to list loopback.auto 2013-02-01 08:45:56.675713 [NOTICE] switch_core.c:1244 Created ip list localnet.auto default (deny) 2013-02-01 08:45:56.675717 [NOTICE] switch_core.c:1247 Adding XXX.XXX.XXX.10/255.255.255.240 (allow) to list localnet.auto 2013-02-01 08:45:56.675731 [NOTICE] switch_utils.c:295 Adding 192.168.42.0/24 (deny) [] to list lan 2013-02-01 08:45:56.675735 [NOTICE] switch_core.c:1343 Adding 192.168.42.0/24 (deny) to list lan 2013-02-01 08:45:56.675740 [NOTICE] switch_utils.c:295 Adding 192.168.42.42/32 (allow) [] to list lan 2013-02-01 08:45:56.675744 [NOTICE] switch_core.c:1343 Adding 192.168.42.42/32 (allow) to list lan 2013-02-01 08:45:56.675792 [NOTICE] switch_utils.c:295 Adding 216.113.106.182/32 (allow) [0000000000 at XXX.XXX.XXX.10] to list domains 2013-02-01 08:45:56.675809 [NOTICE] switch_utils.c:295 Adding YYY.YYY.YYY.YYY/32 (allow) [0000000000 at XXX.XXX.XXX.10] to list domains 2013-02-01 08:45:56.675812 [NOTICE] switch_utils.c:295 Adding 10.5.3.154/32 (allow) [4388999130 at XXX.XXX.XXX.10] to list domains 2013-02-01 08:45:56.675816 [NOTICE] switch_utils.c:295 Adding 192.0.2.0/24 (allow) [brian at XXX.XXX.XXX.10] to list domains 2013-02-01 08:45:56.675820 [NOTICE] switch_utils.c:295 Adding 159.18.161.101/32 (allow) [] to list domains 2013-02-01 08:45:56.675822 [NOTICE] switch_core.c:1343 Adding 159.18.161.101/32 (allow) to list domains 2013-02-01 08:46:25.752905 [WARNING] sofia.c:4719 Ping succeeded thinktel with code 200 - count 1/1/1, state UP recv 476 bytes from udp/[YYY.YYY.YYY.YYY]:5060 at 13:52:12.050223: ------------------------------------------------------------------------ NOTIFY sip:XXX.XXX.XXX.10 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.165:5060;branch=z9hG4bK-1f09d4ac From: "Test" ;tag=bdccc36ed8a29fdeo0 To: Call-ID: 36b982e-fa23d21e at 192.168.3.165 CSeq: 16010 NOTIFY Max-Forwards: 70 Contact: "Test" Event: keep-alive User-Agent: Cisco/SPA112-1.2.1(004) Allow-Events: talk, hold, conference Content-Length: 0 ------------------------------------------------------------------------ send 722 bytes to udp/[YYY.YYY.YYY.YYY]:5060 at 13:52:12.050490: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.165:5060;branch=z9hG4bK-1f09d4ac;received=YYY.YYY.YYY.YYY From: "Test" ;tag=bdccc36ed8a29fdeo0 To: ;tag=He02D01NactjF Call-ID: 36b982e-fa23d21e at 192.168.3.165 CSeq: 16010 NOTIFY Contact: User-Agent: FreeSWITCH-mod_sofia/1.3.13b+git~20130115T041739Z~8b195824db Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 ------------------------------------------------------------------------ send 563 bytes to udp/[206.80.250.100]:5060 at 13:52:25.636789: ------------------------------------------------------------------------ OPTIONS sip:tor.trk.tprm.ca;transport=udp SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.10:5080;rport;branch=z9hG4bK3t15jH7ea3vFp Max-Forwards: 70 From: ;tag=363Q8HXF4jD2c To: Call-ID: 738698b1-e719-1230-dd9e-3085a9a5c78e CSeq: 39527140 OPTIONS User-Agent: FreeSWITCH-mod_sofia/1.3.13b+git~20130115T041739Z~8b195824db Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Content-Length: 0 ------------------------------------------------------------------------ recv 936 bytes from udp/[206.80.250.100]:5060 at 13:52:25.690062: ------------------------------------------------------------------------ SIP/2.0 200 OK Call-ID: 738698b1-e719-1230-dd9e-3085a9a5c78e CSeq: 39527140 OPTIONS From: ;tag=363Q8HXF4jD2c To: ;tag=159.18.161.101+1+49090e+869d06b0 Via: SIP/2.0/UDP XXX.XXX.XXX.10:5080;received=XXX.XXX.XXX.10;rport=5080;branch=z9hG4bK3t15jH7ea3vFp Server: DC-SIP/2.0 Organization: MetaSwitch Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info, as-feature-event Supported: resource-priority, 100rel Allow: INVITE, ACK, CANCEL, BYE, REGISTER, OPTIONS, PRACK, UPDATE, SUBSCRIBE, NOTIFY, REFER, INFO, PUBLISH Accept-Encoding: identity Accept: application/sdp, application/simple-message-summary, message/sipfrag, application/isup, application/x-simple-call-service-info, multipart/mixed, application/broadsoft, application/vq-rtcpxr, application/media_control+xml, application/dtmf-relay, text/plain, application/x-as-feature-event+xml Content-Length: 0 ------------------------------------------------------------------------ recv 476 bytes from udp/[YYY.YYY.YYY.YYY]:5060 at 13:52:27.097518: ------------------------------------------------------------------------ NOTIFY sip:XXX.XXX.XXX.10 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.165:5060;branch=z9hG4bK-6a9f8a02 From: "Test" ;tag=bdccc36ed8a29fdeo0 To: Call-ID: 36b982e-fa23d21e at 192.168.3.165 CSeq: 16011 NOTIFY Max-Forwards: 70 Contact: "Test" Event: keep-alive User-Agent: Cisco/SPA112-1.2.1(004) Allow-Events: talk, hold, conference Content-Length: 0 ------------------------------------------------------------------------ send 722 bytes to udp/[YYY.YYY.YYY.YYY]:5060 at 13:52:27.097816: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.165:5060;branch=z9hG4bK-6a9f8a02;received=YYY.YYY.YYY.YYY From: "Test" ;tag=bdccc36ed8a29fdeo0 To: ;tag=jQSUFUjS7mg5a Call-ID: 36b982e-fa23d21e at 192.168.3.165 CSeq: 16011 NOTIFY Contact: User-Agent: FreeSWITCH-mod_sofia/1.3.13b+git~20130115T041739Z~8b195824db Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 ------------------------------------------------------------------------ recv 883 bytes from udp/[206.80.250.100]:5060 at 13:52:30.080629: ------------------------------------------------------------------------ OPTIONS sip:metaswitch at XXX.XXX.XXX.10:5060;transport=udp SIP/2.0 Record-Route: Record-Route: Via: SIP/2.0/UDP 206.80.250.100;branch=z9hG4bKa6c3.3ec4b036.0 Via: SIP/2.0/UDP 159.18.161.101:5060;received=159.18.161.101;rport=5060;branch=z9hG4bK-c3bc9cddf29534261eaed3e3de3382f41-159.18.161.101-1 Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info, as-feature-event Max-Forwards: 69 Call-ID: E32D6A63 at 159.18.161.101 From: ;tag=159.18.161.101+1+4c3e1e+fe875dfa CSeq: 134426196 OPTIONS Organization: MetaSwitch Supported: resource-priority, 100rel Content-Length: 0 Contact: To: ------------------------------------------------------------------------ send 1052 bytes to udp/[206.80.250.100]:5060 at 13:52:30.080976: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 206.80.250.100;branch=z9hG4bKa6c3.3ec4b036.0 Via: SIP/2.0/UDP 159.18.161.101:5060;received=159.18.161.101;rport=5060;branch=z9hG4bK-c3bc9cddf29534261eaed3e3de3382f41-159.18.161.101-1 Record-Route: Record-Route: From: ;tag=159.18.161.101+1+4c3e1e+fe875dfa To: ;tag=K0jmHp3v4X6Qp Call-ID: E32D6A63 at 159.18.161.101 CSeq: 134426196 OPTIONS Contact: User-Agent: FreeSWITCH-mod_sofia/1.3.13b+git~20130115T041739Z~8b195824db Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 ------------------------------------------------------------------------ recv 476 bytes from udp/[YYY.YYY.YYY.YYY]:5060 at 13:52:42.160855: ------------------------------------------------------------------------ NOTIFY sip:XXX.XXX.XXX.10 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.165:5060;branch=z9hG4bK-9a226a86 From: "Test" ;tag=bdccc36ed8a29fdeo0 To: Call-ID: 36b982e-fa23d21e at 192.168.3.165 CSeq: 16012 NOTIFY Max-Forwards: 70 Contact: "Test" Event: keep-alive User-Agent: Cisco/SPA112-1.2.1(004) Allow-Events: talk, hold, conference Content-Length: 0 ------------------------------------------------------------------------ send 722 bytes to udp/[YYY.YYY.YYY.YYY]:5060 at 13:52:42.161227: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.165:5060;branch=z9hG4bK-9a226a86;received=YYY.YYY.YYY.YYY From: "Test" ;tag=bdccc36ed8a29fdeo0 To: ;tag=m9BDKHm016vaj Call-ID: 36b982e-fa23d21e at 192.168.3.165 CSeq: 16012 NOTIFY Contact: User-Agent: FreeSWITCH-mod_sofia/1.3.13b+git~20130115T041739Z~8b195824db Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 ------------------------------------------------------------------------ recv 1235 bytes from udp/[206.80.250.100]:5060 at 13:52:53.935287: ------------------------------------------------------------------------ INVITE sip:0000000000 at XXX.XXX.XXX.10:5060;transport=udp SIP/2.0 Record-Route: Record-Route: Via: SIP/2.0/UDP 206.80.250.100;branch=z9hG4bKe9a5.eceec2d1.0 Via: SIP/2.0/UDP 159.18.161.101:5060;received=159.18.161.101;rport=5060;branch=z9hG4bK-52137b09f400f252d8590c120ff385c11-159.18.161.101-1 Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info, as-feature-event Max-Forwards: 69 Call-ID: 90B682B7 at 159.18.161.101 From: "Targo" ;tag=159.18.161.101+1+56940a+8443c85e;isup-oli=00 To: CSeq: 484850007 INVITE Expires: 180 Organization: MetaSwitch Supported: resource-priority, 100rel Content-Length: 200 Content-Type: application/sdp Contact: "Targo" ;isup-oli=00 P-Asserted-Identity: "Targo" v=0 o=- 2517108369 2517108369 IN IP4 159.18.161.105 s=- c=IN IP4 206.80.250.102 t=0 0 m=audio 17352 RTP/AVP 18 0 101 a=rtpmap:101 telephone-event/8000 a=ptime:20 a=sendrecv a=nortpproxy:yes ------------------------------------------------------------------------ send 691 bytes to udp/[206.80.250.100]:5060 at 13:52:53.935625: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 206.80.250.100;branch=z9hG4bKe9a5.eceec2d1.0 Via: SIP/2.0/UDP 159.18.161.101:5060;received=159.18.161.101;rport=5060;branch=z9hG4bK-52137b09f400f252d8590c120ff385c11-159.18.161.101-1 Record-Route: Record-Route: From: "Targo" ;tag=159.18.161.101+1+56940a+8443c85e;isup-oli=00 To: Call-ID: 90B682B7 at 159.18.161.101 CSeq: 484850007 INVITE User-Agent: FreeSWITCH-mod_sofia/1.3.13b+git~20130115T041739Z~8b195824db Content-Length: 0 ------------------------------------------------------------------------ send 1027 bytes to udp/[206.80.250.100]:5060 at 13:52:53.956564: ------------------------------------------------------------------------ SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 206.80.250.100;branch=z9hG4bKe9a5.eceec2d1.0 Via: SIP/2.0/UDP 159.18.161.101:5060;received=159.18.161.101;rport=5060;branch=z9hG4bK-52137b09f400f252d8590c120ff385c11-159.18.161.101-1 From: "Targo" ;tag=159.18.161.101+1+56940a+8443c85e;isup-oli=00 To: ;tag=Nj55mc53yFKXD Call-ID: 90B682B7 at 159.18.161.101 CSeq: 484850007 INVITE User-Agent: FreeSWITCH-mod_sofia/1.3.13b+git~20130115T041739Z~8b195824db Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Proxy-Authenticate: Digest realm="159.18.161.101", nonce="ad098bc4-6c76-11e2-bf36-d31cf8b5111c", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ recv 438 bytes from udp/[206.80.250.100]:5060 at 13:52:53.971465: ------------------------------------------------------------------------ ACK sip:0000000000 at XXX.XXX.XXX.10:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 206.80.250.100;branch=z9hG4bKe9a5.eceec2d1.0 From: "Targo" ;tag=159.18.161.101+1+56940a+8443c85e;isup-oli=00 Call-ID: 90B682B7 at 159.18.161.101 To: ;tag=Nj55mc53yFKXD CSeq: 484850007 ACK Max-Forwards: 70 User-Agent: OpenSIPS (1.6.2-notls (x86_64/linux)) Content-Length: 0 ------------------------------------------------------------------------ send 563 bytes to udp/[206.80.250.100]:5060 at 13:52:55.637986: ------------------------------------------------------------------------ OPTIONS sip:tor.trk.tprm.ca;transport=udp SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.10:5080;rport;branch=z9hG4bK43tymcrj7BK2H Max-Forwards: 70 From: ;tag=4FXgaDeK1U3mr To: Call-ID: 85686a26-e719-1230-dd9e-3085a9a5c78e CSeq: 39527141 OPTIONS User-Agent: FreeSWITCH-mod_sofia/1.3.13b+git~20130115T041739Z~8b195824db Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Content-Length: 0 ------------------------------------------------------------------------ recv 936 bytes from udp/[206.80.250.100]:5060 at 13:52:55.691358: ------------------------------------------------------------------------ SIP/2.0 200 OK Call-ID: 85686a26-e719-1230-dd9e-3085a9a5c78e CSeq: 39527141 OPTIONS From: ;tag=4FXgaDeK1U3mr To: ;tag=159.18.161.101+1+576b3e+e1b2a7e8 Via: SIP/2.0/UDP XXX.XXX.XXX.10:5080;received=XXX.XXX.XXX.10;rport=5080;branch=z9hG4bK43tymcrj7BK2H Server: DC-SIP/2.0 Organization: MetaSwitch Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info, as-feature-event Supported: resource-priority, 100rel Allow: INVITE, ACK, CANCEL, BYE, REGISTER, OPTIONS, PRACK, UPDATE, SUBSCRIBE, NOTIFY, REFER, INFO, PUBLISH Accept-Encoding: identity Accept: application/sdp, application/simple-message-summary, message/sipfrag, application/isup, application/x-simple-call-service-info, multipart/mixed, application/broadsoft, application/vq-rtcpxr, application/media_control+xml, application/dtmf-relay, text/plain, application/x-as-feature-event+xml Content-Length: 0 ------------------------------------------------------------------------ recv 1234 bytes from udp/[206.80.250.100]:5060 at 13:52:55.980321: ------------------------------------------------------------------------ INVITE sip:0000000000 at XXX.XXX.XXX.10:5060;transport=udp SIP/2.0 Record-Route: Record-Route: Via: SIP/2.0/UDP 206.80.250.100;branch=z9hG4bKb4c8.f6180ea6.0 Via: SIP/2.0/UDP 159.18.161.101:5060;received=159.18.161.101;rport=5060;branch=z9hG4bK-e2087221c3a759c02140a28bd255fb331-159.18.161.101-1 Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info, as-feature-event Max-Forwards: 69 Call-ID: 675CF1F7 at 159.18.161.101 From: "Targo" ;tag=159.18.161.101+1+57b430+5c3b5434;isup-oli=00 To: CSeq: 946568372 INVITE Expires: 180 Organization: MetaSwitch Supported: resource-priority, 100rel Content-Length: 199 Content-Type: application/sdp Contact: "Targo" ;isup-oli=00 P-Asserted-Identity: "Targo" v=0 o=- 2517110382 2517110382 IN IP4 206.80.250.10 s=- c=IN IP4 206.80.250.107 t=0 0 m=audio 19618 RTP/AVP 18 0 101 a=rtpmap:101 telephone-event/8000 a=ptime:20 a=sendrecv a=nortpproxy:yes ------------------------------------------------------------------------ send 691 bytes to udp/[206.80.250.100]:5060 at 13:52:55.980626: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 206.80.250.100;branch=z9hG4bKb4c8.f6180ea6.0 Via: SIP/2.0/UDP 159.18.161.101:5060;received=159.18.161.101;rport=5060;branch=z9hG4bK-e2087221c3a759c02140a28bd255fb331-159.18.161.101-1 Record-Route: Record-Route: From: "Targo" ;tag=159.18.161.101+1+57b430+5c3b5434;isup-oli=00 To: Call-ID: 675CF1F7 at 159.18.161.101 CSeq: 946568372 INVITE User-Agent: FreeSWITCH-mod_sofia/1.3.13b+git~20130115T041739Z~8b195824db Content-Length: 0 ------------------------------------------------------------------------ send 1027 bytes to udp/[206.80.250.100]:5060 at 13:52:56.001508: ------------------------------------------------------------------------ SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 206.80.250.100;branch=z9hG4bKb4c8.f6180ea6.0 Via: SIP/2.0/UDP 159.18.161.101:5060;received=159.18.161.101;rport=5060;branch=z9hG4bK-e2087221c3a759c02140a28bd255fb331-159.18.161.101-1 From: "Targo" ;tag=159.18.161.101+1+57b430+5c3b5434;isup-oli=00 To: ;tag=pUyyp7N7Ur9FS Call-ID: 675CF1F7 at 159.18.161.101 CSeq: 946568372 INVITE User-Agent: FreeSWITCH-mod_sofia/1.3.13b+git~20130115T041739Z~8b195824db Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Proxy-Authenticate: Digest realm="159.18.161.101", nonce="ae4194d2-6c76-11e2-bf38-d31cf8b5111c", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ recv 438 bytes from udp/[206.80.250.100]:5060 at 13:52:56.016327: ------------------------------------------------------------------------ ACK sip:0000000000 at XXX.XXX.XXX.10:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 206.80.250.100;branch=z9hG4bKb4c8.f6180ea6.0 From: "Targo" ;tag=159.18.161.101+1+57b430+5c3b5434;isup-oli=00 Call-ID: 675CF1F7 at 159.18.161.101 To: ;tag=pUyyp7N7Ur9FS CSeq: 946568372 ACK Max-Forwards: 70 User-Agent: OpenSIPS (1.6.2-notls (x86_64/linux)) Content-Length: 0 ------------------------------------------------------------------------ recv 476 bytes from udp/[YYY.YYY.YYY.YYY]:5060 at 13:52:57.208168: ------------------------------------------------------------------------ NOTIFY sip:XXX.XXX.XXX.10 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.165:5060;branch=z9hG4bK-ad8e6027 From: "Test" ;tag=bdccc36ed8a29fdeo0 To: Call-ID: 36b982e-fa23d21e at 192.168.3.165 CSeq: 16013 NOTIFY Max-Forwards: 70 Contact: "Test" Event: keep-alive User-Agent: Cisco/SPA112-1.2.1(004) Allow-Events: talk, hold, conference Content-Length: 0 ------------------------------------------------------------------------ send 722 bytes to udp/[YYY.YYY.YYY.YYY]:5060 at 13:52:57.208508: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.165:5060;branch=z9hG4bK-ad8e6027;received=YYY.YYY.YYY.YYY From: "Test" ;tag=bdccc36ed8a29fdeo0 To: ;tag=Q4QQr26aS1Z2m Call-ID: 36b982e-fa23d21e at 192.168.3.165 CSeq: 16013 NOTIFY Contact: User-Agent: FreeSWITCH-mod_sofia/1.3.13b+git~20130115T041739Z~8b195824db Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 ------------------------------------------------------------------------ recv 1234 bytes from udp/[206.80.250.100]:5060 at 13:52:58.924338: ------------------------------------------------------------------------ INVITE sip:0000000000 at XXX.XXX.XXX.10:5060;transport=udp SIP/2.0 Record-Route: Record-Route: Via: SIP/2.0/UDP 206.80.250.100;branch=z9hG4bK8d06.28542a74.0 Via: SIP/2.0/UDP 159.18.161.101:5060;received=159.18.161.101;rport=5060;branch=z9hG4bK-9b2e0b9b9cef6d45dd453963e40ddb821-159.18.161.101-1 Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info, as-feature-event Max-Forwards: 69 Call-ID: 5AB35BF4 at 159.18.161.101 From: "Targo" ;tag=159.18.161.101+1+581813+a10fdfe4;isup-oli=00 To: CSeq: 748762263 INVITE Expires: 180 Organization: MetaSwitch Supported: resource-priority, 100rel Content-Length: 199 Content-Type: application/sdp Contact: "Targo" ;isup-oli=00 P-Asserted-Identity: "Targo" v=0 o=- 2517113327 2517113327 IN IP4 206.80.250.11 s=- c=IN IP4 206.80.250.101 t=0 0 m=audio 16120 RTP/AVP 18 0 101 a=rtpmap:101 telephone-event/8000 a=ptime:20 a=sendrecv a=nortpproxy:yes ------------------------------------------------------------------------ send 691 bytes to udp/[206.80.250.100]:5060 at 13:52:58.924779: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 206.80.250.100;branch=z9hG4bK8d06.28542a74.0 Via: SIP/2.0/UDP 159.18.161.101:5060;received=159.18.161.101;rport=5060;branch=z9hG4bK-9b2e0b9b9cef6d45dd453963e40ddb821-159.18.161.101-1 Record-Route: Record-Route: From: "Targo" ;tag=159.18.161.101+1+581813+a10fdfe4;isup-oli=00 To: Call-ID: 5AB35BF4 at 159.18.161.101 CSeq: 748762263 INVITE User-Agent: FreeSWITCH-mod_sofia/1.3.13b+git~20130115T041739Z~8b195824db Content-Length: 0 ------------------------------------------------------------------------ send 1027 bytes to udp/[206.80.250.100]:5060 at 13:52:58.945828: ------------------------------------------------------------------------ SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 206.80.250.100;branch=z9hG4bK8d06.28542a74.0 Via: SIP/2.0/UDP 159.18.161.101:5060;received=159.18.161.101;rport=5060;branch=z9hG4bK-9b2e0b9b9cef6d45dd453963e40ddb821-159.18.161.101-1 From: "Targo" ;tag=159.18.161.101+1+581813+a10fdfe4;isup-oli=00 To: ;tag=rDHgtXQepapNg Call-ID: 5AB35BF4 at 159.18.161.101 CSeq: 748762263 INVITE User-Agent: FreeSWITCH-mod_sofia/1.3.13b+git~20130115T041739Z~8b195824db Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Proxy-Authenticate: Digest realm="159.18.161.101", nonce="b002d600-6c76-11e2-bf3a-d31cf8b5111c", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ recv 438 bytes from udp/[206.80.250.100]:5060 at 13:52:58.960650: ------------------------------------------------------------------------ ACK sip:0000000000 at XXX.XXX.XXX.10:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 206.80.250.100;branch=z9hG4bK8d06.28542a74.0 From: "Targo" ;tag=159.18.161.101+1+581813+a10fdfe4;isup-oli=00 Call-ID: 5AB35BF4 at 159.18.161.101 To: ;tag=rDHgtXQepapNg CSeq: 748762263 ACK Max-Forwards: 70 User-Agent: OpenSIPS (1.6.2-notls (x86_64/linux)) Content-Length: 0 ------------------------------------------------------------------------ recv 476 bytes from udp/[YYY.YYY.YYY.YYY]:5060 at 13:53:12.254036: ------------------------------------------------------------------------ NOTIFY sip:XXX.XXX.XXX.10 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.165:5060;branch=z9hG4bK-fcde8e66 From: "Test" ;tag=bdccc36ed8a29fdeo0 To: Call-ID: 36b982e-fa23d21e at 192.168.3.165 CSeq: 16014 NOTIFY Max-Forwards: 70 Contact: "Test" Event: keep-alive User-Agent: Cisco/SPA112-1.2.1(004) Allow-Events: talk, hold, conference Content-Length: 0 ------------------------------------------------------------------------ send 722 bytes to udp/[YYY.YYY.YYY.YYY]:5060 at 13:53:12.254431: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.165:5060;branch=z9hG4bK-fcde8e66;received=YYY.YYY.YYY.YYY From: "Test" ;tag=bdccc36ed8a29fdeo0 To: ;tag=Spa9Ur8HKKc8B Call-ID: 36b982e-fa23d21e at 192.168.3.165 CSeq: 16014 NOTIFY Contact: User-Agent: FreeSWITCH-mod_sofia/1.3.13b+git~20130115T041739Z~8b195824db Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 ------------------------------------------------------------------------ recv 1234 bytes from udp/[206.80.250.100]:5060 at 13:53:13.508471: ------------------------------------------------------------------------ INVITE sip:0000000000 at XXX.XXX.XXX.10:5060;transport=udp SIP/2.0 Record-Route: Record-Route: Via: SIP/2.0/UDP 206.80.250.100;branch=z9hG4bKe9f9.f210c4e7.0 Via: SIP/2.0/UDP 159.18.161.101:5060;received=159.18.161.101;rport=5060;branch=z9hG4bK-d1ba520ca83cd5dffb53405aa7f6c47b1-159.18.161.101-1 Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info, as-feature-event Max-Forwards: 69 Call-ID: 53A708A5 at 159.18.161.101 From: "Targo" ;tag=159.18.161.101+1+5f7c10+cb84d46e;isup-oli=00 To: CSeq: 379594450 INVITE Expires: 180 Organization: MetaSwitch Supported: resource-priority, 100rel Content-Length: 199 Content-Type: application/sdp Contact: "Targo" ;isup-oli=00 P-Asserted-Identity: "Targo" v=0 o=- 2517127903 2517127903 IN IP4 206.80.250.11 s=- c=IN IP4 206.80.250.107 t=0 0 m=audio 18464 RTP/AVP 18 0 101 a=rtpmap:101 telephone-event/8000 a=ptime:20 a=sendrecv a=nortpproxy:yes ------------------------------------------------------------------------ send 691 bytes to udp/[206.80.250.100]:5060 at 13:53:13.508769: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 206.80.250.100;branch=z9hG4bKe9f9.f210c4e7.0 Via: SIP/2.0/UDP 159.18.161.101:5060;received=159.18.161.101;rport=5060;branch=z9hG4bK-d1ba520ca83cd5dffb53405aa7f6c47b1-159.18.161.101-1 Record-Route: Record-Route: From: "Targo" ;tag=159.18.161.101+1+5f7c10+cb84d46e;isup-oli=00 To: Call-ID: 53A708A5 at 159.18.161.101 CSeq: 379594450 INVITE User-Agent: FreeSWITCH-mod_sofia/1.3.13b+git~20130115T041739Z~8b195824db Content-Length: 0 ------------------------------------------------------------------------ send 1027 bytes to udp/[206.80.250.100]:5060 at 13:53:13.529576: ------------------------------------------------------------------------ SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 206.80.250.100;branch=z9hG4bKe9f9.f210c4e7.0 Via: SIP/2.0/UDP 159.18.161.101:5060;received=159.18.161.101;rport=5060;branch=z9hG4bK-d1ba520ca83cd5dffb53405aa7f6c47b1-159.18.161.101-1 From: "Targo" ;tag=159.18.161.101+1+5f7c10+cb84d46e;isup-oli=00 To: ;tag=tZ31XKSNgv2tQ Call-ID: 53A708A5 at 159.18.161.101 CSeq: 379594450 INVITE User-Agent: FreeSWITCH-mod_sofia/1.3.13b+git~20130115T041739Z~8b195824db Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Proxy-Authenticate: Digest realm="159.18.161.101", nonce="b8b426b4-6c76-11e2-bf3c-d31cf8b5111c", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ recv 438 bytes from udp/[206.80.250.100]:5060 at 13:53:13.544561: ------------------------------------------------------------------------ ACK sip:0000000000 at XXX.XXX.XXX.10:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 206.80.250.100;branch=z9hG4bKe9f9.f210c4e7.0 From: "Targo" ;tag=159.18.161.101+1+5f7c10+cb84d46e;isup-oli=00 Call-ID: 53A708A5 at 159.18.161.101 To: ;tag=tZ31XKSNgv2tQ CSeq: 379594450 ACK Max-Forwards: 70 User-Agent: OpenSIPS (1.6.2-notls (x86_64/linux)) Content-Length: 0 ------------------------------------------------------------------------ recv 1234 bytes from udp/[206.80.250.100]:5060 at 13:53:16.678729: ------------------------------------------------------------------------ INVITE sip:0000000000 at XXX.XXX.XXX.10:5060;transport=udp SIP/2.0 Record-Route: Record-Route: Via: SIP/2.0/UDP 206.80.250.100;branch=z9hG4bK63ab.1987d4c7.0 Via: SIP/2.0/UDP 159.18.161.101:5060;received=159.18.161.101;rport=5060;branch=z9hG4bK-9f466f2c1215f116f7320ac7c59d82841-159.18.161.101-1 Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info, as-feature-event Max-Forwards: 69 Call-ID: 6A3ACEB0 at 159.18.161.101 From: "Targo" ;tag=159.18.161.101+1+61f633+572a68cf;isup-oli=00 To: CSeq: 484765555 INVITE Expires: 180 Organization: MetaSwitch Supported: resource-priority, 100rel Content-Length: 199 Content-Type: application/sdp Contact: "Targo" ;isup-oli=00 P-Asserted-Identity: "Targo" v=0 o=- 2517131077 2517131077 IN IP4 206.80.250.10 s=- c=IN IP4 206.80.250.101 t=0 0 m=audio 15170 RTP/AVP 18 0 101 a=rtpmap:101 telephone-event/8000 a=ptime:20 a=sendrecv a=nortpproxy:yes ------------------------------------------------------------------------ send 691 bytes to udp/[206.80.250.100]:5060 at 13:53:16.679020: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 206.80.250.100;branch=z9hG4bK63ab.1987d4c7.0 Via: SIP/2.0/UDP 159.18.161.101:5060;received=159.18.161.101;rport=5060;branch=z9hG4bK-9f466f2c1215f116f7320ac7c59d82841-159.18.161.101-1 Record-Route: Record-Route: From: "Targo" ;tag=159.18.161.101+1+61f633+572a68cf;isup-oli=00 To: Call-ID: 6A3ACEB0 at 159.18.161.101 CSeq: 484765555 INVITE User-Agent: FreeSWITCH-mod_sofia/1.3.13b+git~20130115T041739Z~8b195824db Content-Length: 0 ------------------------------------------------------------------------ send 1027 bytes to udp/[206.80.250.100]:5060 at 13:53:16.699780: ------------------------------------------------------------------------ SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 206.80.250.100;branch=z9hG4bK63ab.1987d4c7.0 Via: SIP/2.0/UDP 159.18.161.101:5060;received=159.18.161.101;rport=5060;branch=z9hG4bK-9f466f2c1215f116f7320ac7c59d82841-159.18.161.101-1 From: "Targo" ;tag=159.18.161.101+1+61f633+572a68cf;isup-oli=00 To: ;tag=U8vtZeaSD5rDK Call-ID: 6A3ACEB0 at 159.18.161.101 CSeq: 484765555 INVITE User-Agent: FreeSWITCH-mod_sofia/1.3.13b+git~20130115T041739Z~8b195824db Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Proxy-Authenticate: Digest realm="159.18.161.101", nonce="ba97e4b6-6c76-11e2-bf3e-d31cf8b5111c", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ recv 438 bytes from udp/[206.80.250.100]:5060 at 13:53:16.714857: ------------------------------------------------------------------------ ACK sip:0000000000 at XXX.XXX.XXX.10:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 206.80.250.100;branch=z9hG4bK63ab.1987d4c7.0 From: "Targo" ;tag=159.18.161.101+1+61f633+572a68cf;isup-oli=00 Call-ID: 6A3ACEB0 at 159.18.161.101 To: ;tag=U8vtZeaSD5rDK CSeq: 484765555 ACK Max-Forwards: 70 User-Agent: OpenSIPS (1.6.2-notls (x86_64/linux)) Content-Length: 0 ------------------------------------------------------------------------ send 563 bytes to udp/[206.80.250.100]:5060 at 13:53:25.639082: ------------------------------------------------------------------------ OPTIONS sip:tor.trk.tprm.ca;transport=udp SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.10:5080;rport;branch=z9hG4bK5cmQp78N4m9mD Max-Forwards: 70 From: ;tag=5rp9B8ypy4S7K To: Call-ID: 974a3871-e719-1230-dd9e-3085a9a5c78e CSeq: 39527142 OPTIONS User-Agent: FreeSWITCH-mod_sofia/1.3.13b+git~20130115T041739Z~8b195824db Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Content-Length: 0 ------------------------------------------------------------------------ recv 935 bytes from udp/[206.80.250.100]:5060 at 13:53:25.692169: ------------------------------------------------------------------------ SIP/2.0 200 OK Call-ID: 974a3871-e719-1230-dd9e-3085a9a5c78e CSeq: 39527142 OPTIONS From: ;tag=5rp9B8ypy4S7K To: ;tag=159.18.161.101+1+20302+82df7328 Via: SIP/2.0/UDP XXX.XXX.XXX.10:5080;received=XXX.XXX.XXX.10;rport=5080;branch=z9hG4bK5cmQp78N4m9mD Server: DC-SIP/2.0 Organization: MetaSwitch Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info, as-feature-event Supported: resource-priority, 100rel Allow: INVITE, ACK, CANCEL, BYE, REGISTER, OPTIONS, PRACK, UPDATE, SUBSCRIBE, NOTIFY, REFER, INFO, PUBLISH Accept-Encoding: identity Accept: application/sdp, application/simple-message-summary, message/sipfrag, application/isup, application/x-simple-call-service-info, multipart/mixed, application/broadsoft, application/vq-rtcpxr, application/media_control+xml, application/dtmf-relay, text/plain, application/x-as-feature-event+xml Content-Length: 0 ------------------------------------------------------------------------ recv 476 bytes from udp/[YYY.YYY.YYY.YYY]:5060 at 13:53:27.302711: ------------------------------------------------------------------------ NOTIFY sip:XXX.XXX.XXX.10 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.165:5060;branch=z9hG4bK-b256a314 From: "Test" ;tag=bdccc36ed8a29fdeo0 To: Call-ID: 36b982e-fa23d21e at 192.168.3.165 CSeq: 16015 NOTIFY Max-Forwards: 70 Contact: "Test" Event: keep-alive User-Agent: Cisco/SPA112-1.2.1(004) Allow-Events: talk, hold, conference Content-Length: 0 ------------------------------------------------------------------------ send 722 bytes to udp/[YYY.YYY.YYY.YYY]:5060 at 13:53:27.303009: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.165:5060;branch=z9hG4bK-b256a314;received=YYY.YYY.YYY.YYY From: "Test" ;tag=bdccc36ed8a29fdeo0 To: ;tag=vHpK19tvaeF0e Call-ID: 36b982e-fa23d21e at 192.168.3.165 CSeq: 16015 NOTIFY Contact: User-Agent: FreeSWITCH-mod_sofia/1.3.13b+git~20130115T041739Z~8b195824db Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 ------------------------------------------------------------------------ recv 476 bytes from udp/[YYY.YYY.YYY.YYY]:5060 at 13:53:42.351254: ------------------------------------------------------------------------ NOTIFY sip:XXX.XXX.XXX.10 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.165:5060;branch=z9hG4bK-a1beec1f From: "Test" ;tag=bdccc36ed8a29fdeo0 To: Call-ID: 36b982e-fa23d21e at 192.168.3.165 CSeq: 16016 NOTIFY Max-Forwards: 70 Contact: "Test" Event: keep-alive User-Agent: Cisco/SPA112-1.2.1(004) Allow-Events: talk, hold, conference Content-Length: 0 ------------------------------------------------------------------------ send 722 bytes to udp/[YYY.YYY.YYY.YYY]:5060 at 13:53:42.351607: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.165:5060;branch=z9hG4bK-a1beec1f;received=YYY.YYY.YYY.YYY From: "Test" ;tag=bdccc36ed8a29fdeo0 To: ;tag=XtFc34B07p5ja Call-ID: 36b982e-fa23d21e at 192.168.3.165 CSeq: 16016 NOTIFY Contact: User-Agent: FreeSWITCH-mod_sofia/1.3.13b+git~20130115T041739Z~8b195824db Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 ------------------------------------------------------------------------ recv 880 bytes from udp/[206.80.250.100]:5060 at 13:53:46.763491: ------------------------------------------------------------------------ OPTIONS sip:metaswitch at XXX.XXX.XXX.10:5060;transport=udp SIP/2.0 Record-Route: Record-Route: Via: SIP/2.0/UDP 206.80.250.100;branch=z9hG4bK9d4a.f4741622.0 Via: SIP/2.0/UDP 159.18.161.101:5060;received=159.18.161.101;rport=5060;branch=z9hG4bK-7bef601cb4681d5987165959279aa0851-159.18.161.101-1 Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info, as-feature-event Max-Forwards: 69 Call-ID: 51DED8A0 at 159.18.161.101 From: ;tag=159.18.161.101+1+b5232+251fb057 CSeq: 179875557 OPTIONS Organization: MetaSwitch Supported: resource-priority, 100rel Content-Length: 0 Contact: To: ------------------------------------------------------------------------ send 1049 bytes to udp/[206.80.250.100]:5060 at 13:53:46.763847: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 206.80.250.100;branch=z9hG4bK9d4a.f4741622.0 Via: SIP/2.0/UDP 159.18.161.101:5060;received=159.18.161.101;rport=5060;branch=z9hG4bK-7bef601cb4681d5987165959279aa0851-159.18.161.101-1 Record-Route: Record-Route: From: ;tag=159.18.161.101+1+b5232+251fb057 To: ;tag=y3844Zv34ZU5N Call-ID: 51DED8A0 at 159.18.161.101 CSeq: 179875557 OPTIONS Contact: User-Agent: FreeSWITCH-mod_sofia/1.3.13b+git~20130115T041739Z~8b195824db Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 ------------------------------------------------------------------------ send 563 bytes to udp/[206.80.250.100]:5060 at 13:53:55.640109: ------------------------------------------------------------------------ OPTIONS sip:tor.trk.tprm.ca;transport=udp SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.10:5080;rport;branch=z9hG4bK6NDgr2SS1XZ7r Max-Forwards: 70 From: ;tag=61F2D3FtUDgtF To: Call-ID: a92c0359-e719-1230-dd9e-3085a9a5c78e CSeq: 39527143 OPTIONS User-Agent: FreeSWITCH-mod_sofia/1.3.13b+git~20130115T041739Z~8b195824db Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Content-Length: 0 ------------------------------------------------------------------------ recv 936 bytes from udp/[206.80.250.100]:5060 at 13:53:55.693380: ------------------------------------------------------------------------ SIP/2.0 200 OK Call-ID: a92c0359-e719-1230-dd9e-3085a9a5c78e CSeq: 39527143 OPTIONS From: ;tag=61F2D3FtUDgtF To: ;tag=159.18.161.101+1+103e2c+ac2c244a Via: SIP/2.0/UDP XXX.XXX.XXX.10:5080;received=XXX.XXX.XXX.10;rport=5080;branch=z9hG4bK6NDgr2SS1XZ7r Server: DC-SIP/2.0 Organization: MetaSwitch Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info, as-feature-event Supported: resource-priority, 100rel Allow: INVITE, ACK, CANCEL, BYE, REGISTER, OPTIONS, PRACK, UPDATE, SUBSCRIBE, NOTIFY, REFER, INFO, PUBLISH Accept-Encoding: identity Accept: application/sdp, application/simple-message-summary, message/sipfrag, application/isup, application/x-simple-call-service-info, multipart/mixed, application/broadsoft, application/vq-rtcpxr, application/media_control+xml, application/dtmf-relay, text/plain, application/x-as-feature-event+xml Content-Length: 0 ------------------------------------------------------------------------ recv 476 bytes from udp/[YYY.YYY.YYY.YYY]:5060 at 13:53:57.400809: ------------------------------------------------------------------------ NOTIFY sip:XXX.XXX.XXX.10 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.165:5060;branch=z9hG4bK-1b2abeab From: "Test" ;tag=bdccc36ed8a29fdeo0 To: Call-ID: 36b982e-fa23d21e at 192.168.3.165 CSeq: 16017 NOTIFY Max-Forwards: 70 Contact: "Test" Event: keep-alive User-Agent: Cisco/SPA112-1.2.1(004) Allow-Events: talk, hold, conference Content-Length: 0 ------------------------------------------------------------------------ send 722 bytes to udp/[YYY.YYY.YYY.YYY]:5060 at 13:53:57.401180: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.165:5060;branch=z9hG4bK-1b2abeab;received=YYY.YYY.YYY.YYY From: "Test" ;tag=bdccc36ed8a29fdeo0 To: ;tag=Zc2X6tD718HrH Call-ID: 36b982e-fa23d21e at 192.168.3.165 CSeq: 16017 NOTIFY Contact: User-Agent: FreeSWITCH-mod_sofia/1.3.13b+git~20130115T041739Z~8b195824db Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 ------------------------------------------------------------------------ recv 476 bytes from udp/[YYY.YYY.YYY.YYY]:5060 at 13:54:12.459286: ------------------------------------------------------------------------ NOTIFY sip:XXX.XXX.XXX.10 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.165:5060;branch=z9hG4bK-f2383937 From: "Test" ;tag=bdccc36ed8a29fdeo0 To: Call-ID: 36b982e-fa23d21e at 192.168.3.165 CSeq: 16018 NOTIFY Max-Forwards: 70 Contact: "Test" Event: keep-alive User-Agent: Cisco/SPA112-1.2.1(004) Allow-Events: talk, hold, conference Content-Length: 0 ------------------------------------------------------------------------ send 722 bytes to udp/[YYY.YYY.YYY.YYY]:5060 at 13:54:12.459682: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.165:5060;branch=z9hG4bK-f2383937;received=YYY.YYY.YYY.YYY From: "Test" ;tag=bdccc36ed8a29fdeo0 To: ;tag=0NUp8NyaZH8aD Call-ID: 36b982e-fa23d21e at 192.168.3.165 CSeq: 16018 NOTIFY Contact: User-Agent: FreeSWITCH-mod_sofia/1.3.13b+git~20130115T041739Z~8b195824db Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 ------------------------------------------------------------------------ recv 883 bytes from udp/[206.80.250.100]:5060 at 13:54:16.823211: ------------------------------------------------------------------------ OPTIONS sip:metaswitch at XXX.XXX.XXX.10:5060;transport=udp SIP/2.0 Record-Route: Record-Route: Via: SIP/2.0/UDP 206.80.250.100;branch=z9hG4bK3b83.fd2bd7e3.0 Via: SIP/2.0/UDP 159.18.161.101:5060;received=159.18.161.101;rport=5060;branch=z9hG4bK-bb3d6efe408f54c8c4b10d03f420187b1-159.18.161.101-1 Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info, as-feature-event Max-Forwards: 69 Call-ID: 41C790AE at 159.18.161.101 From: ;tag=159.18.161.101+1+1aa238+2547ae69 CSeq: 202885604 OPTIONS Organization: MetaSwitch Supported: resource-priority, 100rel Content-Length: 0 Contact: To: ------------------------------------------------------------------------ send 1052 bytes to udp/[206.80.250.100]:5060 at 13:54:16.823588: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 206.80.250.100;branch=z9hG4bK3b83.fd2bd7e3.0 Via: SIP/2.0/UDP 159.18.161.101:5060;received=159.18.161.101;rport=5060;branch=z9hG4bK-bb3d6efe408f54c8c4b10d03f420187b1-159.18.161.101-1 Record-Route: Record-Route: From: ;tag=159.18.161.101+1+1aa238+2547ae69 To: ;tag=1ymFaHFevtyXr Call-ID: 41C790AE at 159.18.161.101 CSeq: 202885604 OPTIONS Contact: User-Agent: FreeSWITCH-mod_sofia/1.3.13b+git~20130115T041739Z~8b195824db Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 ------------------------------------------------------------------------ send 563 bytes to udp/[206.80.250.100]:5060 at 13:54:25.641105: ------------------------------------------------------------------------ OPTIONS sip:tor.trk.tprm.ca;transport=udp SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.10:5080;rport;branch=z9hG4bK7y68SXaXy6Ntm Max-Forwards: 70 From: ;tag=7a9tFy0Xrp6cB To: Call-ID: bb0dcd53-e719-1230-dd9e-3085a9a5c78e CSeq: 39527144 OPTIONS User-Agent: FreeSWITCH-mod_sofia/1.3.13b+git~20130115T041739Z~8b195824db Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Content-Length: 0 ------------------------------------------------------------------------ recv 936 bytes from udp/[206.80.250.100]:5060 at 13:54:25.695024: ------------------------------------------------------------------------ SIP/2.0 200 OK Call-ID: bb0dcd53-e719-1230-dd9e-3085a9a5c78e CSeq: 39527144 OPTIONS From: ;tag=7a9tFy0Xrp6cB To: ;tag=159.18.161.101+1+1e022d+ff66e02b Via: SIP/2.0/UDP XXX.XXX.XXX.10:5080;received=XXX.XXX.XXX.10;rport=5080;branch=z9hG4bK7y68SXaXy6Ntm Server: DC-SIP/2.0 Organization: MetaSwitch Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info, as-feature-event Supported: resource-priority, 100rel Allow: INVITE, ACK, CANCEL, BYE, REGISTER, OPTIONS, PRACK, UPDATE, SUBSCRIBE, NOTIFY, REFER, INFO, PUBLISH Accept-Encoding: identity Accept: application/sdp, application/simple-message-summary, message/sipfrag, application/isup, application/x-simple-call-service-info, multipart/mixed, application/broadsoft, application/vq-rtcpxr, application/media_control+xml, application/dtmf-relay, text/plain, application/x-as-feature-event+xml Content-Length: 0 ------------------------------------------------------------------------ recv 476 bytes from udp/[YYY.YYY.YYY.YYY]:5060 at 13:54:27.508130: ------------------------------------------------------------------------ NOTIFY sip:XXX.XXX.XXX.10 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.165:5060;branch=z9hG4bK-86d025fb From: "Test" ;tag=bdccc36ed8a29fdeo0 To: Call-ID: 36b982e-fa23d21e at 192.168.3.165 CSeq: 16019 NOTIFY Max-Forwards: 70 Contact: "Test" Event: keep-alive User-Agent: Cisco/SPA112-1.2.1(004) Allow-Events: talk, hold, conference Content-Length: 0 ------------------------------------------------------------------------ send 722 bytes to udp/[YYY.YYY.YYY.YYY]:5060 at 13:54:27.508445: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.165:5060;branch=z9hG4bK-86d025fb;received=YYY.YYY.YYY.YYY From: "Test" ;tag=bdccc36ed8a29fdeo0 To: ;tag=27D8Bc0HS3mgm Call-ID: 36b982e-fa23d21e at 192.168.3.165 CSeq: 16019 NOTIFY Contact: User-Agent: FreeSWITCH-mod_sofia/1.3.13b+git~20130115T041739Z~8b195824db Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 ------------------------------------------------------------------------ recv 476 bytes from udp/[YYY.YYY.YYY.YYY]:5060 at 13:54:42.555691: ------------------------------------------------------------------------ NOTIFY sip:XXX.XXX.XXX.10 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.165:5060;branch=z9hG4bK-a29985e8 From: "Test" ;tag=bdccc36ed8a29fdeo0 To: Call-ID: 36b982e-fa23d21e at 192.168.3.165 CSeq: 16020 NOTIFY Max-Forwards: 70 Contact: "Test" Event: keep-alive User-Agent: Cisco/SPA112-1.2.1(004) Allow-Events: talk, hold, conference Content-Length: 0 ------------------------------------------------------------------------ send 722 bytes to udp/[YYY.YYY.YYY.YYY]:5060 at 13:54:42.556042: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.165:5060;branch=z9hG4bK-a29985e8;received=YYY.YYY.YYY.YYY From: "Test" ;tag=bdccc36ed8a29fdeo0 To: ;tag=3g70D7gNpcB3F Call-ID: 36b982e-fa23d21e at 192.168.3.165 CSeq: 16020 NOTIFY Contact: User-Agent: FreeSWITCH-mod_sofia/1.3.13b+git~20130115T041739Z~8b195824db Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 ------------------------------------------------------------------------ recv 883 bytes from udp/[206.80.250.100]:5060 at 13:54:46.878105: ------------------------------------------------------------------------ OPTIONS sip:metaswitch at XXX.XXX.XXX.10:5060;transport=udp SIP/2.0 Record-Route: Record-Route: Via: SIP/2.0/UDP 206.80.250.100;branch=z9hG4bK54e6.38ffd511.0 Via: SIP/2.0/UDP 159.18.161.101:5060;received=159.18.161.101;rport=5060;branch=z9hG4bK-c5632998431a758e2273660abe361edc1-159.18.161.101-1 Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info, as-feature-event Max-Forwards: 69 Call-ID: 4FAAF32F at 159.18.161.101 From: ;tag=159.18.161.101+1+27b51c+dab9c1ec CSeq: 395018158 OPTIONS Organization: MetaSwitch Supported: resource-priority, 100rel Content-Length: 0 Contact: To: ------------------------------------------------------------------------ send 1052 bytes to udp/[206.80.250.100]:5060 at 13:54:46.878468: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 206.80.250.100;branch=z9hG4bK54e6.38ffd511.0 Via: SIP/2.0/UDP 159.18.161.101:5060;received=159.18.161.101;rport=5060;branch=z9hG4bK-c5632998431a758e2273660abe361edc1-159.18.161.101-1 Record-Route: Record-Route: From: ;tag=159.18.161.101+1+27b51c+dab9c1ec To: ;tag=4S0SF21rKN1NB Call-ID: 4FAAF32F at 159.18.161.101 CSeq: 395018158 OPTIONS Contact: User-Agent: FreeSWITCH-mod_sofia/1.3.13b+git~20130115T041739Z~8b195824db Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 ------------------------------------------------------------------------ send 563 bytes to udp/[206.80.250.100]:5060 at 13:54:55.642098: ------------------------------------------------------------------------ OPTIONS sip:tor.trk.tprm.ca;transport=udp SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.10:5080;rport;branch=z9hG4bK87Z1UrU0UFcDg Max-Forwards: 70 From: ;tag=8K2KHSH1NZvZp To: Call-ID: ccef9765-e719-1230-dd9e-3085a9a5c78e CSeq: 39527145 OPTIONS User-Agent: FreeSWITCH-mod_sofia/1.3.13b+git~20130115T041739Z~8b195824db Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Content-Length: 0 ------------------------------------------------------------------------ recv 936 bytes from udp/[206.80.250.100]:5060 at 13:54:55.695718: ------------------------------------------------------------------------ SIP/2.0 200 OK Call-ID: ccef9765-e719-1230-dd9e-3085a9a5c78e CSeq: 39527145 OPTIONS From: ;tag=8K2KHSH1NZvZp To: ;tag=159.18.161.101+1+2c6939+6638d3ba Via: SIP/2.0/UDP XXX.XXX.XXX.10:5080;received=XXX.XXX.XXX.10;rport=5080;branch=z9hG4bK87Z1UrU0UFcDg Server: DC-SIP/2.0 Organization: MetaSwitch Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info, as-feature-event Supported: resource-priority, 100rel Allow: INVITE, ACK, CANCEL, BYE, REGISTER, OPTIONS, PRACK, UPDATE, SUBSCRIBE, NOTIFY, REFER, INFO, PUBLISH Accept-Encoding: identity Accept: application/sdp, application/simple-message-summary, message/sipfrag, application/isup, application/x-simple-call-service-info, multipart/mixed, application/broadsoft, application/vq-rtcpxr, application/media_control+xml, application/dtmf-relay, text/plain, application/x-as-feature-event+xml Content-Length: 0 ------------------------------------------------------------------------ recv 476 bytes from udp/[YYY.YYY.YYY.YYY]:5060 at 13:54:57.603503: ------------------------------------------------------------------------ NOTIFY sip:XXX.XXX.XXX.10 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.165:5060;branch=z9hG4bK-34147cec From: "Test" ;tag=bdccc36ed8a29fdeo0 To: Call-ID: 36b982e-fa23d21e at 192.168.3.165 CSeq: 16021 NOTIFY Max-Forwards: 70 Contact: "Test" Event: keep-alive User-Agent: Cisco/SPA112-1.2.1(004) Allow-Events: talk, hold, conference Content-Length: 0 ------------------------------------------------------------------------ send 722 bytes to udp/[YYY.YYY.YYY.YYY]:5060 at 13:54:57.603856: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.165:5060;branch=z9hG4bK-34147cec;received=YYY.YYY.YYY.YYY From: "Test" ;tag=bdccc36ed8a29fdeo0 To: ;tag=52SjHXjvgyQ8p Call-ID: 36b982e-fa23d21e at 192.168.3.165 CSeq: 16021 NOTIFY Contact: User-Agent: FreeSWITCH-mod_sofia/1.3.13b+git~20130115T041739Z~8b195824db Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 ------------------------------------------------------------------------ recv 476 bytes from udp/[YYY.YYY.YYY.YYY]:5060 at 13:55:12.652368: ------------------------------------------------------------------------ NOTIFY sip:XXX.XXX.XXX.10 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.165:5060;branch=z9hG4bK-cfbb3f6d From: "Test" ;tag=bdccc36ed8a29fdeo0 To: Call-ID: 36b982e-fa23d21e at 192.168.3.165 CSeq: 16022 NOTIFY Max-Forwards: 70 Contact: "Test" Event: keep-alive User-Agent: Cisco/SPA112-1.2.1(004) Allow-Events: talk, hold, conference Content-Length: 0 ------------------------------------------------------------------------ send 722 bytes to udp/[YYY.YYY.YYY.YYY]:5060 at 13:55:12.652747: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.165:5060;branch=z9hG4bK-cfbb3f6d;received=YYY.YYY.YYY.YYY From: "Test" ;tag=bdccc36ed8a29fdeo0 To: ;tag=6BKBKr3ZD7DUj Call-ID: 36b982e-fa23d21e at 192.168.3.165 CSeq: 16022 NOTIFY Contact: User-Agent: FreeSWITCH-mod_sofia/1.3.13b+git~20130115T041739Z~8b195824db Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 ------------------------------------------------------------------------ recv 882 bytes from udp/[206.80.250.100]:5060 at 13:55:16.929745: ------------------------------------------------------------------------ OPTIONS sip:metaswitch at XXX.XXX.XXX.10:5060;transport=udp SIP/2.0 Record-Route: Record-Route: Via: SIP/2.0/UDP 206.80.250.100;branch=z9hG4bK74c.73203b16.0 Via: SIP/2.0/UDP 159.18.161.101:5060;received=159.18.161.101;rport=5060;branch=z9hG4bK-d9d04678d5e1e5408cae1dfc94bf9d6f1-159.18.161.101-1 Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info, as-feature-event Max-Forwards: 69 Call-ID: 0D9BF8AC at 159.18.161.101 From: ;tag=159.18.161.101+1+364b17+39cf2a52 CSeq: 538287085 OPTIONS Organization: MetaSwitch Supported: resource-priority, 100rel Content-Length: 0 Contact: To: ------------------------------------------------------------------------ send 1051 bytes to udp/[206.80.250.100]:5060 at 13:55:16.930124: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 206.80.250.100;branch=z9hG4bK74c.73203b16.0 Via: SIP/2.0/UDP 159.18.161.101:5060;received=159.18.161.101;rport=5060;branch=z9hG4bK-d9d04678d5e1e5408cae1dfc94bf9d6f1-159.18.161.101-1 Record-Route: Record-Route: From: ;tag=159.18.161.101+1+364b17+39cf2a52 To: ;tag=7mc4mKm3ag4De Call-ID: 0D9BF8AC at 159.18.161.101 CSeq: 538287085 OPTIONS Contact: User-Agent: FreeSWITCH-mod_sofia/1.3.13b+git~20130115T041739Z~8b195824db Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 ------------------------------------------------------------------------ send 563 bytes to udp/[206.80.250.100]:5060 at 13:55:25.643077: ------------------------------------------------------------------------ OPTIONS sip:tor.trk.tprm.ca;transport=udp SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.10:5080;rport;branch=z9hG4bK9gStXKc4rr2ZB Max-Forwards: 70 From: ;tag=9vUcKm24j8jjj To: Call-ID: ded1607b-e719-1230-dd9e-3085a9a5c78e CSeq: 39527146 OPTIONS User-Agent: FreeSWITCH-mod_sofia/1.3.13b+git~20130115T041739Z~8b195824db Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Content-Length: 0 ------------------------------------------------------------------------ recv 936 bytes from udp/[206.80.250.100]:5060 at 13:55:25.696676: ------------------------------------------------------------------------ SIP/2.0 200 OK Call-ID: ded1607b-e719-1230-dd9e-3085a9a5c78e CSeq: 39527146 OPTIONS From: ;tag=9vUcKm24j8jjj To: ;tag=159.18.161.101+1+3ae612+1e63216f Via: SIP/2.0/UDP XXX.XXX.XXX.10:5080;received=XXX.XXX.XXX.10;rport=5080;branch=z9hG4bK9gStXKc4rr2ZB Server: DC-SIP/2.0 Organization: MetaSwitch Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info, as-feature-event Supported: resource-priority, 100rel Allow: INVITE, ACK, CANCEL, BYE, REGISTER, OPTIONS, PRACK, UPDATE, SUBSCRIBE, NOTIFY, REFER, INFO, PUBLISH Accept-Encoding: identity Accept: application/sdp, application/simple-message-summary, message/sipfrag, application/isup, application/x-simple-call-service-info, multipart/mixed, application/broadsoft, application/vq-rtcpxr, application/media_control+xml, application/dtmf-relay, text/plain, application/x-as-feature-event+xml Content-Length: 0 ------------------------------------------------------------------------ recv 476 bytes from udp/[YYY.YYY.YYY.YYY]:5060 at 13:55:27.699532: ------------------------------------------------------------------------ NOTIFY sip:XXX.XXX.XXX.10 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.165:5060;branch=z9hG4bK-daeafcc1 From: "Test" ;tag=bdccc36ed8a29fdeo0 To: Call-ID: 36b982e-fa23d21e at 192.168.3.165 CSeq: 16023 NOTIFY Max-Forwards: 70 Contact: "Test" Event: keep-alive User-Agent: Cisco/SPA112-1.2.1(004) Allow-Events: talk, hold, conference Content-Length: 0 ------------------------------------------------------------------------ send 722 bytes to udp/[YYY.YYY.YYY.YYY]:5060 at 13:55:27.699829: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.165:5060;branch=z9hG4bK-daeafcc1;received=YYY.YYY.YYY.YYY From: "Test" ;tag=bdccc36ed8a29fdeo0 To: ;tag=8X5vpe567rt0S Call-ID: 36b982e-fa23d21e at 192.168.3.165 CSeq: 16023 NOTIFY Contact: User-Agent: FreeSWITCH-mod_sofia/1.3.13b+git~20130115T041739Z~8b195824db Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 ------------------------------------------------------------------------ recv 476 bytes from udp/[YYY.YYY.YYY.YYY]:5060 at 13:55:42.759531: ------------------------------------------------------------------------ NOTIFY sip:XXX.XXX.XXX.10 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.165:5060;branch=z9hG4bK-1504455f From: "Test" ;tag=bdccc36ed8a29fdeo0 To: Call-ID: 36b982e-fa23d21e at 192.168.3.165 CSeq: 16024 NOTIFY Max-Forwards: 70 Contact: "Test" Event: keep-alive User-Agent: Cisco/SPA112-1.2.1(004) Allow-Events: talk, hold, conference Content-Length: 0 ------------------------------------------------------------------------ send 722 bytes to udp/[YYY.YYY.YYY.YYY]:5060 at 13:55:42.759922: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.165:5060;branch=z9hG4bK-1504455f;received=YYY.YYY.YYY.YYY From: "Test" ;tag=bdccc36ed8a29fdeo0 To: ;tag=96yNr9Na51gKN Call-ID: 36b982e-fa23d21e at 192.168.3.165 CSeq: 16024 NOTIFY Contact: User-Agent: FreeSWITCH-mod_sofia/1.3.13b+git~20130115T041739Z~8b195824db Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 ------------------------------------------------------------------------ recv 883 bytes from udp/[206.80.250.100]:5060 at 13:55:46.986769: ------------------------------------------------------------------------ OPTIONS sip:metaswitch at XXX.XXX.XXX.10:5060;transport=udp SIP/2.0 Record-Route: Record-Route: Via: SIP/2.0/UDP 206.80.250.100;branch=z9hG4bK5d7f.9e2c8776.0 Via: SIP/2.0/UDP 159.18.161.101:5060;received=159.18.161.101;rport=5060;branch=z9hG4bK-c16202dac874d2fbfebb75f5a45c49241-159.18.161.101-1 Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info, as-feature-event Max-Forwards: 69 Call-ID: FDB493A8 at 159.18.161.101 From: ;tag=159.18.161.101+1+45592a+bcd61410 CSeq: 218455992 OPTIONS Organization: MetaSwitch Supported: resource-priority, 100rel Content-Length: 0 Contact: To: ------------------------------------------------------------------------ send 1052 bytes to udp/[206.80.250.100]:5060 at 13:55:46.987125: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 206.80.250.100;branch=z9hG4bK5d7f.9e2c8776.0 Via: SIP/2.0/UDP 159.18.161.101:5060;received=159.18.161.101;rport=5060;branch=z9hG4bK-c16202dac874d2fbfebb75f5a45c49241-159.18.161.101-1 Record-Route: Record-Route: From: ;tag=159.18.161.101+1+45592a+bcd61410 To: ;tag=agret46D2a75g Call-ID: FDB493A8 at 159.18.161.101 CSeq: 218455992 OPTIONS Contact: User-Agent: FreeSWITCH-mod_sofia/1.3.13b+git~20130115T041739Z~8b195824db Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 ------------------------------------------------------------------------ send 563 bytes to udp/[206.80.250.100]:5060 at 13:55:55.644081: ------------------------------------------------------------------------ OPTIONS sip:tor.trk.tprm.ca;transport=udp SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.10:5080;rport;branch=z9hG4bKatjKZeX7N1rjQ Max-Forwards: 70 From: ;tag=a6m5mFK8FH94D To: Call-ID: f0b32a5c-e719-1230-dd9e-3085a9a5c78e CSeq: 39527147 OPTIONS User-Agent: FreeSWITCH-mod_sofia/1.3.13b+git~20130115T041739Z~8b195824db Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Content-Length: 0 ------------------------------------------------------------------------ recv 936 bytes from udp/[206.80.250.100]:5060 at 13:55:55.697161: ------------------------------------------------------------------------ SIP/2.0 200 OK Call-ID: f0b32a5c-e719-1230-dd9e-3085a9a5c78e CSeq: 39527147 OPTIONS From: ;tag=a6m5mFK8FH94D To: ;tag=159.18.161.101+1+49b037+78f7fdce Via: SIP/2.0/UDP XXX.XXX.XXX.10:5080;received=XXX.XXX.XXX.10;rport=5080;branch=z9hG4bKatjKZeX7N1rjQ Server: DC-SIP/2.0 Organization: MetaSwitch Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info, as-feature-event Supported: resource-priority, 100rel Allow: INVITE, ACK, CANCEL, BYE, REGISTER, OPTIONS, PRACK, UPDATE, SUBSCRIBE, NOTIFY, REFER, INFO, PUBLISH Accept-Encoding: identity Accept: application/sdp, application/simple-message-summary, message/sipfrag, application/isup, application/x-simple-call-service-info, multipart/mixed, application/broadsoft, application/vq-rtcpxr, application/media_control+xml, application/dtmf-relay, text/plain, application/x-as-feature-event+xml Content-Length: 0 ------------------------------------------------------------------------ recv 476 bytes from udp/[YYY.YYY.YYY.YYY]:5060 at 13:55:57.808917: ------------------------------------------------------------------------ NOTIFY sip:XXX.XXX.XXX.10 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.165:5060;branch=z9hG4bK-80636b39 From: "Test" ;tag=bdccc36ed8a29fdeo0 To: Call-ID: 36b982e-fa23d21e at 192.168.3.165 CSeq: 16025 NOTIFY Max-Forwards: 70 Contact: "Test" Event: keep-alive User-Agent: Cisco/SPA112-1.2.1(004) Allow-Events: talk, hold, conference Content-Length: 0 ------------------------------------------------------------------------ send 722 bytes to udp/[YYY.YYY.YYY.YYY]:5060 at 13:55:57.809287: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.165:5060;branch=z9hG4bK-80636b39;received=YYY.YYY.YYY.YYY From: "Test" ;tag=bdccc36ed8a29fdeo0 To: ;tag=BSH7UZQHZKXrc Call-ID: 36b982e-fa23d21e at 192.168.3.165 CSeq: 16025 NOTIFY Contact: User-Agent: FreeSWITCH-mod_sofia/1.3.13b+git~20130115T041739Z~8b195824db Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 ------------------------------------------------------------------------ recv 475 bytes from udp/[YYY.YYY.YYY.YYY]:5060 at 13:56:12.856499: ------------------------------------------------------------------------ NOTIFY sip:XXX.XXX.XXX.10 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.165:5060;branch=z9hG4bK-7039dd6 From: "Test" ;tag=bdccc36ed8a29fdeo0 To: Call-ID: 36b982e-fa23d21e at 192.168.3.165 CSeq: 16026 NOTIFY Max-Forwards: 70 Contact: "Test" Event: keep-alive User-Agent: Cisco/SPA112-1.2.1(004) Allow-Events: talk, hold, conference Content-Length: 0 ------------------------------------------------------------------------ send 721 bytes to udp/[YYY.YYY.YYY.YYY]:5060 at 13:56:12.856940: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.165:5060;branch=z9hG4bK-7039dd6;received=YYY.YYY.YYY.YYY From: "Test" ;tag=bdccc36ed8a29fdeo0 To: ;tag=c2a0Xt8mvvKBr Call-ID: 36b982e-fa23d21e at 192.168.3.165 CSeq: 16026 NOTIFY Contact: User-Agent: FreeSWITCH-mod_sofia/1.3.13b+git~20130115T041739Z~8b195824db Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 ------------------------------------------------------------------------ From POlsson at enghouse.com Fri Feb 1 18:12:56 2013 From: POlsson at enghouse.com (Peter Olsson) Date: Fri, 1 Feb 2013 15:12:56 +0000 Subject: [Freeswitch-users] Execute app in sync mode when using event sockets Message-ID: <1FFF97C269757C458224B7C895F35F15214D5A@cantor.std.visionutv.se> My best suggestion is to "think different". If you're using ESL in inbound mode, you probably don't want to block the socket for anything, since you might receive events for other channels etc. The best way is to use it asynchronous - as intended. Send the speak command, and then trigger on the event that is sent when the playback is finished. In my use case I use it for an IVR system, I have the ESL lib, and then I create one thread per new channel (could be done differently, but it was the easiest approach right now). Then I queue the events for the separate channels back into their threads (separated per channel uuid). In the thread I do the things I want asynchronously, then I wait for events to be queued to my thread, and I handle these events. When I get the reply that the playback is finished, I do the next step etc. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Hector Geraldino Skickat: den 31 januari 2013 20:19 Till: FreeSWITCH Users Help ?mne: [Freeswitch-users] Execute app in sync mode when using event sockets Hi, I have a java app that uses mod_event_socket (inbound mode) to control FreeSWITCH. I can send api commands which, by default, are synchronous, so the thread locks until the command is successfully executed. The problem I'm facing is when I have to execute commands that are in other modules (like playback, or speak). As they are not api commands, I must send an execute appName + arguments, which by nature is executed in async mode. What I want to achieve is to execute an speak command and wait until the TTS operation ends (blocking the thread). This sounds like a trivial use case: play some audio (TTS, file) and then execute the rest of the logic. It is easy to accomplish on the dialplan, but I just can't figure out how to do it on an ESL application. Here's what I've tried so far: + send execute speak command + api sleep and wait for the CHANNEL_EXECUTE_COMPLETE (to simulate a lock [sleep] and continue [break after event is received]), but the event arrives after the sleep command completes. + use uuid_broadcast uuid speak::args, doesn't work Any hints, ideas are welcomed. Thank you! Hector !DSPAM:510ac0b632761097822086! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130201/5aee7382/attachment.html From drk at drkngs.net Fri Feb 1 22:59:35 2013 From: drk at drkngs.net (Dave R. Kompel) Date: Fri, 01 Feb 2013 11:59:35 -0800 Subject: [Freeswitch-users] ESL and Event Socket with secure connection In-Reply-To: Message-ID: <20130201195935.3a334a17@mail.tritonwest.net> It's just as easy on windows, most VPN protocols are there out of the box. --Dave _____ From: Roger Castaldo [mailto:roger.castaldo at gmail.com] To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] Sent: Fri, 01 Feb 2013 10:01:48 -0800 Subject: Re: [Freeswitch-users] ESL and Event Socket with secure connection If you are not using ESL on the server itself I would suggest using ssl or ssh tunneling to handle it, or even a vpn connection would work as well. If you are running on linux servers, ssl tunneling and ssh tunneling are relatively easy to setup. Windows, don't know. On Fri, Feb 1, 2013 at 12:53 PM, Eduardo Nunes Pereira wrote: How is the best way to implement connection between ESL and Event Socket with secure connection, something like HTTPS. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130201/ee4dbc7d/attachment.html From curriegrad2004 at gmail.com Fri Feb 1 23:03:39 2013 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Fri, 1 Feb 2013 12:03:39 -0800 Subject: [Freeswitch-users] Friday Free For All on Today? Message-ID: Is the Friday Free For All still going on today? From krice at freeswitch.org Fri Feb 1 23:09:50 2013 From: krice at freeswitch.org (Ken Rice) Date: Fri, 01 Feb 2013 14:09:50 -0600 Subject: [Freeswitch-users] Friday Free For All on Today? In-Reply-To: Message-ID: Yes sounds like a good time to start it to me On 2/1/13 2:03 PM, "curriegrad2004" wrote: > Is the Friday Free For All still going on today? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch From steveayre at gmail.com Fri Feb 1 23:32:02 2013 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 1 Feb 2013 20:32:02 +0000 Subject: [Freeswitch-users] ESL and Event Socket with secure connection In-Reply-To: <20130201195935.3a334a17@mail.tritonwest.net> References: <20130201195935.3a334a17@mail.tritonwest.net> Message-ID: I suspect he more meant stunnel-like stuff for tunnelling the connection via SSH. Putty (or plink) allow you to tunnel connections via SSH very easily. But making sure it's running all the time and reconnecting automatically would be the hard part. You could fork() and exec() it from your connecting program. A VPN such as OpenVPN may be easier, but you are probably then going to have to connect via a central hub rather than directly then. ESL itself is a plaintext protocol and has no encryption or identity verification of its own, though it would be nice if a TLS option could be added. -Steve On 1 February 2013 19:59, Dave R. Kompel wrote: > ** > It's just as easy on windows, most VPN protocols are there out of the box. > > --Dave > > ------------------------------ > *From:* Roger Castaldo [mailto:roger.castaldo at gmail.com] > *To:* FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] > *Sent:* Fri, 01 Feb 2013 10:01:48 -0800 > *Subject:* Re: [Freeswitch-users] ESL and Event Socket with secure > connection > > > If you are not using ESL on the server itself I would suggest using ssl or > ssh tunneling to handle it, or even a vpn connection would work as well. > If you are running on linux servers, ssl tunneling and ssh tunneling are > relatively easy to setup. Windows, don't know. > > > On Fri, Feb 1, 2013 at 12:53 PM, Eduardo Nunes Pereira < > eduardonunesp at gmail.com> wrote: > >> How is the best way to implement connection between ESL and Event >> Socket with secure connection, something like HTTPS. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130201/632246ed/attachment-0001.html From steveayre at gmail.com Fri Feb 1 23:33:31 2013 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 1 Feb 2013 20:33:31 +0000 Subject: [Freeswitch-users] problems with freeswitch + zrtp in proxy-media mode In-Reply-To: <2DBB65A0-2551-4FE8-88E0-CB4624325733@edge-net.net> References: <2DBB65A0-2551-4FE8-88E0-CB4624325733@edge-net.net> Message-ID: Is there a reason you're using proxy media mode? Does it work in normal media handling mode? (ie not proxy and not bypass) -Steve On 1 February 2013 14:15, Eli Burke wrote: > Yes, inbound-zrtp-passthru is true. I think it gets set automatically when > you turn on ZRTP. > > The config I'm testing with is a stock 'enable_zrtp=yes' build of HEAD > circa last week, with proxy-media=true added to Local_Extension in the > default dial plan. > > -Eli > > On Feb 1, 2013, at 6:00 AM, royj at yandex.ru wrote: > > > Is there in profile > > > > On Thu, 31 Jan 2013 17:43:34 -0500 > > Eli Burke wrote: > > > >> I'm trying to use Freeswitch with an open source voip client called > Linphone, but cannot manage to establish functioning ZRTP sessions. The > problem *seems* to be related to the RTP stream's SSRC. I'm trying to > conclusively determine if the problem lies with Linphone, Freeswitch, or > one of the underlying zrtp implementations (libzrtp and zrtpcpp). > >> > >> Here's what I've determined so far: > >> 1) ZRTP support is compiled and functioning. I can establish ZRTP > sessions between two iPhones running Groundwire. > >> 2) linphone + ZRTP works correctly if I make a one legged call to a > Freeswitch echo extension > >> 3) linphone + ZRTP works correctly if I turn on bypass-media > >> 4) linphone gets a CRC error when trying to verify the ZRTP packet on > any two-legged call > >> 5) disabling the CRC check results in a successful ZRTP session, but > the audio is extremely garbled > >> > >> Is Freeswitch modifying the ZRTP packets when running in proxy mode? If > so, how is it that some clients are able to handle this situation? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130201/657adf88/attachment.html From steveayre at gmail.com Fri Feb 1 23:38:00 2013 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 1 Feb 2013 20:38:00 +0000 Subject: [Freeswitch-users] Choice of codecs for incoming call In-Reply-To: <510BA47E.2070906@gmail.com> References: <510BA47E.2070906@gmail.com> Message-ID: The client sends the list of codecs it supports in the INVITE (PCMA+PCMU). FS will check check that against the list of codecs it supports, and select just *one* codec that's supported by both. That's returned by FS in the SDP of the 183 (if any) and 200. If there's no codec supported by both sides then you get 488. There are parameters that influence when and which codec is selected, but you end up with only one codec in use for any particular call. In this case it's selected the first supported codec, PCMU. -Steve * * On 1 February 2013 11:18, ????????? ???????? wrote: > Hello. > > My FS receive INVITE with 2 codecs in SDP: PCMA and PCMU. After hunting > dialplan, this call forwarding to IVR, and I see next message in log: > > 2013-01-31 01:23:22.388780 [DEBUG] sofia_glue.c:5094 Audio Codec Compare > [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] > 2013-01-31 01:23:22.388780 [DEBUG] sofia_glue.c:3077 Set Codec > sofia/internal/9089080908 at PSTN PCMU/8000 20 ms 160 samples 64000 bits > 2013-01-31 01:23:22.388780 [DEBUG] switch_core_codec.c:111 > sofia/internal/9089080908 at PSTN Original read codec set to PCMU:0 > > And FS send 200 OK to remote side with one codec in SDP: PCMU. > > My parameters: > > In internal.xml (profile) > > > > In vars.xml > data="global_codec_prefs=PCMU,PCMA,G729,G722,GSM"/> > > > > This is normal behavior?How can I manage set of codec in inbound calls? > > Best regards, Alexander. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130201/b88c5224/attachment.html From steveayre at gmail.com Fri Feb 1 23:41:19 2013 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 1 Feb 2013 20:41:19 +0000 Subject: [Freeswitch-users] Choice of codecs for incoming call In-Reply-To: <510BA47E.2070906@gmail.com> References: <510BA47E.2070906@gmail.com> Message-ID: > > How can I manage set of codec in inbound calls? Your previous configuration used the global list for both parameters. global_codec_prefs isn't actually then used, there's a codec-prefs param for convenience only that sets both inbound and outbound at the same time and isn't actually needed if you're setting both. -Steve On 1 February 2013 11:18, ????????? ???????? wrote: > Hello. > > My FS receive INVITE with 2 codecs in SDP: PCMA and PCMU. After hunting > dialplan, this call forwarding to IVR, and I see next message in log: > > 2013-01-31 01:23:22.388780 [DEBUG] sofia_glue.c:5094 Audio Codec Compare > [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] > 2013-01-31 01:23:22.388780 [DEBUG] sofia_glue.c:3077 Set Codec > sofia/internal/9089080908 at PSTN PCMU/8000 20 ms 160 samples 64000 bits > 2013-01-31 01:23:22.388780 [DEBUG] switch_core_codec.c:111 > sofia/internal/9089080908 at PSTN Original read codec set to PCMU:0 > > And FS send 200 OK to remote side with one codec in SDP: PCMU. > > My parameters: > > In internal.xml (profile) > > > > In vars.xml > data="global_codec_prefs=PCMU,PCMA,G729,G722,GSM"/> > > > > This is normal behavior?How can I manage set of codec in inbound calls? > > Best regards, Alexander. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130201/678ef9ed/attachment.html From benoit.raymond at amiconcept.com Sat Feb 2 00:12:03 2013 From: benoit.raymond at amiconcept.com (Benoit Raymond) Date: Fri, 1 Feb 2013 16:12:03 -0500 Subject: [Freeswitch-users] [CS_NEW] [WRONG_CALL_STATE] with incoming calls In-Reply-To: <510BD603.50303@targointernet.com> References: <51092BBA.8030001@targointernet.com> <510A865C.505@targointernet.com> <06E0038D-469F-47F1-8449-C6B590AAECA0@gmail.com> <510AA553.9060105@targointernet.com> <510BD603.50303@targointernet.com> Message-ID: <0d2901ce00c0$c8bfb1d0$5a3f1570$@amiconcept.com> Fred, How many ethernet interfaces do you have on this host (ifconfig ?a)? If you have multiple interfaces, you may want to hardcode the IP in Freeswitch instead of relying on local_ip_v4 variable as FS may bind to another of your interface. Are you using IPv4 or IPv6? Benoit Raymond AMI Concept Inc. Solutions VoIP Affaire http://www.amiconcept.com De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Frederick Pruneau Envoy? : 1 f?vrier 2013 09:50 ? : freeswitch-users at lists.freeswitch.org Objet : Re: [Freeswitch-users] [CS_NEW] [WRONG_CALL_STATE] with incoming calls I think I have enabled siptrace correctly in my log file. Let me know if i'm wrong because it's the first time I enable this option. It was enable in the console but not in log file. I have hide the first three numbers of my ip address. IP addresses look like this in log file : XXX.XXX.XXX.10. I tried to grep the old IP in config directory but got no result. I'm running the latest release. Thanks! Fred Le 2013-01-31 21:31, Anthony Minessale a ?crit : the best way to change the ip using the default config is to explicitly set local_ip_v4 in vars.xml and leave all the other settings as default. The trace is missing the siptrace which is necessary to diagnose the problem and its better as an attachment so it does not flood the screen. On Thu, Jan 31, 2013 at 11:09 AM, Frederick Pruneau wrote: I changed sip-ip and rtp-ip settings for my ip address. I restarted freeswitch. The problem is still there. My FS server is not behind nat. Thanks for your help! Fred Le 2013-01-31 11:46, Steven Ayre a ?crit : Check the sip-ip rtp-ip settings on your Sofia profiles. They may be using a variable such as local_ip4, if so check that too. Did you restart FS or at least the relevant profiles after changing the IP? Note that if you're not setting it explicitly it may be autodetecting your IP, in which case that might not happen correctly if your server now has multiple IPs. Or using the old one if FS wasn't restarted. Also remember Sofia cannot listen on 0.0.0.0 If your FS is behind NAT then that also may imply a few other settings need checking too (ext-sip-ip etc). Steve On 31 Jan 2013, at 14:57, Frederick Pruneau wrote: Hi Anthony, First of all, thanks for your reply! I appreciate your help! I checked conf files that could have changed since the new ip address has been set. I tried to rollback as far as I can but did not find the option that makes incoming calls not working. I checked also my network settings. Everything seems to be good. I'm probably missing something in freeswitch config files. Is there a file (or files) that I need to modify and put my new IP? Thanks again. Fred Le 2013-01-30 20:07, Anthony Minessale a ?crit : That means when the call was challenged for auth, it never replied with the new INVITE with www-auth header. You may have some network settings wrong making the followup message go to the wrong host. On Wed, Jan 30, 2013 at 8:18 AM, Frederick Pruneau wrote: Hi guys! I have a problem with my freeswitch server and I need some help. All incoming calls are not working since yesterday. All I did is to change the IP address. The only error message I have is this one: [WARNING] switch_core_state_machine.c:514 7b24f8fa-6ae6-11e2-a6be-25229cd2cfbb sofia/internal/XXX at XXX.XXX.XXX.XXX:5060 Abandoned [NOTICE] switch_core_state_machine.c:517 Hangup sofia/internal/XXX at XXX.XXX.XXX.XXX:5060 [CS_NEW] [WRONG_CALL_STATE] What does Wrong call state mean and what can I do to fix this issue? Everything else is working. I can make outgoing calls and call other extensions. Thanks in advance. Fred _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130201/3a39c5cd/attachment-0001.html From msc at freeswitch.org Sat Feb 2 03:25:55 2013 From: msc at freeswitch.org (Michael Collins) Date: Fri, 1 Feb 2013 16:25:55 -0800 Subject: [Freeswitch-users] FreeSWITCH Mirror on Github In-Reply-To: <335647C8-86F0-40BA-8BB6-F21476E71E07@gmail.com> References: <335647C8-86F0-40BA-8BB6-F21476E71E07@gmail.com> Message-ID: It was created by Andrew Thompson (IRC: vagabond) but I think he gave the login info to Ken and me. We'll check it out. -MC On Fri, Feb 1, 2013 at 3:04 AM, Steven Ayre wrote: > Who owns the account? > > 'The World's First Cross-Platform Scalable FREE Multi-Protocol' > > > Perhaps it'd be worth putting a note to that effect in that blurb ^ at the > top of the github page, directing them to the correct place. > > > -Steve > > > > > On 31 Jan 2013, at 21:26, Ken Rice wrote: > > Hey Guy, > > As many of you know there is a Github mirror of the GIT repo at > https://github.com/FreeSWITCH/FreeSWITCH . > > This is intended as a READ ONLY mirror of our master repos. I have noticed > that some people are actually doing push request and opening issues there. > Please do not do that. The Core dev team rarely (if ever) reviews the other > functions that Github provides. If you have an issue or a possible patch, > please open a jira and attach the patch. > > If you have already patched your local copy, using the ?git diff? command > to get the patch will work, just pipe its output to a .txt file and attach > to your Jira. > > Thanks > Ken > > -- > Ken > *http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > *irc.freenode.net #freeswitch > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130201/18ec3dac/attachment.html From mustafin.aleksandr at gmail.com Sat Feb 2 07:47:10 2013 From: mustafin.aleksandr at gmail.com (=?UTF-8?B?0JDQu9C10LrRgdCw0L3QtNGAINCc0YPRgdGC0LDRhNC40L0=?=) Date: Sat, 02 Feb 2013 10:47:10 +0600 Subject: [Freeswitch-users] Choice of codecs for incoming call In-Reply-To: References: <510BA47E.2070906@gmail.com> Message-ID: <510C9A4E.9010202@gmail.com> Hello, Steven. Thank you for reply. Maybe my DID-provider is buggy, but I've succesfull calls in the other pbx with the same number. Short examples below - second example is work. FS (1st test) ----> INVITE (PCMU,PCMA) <---- 100 Trying <---- 200 OK (PCMU) ----> ACK ----> Re-INVITE (PCMA) <---- 100 Trying <---- 488 Not Acceptable Here CG (2nd test) ----> INVITE (PCMU,PCMA) <---- 100 Trying <---- 200 OK (PCMU, PCMA) ----> ACK ----> Re-INVITE (PCMA) <---- 100 Trying <---- 200 OK (PCMA) ----> ACK 02.02.2013 02:41, Steven Ayre ?????: >> How can I manage set of codec in inbound calls? > > > > > > > Your previous configuration used the global list for both parameters. > > global_codec_prefs isn't actually then used, there's a codec-prefs param > for convenience only that sets both inbound and outbound at the same time > and isn't actually needed if you're setting both. > > -Steve > > > > On 1 February 2013 11:18, ????????? ???????? > wrote: > >> Hello. >> >> My FS receive INVITE with 2 codecs in SDP: PCMA and PCMU. After hunting >> dialplan, this call forwarding to IVR, and I see next message in log: >> >> 2013-01-31 01:23:22.388780 [DEBUG] sofia_glue.c:5094 Audio Codec Compare >> [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] >> 2013-01-31 01:23:22.388780 [DEBUG] sofia_glue.c:3077 Set Codec >> sofia/internal/9089080908 at PSTN PCMU/8000 20 ms 160 samples 64000 bits >> 2013-01-31 01:23:22.388780 [DEBUG] switch_core_codec.c:111 >> sofia/internal/9089080908 at PSTN Original read codec set to PCMU:0 >> >> And FS send 200 OK to remote side with one codec in SDP: PCMU. >> >> My parameters: >> >> In internal.xml (profile) >> >> >> >> In vars.xml >> > data="global_codec_prefs=PCMU,PCMA,G729,G722,GSM"/> >> >> >> >> This is normal behavior?How can I manage set of codec in inbound calls? >> >> Best regards, Alexander. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130202/8c01dbf8/attachment.html From brian at freeswitch.org Sun Feb 3 00:01:55 2013 From: brian at freeswitch.org (Brian West) Date: Sat, 2 Feb 2013 15:01:55 -0600 Subject: [Freeswitch-users] Choice of codecs for incoming call In-Reply-To: <6700609.14174.1359789999484.JavaMail.mobile-sync@oekv10> References: <510BA47E.2070906@gmail.com> <6700609.14174.1359789999484.JavaMail.mobile-sync@oekv10> Message-ID: <-3405763812603204302@unknownmsgid> You can't reinvite our 200ok has only pcmu so yes it will 488 Sent from my iPhone On Feb 1, 2013, at 10:56 PM, "????????? ????????" < mustafin.aleksandr at gmail.com> wrote: Hello, Steven. Thank you for reply. Maybe my DID-provider is buggy, but I've succesfull calls in the other pbx with the same number. Short examples below - second example is work. FS (1st test) ----> INVITE (PCMU,PCMA) <---- 100 Trying <---- 200 OK (PCMU) ----> ACK ----> Re-INVITE (PCMA) <---- 100 Trying <---- 488 Not Acceptable Here CG (2nd test) ----> INVITE (PCMU,PCMA) <---- 100 Trying <---- 200 OK (PCMU, PCMA) ----> ACK ----> Re-INVITE (PCMA) <---- 100 Trying <---- 200 OK (PCMA) ----> ACK 02.02.2013 02:41, Steven Ayre ?????: How can I manage set of codec in inbound calls? Your previous configuration used the global list for both parameters. global_codec_prefs isn't actually then used, there's a codec-prefs param for convenience only that sets both inbound and outbound at the same time and isn't actually needed if you're setting both. -Steve On 1 February 2013 11:18, ????????? ???????? wrote: Hello. My FS receive INVITE with 2 codecs in SDP: PCMA and PCMU. After hunting dialplan, this call forwarding to IVR, and I see next message in log: 2013-01-31 01:23:22.388780 [DEBUG] sofia_glue.c:5094 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2013-01-31 01:23:22.388780 [DEBUG] sofia_glue.c:3077 Set Codec sofia/internal/9089080908 at PSTN PCMU/8000 20 ms 160 samples 64000 bits 2013-01-31 01:23:22.388780 [DEBUG] switch_core_codec.c:111 sofia/internal/9089080908 at PSTN Original read codec set to PCMU:0 And FS send 200 OK to remote side with one codec in SDP: PCMU. My parameters: In internal.xml (profile) In vars.xml This is normal behavior?How can I manage set of codec in inbound calls? Best regards, Alexander. _________________________________________________________________________ Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130202/ad863b03/attachment-0001.html From peter.kosztolanyi at gmail.com Sat Feb 2 21:08:33 2013 From: peter.kosztolanyi at gmail.com (=?ISO-8859-1?Q?Kosztol=E1nyi_P=E9ter?=) Date: Sat, 2 Feb 2013 19:08:33 +0100 Subject: [Freeswitch-users] problems with freeswitch + zrtp in proxy-media mode In-Reply-To: References: <2DBB65A0-2551-4FE8-88E0-CB4624325733@edge-net.net> Message-ID: Hi, I have exactly the same problem as Eli described below. Eli, I'm glad too that finally not I'm the only guy in this situation. In my case the proxy media mode is required for end-to-end security but bypass-media is not an option because the endpoints are in different subnets. The normal media handling mode (not proxy and not bypass) works well however in this case there are different SAS on the two endpoints. Probably it's normal because FS opens a new session and the first SAS is not transferred to the other side automatically. But what I'm more interested in what "inbound-zrtp-passthru" is actually doing? I couldn't find proper docs only the codebase at http://fisheye.freeswitch.org/changelog/freeswitch.git?cs=23f8967c32e63050af93b43c66ec94e7b404b063 I can see two entries in my logs relating to this: 2013-02-02 18:26:14.925953 [DEBUG] mod_sofia.c:5031 [zrtp_passthru] Setting a-leg inherit_codec=true 2013-02-02 18:26:14.925953 [DEBUG] mod_sofia.c:5034 [zrtp_passthru] Setting b-leg absolute_codec_string='PCMU at 8000h@20i at 64000b' Meanwhile someone on the linphone dev mailing list said that linphone does not support "zrtp-hash" attribute in SIP/SDP. May it has any effect in FS media proxy with zrtp? -Peter On Fri, Feb 1, 2013 at 9:33 PM, Steven Ayre wrote: > Is there a reason you're using proxy media mode? > > Does it work in normal media handling mode? (ie not proxy and not bypass) > > -Steve > > > > On 1 February 2013 14:15, Eli Burke wrote: > >> Yes, inbound-zrtp-passthru is true. I think it gets set automatically >> when you turn on ZRTP. >> >> The config I'm testing with is a stock 'enable_zrtp=yes' build of HEAD >> circa last week, with proxy-media=true added to Local_Extension in the >> default dial plan. >> >> -Eli >> >> On Feb 1, 2013, at 6:00 AM, royj at yandex.ru wrote: >> >> > Is there in profile >> > >> > On Thu, 31 Jan 2013 17:43:34 -0500 >> > Eli Burke wrote: >> > >> >> I'm trying to use Freeswitch with an open source voip client called >> Linphone, but cannot manage to establish functioning ZRTP sessions. The >> problem *seems* to be related to the RTP stream's SSRC. I'm trying to >> conclusively determine if the problem lies with Linphone, Freeswitch, or >> one of the underlying zrtp implementations (libzrtp and zrtpcpp). >> >> >> >> Here's what I've determined so far: >> >> 1) ZRTP support is compiled and functioning. I can establish ZRTP >> sessions between two iPhones running Groundwire. >> >> 2) linphone + ZRTP works correctly if I make a one legged call to a >> Freeswitch echo extension >> >> 3) linphone + ZRTP works correctly if I turn on bypass-media >> >> 4) linphone gets a CRC error when trying to verify the ZRTP packet on >> any two-legged call >> >> 5) disabling the CRC check results in a successful ZRTP session, but >> the audio is extremely garbled >> >> >> >> Is Freeswitch modifying the ZRTP packets when running in proxy mode? >> If so, how is it that some clients are able to handle this situation? >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130202/9ddab68a/attachment.html From moises.silva at gmail.com Sun Feb 3 08:23:16 2013 From: moises.silva at gmail.com (Moises Silva) Date: Sun, 3 Feb 2013 00:23:16 -0500 Subject: [Freeswitch-users] Dequeue DTMF from FreeTDM Channel In-Reply-To: References: Message-ID: > > >> On Tue, Jan 22, 2013 at 6:19 PM, Subhash wrote: >> >>> Ok. >>> We did the test again,pasted the freeswitch log and uploaded the >>> wireshark of the both success and failed call traces.Please let me know if >>> you have problem in accessing the URLs . >>> >>> http://pastebin.freeswitch.org/20484 >>> >>> http://www.cloudshark.org/captures/6dc99dc2f68c -- Success >>> >>> http://www.cloudshark.org/captures/27b88b2c8ca4--- Failure >>> >> I don't see anything wrong in the traces. First call (the one you flagged as success), is an incoming call on the ISDN link, almost immediately after the call is connected to the SIP end, FreeSWITCH receives 9199 from the RTP stream, and 71 from the TDM end. Those DTMF digits are passed from RTP to TDM and viceversa as expected. The call is then finally hung up Second call (the one you flagged as failed) is pretty much the same, the DTMF 9 that you see is in the RTP wireshark trace, so there is nothing wrong here. That DTMF is there. The other end is sending that 9, the only difference is that this time the far end of the TDM side hangs up the call almost immediately after re-generating that 9 received in the RTP side into the TDM side. *Moises Silva **Manager, Software Engineering*** msilva at sangoma.com Sangoma Technologies 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada t. +1 800 388 2475 (N. America) t. +1 905 474 1990 x128 f. +1 905 474 9223 ** Products | Solutions | Events | Contact | Wiki | Facebook | Twitter`| | YouTube -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130203/392721a7/attachment.html From vbvbrj at gmail.com Sun Feb 3 21:24:16 2013 From: vbvbrj at gmail.com (Mimiko) Date: Sun, 03 Feb 2013 20:24:16 +0200 Subject: [Freeswitch-users] Get recorded file from callcenter via API. Message-ID: <510EAB50.80308@gmail.com> Hello. How to retrieve a recorded audio file by callcenter using API call to FS? Or by other meaning except using direct file access. -- Mimiko desu. From POlsson at enghouse.com Sun Feb 3 22:49:02 2013 From: POlsson at enghouse.com (Peter Olsson) Date: Sun, 3 Feb 2013 19:49:02 +0000 Subject: [Freeswitch-users] Get recorded file from callcenter via API. In-Reply-To: <510EAB50.80308@gmail.com> References: <510EAB50.80308@gmail.com> Message-ID: <1FFF97C269757C458224B7C895F35F15215DC1@cantor.std.visionutv.se> You could use an API call to make an outgoing call to a telephone, and then play the file. If you need to get the actual file, you wil need do make something on your own (as far as I know), if you don't want to do direct file access. /Peter ________________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för Mimiko [vbvbrj at gmail.com] Skickat: den 3 februari 2013 19:24 Till: FreeSWITCH Users Help ?mne: [Freeswitch-users] Get recorded file from callcenter via API. Hello. How to retrieve a recorded audio file by callcenter using API call to FS? Or by other meaning except using direct file access. -- Mimiko desu. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 !DSPAM:510ea8bd32766001739750! From drk at drkngs.net Sun Feb 3 23:49:55 2013 From: drk at drkngs.net (Dave R. Kompel) Date: Sun, 03 Feb 2013 12:49:55 -0800 Subject: [Freeswitch-users] ESL Client Library Managed wrappers (Windows) Message-ID: <20130203204955.8db66002@mail.tritonwest.net> I just spent the good part of this wekend trying to get them compilied, tested and working, it's been over 2 years since I had to build them, and it's a bitch. To save ppl time, I put the binaries up, and made them available if anyone needs them. You can grab them at: http://download.drknetworking.com/eslmanaged.zip --Dave -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130203/4e08bd06/attachment.html From david.villasmil.work at gmail.com Mon Feb 4 04:54:24 2013 From: david.villasmil.work at gmail.com (David Villasmil) Date: Mon, 4 Feb 2013 02:54:24 +0100 Subject: [Freeswitch-users] ESL outbound race condition? Message-ID: Hello guys, I've writing an app using outbound socket, when the channel is created from FS to my app, I get an event called "SOCKET_DATA", I was expecting a CHANNEL_CALLSTATE or similar. I read somewhere that it is related to a race condition? I'd love some help! Thanks! David -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130204/c5389b79/attachment.html From gassaad at emassembly.com Mon Feb 4 07:30:14 2013 From: gassaad at emassembly.com (George Assaad) Date: Sun, 3 Feb 2013 23:30:14 -0500 Subject: [Freeswitch-users] spandsp_start_fax_detect how to use? Message-ID: Hello, I have one DID that I would like to detect the tone 1- if fax goes to extension 1111. 2- If it is is not fax then t goes to extension 2222 How do that using spandsp_start_fax_detect ? I am looking for the Inbound dialplan Thanks in advance, George -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130203/2bfc5a39/attachment.html From POlsson at enghouse.com Mon Feb 4 08:46:44 2013 From: POlsson at enghouse.com (Peter Olsson) Date: Mon, 4 Feb 2013 05:46:44 +0000 Subject: [Freeswitch-users] ESL outbound race condition? In-Reply-To: References: Message-ID: What version of FS are you running? This sounds like an old bug to me. /Peter 4 feb 2013 kl. 03:06 skrev "David Villasmil" : > Hello guys, > > I've writing an app using outbound socket, when the channel is created from FS to my app, I get an event called "SOCKET_DATA", I was expecting a CHANNEL_CALLSTATE or similar. > > I read somewhere that it is related to a race condition? > > I'd love some help! > > Thanks! > > David > !DSPAM:510f134332762069512667! > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > !DSPAM:510f134332762069512667! From vbvbrj at gmail.com Mon Feb 4 09:01:17 2013 From: vbvbrj at gmail.com (Mimiko) Date: Mon, 04 Feb 2013 08:01:17 +0200 Subject: [Freeswitch-users] Get recorded file from callcenter via API. In-Reply-To: <1FFF97C269757C458224B7C895F35F15215DC1@cantor.std.visionutv.se> References: <510EAB50.80308@gmail.com> <1FFF97C269757C458224B7C895F35F15215DC1@cantor.std.visionutv.se> Message-ID: <510F4EAD.3040807@gmail.com> On 03.02.2013 21:49, Peter Olsson wrote: > You could use an API call to make an outgoing call to a telephone, and then play the file. > > If you need to get the actual file, you wil need do make something on your own (as far as I know), if you don't want to do direct file access. > > /Peter Oh, may be starting a lua script from API CLI to get that file. :) But I am not usre if this is possible. -- Mimiko desu. From POlsson at enghouse.com Mon Feb 4 09:49:29 2013 From: POlsson at enghouse.com (Peter Olsson) Date: Mon, 4 Feb 2013 06:49:29 +0000 Subject: [Freeswitch-users] Get recorded file from callcenter via API. In-Reply-To: <510F4EAD.3040807@gmail.com> References: <510EAB50.80308@gmail.com> <1FFF97C269757C458224B7C895F35F15215DC1@cantor.std.visionutv.se>, <510F4EAD.3040807@gmail.com> Message-ID: <2D54617F-C018-491F-B19F-9FAA02D5412D@visionutveckling.se> You need to be more specific... What exactly do you want to do? Play the file as audio inside a call, or grab the actual file from server? /Peter 4 feb 2013 kl. 07:06 skrev "Mimiko" : > On 03.02.2013 21:49, Peter Olsson wrote: >> You could use an API call to make an outgoing call to a telephone, and then play the file. >> >> If you need to get the actual file, you wil need do make something on your own (as far as I know), if you don't want to do direct file access. >> >> /Peter > > Oh, may be starting a lua script from API CLI to get that file. :) > > But I am not usre if this is possible. > > -- > Mimiko desu. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > !DSPAM:510f4b4f32762028220902! > From vbvbrj at gmail.com Mon Feb 4 10:35:25 2013 From: vbvbrj at gmail.com (Mimiko) Date: Mon, 04 Feb 2013 09:35:25 +0200 Subject: [Freeswitch-users] Get recorded file from callcenter via API. In-Reply-To: <2D54617F-C018-491F-B19F-9FAA02D5412D@visionutveckling.se> References: <510EAB50.80308@gmail.com> <1FFF97C269757C458224B7C895F35F15215DC1@cantor.std.visionutv.se>, <510F4EAD.3040807@gmail.com> <2D54617F-C018-491F-B19F-9FAA02D5412D@visionutveckling.se> Message-ID: <510F64BD.6070107@gmail.com> On 04.02.2013 08:49, Peter Olsson wrote: > You need to be more specific... What exactly do you want to do? Play the file as audio inside a call, or grab the actual file from server? Get the file from the recordings directory storage to play inline on a web page for listening what was recorded. For now I use direct file access from PHP sending file to client. But this breaks the security to record storage directory where both, httpd' user and FS' user have access to same directory. -- Mimiko desu. From fdelawarde at wirelessmundi.com Mon Feb 4 13:54:07 2013 From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde) Date: Mon, 04 Feb 2013 11:54:07 +0100 Subject: [Freeswitch-users] DTMF duration unit Message-ID: <1359975247.8347.78.camel@luna.madrid.commsmundi.com> Hi all, I don't clearly get what unit we should use in dtmf-duration parameters (core min-dtmf-duration / max-dtmf-duration and sofia dtmf-duration). The variable name and some parts of the code/wiki imply it's in milliseconds (for example in sofia.c, we can see a line saying profile->dtmf_duration = 100;). However, the default values in switch_types.h suggest the values should be based on timestamp like in RTP. Something like: minimum = 400 @ 8kHz => 50ms default = 2000 @ 8kHz => 250ms maximum = 192000 @ 8kHz => 24s If the later is the right one, should I change values when using WB codecs that are NOT sampling at 8kHz? Also, is the "unit" consistent across all endpoint modules? Thanks, Fran?ois. From rostotskiy.a at gmail.com Mon Feb 4 14:42:46 2013 From: rostotskiy.a at gmail.com (Anton Rostotskiy) Date: Mon, 4 Feb 2013 13:42:46 +0200 Subject: [Freeswitch-users] building mod_opal and v1.2stable In-Reply-To: <009201cdfccc$5af22900$10d67b00$@center.com> References: <000701cdf4f4$cee11ef0$6ca35cd0$@center.com> <4A8B8CFC-2B43-45F1-AD6E-99ADA1E9DC75@gmail.com> <004101cdfa60$25b1b4e0$71151ea0$@center.com> <009201cdfccc$5af22900$10d67b00$@center.com> Message-ID: Hi Gilbert, I built mod_opal on CentOS 6.3 x86_64 and FS 1.2 stable after all, using opal and ptlib of 3.13.0 and 2.13.0 versions, but have the same problem as you described. 2013/1/27 Gilbert Amar > Hello,**** > > ** ** > > I did fill two Jira bugs, one for mod_323 and another for mod_opal.**** > > Just wanted to remind that since those two do not work, at least for me, > there is no more any support in FS 1.2 stable for H323.**** > > ** ** > > So I ask again did someone succeeded in having any of those 2 working > with 1.2 stable on a Linux platform ?**** > > ** ** > > Gilbert**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael > Collins > *Sent:* Friday, January 25, 2013 12:28 PM > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] building mod_opal and v1.2stable**** > > ** ** > > Gilbert, > > Thank you for collecting all of this information. You did quite a bit of > work and that will help the developers figure out what's going on. Would > you mind creating a ticket at jira.freeswitch.org and adding all this > information? That will help the developers keep track of the issue and > avoid any unnecessary delays. > > Thanks! > -MC**** > > On Thu, Jan 24, 2013 at 10:25 AM, Gilbert Amar wrote:** > ** > > Hello,**** > > **** > > I succeeded in building mod_opal, but it crash FreeSWITCH (segmentation > default) after simple test.**** > > **** > > Here are the steps on Debian "squeeze".**** > > After getting all required packages including swig**** > > **** > > Advice, check that you do not have any version of ptlib or h323plus or > opal already somewhere.**** > > use find /- name "*libpt*.so"**** > > use find /- name "*libopal*.so"**** > > **** > > **** > > **** > > My FreeSWITCH is in /usr/local/src/ and I am root when I build.**** > > **** > > cd /usr/local/src/**** > > svn co > https://opalvoip.svn.sourceforge.net/svnroot/opalvoip/ptlib/tags/v2_12_0/ptlib-2.12.0 > **** > > **** > > cd /usr/local/src/ptlib-2.12.0**** > > ./configure --disable-plugins --disable-v4l2 --prefix=/opt/lib**** > > make && make install**** > > **** > > export PKG_CONFIG_PATH=/opt/lib/lib/pkgconfig **** > > **** > > **** > > cd /usr/local/src/**** > > svn co > https://opalvoip.svn.sourceforge.net/svnroot/opalvoip/opal/tags/v3_12_0opal-3.12.0 > **** > > **** > > cd /usr/local/src/opal-3.12.0**** > > ./configure --disable-plugins --prefix=/opt/lib**** > > make && make install**** > > **** > > **** > > As the install fails to copy include/ep/**** > > **** > > cd /usr/local/src/opal-3.12.0**** > > mkdir -p /opt/lib/include/opal/ep ; chmod 755 /opt/lib/include/opal/ep ; ( > for fn in include/ep/*.h ; do /usr/bin/install -c -m 644 $fn > /opt/lib/include/opal/ep ; done);**** > > **** > > Please notice that the mkdir line is one long line.**** > > **** > > Add mod_opal to your modules.conf**** > > **** > > Go to your mod_opal src dir in my case **** > > cd /usr/local/src/freeswitch/src/mod/endpoints/mod_opal**** > > **** > > Edit mod_opal.h**** > > You need to change 2 lines**** > > **** > > #include **** > > to**** > > #include **** > > **** > > then **** > > #define HAVE_T38 (OPAL_CHECK_VERSION(3,11,2) && OPAL_T38_CAPABILITY)**** > > to**** > > #define HAVE_T38 0**** > > **** > > **** > > Now you can build mod_opal type make **** > > Go back to your FS dir and do make and make install to check all is ok.*** > * > > Before launch FreeSWITCH you need to set LD_LIBRARY_PATH**** > > **** > > export LD_LIBRARY_PATH=/opt/lib/lib**** > > start your FreeSWITCH **** > > **** > > Then try to call extension 9197 it should work despite the errors lines > like **** > > 2013-01-24 03:12:25.597058 [WARNING] switch_core_codec.c:802 Codec is not > initialized!**** > > 2013-01-24 03:12:25.597058 [WARNING] switch_core_codec.c:802 Codec is not > initialized!**** > > 2013-01-24 03:12:25.597058 [ERR] switch_core_timer.c:117 Timer is not > properly configured.**** > > **** > > In my case I have a sip phone registered to FS at extension 1000**** > > If I call from openphone this extension**** > > the call get thru, I pick up but mod_opal isn't aware of this and when I > hang up on the sip phone FreeSWITCH crash.**** > > **** > > **** > > Segmentation fault /opt/freeswitch-v1.2stable/bin/freeswitch -conf > /home/center/freeswitch/conf -log /home/center/freeswitch/log -db > /home/center/freeswitch/db -sounds /home/center/freeswitch/sounds -nonat > -nonatmap**** > > **** > > **** > > Gilbert**** > > **** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Steven Ayre > *Sent:* Wednesday, January 23, 2013 3:26 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] building mod_opal and v1.2stable**** > > **** > > Do you mean you built opal in /root/opal? You shouldn't build things as > root, but that's not the reason for your troubles.**** > > **** > > I have to say I'm a bit puzzled, it looks like you've installed opal to > multiple locations.**** > > **** > > /usr/local/freeswitch/opal looks the best match (opal/localep.h) but it's > in the wrong location.**** > > **** > > /usr/local/src/freeswitch/libs/opal/include/ep/localep.h**** > > Not sure why it's installed here, opal isn't shipped under libs/**** > > **** > > /usr/local/include/ep/localep.h**** > > /usr/include/ep/localep.h**** > > Close but the directory is wrong as it needs to be opal/localep.h, not > ep/localep.h**** > > **** > > /usr/include/opal/localep.h**** > > Should be the correct filename, but you don't have a copy.**** > > **** > > It could be a change in the opal library but I find that unlikely since > it'd break apps using the library like FS. More likely there's a problem > with the prefixes given to opal's configure command.**** > > **** > > -Steve**** > > **** > > **** > > **** > > On 23 January 2013 10:46, Anton Rostotskiy wrote: > **** > > Search results:**** > > **** > > /usr/local/freeswitch/opal/include/opal/localep.h**** > > /usr/local/src/freeswitch/libs/opal/include/ep/localep.h**** > > /usr/local/include/ep/localep.h**** > > /usr/include/ep/localep.h**** > > /root/opal/include/ep/localep.h**** > > **** > > As I said earlier, I installed ptlib and opal in /root directory.**** > > **** > > 2013/1/23 Steven Ayre **** > > It's localep.h, not local.h:**** > > /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.h:41:26: > error: opal/localep.h: No such file or directory**** > > **** > > find / -name localep.h**** > > **** > > Sorry, seems my phone did an annoying autouncorrection!**** > > **** > > -Steve**** > > **** > > **** > > **** > > On 23 January 2013 08:16, Anton Rostotskiy wrote: > **** > > Thanks for reply, Steven!**** > > Tried to search local.h file, but search returned nothing. That are all > local.h files in my system:**** > > **** > > /usr/local/src/freeswitch/libs/spandsp/src/gsm0610_local.h**** > > /usr/local/src/freeswitch/libs/spandsp/src/t30_local.h**** > > /usr/local/src/freeswitch/libs/spandsp/src/t42_t43_local.h**** > > **** > > About -devel packages - at first tried to install ptlib, opal and > -devel from yum but their versions have been too low for mod_opal.**** > > **** > > 2013/1/22 Steven Ayre **** > > Run:**** > > find / -name local.h**** > > **** > > It's failing to include opal/local.h. All the other errors are because its > missing the definitions in that file.**** > > **** > > Normally I would say to instal the -devel package, but since you built > from svn that doesn't apply. Perhaps it's not installed or installed to the > wrong place. The find command will tell us where that file is.**** > > > Steve on iPhone**** > > > On 22 Jan 2013, at 16:07, Anton Rostotskiy wrote: > **** > > Also tried to install mod_opal and also failed.**** > > CentOS 6.3 x86_64**** > > ptlib and opal are the last version obtained from svn as it is said in > wiki .**** > > but just a bunch of errors after make mod_opal > http://pastebin.com/HHUPLCLV**** > > **** > > 2013/1/18 Steven Ayre **** > > The latest version of opal in Debian (even in Sid) isn't new enough for > mod_opal. It needed various features added that are only only available in > the svn version.**** > > **** > > The wiki page states you have to install ptlib and opal from svn - this is > why.**** > > http://wiki.freeswitch.org/wiki/Mod_opal > > Of course that can then cause conflicts unless you package it yourself or > install a dummy package, or problems with other libraries/apps that are > built against the libopal that's already packaged.**** > > **** > > Ptlib and h323plus/opal are also notorious for having to get matching > versions. If you don't pair exactly the right version of ptlib with the > specific version of h323plus/opal you can get instability problems. So the > official Debian ptlib package probably wouldn't be good with opal from > svn...**** > > **** > > As for mod_h323, it does compile and work, at least on Wheezy and possibly > Squeeze too. I've never had any luck getting mod_opal working but have got > mod_h323 working. I don't actively use it though - all my traffic is > currently SIP.**** > > **** > > Another option is to run Yate as a SIP->H323 proxy. It only handles the > signalling, media still goes directly caller->FS. It worked quite well, > although it would occasionally crash - but at least that only dropped the > H323 calls and didn't take FS down with it, and Yate automatically > restarted. That was on Lenny so a long time ago...**** > > > Steve on iPhone**** > > **** > > **** > > > On 17 Jan 2013, at 20:54, "Gilbert Amar" wrote:**** > > Hi, > > I am trying to build mod_opal in FS v1.2stable, using ptlib 2.10.7 and opal > 3.10.7 on a debian > Ptlib and opal get build with no error, but when trying mod_opal I get > > Compiling > /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp... > quiet_libtool: compile: g++ -g -ggdb > -I/usr/local/src/freeswitch/libs/curl/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/libs/libteletone/src > -I/usr/local/src/freeswitch/libs/stfu -fPIC -fvisibility=hidden > -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE > -DP_64BIT > -DPTRACING=1 -D_REENTRANT -D_GNU_SOURCE=1 -fno-exceptions > -I/usr/local/include -I/usr/local/include/opal -I/usr/include/SDL > -DHAVE_CONFIG_H -c > /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp -fPIC > -DPIC -o .libs/mod_opal.o > /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp: In > member > function 'switch_status_t FSConnection::read_frame(const OpalMediaType&, > switch_frame_t**, switch_io_flag_t)': > /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp:1208: > error: 'class OpalCall' has no member named 'IsSwitchingT38' > /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp: In > member > function 'switch_status_t FSConnection::write_frame(const OpalMediaType&, > const switch_frame_t*, switch_io_flag_t)': > /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp:1225: > error: 'class OpalCall' has no member named 'IsSwitchingT38' > make[3]: *** [mod_opal.lo] Error 1 > make[2]: *** [install] Error 1 > make[1]: *** [mod_opal-install] Error 1 > make: *** [mod_opal-install] Error 2 > > > Has anyone succeded in building mod_opal ? > > Should I go with mod_h323 instead ? > > Thanks for your help > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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**** > > > > **** > > **** > > -- > ? ?????????, ????? ????????? **** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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**** > > > > **** > > **** > > -- > ? ?????????, ????? ????????? **** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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**** > > **** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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**** > > > > **** > > **** > > -- > ? ?????????, ????? ????????? **** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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**** > > **** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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**** > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- ? 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URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130204/10cdaf1e/attachment-0001.html From avi at avimarcus.net Mon Feb 4 14:57:39 2013 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 4 Feb 2013 13:57:39 +0200 Subject: [Freeswitch-users] SPA-2102 Gives 486 instead of Call Waiting In-Reply-To: <4522496924514533582@unknownmsgid> References: <3010184978485284118@unknownmsgid> <4522496924514533582@unknownmsgid> Message-ID: It's on debug and syslog, level 3 and "full" on line 1. I seemed to only get this when calling line 1 while it was on the phone. http://pastebin.freeswitch.org/20547 Should I have gotten more? -Avi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130204/2a9d2bac/attachment.html From steveayre at gmail.com Mon Feb 4 15:55:24 2013 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 4 Feb 2013 12:55:24 +0000 Subject: [Freeswitch-users] building mod_opal and v1.2stable In-Reply-To: References: <000701cdf4f4$cee11ef0$6ca35cd0$@center.com> <4A8B8CFC-2B43-45F1-AD6E-99ADA1E9DC75@gmail.com> <004101cdfa60$25b1b4e0$71151ea0$@center.com> <009201cdfccc$5af22900$10d67b00$@center.com> Message-ID: I found this on the opal mailing list archive: The "endpoint" files (local, IVR, PCSS etc) were moved to an ep directory. It is not "one level higher". So if you had #include it is now #include So it looks like this is an actual change in the opal library and mod_opal will need to be patched. I've added that info to the Jira ticket http://jira.freeswitch.org/browse/FS-5054. I think that part may have been easy to overlook as the Jira title says it builds ok and the header issue is buried within the description of the instability problems. Possibly it should've had its own Jira separate to the crash. -Steve On 4 February 2013 11:42, Anton Rostotskiy wrote: > Hi Gilbert, > I built mod_opal on CentOS 6.3 x86_64 and FS 1.2 stable after all, using > opal and ptlib of 3.13.0 and 2.13.0 versions, but have the same problem as > you described. > > > 2013/1/27 Gilbert Amar > >> Hello,**** >> >> ** ** >> >> I did fill two Jira bugs, one for mod_323 and another for mod_opal.**** >> >> Just wanted to remind that since those two do not work, at least for me, >> there is no more any support in FS 1.2 stable for H323.**** >> >> ** ** >> >> So I ask again did someone succeeded in having any of those 2 working >> with 1.2 stable on a Linux platform ?**** >> >> ** ** >> >> Gilbert**** >> >> ** ** >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael >> Collins >> *Sent:* Friday, January 25, 2013 12:28 PM >> >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] building mod_opal and v1.2stable**** >> >> ** ** >> >> Gilbert, >> >> Thank you for collecting all of this information. You did quite a bit of >> work and that will help the developers figure out what's going on. Would >> you mind creating a ticket at jira.freeswitch.org and adding all this >> information? That will help the developers keep track of the issue and >> avoid any unnecessary delays. >> >> Thanks! >> -MC**** >> >> On Thu, Jan 24, 2013 at 10:25 AM, Gilbert Amar wrote:* >> *** >> >> Hello,**** >> >> **** >> >> I succeeded in building mod_opal, but it crash FreeSWITCH (segmentation >> default) after simple test.**** >> >> **** >> >> Here are the steps on Debian "squeeze".**** >> >> After getting all required packages including swig**** >> >> **** >> >> Advice, check that you do not have any version of ptlib or h323plus or >> opal already somewhere.**** >> >> use find /- name "*libpt*.so"**** >> >> use find /- name "*libopal*.so"**** >> >> **** >> >> **** >> >> **** >> >> My FreeSWITCH is in /usr/local/src/ and I am root when I build.**** >> >> **** >> >> cd /usr/local/src/**** >> >> svn co >> https://opalvoip.svn.sourceforge.net/svnroot/opalvoip/ptlib/tags/v2_12_0/ptlib-2.12.0 >> **** >> >> **** >> >> cd /usr/local/src/ptlib-2.12.0**** >> >> ./configure --disable-plugins --disable-v4l2 --prefix=/opt/lib**** >> >> make && make install**** >> >> **** >> >> export PKG_CONFIG_PATH=/opt/lib/lib/pkgconfig **** >> >> **** >> >> **** >> >> cd /usr/local/src/**** >> >> svn co >> https://opalvoip.svn.sourceforge.net/svnroot/opalvoip/opal/tags/v3_12_0opal-3.12.0 >> **** >> >> **** >> >> cd /usr/local/src/opal-3.12.0**** >> >> ./configure --disable-plugins --prefix=/opt/lib**** >> >> make && make install**** >> >> **** >> >> **** >> >> As the install fails to copy include/ep/**** >> >> **** >> >> cd /usr/local/src/opal-3.12.0**** >> >> mkdir -p /opt/lib/include/opal/ep ; chmod 755 /opt/lib/include/opal/ep ; >> ( for fn in include/ep/*.h ; do /usr/bin/install -c -m 644 $fn >> /opt/lib/include/opal/ep ; done);**** >> >> **** >> >> Please notice that the mkdir line is one long line.**** >> >> **** >> >> Add mod_opal to your modules.conf**** >> >> **** >> >> Go to your mod_opal src dir in my case **** >> >> cd /usr/local/src/freeswitch/src/mod/endpoints/mod_opal**** >> >> **** >> >> Edit mod_opal.h**** >> >> You need to change 2 lines**** >> >> **** >> >> #include **** >> >> to**** >> >> #include **** >> >> **** >> >> then **** >> >> #define HAVE_T38 (OPAL_CHECK_VERSION(3,11,2) && OPAL_T38_CAPABILITY)**** >> >> to**** >> >> #define HAVE_T38 0**** >> >> **** >> >> **** >> >> Now you can build mod_opal type make **** >> >> Go back to your FS dir and do make and make install to check all is ok.** >> ** >> >> Before launch FreeSWITCH you need to set LD_LIBRARY_PATH**** >> >> **** >> >> export LD_LIBRARY_PATH=/opt/lib/lib**** >> >> start your FreeSWITCH **** >> >> **** >> >> Then try to call extension 9197 it should work despite the errors lines >> like **** >> >> 2013-01-24 03:12:25.597058 [WARNING] switch_core_codec.c:802 Codec is not >> initialized!**** >> >> 2013-01-24 03:12:25.597058 [WARNING] switch_core_codec.c:802 Codec is not >> initialized!**** >> >> 2013-01-24 03:12:25.597058 [ERR] switch_core_timer.c:117 Timer is not >> properly configured.**** >> >> **** >> >> In my case I have a sip phone registered to FS at extension 1000**** >> >> If I call from openphone this extension**** >> >> the call get thru, I pick up but mod_opal isn't aware of this and when I >> hang up on the sip phone FreeSWITCH crash.**** >> >> **** >> >> **** >> >> Segmentation fault /opt/freeswitch-v1.2stable/bin/freeswitch -conf >> /home/center/freeswitch/conf -log /home/center/freeswitch/log -db >> /home/center/freeswitch/db -sounds /home/center/freeswitch/sounds -nonat >> -nonatmap**** >> >> **** >> >> **** >> >> Gilbert**** >> >> **** >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Steven Ayre >> *Sent:* Wednesday, January 23, 2013 3:26 AM >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] building mod_opal and v1.2stable**** >> >> **** >> >> Do you mean you built opal in /root/opal? You shouldn't build things as >> root, but that's not the reason for your troubles.**** >> >> **** >> >> I have to say I'm a bit puzzled, it looks like you've installed opal to >> multiple locations.**** >> >> **** >> >> /usr/local/freeswitch/opal looks the best match (opal/localep.h) but it's >> in the wrong location.**** >> >> **** >> >> /usr/local/src/freeswitch/libs/opal/include/ep/localep.h**** >> >> Not sure why it's installed here, opal isn't shipped under libs/**** >> >> **** >> >> /usr/local/include/ep/localep.h**** >> >> /usr/include/ep/localep.h**** >> >> Close but the directory is wrong as it needs to be opal/localep.h, not >> ep/localep.h**** >> >> **** >> >> /usr/include/opal/localep.h**** >> >> Should be the correct filename, but you don't have a copy.**** >> >> **** >> >> It could be a change in the opal library but I find that unlikely since >> it'd break apps using the library like FS. More likely there's a problem >> with the prefixes given to opal's configure command.**** >> >> **** >> >> -Steve**** >> >> **** >> >> **** >> >> **** >> >> On 23 January 2013 10:46, Anton Rostotskiy >> wrote:**** >> >> Search results:**** >> >> **** >> >> /usr/local/freeswitch/opal/include/opal/localep.h**** >> >> /usr/local/src/freeswitch/libs/opal/include/ep/localep.h**** >> >> /usr/local/include/ep/localep.h**** >> >> /usr/include/ep/localep.h**** >> >> /root/opal/include/ep/localep.h**** >> >> **** >> >> As I said earlier, I installed ptlib and opal in /root directory.**** >> >> **** >> >> 2013/1/23 Steven Ayre **** >> >> It's localep.h, not local.h:**** >> >> /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.h:41:26: >> error: opal/localep.h: No such file or directory**** >> >> **** >> >> find / -name localep.h**** >> >> **** >> >> Sorry, seems my phone did an annoying autouncorrection!**** >> >> **** >> >> -Steve**** >> >> **** >> >> **** >> >> **** >> >> On 23 January 2013 08:16, Anton Rostotskiy >> wrote:**** >> >> Thanks for reply, Steven!**** >> >> Tried to search local.h file, but search returned nothing. That are all >> local.h files in my system:**** >> >> **** >> >> /usr/local/src/freeswitch/libs/spandsp/src/gsm0610_local.h**** >> >> /usr/local/src/freeswitch/libs/spandsp/src/t30_local.h**** >> >> /usr/local/src/freeswitch/libs/spandsp/src/t42_t43_local.h**** >> >> **** >> >> About -devel packages - at first tried to install ptlib, opal and >> -devel from yum but their versions have been too low for mod_opal.**** >> >> **** >> >> 2013/1/22 Steven Ayre **** >> >> Run:**** >> >> find / -name local.h**** >> >> **** >> >> It's failing to include opal/local.h. All the other errors are because >> its missing the definitions in that file.**** >> >> **** >> >> Normally I would say to instal the -devel package, but since you built >> from svn that doesn't apply. Perhaps it's not installed or installed to the >> wrong place. The find command will tell us where that file is.**** >> >> >> Steve on iPhone**** >> >> >> On 22 Jan 2013, at 16:07, Anton Rostotskiy >> wrote:**** >> >> Also tried to install mod_opal and also failed.**** >> >> CentOS 6.3 x86_64**** >> >> ptlib and opal are the last version obtained from svn as it is said in >> wiki .**** >> >> but just a bunch of errors after make mod_opal >> http://pastebin.com/HHUPLCLV**** >> >> **** >> >> 2013/1/18 Steven Ayre **** >> >> The latest version of opal in Debian (even in Sid) isn't new enough for >> mod_opal. It needed various features added that are only only available in >> the svn version.**** >> >> **** >> >> The wiki page states you have to install ptlib and opal from svn - this >> is why.**** >> >> http://wiki.freeswitch.org/wiki/Mod_opal >> >> Of course that can then cause conflicts unless you package it yourself or >> install a dummy package, or problems with other libraries/apps that are >> built against the libopal that's already packaged.**** >> >> **** >> >> Ptlib and h323plus/opal are also notorious for having to get matching >> versions. If you don't pair exactly the right version of ptlib with the >> specific version of h323plus/opal you can get instability problems. So the >> official Debian ptlib package probably wouldn't be good with opal from >> svn...**** >> >> **** >> >> As for mod_h323, it does compile and work, at least on Wheezy and >> possibly Squeeze too. I've never had any luck getting mod_opal working but >> have got mod_h323 working. I don't actively use it though - all my traffic >> is currently SIP.**** >> >> **** >> >> Another option is to run Yate as a SIP->H323 proxy. It only handles the >> signalling, media still goes directly caller->FS. It worked quite well, >> although it would occasionally crash - but at least that only dropped the >> H323 calls and didn't take FS down with it, and Yate automatically >> restarted. That was on Lenny so a long time ago...**** >> >> >> Steve on iPhone**** >> >> **** >> >> **** >> >> >> On 17 Jan 2013, at 20:54, "Gilbert Amar" wrote:**** >> >> Hi, >> >> I am trying to build mod_opal in FS v1.2stable, using ptlib 2.10.7 and >> opal >> 3.10.7 on a debian >> Ptlib and opal get build with no error, but when trying mod_opal I get >> >> Compiling >> /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp... >> quiet_libtool: compile: g++ -g -ggdb >> -I/usr/local/src/freeswitch/libs/curl/include >> -I/usr/local/src/freeswitch/src/include >> -I/usr/local/src/freeswitch/src/include >> -I/usr/local/src/freeswitch/libs/libteletone/src >> -I/usr/local/src/freeswitch/libs/stfu -fPIC -fvisibility=hidden >> -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE >> -DP_64BIT >> -DPTRACING=1 -D_REENTRANT -D_GNU_SOURCE=1 -fno-exceptions >> -I/usr/local/include -I/usr/local/include/opal -I/usr/include/SDL >> -DHAVE_CONFIG_H -c >> /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp -fPIC >> -DPIC -o .libs/mod_opal.o >> /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp: In >> member >> function 'switch_status_t FSConnection::read_frame(const OpalMediaType&, >> switch_frame_t**, switch_io_flag_t)': >> /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp:1208: >> error: 'class OpalCall' has no member named 'IsSwitchingT38' >> /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp: In >> member >> function 'switch_status_t FSConnection::write_frame(const OpalMediaType&, >> const switch_frame_t*, switch_io_flag_t)': >> /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp:1225: >> error: 'class OpalCall' has no member named 'IsSwitchingT38' >> make[3]: *** [mod_opal.lo] Error 1 >> make[2]: *** [install] Error 1 >> make[1]: *** [mod_opal-install] Error 1 >> make: *** [mod_opal-install] Error 2 >> >> >> Has anyone succeded in building mod_opal ? >> >> Should I go with mod_h323 instead ? >> >> Thanks for your help >> >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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**** >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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**** >> >> >> >> **** >> >> **** >> >> -- >> ? ?????????, ????? ????????? **** >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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**** >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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**** >> >> >> >> **** >> >> **** >> >> -- >> ? ?????????, ????? ????????? **** >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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**** >> >> **** >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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**** >> >> >> >> **** >> >> **** >> >> -- >> ? ?????????, ????? ????????? **** >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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**** >> >> **** >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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**** >> >> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org**** >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > > -- > ? ?????????, ????? ????????? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130204/8f58f0de/attachment-0001.html From david.villasmil.work at gmail.com Mon Feb 4 16:01:01 2013 From: david.villasmil.work at gmail.com (David Villasmil) Date: Mon, 4 Feb 2013 14:01:01 +0100 Subject: [Freeswitch-users] ESL outbound race condition? In-Reply-To: References: Message-ID: Hello Peter, freeswitch at internal> version FreeSWITCH Version 1.3.13b+git~20130201T013805Z~f5f8e90f01 (git f5f8e90 2013-02-01 01:38:05Z) Thanks for your help. David On Mon, Feb 4, 2013 at 6:46 AM, Peter Olsson wrote: > What version of FS are you running? This sounds like an old bug to me. > > /Peter > > 4 feb 2013 kl. 03:06 skrev "David Villasmil" < > david.villasmil.work at gmail.com>: > > > Hello guys, > > > > I've writing an app using outbound socket, when the channel is created > from FS to my app, I get an event called "SOCKET_DATA", I was expecting a > CHANNEL_CALLSTATE or similar. > > > > I read somewhere that it is related to a race condition? > > > > I'd love some help! > > > > Thanks! > > > > David > > !DSPAM:510f134332762069512667! > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > 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 > > > > > > !DSPAM:510f134332762069512667! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130204/b76d865c/attachment.html From stargray at bigmir.net Mon Feb 4 16:50:20 2013 From: stargray at bigmir.net (Anton Vojlenko) Date: Mon, 4 Feb 2013 15:50:20 +0200 Subject: [Freeswitch-users] ESL and DEBUG messages In-Reply-To: References: Message-ID: ****I can't solve this problem alone. In fs_cli i can see DEBUG messages: freeswitch at internal> skypopen sk1 GET USERSTATUS Using interface: globals.SKYPOPEN_INTERFACES[1].name=|||sk1||| 2013-02-04 15:31:44.565590 [DEBUG] skypopen_protocol.c:1732 [|] [DEBUG_SKYPE 1732 ][sk1 ][IDLE,IDLE] SENDING: |||GET USERSTATUS|||| 2013-02-04 15:31:44.565590 [DEBUG] skypopen_protocol.c:207 [|] [DEBUG_SKYPE 207 ][sk1 ][IDLE,IDLE] READING: |||USERSTATUS ONLINE||| But when i execute perl script i can't see them: #perl /home/checker.pl Using interface: globals.SKYPOPEN_INTERFACES[1].name=|||sk1||| #cat /home/checker.pl #!/usr/bin/perl use strict; use warnings; require ESL; #ESL::eslSetLogLevel(7); my $host = "127.0.0.1"; my $port = "8021"; my $pass = "ClueCon"; my $con = new ESL::ESLconnection($host, $port, $pass); $con->api("console loglevel 9"); $con->api("fsctl loglevel 9"); my $status = $con->api("skypopen sk1 GET USERSTATUS")->getBody(); print "$status"; $con->disconnect() 2013/1/23 Anton Vojlenko > Giovanni, I can see Skype API answers only when i using fs_cli. With perl > script I can't see them. > > > 2013/1/22 Giovanni Maruzzelli > >> On Tue, Jan 22, 2013 at 12:51 PM, Anton Vojlenko >> wrote: >> > Hello, >> > >> > I want to check user status from the Skype client instance, but I can't >> see >> > Skype API answers from it. How to fix this? >> >> Have you tried to give: >> >> "fsctl loglevel 9" in addition to "console loglevel 9" ? >> >> >> >> > Script example: >> > >> > #!/usr/bin/perl >> > use strict; >> > use warnings; >> > require ESL; >> > >> > my $host = "localhost"; >> > my $port = "8021"; >> > my $pass = "somepass"; >> > my $con = new ESL::ESLconnection($host, $port, $pass); >> > >> > $con->api("console loglevel 9"); >> > my $status = $con->api("skypopen sk1 GET USERSTATUS")->getBody(); >> > print $status; >> > $con->disconnect(); >> > >> >> -- >> Sincerely, >> >> Giovanni Maruzzelli >> Cell : +39-347-2665618 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130204/7aee7b61/attachment.html From gmaruzz at celliax.org Mon Feb 4 16:59:12 2013 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Mon, 4 Feb 2013 14:59:12 +0100 Subject: [Freeswitch-users] ESL and DEBUG messages In-Reply-To: References: Message-ID: Please, can someone chime in and tell if something has to change in the skype module, or if the perl script has to be modified? -giovanni On Mon, Feb 4, 2013 at 2:50 PM, Anton Vojlenko wrote: > ****I > can't solve this problem alone. > In fs_cli i can see DEBUG messages: > freeswitch at internal> skypopen sk1 GET USERSTATUS > Using interface: globals.SKYPOPEN_INTERFACES[1].name=|||sk1||| > > 2013-02-04 15:31:44.565590 [DEBUG] skypopen_protocol.c:1732 [|] > [DEBUG_SKYPE 1732 ][sk1 ][IDLE,IDLE] SENDING: |||GET > USERSTATUS|||| > 2013-02-04 15:31:44.565590 [DEBUG] skypopen_protocol.c:207 [|] > [DEBUG_SKYPE 207 ][sk1 ][IDLE,IDLE] READING: |||USERSTATUS > ONLINE||| > > But when i execute perl script i can't see them: > #perl /home/checker.pl > Using interface: globals.SKYPOPEN_INTERFACES[1].name=|||sk1||| > > #cat /home/checker.pl > > #!/usr/bin/perl > use strict; > use warnings; > require ESL; > #ESL::eslSetLogLevel(7); > > my $host = "127.0.0.1"; > my $port = "8021"; > my $pass = "ClueCon"; > > my $con = new ESL::ESLconnection($host, $port, $pass); > > $con->api("console loglevel 9"); > $con->api("fsctl loglevel 9"); > > my $status = $con->api("skypopen sk1 GET USERSTATUS")->getBody(); > print "$status"; > $con->disconnect() > > > > 2013/1/23 Anton Vojlenko > >> Giovanni, I can see Skype API answers only when i using fs_cli. With perl >> script I can't see them. >> >> >> 2013/1/22 Giovanni Maruzzelli >> >>> On Tue, Jan 22, 2013 at 12:51 PM, Anton Vojlenko >>> wrote: >>> > Hello, >>> > >>> > I want to check user status from the Skype client instance, but I >>> can't see >>> > Skype API answers from it. How to fix this? >>> >>> Have you tried to give: >>> >>> "fsctl loglevel 9" in addition to "console loglevel 9" ? >>> >>> >>> >>> > Script example: >>> > >>> > #!/usr/bin/perl >>> > use strict; >>> > use warnings; >>> > require ESL; >>> > >>> > my $host = "localhost"; >>> > my $port = "8021"; >>> > my $pass = "somepass"; >>> > my $con = new ESL::ESLconnection($host, $port, $pass); >>> > >>> > $con->api("console loglevel 9"); >>> > my $status = $con->api("skypopen sk1 GET USERSTATUS")->getBody(); >>> > print $status; >>> > $con->disconnect(); >>> > >>> >>> -- >>> Sincerely, >>> >>> Giovanni Maruzzelli >>> Cell : +39-347-2665618 >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130204/0153c2af/attachment-0001.html From steveayre at gmail.com Mon Feb 4 17:32:03 2013 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 4 Feb 2013 14:32:03 +0000 Subject: [Freeswitch-users] ESL and DEBUG messages In-Reply-To: References: Message-ID: 'fsctl loglevel 9' controls the core logging, all logs are routed through here and dropped if they're below the logging level set there. On top of that modules that handle log messages separately filter log levels. 'console loglevel 9' controls mod_console which is only for when freeswitch is started in the foreground. You don't want to use either of the above. Mod_event_socket controls logging via ESL connections. You need to issue the 'log' ESL command to set the log level on a per-connection basis (this is what /log does in fs_cli). See http://wiki.freeswitch.org/wiki/Mod_event_socket#log Note log is an ESL protocol command, not an api/app. -Steve On 4 February 2013 13:50, Anton Vojlenko wrote: > ****I > can't solve this problem alone. > In fs_cli i can see DEBUG messages: > freeswitch at internal> skypopen sk1 GET USERSTATUS > Using interface: globals.SKYPOPEN_INTERFACES[1].name=|||sk1||| > > 2013-02-04 15:31:44.565590 [DEBUG] skypopen_protocol.c:1732 [|] > [DEBUG_SKYPE 1732 ][sk1 ][IDLE,IDLE] SENDING: |||GET > USERSTATUS|||| > 2013-02-04 15:31:44.565590 [DEBUG] skypopen_protocol.c:207 [|] > [DEBUG_SKYPE 207 ][sk1 ][IDLE,IDLE] READING: |||USERSTATUS > ONLINE||| > > But when i execute perl script i can't see them: > #perl /home/checker.pl > Using interface: globals.SKYPOPEN_INTERFACES[1].name=|||sk1||| > > #cat /home/checker.pl > > #!/usr/bin/perl > use strict; > use warnings; > require ESL; > #ESL::eslSetLogLevel(7); > > my $host = "127.0.0.1"; > my $port = "8021"; > my $pass = "ClueCon"; > > my $con = new ESL::ESLconnection($host, $port, $pass); > > $con->api("console loglevel 9"); > $con->api("fsctl loglevel 9"); > > my $status = $con->api("skypopen sk1 GET USERSTATUS")->getBody(); > print "$status"; > $con->disconnect() > > > > 2013/1/23 Anton Vojlenko > >> Giovanni, I can see Skype API answers only when i using fs_cli. With perl >> script I can't see them. >> >> >> 2013/1/22 Giovanni Maruzzelli >> >>> On Tue, Jan 22, 2013 at 12:51 PM, Anton Vojlenko >>> wrote: >>> > Hello, >>> > >>> > I want to check user status from the Skype client instance, but I >>> can't see >>> > Skype API answers from it. How to fix this? >>> >>> Have you tried to give: >>> >>> "fsctl loglevel 9" in addition to "console loglevel 9" ? >>> >>> >>> >>> > Script example: >>> > >>> > #!/usr/bin/perl >>> > use strict; >>> > use warnings; >>> > require ESL; >>> > >>> > my $host = "localhost"; >>> > my $port = "8021"; >>> > my $pass = "somepass"; >>> > my $con = new ESL::ESLconnection($host, $port, $pass); >>> > >>> > $con->api("console loglevel 9"); >>> > my $status = $con->api("skypopen sk1 GET USERSTATUS")->getBody(); >>> > print $status; >>> > $con->disconnect(); >>> > >>> >>> -- >>> Sincerely, >>> >>> Giovanni Maruzzelli >>> Cell : +39-347-2665618 >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130204/9fa1ff86/attachment.html From gmaruzz at celliax.org Mon Feb 4 17:40:31 2013 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Mon, 4 Feb 2013 15:40:31 +0100 Subject: [Freeswitch-users] ESL and DEBUG messages In-Reply-To: References: Message-ID: Thanks Steve! On Mon, Feb 4, 2013 at 3:32 PM, Steven Ayre wrote: > 'fsctl loglevel 9' controls the core logging, all logs are routed through > here and dropped if they're below the logging level set there. On top of > that modules that handle log messages separately filter log levels. > > 'console loglevel 9' controls mod_console which is only for when > freeswitch is started in the foreground. > > You don't want to use either of the above. > > Mod_event_socket controls logging via ESL connections. You need to issue > the 'log' ESL command to set the log level on a per-connection basis (this > is what /log does in fs_cli). > > See http://wiki.freeswitch.org/wiki/Mod_event_socket#log > > Note log is an ESL protocol command, not an api/app. > > -Steve > > > > On 4 February 2013 13:50, Anton Vojlenko wrote: > >> ****I >> can't solve this problem alone. >> In fs_cli i can see DEBUG messages: >> freeswitch at internal> skypopen sk1 GET USERSTATUS >> Using interface: globals.SKYPOPEN_INTERFACES[1].name=|||sk1||| >> >> 2013-02-04 15:31:44.565590 [DEBUG] skypopen_protocol.c:1732 [|] >> [DEBUG_SKYPE 1732 ][sk1 ][IDLE,IDLE] SENDING: |||GET >> USERSTATUS|||| >> 2013-02-04 15:31:44.565590 [DEBUG] skypopen_protocol.c:207 [|] >> [DEBUG_SKYPE 207 ][sk1 ][IDLE,IDLE] READING: |||USERSTATUS >> ONLINE||| >> >> But when i execute perl script i can't see them: >> #perl /home/checker.pl >> Using interface: globals.SKYPOPEN_INTERFACES[1].name=|||sk1||| >> >> #cat /home/checker.pl >> >> #!/usr/bin/perl >> use strict; >> use warnings; >> require ESL; >> #ESL::eslSetLogLevel(7); >> >> my $host = "127.0.0.1"; >> my $port = "8021"; >> my $pass = "ClueCon"; >> >> my $con = new ESL::ESLconnection($host, $port, $pass); >> >> $con->api("console loglevel 9"); >> $con->api("fsctl loglevel 9"); >> >> my $status = $con->api("skypopen sk1 GET USERSTATUS")->getBody(); >> print "$status"; >> $con->disconnect() >> >> >> >> 2013/1/23 Anton Vojlenko >> >>> Giovanni, I can see Skype API answers only when i using fs_cli. With >>> perl script I can't see them. >>> >>> >>> 2013/1/22 Giovanni Maruzzelli >>> >>>> On Tue, Jan 22, 2013 at 12:51 PM, Anton Vojlenko >>>> wrote: >>>> > Hello, >>>> > >>>> > I want to check user status from the Skype client instance, but I >>>> can't see >>>> > Skype API answers from it. How to fix this? >>>> >>>> Have you tried to give: >>>> >>>> "fsctl loglevel 9" in addition to "console loglevel 9" ? >>>> >>>> >>>> >>>> > Script example: >>>> > >>>> > #!/usr/bin/perl >>>> > use strict; >>>> > use warnings; >>>> > require ESL; >>>> > >>>> > my $host = "localhost"; >>>> > my $port = "8021"; >>>> > my $pass = "somepass"; >>>> > my $con = new ESL::ESLconnection($host, $port, $pass); >>>> > >>>> > $con->api("console loglevel 9"); >>>> > my $status = $con->api("skypopen sk1 GET USERSTATUS")->getBody(); >>>> > print $status; >>>> > $con->disconnect(); >>>> > >>>> >>>> -- >>>> Sincerely, >>>> >>>> Giovanni Maruzzelli >>>> Cell : +39-347-2665618 >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>>> >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130204/ce0cfa37/attachment-0001.html From stargray at bigmir.net Mon Feb 4 17:58:19 2013 From: stargray at bigmir.net (Anton Vojlenko) Date: Mon, 4 Feb 2013 16:58:19 +0200 Subject: [Freeswitch-users] ESL and DEBUG messages In-Reply-To: References: Message-ID: Thank you, Steve. #!/usr/bin/perl use strict; use warnings; require ESL; #ESL::eslSetLogLevel(7); my $host = "127.0.0.1"; my $port = "8021"; my $pass = "ClueCon"; my $con = new ESL::ESLconnection($host, $port, $pass); *$con->log(7);* my $status = $con->api("skypopen sk1 GET USERSTATUS")->getBody(); print "$status"; $con->disconnect(); Am i right? 2013/2/4 Giovanni Maruzzelli > Thanks Steve! > > > On Mon, Feb 4, 2013 at 3:32 PM, Steven Ayre wrote: > >> 'fsctl loglevel 9' controls the core logging, all logs are routed through >> here and dropped if they're below the logging level set there. On top of >> that modules that handle log messages separately filter log levels. >> >> 'console loglevel 9' controls mod_console which is only for when >> freeswitch is started in the foreground. >> >> You don't want to use either of the above. >> >> Mod_event_socket controls logging via ESL connections. You need to issue >> the 'log' ESL command to set the log level on a per-connection basis (this >> is what /log does in fs_cli). >> >> See http://wiki.freeswitch.org/wiki/Mod_event_socket#log >> >> Note log is an ESL protocol command, not an api/app. >> >> -Steve >> >> >> >> On 4 February 2013 13:50, Anton Vojlenko wrote: >> >>> ****I >>> can't solve this problem alone. >>> In fs_cli i can see DEBUG messages: >>> freeswitch at internal> skypopen sk1 GET USERSTATUS >>> Using interface: globals.SKYPOPEN_INTERFACES[1].name=|||sk1||| >>> >>> 2013-02-04 15:31:44.565590 [DEBUG] skypopen_protocol.c:1732 [|] >>> [DEBUG_SKYPE 1732 ][sk1 ][IDLE,IDLE] SENDING: |||GET >>> USERSTATUS|||| >>> 2013-02-04 15:31:44.565590 [DEBUG] skypopen_protocol.c:207 [|] >>> [DEBUG_SKYPE 207 ][sk1 ][IDLE,IDLE] READING: |||USERSTATUS >>> ONLINE||| >>> >>> But when i execute perl script i can't see them: >>> #perl /home/checker.pl >>> Using interface: globals.SKYPOPEN_INTERFACES[1].name=|||sk1||| >>> >>> #cat /home/checker.pl >>> >>> #!/usr/bin/perl >>> use strict; >>> use warnings; >>> require ESL; >>> #ESL::eslSetLogLevel(7); >>> >>> my $host = "127.0.0.1"; >>> my $port = "8021"; >>> my $pass = "ClueCon"; >>> >>> my $con = new ESL::ESLconnection($host, $port, $pass); >>> >>> $con->api("console loglevel 9"); >>> $con->api("fsctl loglevel 9"); >>> >>> my $status = $con->api("skypopen sk1 GET USERSTATUS")->getBody(); >>> print "$status"; >>> $con->disconnect() >>> >>> >>> >>> 2013/1/23 Anton Vojlenko >>> >>>> Giovanni, I can see Skype API answers only when i using fs_cli. With >>>> perl script I can't see them. >>>> >>>> >>>> 2013/1/22 Giovanni Maruzzelli >>>> >>>>> On Tue, Jan 22, 2013 at 12:51 PM, Anton Vojlenko >>>>> wrote: >>>>> > Hello, >>>>> > >>>>> > I want to check user status from the Skype client instance, but I >>>>> can't see >>>>> > Skype API answers from it. How to fix this? >>>>> >>>>> Have you tried to give: >>>>> >>>>> "fsctl loglevel 9" in addition to "console loglevel 9" ? >>>>> >>>>> >>>>> >>>>> > Script example: >>>>> > >>>>> > #!/usr/bin/perl >>>>> > use strict; >>>>> > use warnings; >>>>> > require ESL; >>>>> > >>>>> > my $host = "localhost"; >>>>> > my $port = "8021"; >>>>> > my $pass = "somepass"; >>>>> > my $con = new ESL::ESLconnection($host, $port, $pass); >>>>> > >>>>> > $con->api("console loglevel 9"); >>>>> > my $status = $con->api("skypopen sk1 GET USERSTATUS")->getBody(); >>>>> > print $status; >>>>> > $con->disconnect(); >>>>> > >>>>> >>>>> -- >>>>> Sincerely, >>>>> >>>>> Giovanni Maruzzelli >>>>> Cell : +39-347-2665618 >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> 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 >>>>> >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > > -- > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130204/546db5cb/attachment.html From steveayre at gmail.com Mon Feb 4 18:35:33 2013 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 4 Feb 2013 15:35:33 +0000 Subject: [Freeswitch-users] ESL and DEBUG messages In-Reply-To: References: Message-ID: I believe so. They're probably received as the LOG event, which is unfortunately undocumented http://wiki.freeswitch.org/wiki/Event_list#LOG -Steve On 4 February 2013 14:58, Anton Vojlenko wrote: > Thank you, Steve. > > > #!/usr/bin/perl > use strict; > use warnings; > require ESL; > #ESL::eslSetLogLevel(7); > > my $host = "127.0.0.1"; > my $port = "8021"; > my $pass = "ClueCon"; > my $con = new ESL::ESLconnection($host, $port, $pass); > > *$con->log(7);* > > my $status = $con->api("skypopen sk1 GET USERSTATUS")->getBody(); > print "$status"; > $con->disconnect(); > > Am i right? > > > 2013/2/4 Giovanni Maruzzelli > >> Thanks Steve! >> >> >> On Mon, Feb 4, 2013 at 3:32 PM, Steven Ayre wrote: >> >>> 'fsctl loglevel 9' controls the core logging, all logs are routed >>> through here and dropped if they're below the logging level set there. On >>> top of that modules that handle log messages separately filter log levels. >>> >>> 'console loglevel 9' controls mod_console which is only for when >>> freeswitch is started in the foreground. >>> >>> You don't want to use either of the above. >>> >>> Mod_event_socket controls logging via ESL connections. You need to issue >>> the 'log' ESL command to set the log level on a per-connection basis (this >>> is what /log does in fs_cli). >>> >>> See http://wiki.freeswitch.org/wiki/Mod_event_socket#log >>> >>> Note log is an ESL protocol command, not an api/app. >>> >>> -Steve >>> >>> >>> >>> On 4 February 2013 13:50, Anton Vojlenko wrote: >>> >>>> ****I >>>> can't solve this problem alone. >>>> In fs_cli i can see DEBUG messages: >>>> freeswitch at internal> skypopen sk1 GET USERSTATUS >>>> Using interface: globals.SKYPOPEN_INTERFACES[1].name=|||sk1||| >>>> >>>> 2013-02-04 15:31:44.565590 [DEBUG] skypopen_protocol.c:1732 [|] >>>> [DEBUG_SKYPE 1732 ][sk1 ][IDLE,IDLE] SENDING: |||GET >>>> USERSTATUS|||| >>>> 2013-02-04 15:31:44.565590 [DEBUG] skypopen_protocol.c:207 [|] >>>> [DEBUG_SKYPE 207 ][sk1 ][IDLE,IDLE] READING: |||USERSTATUS >>>> ONLINE||| >>>> >>>> But when i execute perl script i can't see them: >>>> #perl /home/checker.pl >>>> Using interface: globals.SKYPOPEN_INTERFACES[1].name=|||sk1||| >>>> >>>> #cat /home/checker.pl >>>> >>>> #!/usr/bin/perl >>>> use strict; >>>> use warnings; >>>> require ESL; >>>> #ESL::eslSetLogLevel(7); >>>> >>>> my $host = "127.0.0.1"; >>>> my $port = "8021"; >>>> my $pass = "ClueCon"; >>>> >>>> my $con = new ESL::ESLconnection($host, $port, $pass); >>>> >>>> $con->api("console loglevel 9"); >>>> $con->api("fsctl loglevel 9"); >>>> >>>> my $status = $con->api("skypopen sk1 GET USERSTATUS")->getBody(); >>>> print "$status"; >>>> $con->disconnect() >>>> >>>> >>>> >>>> 2013/1/23 Anton Vojlenko >>>> >>>>> Giovanni, I can see Skype API answers only when i using fs_cli. With >>>>> perl script I can't see them. >>>>> >>>>> >>>>> 2013/1/22 Giovanni Maruzzelli >>>>> >>>>>> On Tue, Jan 22, 2013 at 12:51 PM, Anton Vojlenko >>>>>> wrote: >>>>>> > Hello, >>>>>> > >>>>>> > I want to check user status from the Skype client instance, but I >>>>>> can't see >>>>>> > Skype API answers from it. How to fix this? >>>>>> >>>>>> Have you tried to give: >>>>>> >>>>>> "fsctl loglevel 9" in addition to "console loglevel 9" ? >>>>>> >>>>>> >>>>>> >>>>>> > Script example: >>>>>> > >>>>>> > #!/usr/bin/perl >>>>>> > use strict; >>>>>> > use warnings; >>>>>> > require ESL; >>>>>> > >>>>>> > my $host = "localhost"; >>>>>> > my $port = "8021"; >>>>>> > my $pass = "somepass"; >>>>>> > my $con = new ESL::ESLconnection($host, $port, $pass); >>>>>> > >>>>>> > $con->api("console loglevel 9"); >>>>>> > my $status = $con->api("skypopen sk1 GET USERSTATUS")->getBody(); >>>>>> > print $status; >>>>>> > $con->disconnect(); >>>>>> > >>>>>> >>>>>> -- >>>>>> Sincerely, >>>>>> >>>>>> Giovanni Maruzzelli >>>>>> Cell : +39-347-2665618 >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> 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 >>>>>> >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> >> -- >> Sincerely, >> >> Giovanni Maruzzelli >> Cell : +39-347-2665618 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130204/1003667c/attachment-0001.html From jpyle at fidelityvoice.com Mon Feb 4 19:52:01 2013 From: jpyle at fidelityvoice.com (Jeff Pyle) Date: Mon, 04 Feb 2013 11:52:01 -0500 Subject: [Freeswitch-users] SIP SRV failover behavior Message-ID: <510FE731.9080601@fidelityvoice.com> Hello, I use an SRV record as the proxy value for a sofia gateway. This particular record is configured for failover rather than load balancing. In other words, there is only one record per priority. I've verified Freeswitch will advance to the second host if the first becomes unavailable. It seems Freeswitch waits around 10 minutes before it tries the first host again. Is this delay a configurable value? Is there some kind of internal blacklist the failed host occupies before this timer expires? Is this list manageable from the CLI? Besides a 408, are there other response codes that will land a host on this apparent blacklist? - Jeff From yura407 at gmail.com Mon Feb 4 19:59:59 2013 From: yura407 at gmail.com (Yuriy Ostapchuk) Date: Mon, 4 Feb 2013 11:59:59 -0500 Subject: [Freeswitch-users] Routing messages in mod_sms chatplan Message-ID: Hi, I have a setup where I need to route incoming sms messages from mod_gsmopen to different sip or dingaling extensions, depending on sender's phone number and other conditions. Is there a way to do this directly in chatplan without executing an external script? Basically, I'm looking for a functionality similar to 'transfer' application in dialplan. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130204/70531b8d/attachment.html From avi at avimarcus.net Mon Feb 4 20:11:52 2013 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 4 Feb 2013 19:11:52 +0200 Subject: [Freeswitch-users] SIP SRV failover behavior In-Reply-To: <510FE731.9080601@fidelityvoice.com> References: <510FE731.9080601@fidelityvoice.com> Message-ID: Not really the answer, but... There's a gateway param called "ping", e.g: Then if it's marked down (via 202 or 404 perhaps), any bridge string referencing that gateway will know it's down before contact it and move on. (with error code NETWORK_OUT_OF_ORDER) You can see that showing up in the logs, and I think in the gateway status. I don't see a command to manually re-ping the gateway. -Avi On Mon, Feb 4, 2013 at 6:52 PM, Jeff Pyle wrote: > Hello, > > I use an SRV record as the proxy value for a sofia gateway. This > particular record is configured for failover rather than load > balancing. In other words, there is only one record per priority. I've > verified Freeswitch will advance to the second host if the first becomes > unavailable. > > It seems Freeswitch waits around 10 minutes before it tries the first > host again. Is this delay a configurable value? > > Is there some kind of internal blacklist the failed host occupies before > this timer expires? Is this list manageable from the CLI? > > Besides a 408, are there other response codes that will land a host on > this apparent blacklist? > > > > - Jeff > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130204/b61b7613/attachment.html From michel.brabants at gmail.com Mon Feb 4 20:31:38 2013 From: michel.brabants at gmail.com (Michel Brabants) Date: Mon, 4 Feb 2013 18:31:38 +0100 Subject: [Freeswitch-users] SIP SRV failover behavior In-Reply-To: References: <510FE731.9080601@fidelityvoice.com> Message-ID: Hello, to solve it temporarily (I also don't know of a manual ping), you can ofcourse kill the gateway "sofia ... killgw" and bring it back up using a rescan of the profile ("sofia profile .... rescan reloadxml"). I suppose that the options-ping, like suggested below should work. I don't have any problem as far as I can say and I did quite some failover-tests. Kind regards, Michel On Mon, Feb 4, 2013 at 6:11 PM, Avi Marcus wrote: > Not really the answer, but... > > There's a gateway param called "ping", e.g: > > > > > Then if it's marked down (via 202 or 404 perhaps), > any bridge string referencing that gateway will know it's down before > contact it and move on. (with error code NETWORK_OUT_OF_ORDER) > > You can see that showing up in the logs, and I think in the gateway > status. I don't see a command to manually re-ping the gateway. > > -Avi > > On Mon, Feb 4, 2013 at 6:52 PM, Jeff Pyle wrote: > >> Hello, >> >> I use an SRV record as the proxy value for a sofia gateway. This >> particular record is configured for failover rather than load >> balancing. In other words, there is only one record per priority. I've >> verified Freeswitch will advance to the second host if the first becomes >> unavailable. >> >> It seems Freeswitch waits around 10 minutes before it tries the first >> host again. Is this delay a configurable value? >> >> Is there some kind of internal blacklist the failed host occupies before >> this timer expires? Is this list manageable from the CLI? >> >> Besides a 408, are there other response codes that will land a host on >> this apparent blacklist? >> >> >> >> - Jeff >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130204/507a828e/attachment.html From rnbrady at gmail.com Mon Feb 4 21:08:22 2013 From: rnbrady at gmail.com (Richard Brady) Date: Mon, 4 Feb 2013 18:08:22 +0000 Subject: [Freeswitch-users] adding In-Reply-To in sip header not working In-Reply-To: <0F4F8E19-73FC-460F-A5AB-B7A2CF8CD78B@mgtech.com> References: <0F4F8E19-73FC-460F-A5AB-B7A2CF8CD78B@mgtech.com> Message-ID: That Call-ID looks fine. Just to be clear, you are sending this back to the same provider it came from right? You'll need to post a trace (anonymised) in order for me to help further. Richard On 29 January 2013 19:31, Mario G wrote: > Thanks, I tried both suggestions but no love. When I used * > sip_h_In-Reply-To**=${sip_call_id} *used as below, the trace showed all > normal but the cell phone does not ring at all. When I removed everything > the cell rang but the original number was not passed. BTW, the sip_call_id > was translated to (#s altered): > sip_h_In-Reply-To=3912345-9123456295-612341 at msw1.telengy.net, could that > be an issue with ATT not liking it? > > Mario G > > > > On Jan 27, 2013, at 3:02 PM, Richard Brady wrote: > > Ok, nifty. They are letting you present a number you do not own as Caller > ID on an outbound call if that outbound call is a forwarded leg of an > inbound call. > > They do this by looking the In-Reply-To header of the INVITE for the > forwarded leg, which should contain the Call-ID of the orignal leg. > > So you need to copy the *Call-ID* in order to authorize the *Caller ID*. > > A couple things: > > 1. From the docs: *effective_caller_id_name Sets the effective callerid > name. This is automatically exported to the B-leg; however, it is not > valid in an origination string. In other words, set this before calling > bridge, otherwise use origination_caller_id_name* > > 2. You shouldn't care about 1 above as it should be copied across from the > A leg by default and you are not modifying it, so > remove effective_caller_id_name and don't bother > with origination_caller_id_name either. > > 3. You should use sip_h_ not sip_rh_ because you want the header in the > new INVITE going out. > > Perhaps try: > > > > Hope this helps. > > Richard > > > On 21 January 2013 19:54, Mario G wrote: > >> Thanks, apparently I had it wrong, the doc below states that the PBX must >> support it incoming, they pointed me to using effective_caller_id which I >> added to the bridge but it still does not work. Would love to fix this >> since the cell phones currently have no idea who is calling. >> Mario G >> >> > "{originate_timeout=45,alert_info=n=${lua_ringtone}}${group_call(bria@ >> ${domain_name}+A)},${group_call(deskphone@ >> ${domain_name}+A)},[leg_delay_start=20,leg_timeout=23,effective_caller_id_number=${caller_id_number}]sofia/gateway/${dial_gateway}/19161234567" >> /> >> >> >> - >> >> Please note that this feature is ONLY AVAILABLE for customers using a >> SIP PBX that either supports (or allows the configuring of) the >> "in-reply-to" header (defined by RFC 3261) for incoming calls which are >> forwarded to an outbound trunk. In these instances Callcentric will >> "Pass-Through" the CallerID from the original call which was received to >> the outbound bridged/forwarded call. >> >> On Jan 19, 2013, at 5:13 PM, Richard Brady wrote: >> >> On 20 January 2013 00:06, Mario G wrote: >> >>> I never did this so I must be missing something, I tried both below but >>> the bridge then fails. Can anyone shed some light on what I am doing wrong. >>> My ITSP now supports in-reply-to so I can pass the caller ID to a forwarded >>> call from FS. >>> >> >> In-Reply-To should contain a Call-ID not a caller ID. They are very >> different. >> >> The following would make a bit more sense, but still not a lot: >> >> >> >> Using In-Reply-To in a response doesn't seem right to me. I would expect >> it to appear in an INVITE. For for example, you get a missed call and you >> call the person back, then the INVITE for the callback would have a new >> Call-ID but the original Call-ID in the In-Reply-To header. That said, I >> have no idea what your ITSPs intended use for the header is. >> >> Richard >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130204/1a822862/attachment-0001.html From mario_fs at mgtech.com Mon Feb 4 23:41:28 2013 From: mario_fs at mgtech.com (Mario G) Date: Mon, 4 Feb 2013 12:41:28 -0800 Subject: [Freeswitch-users] [Solved] Callgroup keeps ringing after answer - changed from shared extensions In-Reply-To: <33B7D98B-6002-43CB-B8B7-DEFB30B34B2C@mgtech.com> References: <33B7D98B-6002-43CB-B8B7-DEFB30B34B2C@mgtech.com> Message-ID: <22E6FF8C-62AE-41DD-8CCA-01DCBC5A55D6@mgtech.com> I am sharing this "hair puller" problem in case someone else runs into it. It took 9 months to resolve. I tried to find something different in traces for good/bad calls but nothing. Updated FreeSwitch stable and head many times, no luck. I posted here in April but no responses and found only 1 similar issues searching the web. Intermittent Problems: Started when all SPA962s changed from shared IDs to unique IDs. 1 - A call is answered and heard but other extensions keep ringing. 2 - A call is answered but there is silence and other extensions keep ringing, pick up one by one, each silence. Solution: Replaced Cisco/Linksys SPA962s with Yealink T32s (UDP) and iPads (TCP) and have not seen this problem for a 2 months so we're pretty sure it's gone. BTW we really like the T32s (bought based on comments here). Yes I checked and tried many different SPA962 settings, good riddance. Mario G On Apr 16, 2012, at 2:30 PM, Mario G wrote: > Been at this for 3 weeks now, hope someone can help.... This started after replacing shared extensions with unique users and updating from an early 2011 git. > > Ring group via "pointer" works fine as it always has. However, answer via callgroup variable intermittently (30-40%) leaves one or mores phone ringing even though the the call was answered at another extension. The darn thing is that I can't reproduce it myself to get a trace. Any help/suggestions greatly appreciated. Phones are SPA962 (UDP) and Bria on iPad (TCP, same problems) and they all stopped ringing when they shared IDs. > Mario G > > default.xml ring-group has (abbreviated to save space)------------------: > > > > > ..... more users > > > > > > > > > > > > Extension----------------------: > > User: > > > > > > > > > > > > > > > > > Bridge command-----------------: > > > > > > > > > > > > > > > _________________________________________________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130204/cf129dc7/attachment.html From mario_fs at mgtech.com Mon Feb 4 23:44:44 2013 From: mario_fs at mgtech.com (Mario G) Date: Mon, 4 Feb 2013 12:44:44 -0800 Subject: [Freeswitch-users] adding In-Reply-To in sip header not working In-Reply-To: References: <0F4F8E19-73FC-460F-A5AB-B7A2CF8CD78B@mgtech.com> Message-ID: <1769A24D-BC79-4CFD-AA88-0A1E81E346BA@mgtech.com> Yes, in and out are the same provider, although in/out may be different accounts, could that be the ticket? Mario G On Feb 4, 2013, at 10:08 AM, Richard Brady wrote: > That Call-ID looks fine. Just to be clear, you are sending this back to the same provider it came from right? > > You'll need to post a trace (anonymised) in order for me to help further. > > Richard > > On 29 January 2013 19:31, Mario G wrote: > Thanks, I tried both suggestions but no love. When I used sip_h_In-Reply-To=${sip_call_id} used as below, the trace showed all normal but the cell phone does not ring at all. When I removed everything the cell rang but the original number was not passed. BTW, the sip_call_id was translated to (#s altered): > sip_h_In-Reply-To=3912345-9123456295-612341 at msw1.telengy.net, could that be an issue with ATT not liking it? > > Mario G > > > > On Jan 27, 2013, at 3:02 PM, Richard Brady wrote: > >> Ok, nifty. They are letting you present a number you do not own as Caller ID on an outbound call if that outbound call is a forwarded leg of an inbound call. >> >> They do this by looking the In-Reply-To header of the INVITE for the forwarded leg, which should contain the Call-ID of the orignal leg. >> >> So you need to copy the Call-ID in order to authorize the Caller ID. >> >> A couple things: >> >> 1. From the docs: effective_caller_id_name Sets the effective callerid name. This is automatically exported to the B-leg; however, it is not valid in an origination string. In other words, set this before calling bridge, otherwise use origination_caller_id_name >> >> 2. You shouldn't care about 1 above as it should be copied across from the A leg by default and you are not modifying it, so remove effective_caller_id_name and don't bother with origination_caller_id_name either. >> >> 3. You should use sip_h_ not sip_rh_ because you want the header in the new INVITE going out. >> >> Perhaps try: >> >> >> >> Hope this helps. >> >> Richard >> >> >> On 21 January 2013 19:54, Mario G wrote: >> Thanks, apparently I had it wrong, the doc below states that the PBX must support it incoming, they pointed me to using effective_caller_id which I added to the bridge but it still does not work. Would love to fix this since the cell phones currently have no idea who is calling. >> Mario G >> >> >> >> Please note that this feature is ONLY AVAILABLE for customers using a SIP PBX that either supports (or allows the configuring of) the "in-reply-to" header (defined by RFC 3261) for incoming calls which are forwarded to an outbound trunk. In these instances Callcentric will "Pass-Through" the CallerID from the original call which was received to the outbound bridged/forwarded call. >> >> On Jan 19, 2013, at 5:13 PM, Richard Brady wrote: >> >>> On 20 January 2013 00:06, Mario G wrote: >>> I never did this so I must be missing something, I tried both below but the bridge then fails. Can anyone shed some light on what I am doing wrong. My ITSP now supports in-reply-to so I can pass the caller ID to a forwarded call from FS. >>> >>> In-Reply-To should contain a Call-ID not a caller ID. They are very different. >>> >>> The following would make a bit more sense, but still not a lot: >>> >>> >>> >>> Using In-Reply-To in a response doesn't seem right to me. I would expect it to appear in an INVITE. For for example, you get a missed call and you call the person back, then the INVITE for the callback would have a new Call-ID but the original Call-ID in the In-Reply-To header. That said, I have no idea what your ITSPs intended use for the header is. >>> >>> Richard >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130204/b2898dab/attachment-0001.html From POlsson at enghouse.com Mon Feb 4 23:58:43 2013 From: POlsson at enghouse.com (Peter Olsson) Date: Mon, 4 Feb 2013 20:58:43 +0000 Subject: [Freeswitch-users] ESL outbound race condition? In-Reply-To: References: , Message-ID: <1FFF97C269757C458224B7C895F35F15216FCC@cantor.std.visionutv.se> In that case, please report this to Jira, and make sure to attach all needed information. /Peter ________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r David Villasmil [david.villasmil.work at gmail.com] Skickat: den 4 februari 2013 14:01 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] ESL outbound race condition? Hello Peter, freeswitch at internal> version FreeSWITCH Version 1.3.13b+git~20130201T013805Z~f5f8e90f01 (git f5f8e90 2013-02-01 01:38:05Z) Thanks for your help. David On Mon, Feb 4, 2013 at 6:46 AM, Peter Olsson > wrote: What version of FS are you running? This sounds like an old bug to me. /Peter 4 feb 2013 kl. 03:06 skrev "David Villasmil" >: > Hello guys, > > I've writing an app using outbound socket, when the channel is created from FS to my app, I get an event called "SOCKET_DATA", I was expecting a CHANNEL_CALLSTATE or similar. > > I read somewhere that it is related to a race condition? > > I'd love some help! > > Thanks! > > David > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > !DSPAM:510f134332762069512667! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 !DSPAM:510fbc7e32761660255216! From 8f27e956 at gmail.com Tue Feb 5 00:04:03 2013 From: 8f27e956 at gmail.com (S. Scott) Date: Mon, 4 Feb 2013 16:04:03 -0500 Subject: [Freeswitch-users] [Solved] Callgroup keeps ringing after answer - changed from shared extensions In-Reply-To: <22E6FF8C-62AE-41DD-8CCA-01DCBC5A55D6@mgtech.com> References: <33B7D98B-6002-43CB-B8B7-DEFB30B34B2C@mgtech.com> <22E6FF8C-62AE-41DD-8CCA-01DCBC5A55D6@mgtech.com> Message-ID: <6578849728373269042@unknownmsgid> May be/may not be your issue/solution, but certain SPA config/use conditions result in the SPA utilizing ("listening" -- additionally -- on) a second sip signaling port, which is usually prime+2; therefore, (default) prime 5060, then secondary would be 5062. The fs config would then need to "aim" the secondary features/functions at spa:5062. I once ran into this with a Panasonic switch and the SPA. Regardless, though, they are a pain. ????? iThing: Big thumbs & little keys. Please excuse typo, spelling and grammar errors ? Thought of the Day ? "With all this manure, there must be a pony in here somewhere.? On 2013-02-04, at 15:44, Mario G wrote: I am sharing this "hair puller" problem in case someone else runs into it. It took 9 months to resolve. I tried to find something different in traces for good/bad calls but nothing. Updated FreeSwitch stable and head many times, no luck. I posted here in April but no responses and found only 1 similar issues searching the web. *Intermittent Problems*: Started when all SPA962s changed from shared IDs to unique IDs. *1* - A call is answered and heard but other extensions keep ringing. *2* - A call is answered but there is silence and other extensions keep ringing, pick up one by one, each silence. *Solution*: Replaced Cisco/Linksys SPA962s with Yealink T32s (UDP) and iPads (TCP) and have not seen this problem for a 2 months so we're pretty sure it's gone. BTW we really like the T32s (bought based on comments here). Yes I checked and tried many different SPA962 settings, good riddance. Mario G On Apr 16, 2012, at 2:30 PM, Mario G wrote: Been at this for 3 weeks now, hope someone can help.... This started after replacing shared extensions with unique users and updating from an early 2011 git. Ring group via "pointer" works fine as it always has. However, answer via callgroup variable intermittently (30-40%) leaves one or mores phone ringing even though the the call was answered at another extension. The darn thing is that I can't reproduce it myself to get a trace. Any help/suggestions greatly appreciated. Phones are SPA962 (UDP) and Bria on iPad (TCP, same problems) and they all stopped ringing when they shared IDs. Mario G default.xml ring-group has (abbreviated to save space)------------------: ..... more users Extension----------------------: User: Bridge command-----------------: _________________________________________________________________________ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130204/be515f42/attachment-0001.html From darcy at voice2net.ca Mon Feb 4 23:51:57 2013 From: darcy at voice2net.ca (Darcy Primrose) Date: Mon, 4 Feb 2013 15:51:57 -0500 Subject: [Freeswitch-users] valet park Message-ID: <008701ce0319$7bec1be0$73c453a0$@ca> When I place a call in valet apark, and then retrieve it, it stays bridges to the valet_slot Dial plan: CALL STATUS AFTER PICKUP: freeswitch at internal> show calls uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,des t,presence_id,presence_data,callstate,callee_name,callee_num,callee_directio n,call_uuid,hostname,sent_callee_name,sent_callee_num,b_uuid,b_direction,b_c reated,b_created_epoch,b_name,b_state,b_cid_name,b_cid_num,b_ip_addr,b_dest, b_presence_id,b_presence_data,b_callstate,b_callee_name,b_callee_num,b_calle e_direction,b_sent_callee_name,b_sent_callee_num,call_created_epoch 54b084a8-6f0c-11e2-9612-1dd513f41a4b,inbound,2013-02-04 16:49:12,1360010952,sofia/internal/101 at 192.168.209.49,CS_SOFT_EXECUTE,ext101 ,101,192.168.209.23,6001,6001 at 192.168.209.49,,ACTIVE,,,SEND,54b084a8-6f0c-11 e2-9612-1dd513f41a4b,localhost.localdomain,voice2net,6137041248,4f6e82e2-6f0 c-11e2-95e8-1dd513f41a4b,inbound,2013-02-04 16:49:03,1360010943,sofia/external/6137041248 at 98.158.129.14,CS_EXCHANGE_MEDI A,voice2net,6137041248,98.158.129.14,6001,6001 at 192.168.209.49,,ACTIVE,voice2 net,6137041248,SEND,ext101,101,1360010952 1 total. VERSION: freeswitch at internal> version FreeSWITCH Version 1.3.13b+git~20130119T155523Z~6cb3be7d8b (git 6cb3be7 2013-01-19 15:55:23Z) All aspects of the call work, we want to use the blf with the valet park and this does work, however, because the call does not drop, the valet_park light stays on, this unnecessarily ties up parking spots. Any help or suggestions would be greatly appreciated. Darcy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130204/74e37f3a/attachment.html From msc at freeswitch.org Tue Feb 5 03:05:48 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 4 Feb 2013 18:05:48 -0600 Subject: [Freeswitch-users] valet park In-Reply-To: <008701ce0319$7bec1be0$73c453a0$@ca> References: <008701ce0319$7bec1be0$73c453a0$@ca> Message-ID: Can you get a debug log of the call flowing through the system and drop it on pastebin? Also, once you get the call up please do this: show channels as xml And paste that output as well. -MC On Mon, Feb 4, 2013 at 2:51 PM, Darcy Primrose wrote: > > uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,hostname,sent_callee_name,sent_callee_num,b_uuid,b_direction,b_created,b_created_epoch,b_name,b_state,b_cid_name,b_cid_num,b_ip_addr,b_dest,b_presence_id,b_presence_data,b_callstate,b_callee_name,b_callee_num,b_callee_direction,b_sent_callee_name,b_sent_callee_num,call_created_epoch > **** > > 54b084a8-6f0c-11e2-9612-1dd513f41a4b,inbound,2013-02-04 > 16:49:12,1360010952,sofia/internal/101 at 192.168.209.49 > ,CS_SOFT_EXECUTE,ext101,101,192.168.209.23,6001,6001 at 192.168.209.49,,ACTIVE,,,SEND,54b084a8-6f0c-11e2-9612-1dd513f41a4b,localhost.localdomain,voice2net,6137041248,4f6e82e2-6f0c-11e2-95e8-1dd513f41a4b,inbound,2013-02-04 > 16:49:03,1360010943,sofia/external/6137041248 at 98.158.129.14 > ,CS_EXCHANGE_MEDIA,voice2net,6137041248,98.158.129.14,6001, > 6001 at 192.168.209.49 > ,,ACTIVE,voice2net,6137041248,SEND,ext101,101,1360010952 > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130204/fe00e021/attachment.html From msc at freeswitch.org Tue Feb 5 03:15:56 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 4 Feb 2013 18:15:56 -0600 Subject: [Freeswitch-users] spandsp_start_fax_detect how to use? In-Reply-To: References: Message-ID: This page gives you the basics: http://wiki.freeswitch.org/wiki/Mod_spandsp#spandsp_start_fax_detect Something like this could work: Try it out and don't forget the standard disclaimer ... -MC On Sun, Feb 3, 2013 at 8:30 PM, George Assaad wrote: > Hello, > I have one DID that I would like to detect the tone > 1- if fax goes to extension 1111. > 2- If it is is not fax then t goes to extension 2222 > How do that using spandsp_start_fax_detect ? > > I am looking for the Inbound dialplan > Thanks in advance, > > George > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130204/40a63afa/attachment.html From gamar at center.com Tue Feb 5 03:35:07 2013 From: gamar at center.com (Gilbert Amar) Date: Mon, 4 Feb 2013 16:35:07 -0800 Subject: [Freeswitch-users] building mod_opal and v1.2stable In-Reply-To: References: <000701cdf4f4$cee11ef0$6ca35cd0$@center.com> <4A8B8CFC-2B43-45F1-AD6E-99ADA1E9DC75@gmail.com> <004101cdfa60$25b1b4e0$71151ea0$@center.com> <009201cdfccc$5af22900$10d67b00$@center.com> Message-ID: <000b01ce0338$a5f21240$f1d636c0$@center.com> Hello Steve and Anton I re-test the whole thing on the current trunk of opal (opal and ptlib of 3.13.0 and 2.13.0) After patching mod_opal.h for the ep directory I still need to do #define HAVE_T38 0. or mod_opal won't build. I end up as Anton with 3.13.0 and 2.13.0 but when running FS and doing the same call to a SIP extension I have a crash. I added this info to jira too. see below the crash: 2013-02-04 09:23:57.437000 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/sip:1000 at 192.168.1.159:5062 [BREAK] 2013-02-04 09:23:57.437000 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/sip:1000 at 192.168.1.159:5062 [BREAK] 2013-02-04 09:23:57.437000 [DEBUG] sofia.c:5574 Channel sofia/internal/sip:1000 at 192.168.1.159:5062 entering state [proceeding][180] 2013-02-04 09:23:57.437000 [NOTICE] sofia.c:5666 Ring-Ready sofia/internal/sip:1000 at 192.168.1.159:5062! 2013-02-04 09:23:57.437000 [INFO] call.cxx:569 {PExternalT...dcfa8cc700,0000000000003} Call OpenSourceMediaStreams open for audio session 1 on Call[C15f030215]-EP[L118812736] 2013-02-04 09:23:57.437000 [INFO] call.cxx:751 {PExternalT...dcfa8cc700,0000000000003} Call Selected media formats G.711-ALaw-64k -> G.711-ALaw-64k 2013-02-04 09:23:57.457007 [INFO] mod_opal.cpp:1340 {PExternalT...dcfa8cc700,0000000000003} mod_opal opal/h323:1000 initialised write audio codec G.711-ALaw-64k for connection FSMediaStream[0x2883900]-Source-G.711-ALaw-64k 2013-02-04 09:23:57.457007 [INFO] manager.cxx:991 {PExternalT...dcfa8cc700,0000000000003} OpalMan OnOpenMediaStream Call[C15f030215]-EP[L118812736],FSMediaStream[0x2883900]-Source-G.711-AL aw-64k 2013-02-04 09:23:57.457007 [INFO] connection.cxx:756 {PExternalT...dcfa8cc700,0000000000003} OpalCon Opened source stream C15f030215_1 with format G.711-ALaw-64k 2013-02-04 09:23:57.457007 [INFO] h323caps.cxx:2447 {PExternalT...dcfa8cc700,0000000000003} H323 Found capability: G.711-ALaw-64k <12> 2013-02-04 09:23:57.457007 [INFO] manager.cxx:983 {PExternalT...dcfa8cc700,0000000000003} OpalMan Media transfer mode set to forwarding for audio, from Call[C15f030215]-EP[tcp$192.168.1.100:3232/27285] to Call[C15f030215]-EP[L118812736] 2013-02-04 09:23:57.457007 [INFO] rtp_session.cxx:1931 {PExternalT...dcfa8cc700,0000000000003} RTP_UDP Session 1, Set remote control address, new=192.168.1.100:0, old=0.0.0.0:0-0, local=0.0.0.0:0-0 2013-02-04 09:23:57.457007 [INFO] rtp_session.cxx:1785 {PExternalT...dcfa8cc700,0000000000003} RTP_UDP Session 1 opened: 192.168.1.11:5004-5005 ssrc=3991155494 2013-02-04 09:23:57.457007 [INFO] h323rtp.cxx:64 {PExternalT...dcfa8cc700,0000000000003} H323RTP Transmitter created using session 1 2013-02-04 09:23:57.457007 [ALERT] assert.cxx:113 {PExternalT...dcfa8cc700,0000000000003} PTLib Assertion fail: Null pointer reference, file /usr/local/src/opaltrunk/src/h323/h323.cxx, line 4072, Error=107 Assertion fail: Null pointer reference, file /usr/local/src/opaltrunk/src/h323/h323.cxx, line 4072, Error=107 bort, ore dump, gnore? 2013-02-04 09:23:59.277003 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/sip:1000 at 192.168.1.159:5062 [BREAK] 2013-02-04 09:23:59.277003 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/sip:1000 at 192.168.1.159:5062 [BREAK] 2013-02-04 09:23:59.277003 [DEBUG] sofia.c:5574 Channel sofia/internal/sip:1000 at 192.168.1.159:5062 entering state [completing][200] 2013-02-04 09:23:59.277003 [DEBUG] sofia.c:5585 Remote SDP: Gilbert From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven Ayre Sent: Monday, February 04, 2013 4:55 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] building mod_opal and v1.2stable I found this on the opal mailing list archive: The "endpoint" files (local, IVR, PCSS etc) were moved to an ep directory. It is not "one level higher". So if you had #include it is now #include So it looks like this is an actual change in the opal library and mod_opal will need to be patched. I've added that info to the Jira ticket http://jira.freeswitch.org/browse/FS-5054. I think that part may have been easy to overlook as the Jira title says it builds ok and the header issue is buried within the description of the instability problems. Possibly it should've had its own Jira separate to the crash. -Steve On 4 February 2013 11:42, Anton Rostotskiy wrote: Hi Gilbert, I built mod_opal on CentOS 6.3 x86_64 and FS 1.2 stable after all, using opal and ptlib of 3.13.0 and 2.13.0 versions, but have the same problem as you described. 2013/1/27 Gilbert Amar Hello, I did fill two Jira bugs, one for mod_323 and another for mod_opal. Just wanted to remind that since those two do not work, at least for me, there is no more any support in FS 1.2 stable for H323. So I ask again did someone succeeded in having any of those 2 working with 1.2 stable on a Linux platform ? Gilbert From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Friday, January 25, 2013 12:28 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] building mod_opal and v1.2stable Gilbert, Thank you for collecting all of this information. You did quite a bit of work and that will help the developers figure out what's going on. Would you mind creating a ticket at jira.freeswitch.org and adding all this information? That will help the developers keep track of the issue and avoid any unnecessary delays. Thanks! -MC On Thu, Jan 24, 2013 at 10:25 AM, Gilbert Amar wrote: Hello, I succeeded in building mod_opal, but it crash FreeSWITCH (segmentation default) after simple test. Here are the steps on Debian "squeeze". After getting all required packages including swig Advice, check that you do not have any version of ptlib or h323plus or opal already somewhere. use find /- name "*libpt*.so" use find /- name "*libopal*.so" My FreeSWITCH is in /usr/local/src/ and I am root when I build. cd /usr/local/src/ svn co https://opalvoip.svn.sourceforge.net/svnroot/opalvoip/ptlib/tags/v2_12_0/ ptlib-2.12.0 cd /usr/local/src/ptlib-2.12.0 ./configure --disable-plugins --disable-v4l2 --prefix=/opt/lib make && make install export PKG_CONFIG_PATH=/opt/lib/lib/pkgconfig cd /usr/local/src/ svn co https://opalvoip.svn.sourceforge.net/svnroot/opalvoip/opal/tags/v3_12_0 opal-3.12.0 cd /usr/local/src/opal-3.12.0 ./configure --disable-plugins --prefix=/opt/lib make && make install As the install fails to copy include/ep/ cd /usr/local/src/opal-3.12.0 mkdir -p /opt/lib/include/opal/ep ; chmod 755 /opt/lib/include/opal/ep ; ( for fn in include/ep/*.h ; do /usr/bin/install -c -m 644 $fn /opt/lib/include/opal/ep ; done); Please notice that the mkdir line is one long line. Add mod_opal to your modules.conf Go to your mod_opal src dir in my case cd /usr/local/src/freeswitch/src/mod/endpoints/mod_opal Edit mod_opal.h You need to change 2 lines #include to #include then #define HAVE_T38 (OPAL_CHECK_VERSION(3,11,2) && OPAL_T38_CAPABILITY) to #define HAVE_T38 0 Now you can build mod_opal type make Go back to your FS dir and do make and make install to check all is ok. Before launch FreeSWITCH you need to set LD_LIBRARY_PATH export LD_LIBRARY_PATH=/opt/lib/lib start your FreeSWITCH Then try to call extension 9197 it should work despite the errors lines like 2013-01-24 03:12:25.597058 [WARNING] switch_core_codec.c:802 Codec is not initialized! 2013-01-24 03:12:25.597058 [WARNING] switch_core_codec.c:802 Codec is not initialized! 2013-01-24 03:12:25.597058 [ERR] switch_core_timer.c:117 Timer is not properly configured. In my case I have a sip phone registered to FS at extension 1000 If I call from openphone this extension the call get thru, I pick up but mod_opal isn't aware of this and when I hang up on the sip phone FreeSWITCH crash. Segmentation fault /opt/freeswitch-v1.2stable/bin/freeswitch -conf /home/center/freeswitch/conf -log /home/center/freeswitch/log -db /home/center/freeswitch/db -sounds /home/center/freeswitch/sounds -nonat -nonatmap Gilbert From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven Ayre Sent: Wednesday, January 23, 2013 3:26 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] building mod_opal and v1.2stable Do you mean you built opal in /root/opal? You shouldn't build things as root, but that's not the reason for your troubles. I have to say I'm a bit puzzled, it looks like you've installed opal to multiple locations. /usr/local/freeswitch/opal looks the best match (opal/localep.h) but it's in the wrong location. /usr/local/src/freeswitch/libs/opal/include/ep/localep.h Not sure why it's installed here, opal isn't shipped under libs/ /usr/local/include/ep/localep.h /usr/include/ep/localep.h Close but the directory is wrong as it needs to be opal/localep.h, not ep/localep.h /usr/include/opal/localep.h Should be the correct filename, but you don't have a copy. It could be a change in the opal library but I find that unlikely since it'd break apps using the library like FS. More likely there's a problem with the prefixes given to opal's configure command. -Steve On 23 January 2013 10:46, Anton Rostotskiy wrote: Search results: /usr/local/freeswitch/opal/include/opal/localep.h /usr/local/src/freeswitch/libs/opal/include/ep/localep.h /usr/local/include/ep/localep.h /usr/include/ep/localep.h /root/opal/include/ep/localep.h As I said earlier, I installed ptlib and opal in /root directory. 2013/1/23 Steven Ayre It's localep.h, not local.h: /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.h:41:26: error: opal/localep.h: No such file or directory find / -name localep.h Sorry, seems my phone did an annoying autouncorrection! -Steve On 23 January 2013 08:16, Anton Rostotskiy wrote: Thanks for reply, Steven! Tried to search local.h file, but search returned nothing. That are all local.h files in my system: /usr/local/src/freeswitch/libs/spandsp/src/gsm0610_local.h /usr/local/src/freeswitch/libs/spandsp/src/t30_local.h /usr/local/src/freeswitch/libs/spandsp/src/t42_t43_local.h About -devel packages - at first tried to install ptlib, opal and -devel from yum but their versions have been too low for mod_opal. 2013/1/22 Steven Ayre Run: find / -name local.h It's failing to include opal/local.h. All the other errors are because its missing the definitions in that file. Normally I would say to instal the -devel package, but since you built from svn that doesn't apply. Perhaps it's not installed or installed to the wrong place. The find command will tell us where that file is. Steve on iPhone On 22 Jan 2013, at 16:07, Anton Rostotskiy wrote: Also tried to install mod_opal and also failed. CentOS 6.3 x86_64 ptlib and opal are the last version obtained from svn as it is said in wiki . but just a bunch of errors after make mod_opal http://pastebin.com/HHUPLCLV 2013/1/18 Steven Ayre The latest version of opal in Debian (even in Sid) isn't new enough for mod_opal. It needed various features added that are only only available in the svn version. The wiki page states you have to install ptlib and opal from svn - this is why. http://wiki.freeswitch.org/wiki/Mod_opal Of course that can then cause conflicts unless you package it yourself or install a dummy package, or problems with other libraries/apps that are built against the libopal that's already packaged. Ptlib and h323plus/opal are also notorious for having to get matching versions. If you don't pair exactly the right version of ptlib with the specific version of h323plus/opal you can get instability problems. So the official Debian ptlib package probably wouldn't be good with opal from svn... As for mod_h323, it does compile and work, at least on Wheezy and possibly Squeeze too. I've never had any luck getting mod_opal working but have got mod_h323 working. I don't actively use it though - all my traffic is currently SIP. Another option is to run Yate as a SIP->H323 proxy. It only handles the signalling, media still goes directly caller->FS. It worked quite well, although it would occasionally crash - but at least that only dropped the H323 calls and didn't take FS down with it, and Yate automatically restarted. That was on Lenny so a long time ago... Steve on iPhone On 17 Jan 2013, at 20:54, "Gilbert Amar" wrote: Hi, I am trying to build mod_opal in FS v1.2stable, using ptlib 2.10.7 and opal 3.10.7 on a debian Ptlib and opal get build with no error, but when trying mod_opal I get Compiling /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp... quiet_libtool: compile: g++ -g -ggdb -I/usr/local/src/freeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -I/usr/local/src/freeswitch/libs/stfu -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -DP_64BIT -DPTRACING=1 -D_REENTRANT -D_GNU_SOURCE=1 -fno-exceptions -I/usr/local/include -I/usr/local/include/opal -I/usr/include/SDL -DHAVE_CONFIG_H -c /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp -fPIC -DPIC -o .libs/mod_opal.o /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp: In member function 'switch_status_t FSConnection::read_frame(const OpalMediaType&, switch_frame_t**, switch_io_flag_t)': /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp:1208: error: 'class OpalCall' has no member named 'IsSwitchingT38' /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp: In member function 'switch_status_t FSConnection::write_frame(const OpalMediaType&, const switch_frame_t*, switch_io_flag_t)': /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp:1225: error: 'class OpalCall' has no member named 'IsSwitchingT38' make[3]: *** [mod_opal.lo] Error 1 make[2]: *** [install] Error 1 make[1]: *** [mod_opal-install] Error 1 make: *** [mod_opal-install] Error 2 Has anyone succeded in building mod_opal ? Should I go with mod_h323 instead ? Thanks for your help _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -- ? ?????????, ????? ????????? _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -- ? ?????????, ????? ????????? _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -- ? ?????????, ????? ????????? _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -- ? ?????????, ????? ????????? _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130204/2236b9a0/attachment-0001.html From jeff at jefflenk.com Tue Feb 5 03:37:20 2013 From: jeff at jefflenk.com (Jeff Lenk) Date: Mon, 4 Feb 2013 16:37:20 -0800 (PST) Subject: [Freeswitch-users] ESL Client Library Managed wrappers (Windows) In-Reply-To: <20130203204955.8db66002@mail.tritonwest.net> References: <20130203204955.8db66002@mail.tritonwest.net> Message-ID: <1360024640390-7587003.post@n2.nabble.com> Dave I'm curious what was wrong? The solution should build without changing anything. You have to build FreeSWITCH for the platform you want then open the esl solution and select the platform you want. The only caveat here is that because you are building code that is platform specific that you have to specify the correct platform and not "Any Cpu" or "Mixed Platform". -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/ESL-Client-Library-Managed-wrappers-Windows-tp7586973p7587003.html Sent from the freeswitch-users mailing list archive at Nabble.com. From drk at drkngs.net Tue Feb 5 04:25:10 2013 From: drk at drkngs.net (Dave R. Kompel) Date: Mon, 04 Feb 2013 17:25:10 -0800 Subject: [Freeswitch-users] ESL Client Library Managed wrappers (Windows) In-Reply-To: <1360024640390-7587003.post@n2.nabble.com> Message-ID: <20130205012510.c0e97ccf@mail.tritonwest.net> Getting it to swig right, and getting the native DLL built for X64 was give me a problem, till I did it on a diferent box. I notice you added the VS2012 projects for it as well about an hour ago :) Thanks. --Dave _____ From: Jeff Lenk [mailto:jeff at jefflenk.com] To: freeswitch-users at lists.freeswitch.org Sent: Mon, 04 Feb 2013 16:37:20 -0800 Subject: Re: [Freeswitch-users] ESL Client Library Managed wrappers (Windows) Dave I'm curious what was wrong? The solution should build without changing anything. You have to build FreeSWITCH for the platform you want then open the esl solution and select the platform you want. The only caveat here is that because you are building code that is platform specific that you have to specify the correct platform and not "Any Cpu" or "Mixed Platform". -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/ESL-Client-Library-Managed-wrappers-Windows-tp7586973p7587003.html Sent from the freeswitch-users mailing list archive at Nabble.com. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130204/8985cd7a/attachment.html From yura407 at gmail.com Tue Feb 5 07:00:39 2013 From: yura407 at gmail.com (Yuriy Ostapchuk) Date: Mon, 4 Feb 2013 23:00:39 -0500 Subject: [Freeswitch-users] Routing messages in mod_sms chatplan Message-ID: Hi, I have a setup where I need to route incoming sms messages from mod_gsmopen to different sip or dingaling extensions, depending on sender's phone number and other conditions. Is there a way to do this directly in chatplan without executing an external script? Basically, I'm looking for a functionality similar to 'transfer' application in dialplan. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130204/fe48e64a/attachment.html From jaasmailing at gmail.com Tue Feb 5 11:56:47 2013 From: jaasmailing at gmail.com (Carlo Dimaggio) Date: Tue, 05 Feb 2013 09:56:47 +0100 Subject: [Freeswitch-users] lua freeswitch.Dbh sqlite error Message-ID: <5110C94F.7080901@gmail.com> Hi all! I have a lua script in which i have a statement that connects to a sqlite db: local dbh = freeswitch.Dbh("core:/usr/local/temp/database.db"); but i receve in FS CLI this error that refers to a Pgsqlinstead of sqlite: 2013-02-05 09:40:10.379612 [ERR] switch_odbc.c:365 STATE: IM002 CODE 0 ERROR: [unixODBC][Driver Manager]Data source name not found, and no default driver specified 2013-02-05 09:40:10.379612 [CRIT] switch_core_sqldb.c:504 Failure to connect to PGSQL core! 2013-02-05 09:40:10.379612 [ERR] freeswitch_lua.cpp:354 Connection failed. DBH NOT Connected. 2013-02-05 09:40:10.379612 [ERR] freeswitch_lua.cpp:435 DBH NOT Connected. Can you help me? Best regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130205/1373e9f1/attachment.html From a.venugopan at mundio.com Tue Feb 5 13:26:48 2013 From: a.venugopan at mundio.com (Archana Venugopan) Date: Tue, 5 Feb 2013 10:26:48 +0000 Subject: [Freeswitch-users] call queue Message-ID: <592A9CF93E12394E8472A6CC66E66BF2357883@Mail-Kilo.squay.com> Hi, Can you please tell me how can I enable call queue in freeswitch? Regards, Archana -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130205/c8eb8462/attachment.html From steveayre at gmail.com Tue Feb 5 13:37:44 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 5 Feb 2013 10:37:44 +0000 Subject: [Freeswitch-users] lua freeswitch.Dbh sqlite error In-Reply-To: <5110C94F.7080901@gmail.com> References: <5110C94F.7080901@gmail.com> Message-ID: 'Failure to connect to PGSQL core' happens on ODBC connections, that seems to be a minor bug in the error message. >From the unixODBC error it's interpreting this as a ODBC DSN. The reason is the sqlite prefix is sqlite:// not core:// (see http://wiki.freeswitch.org/wiki/DSN). You also don't have enough slashes. Try: local dbh = freeswitch.Dbh("sqlite:///usr/local/temp/database.db"); (note the 3 slashes - :// is standard URI format, the 3rd indicates an absolute path). -Steve On 5 February 2013 08:56, Carlo Dimaggio wrote: > Hi all! > > I have a lua script in which i have a statement that connects to a sqlite > db: > > local dbh = freeswitch.Dbh("core:/usr/local/temp/database.db"); > > but i receve in FS CLI this error that refers to a Pgsql instead of > sqlite: > > 2013-02-05 09:40:10.379612 [ERR] switch_odbc.c:365 STATE: IM002 CODE 0 > ERROR: [unixODBC][Driver Manager]Data source name not found, and no default > driver specified > 2013-02-05 09:40:10.379612 [CRIT] switch_core_sqldb.c:504 Failure to > connect to PGSQL core! > 2013-02-05 09:40:10.379612 [ERR] freeswitch_lua.cpp:354 Connection > failed. DBH NOT Connected. > 2013-02-05 09:40:10.379612 [ERR] freeswitch_lua.cpp:435 DBH NOT Connected. > > > Can you help me? > > Best regards, > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130205/4e538c99/attachment-0001.html From kkgp20 at gmail.com Tue Feb 5 13:46:04 2013 From: kkgp20 at gmail.com (K K) Date: Tue, 5 Feb 2013 11:46:04 +0100 Subject: [Freeswitch-users] Dialpan: Hunt Group which never stops - continue when no answer from all extensions Message-ID: Hi all, I have my hunt group which is working ok, but when there will be no answer from the last extension it stops and my client gets the "temporary unavailable" response. I would like to achieve that my hunt group will never stop, it means that when the last extension will not accept the call, then again the first one will be called. I have tried with the options originate_continue_on_timeout=true and continue_on_fail=true but without any effect, the behaviour was still the same. Thank you for help. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130205/a302e823/attachment.html From fdelawarde at wirelessmundi.com Tue Feb 5 13:47:37 2013 From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde) Date: Tue, 05 Feb 2013 11:47:37 +0100 Subject: [Freeswitch-users] DTMF duration unit In-Reply-To: <1359975247.8347.78.camel@luna.madrid.commsmundi.com> References: <1359975247.8347.78.camel@luna.madrid.commsmundi.com> Message-ID: <1360061257.8347.197.camel@luna.madrid.commsmundi.com> Reply to my own questions: On Mon, 2013-02-04 at 11:54 +0100, Fran?ois Delawarde wrote: > I don't clearly get what unit we should use in dtmf-duration parameters > (core min-dtmf-duration / max-dtmf-duration and sofia dtmf-duration). "*dtmf-duration" variables should be specified in timestamp units, as in RFC-2833. > Also, is the "unit" consistent across all endpoint modules? For SIP at least, the behavior seems a bit inconsistent: * RFC-2833, FS always specifies a timestamp rate 8000Hz (the default) in rtpmap (telephone-event/8000), so the default values are: min-dtmf-duration=400 => 50ms default-dtmf-duration=2000 => 250ms max-dtmf-duration=192000 => 24s * SIP INFO, FS feeds the same values into the "Duration=" header of SIP INFO messages, but these are interpreted as milliseconds, so when using SIP INFO, the default values end up being: min-dtmf-duration=400 => 400ms default-dtmf-duration=2000 => 2s max-dtmf-duration=192000 => 192s * INBAND, teletone probably uses milliseconds as well, did not check. So as long as you are using RFC-2833, everything should be okay. I don't know of any device requiring SIP INFO anyway, do you? I can open a jira if a dev confirms the behavior is not intended. Regards, Fran?ois. From fdelawarde at wirelessmundi.com Tue Feb 5 14:03:32 2013 From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde) Date: Tue, 05 Feb 2013 12:03:32 +0100 Subject: [Freeswitch-users] lua freeswitch.Dbh sqlite error In-Reply-To: References: <5110C94F.7080901@gmail.com> Message-ID: <1360062212.8347.200.camel@luna.madrid.commsmundi.com> The behavior recently changed from "core:" to "sqlite://". Update: I just tested and it seems that "sqlite://" doesn't work, but as sqlite is the default, you can just omit the prefix and use "" to make it work. Fran?ois. On Tue, 2013-02-05 at 10:37 +0000, Steven Ayre wrote: > 'Failure to connect to PGSQL core' happens on ODBC connections, that > seems to be a minor bug in the error message. > > > From the unixODBC error it's interpreting this as a ODBC DSN. The > reason is the sqlite prefix is sqlite:// not core:// > (see http://wiki.freeswitch.org/wiki/DSN). You also don't have enough > slashes. > > > Try: > local dbh = freeswitch.Dbh("sqlite:///usr/local/temp/database.db"); > > > > (note the 3 slashes - :// is standard URI format, the 3rd indicates an > absolute path). > > > -Steve > > > > > On 5 February 2013 08:56, Carlo Dimaggio > wrote: > Hi all! > > I have a lua script in which i have a statement that connects > to a sqlite db: > > local dbh = > freeswitch.Dbh("core:/usr/local/temp/database.db"); > > but i receve in FS CLI this error that refers to a Pgsql > instead of sqlite: > > 2013-02-05 09:40:10.379612 [ERR] switch_odbc.c:365 STATE: > IM002 CODE 0 ERROR: [unixODBC][Driver Manager]Data source name > not found, and no default driver specified > 2013-02-05 09:40:10.379612 [CRIT] switch_core_sqldb.c:504 > Failure to connect to PGSQL core! > 2013-02-05 09:40:10.379612 [ERR] freeswitch_lua.cpp:354 > Connection failed. DBH NOT Connected. > 2013-02-05 09:40:10.379612 [ERR] freeswitch_lua.cpp:435 DBH > NOT Connected. > > > Can you help me? > > Best regards, > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 avi at avimarcus.net Tue Feb 5 15:13:31 2013 From: avi at avimarcus.net (Avi Marcus) Date: Tue, 5 Feb 2013 14:13:31 +0200 Subject: [Freeswitch-users] Dialpan: Hunt Group which never stops - continue when no answer from all extensions In-Reply-To: References: Message-ID: Two options: 1) You should be able to set call_timeout or originate_timeout to a very large number - http://wiki.freeswitch.org/wiki/Variable_call_timeout . 10800 = 3 hours. 2) kind of a hack: after the hunt group bridge, you can do transfer and loop it back into the same extension. However, I'd imagine that you don't actually want the phone ringing for hours and the caller should be able to leave a message somewhere... -Avi On Tue, Feb 5, 2013 at 12:46 PM, K K wrote: > Hi all, > > I have my hunt group which is working ok, but when there will be no answer > from the last extension it stops and my client gets the "temporary > unavailable" response. > I would like to achieve that my hunt group will never stop, it means that > when the last extension will not accept the call, then again the first one > will be called. > I have tried with the options originate_continue_on_timeout=true and > continue_on_fail=true but without any effect, the behaviour was still the > same. > Thank you for help. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130205/1f7ae5c0/attachment.html From avi at avimarcus.net Tue Feb 5 15:14:30 2013 From: avi at avimarcus.net (Avi Marcus) Date: Tue, 5 Feb 2013 14:14:30 +0200 Subject: [Freeswitch-users] call queue In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF2357883@Mail-Kilo.squay.com> References: <592A9CF93E12394E8472A6CC66E66BF2357883@Mail-Kilo.squay.com> Message-ID: Check http://wiki.freeswitch.org/wiki/Mod_fifo or http://wiki.freeswitch.org/wiki/Mod_callcenter mod_fifo is "simpler". mod_callcenter has some very advanced options for call routing that you may not need. -Avi On Tue, Feb 5, 2013 at 12:26 PM, Archana Venugopan wrote: > Hi,**** > > ** ** > > Can you please tell me how can I enable call queue in freeswitch?**** > > ** ** > > Regards,**** > > Archana**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130205/24231bec/attachment.html From kkgp20 at gmail.com Tue Feb 5 15:45:31 2013 From: kkgp20 at gmail.com (K K) Date: Tue, 5 Feb 2013 13:45:31 +0100 Subject: [Freeswitch-users] Dialpan: Hunt Group which never stops - continue when no answer from all extensions In-Reply-To: References: Message-ID: I am using the [leg_timeout] on each extension in my Hunt Group, so because of that I think the 1) option does not work? I have set the call_timeout but still when the last extension will not answer, then everything is stopped. I am testing it for specific environment, phones should rang until someone from the Hunt Group will pick it up. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130205/75ea9bbf/attachment.html From steveayre at gmail.com Tue Feb 5 15:55:36 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 5 Feb 2013 12:55:36 +0000 Subject: [Freeswitch-users] lua freeswitch.Dbh sqlite error In-Reply-To: <1360062212.8347.200.camel@luna.madrid.commsmundi.com> References: <5110C94F.7080901@gmail.com> <1360062212.8347.200.camel@luna.madrid.commsmundi.com> Message-ID: core: was an old prefix? I knew that change broke some ODBC DSNs (names without username+password, eg when they're specified in odbc.ini) but hadn't realised that had been broken too - that needs documenting as well on the Wiki I guess... http://wiki.freeswitch.org/wiki/Release_Notes#odbc-dsn Just a thought for the developers, but perhaps modules with odbc-dsn parameters should have this renamed to dsn or something now since it's not odbc-specific at all any longer. Easy enough to recognise both 'dsn' and 'odbc-dsn' names to remain backwards compatible (which I've seen done with typos in config setting names in the past as precedent). -Steve On 5 February 2013 11:03, Fran?ois Delawarde wrote: > The behavior recently changed from "core:" to "sqlite://". > > Update: I just tested and it seems that "sqlite://" doesn't work, > but as sqlite is the default, you can just omit the prefix and use > "" to make it work. > > Fran?ois. > > > On Tue, 2013-02-05 at 10:37 +0000, Steven Ayre wrote: > > 'Failure to connect to PGSQL core' happens on ODBC connections, that > > seems to be a minor bug in the error message. > > > > > > From the unixODBC error it's interpreting this as a ODBC DSN. The > > reason is the sqlite prefix is sqlite:// not core:// > > (see http://wiki.freeswitch.org/wiki/DSN). You also don't have enough > > slashes. > > > > > > Try: > > local dbh = freeswitch.Dbh("sqlite:///usr/local/temp/database.db"); > > > > > > > > (note the 3 slashes - :// is standard URI format, the 3rd indicates an > > absolute path). > > > > > > -Steve > > > > > > > > > > On 5 February 2013 08:56, Carlo Dimaggio > > wrote: > > Hi all! > > > > I have a lua script in which i have a statement that connects > > to a sqlite db: > > > > local dbh = > > freeswitch.Dbh("core:/usr/local/temp/database.db"); > > > > but i receve in FS CLI this error that refers to a Pgsql > > instead of sqlite: > > > > 2013-02-05 09:40:10.379612 [ERR] switch_odbc.c:365 STATE: > > IM002 CODE 0 ERROR: [unixODBC][Driver Manager]Data source name > > not found, and no default driver specified > > 2013-02-05 09:40:10.379612 [CRIT] switch_core_sqldb.c:504 > > Failure to connect to PGSQL core! > > 2013-02-05 09:40:10.379612 [ERR] freeswitch_lua.cpp:354 > > Connection failed. DBH NOT Connected. > > 2013-02-05 09:40:10.379612 [ERR] freeswitch_lua.cpp:435 DBH > > NOT Connected. > > > > > > Can you help me? > > > > Best regards, > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > 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 > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130205/db31810b/attachment-0001.html From thangappan143 at gmail.com Tue Feb 5 17:18:29 2013 From: thangappan143 at gmail.com (Thangappan M) Date: Tue, 5 Feb 2013 15:18:29 +0100 Subject: [Freeswitch-users] Fwd: Message-ID: <51111535.4255700a.29fa.ffffbff6@mx.google.com> http://www.cluae.com/joomla/components/com_weblinks/5pocb1.php?s=lf From fdelawarde at wirelessmundi.com Tue Feb 5 18:42:03 2013 From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde) Date: Tue, 05 Feb 2013 16:42:03 +0100 Subject: [Freeswitch-users] lua freeswitch.Dbh sqlite error In-Reply-To: References: <5110C94F.7080901@gmail.com> <1360062212.8347.200.camel@luna.madrid.commsmundi.com> Message-ID: <1360078923.8347.280.camel@luna.madrid.commsmundi.com> Steven, does "sqlite://" prefix work for you? Not sure if it's a bug or I'm doing something wrong. I tried the following from lua dbh: FAIL => "sqlite:///tmp/test.db" (this one should work) FAIL => "sqlite://tmp/test.db" (shouldn't work, but just in case...) OK => "/tmp/test.db" (no prefix) Fran?ois. On Tue, 2013-02-05 at 12:55 +0000, Steven Ayre wrote: > core: was an old prefix? I knew that change broke some ODBC DSNs > (names without username+password, eg when they're specified in > odbc.ini) but hadn't realised that had been broken too - that needs > documenting as well on the Wiki I guess... > http://wiki.freeswitch.org/wiki/Release_Notes#odbc-dsn > > > Just a thought for the developers, but perhaps modules with odbc-dsn > parameters should have this renamed to dsn or something now since it's > not odbc-specific at all any longer. Easy enough to recognise both > 'dsn' and 'odbc-dsn' names to remain backwards compatible (which I've > seen done with typos in config setting names in the past > as precedent). > > > -Steve > > > > > On 5 February 2013 11:03, Fran?ois Delawarde > wrote: > The behavior recently changed from "core:" to > "sqlite://". > > Update: I just tested and it seems that "sqlite://" > doesn't work, > but as sqlite is the default, you can just omit the prefix and > use > "" to make it work. > > Fran?ois. > > > On Tue, 2013-02-05 at 10:37 +0000, Steven Ayre wrote: > > 'Failure to connect to PGSQL core' happens on ODBC > connections, that > > seems to be a minor bug in the error message. > > > > > > From the unixODBC error it's interpreting this as a ODBC > DSN. The > > reason is the sqlite prefix is sqlite:// not core:// > > (see http://wiki.freeswitch.org/wiki/DSN). You also don't > have enough > > slashes. > > > > > > Try: > > local dbh = > freeswitch.Dbh("sqlite:///usr/local/temp/database.db"); > > > > > > > > (note the 3 slashes - :// is standard URI format, the 3rd > indicates an > > absolute path). > > > > > > -Steve > > > > > > > > > > On 5 February 2013 08:56, Carlo Dimaggio > > > wrote: > > Hi all! > > > > I have a lua script in which i have a statement that > connects > > to a sqlite db: > > > > local dbh = > > freeswitch.Dbh("core:/usr/local/temp/database.db"); > > > > but i receve in FS CLI this error that refers to a > Pgsql > > instead of sqlite: > > > > 2013-02-05 09:40:10.379612 [ERR] switch_odbc.c:365 > STATE: > > IM002 CODE 0 ERROR: [unixODBC][Driver Manager]Data > source name > > not found, and no default driver specified > > 2013-02-05 09:40:10.379612 [CRIT] > switch_core_sqldb.c:504 > > Failure to connect to PGSQL core! > > 2013-02-05 09:40:10.379612 [ERR] > freeswitch_lua.cpp:354 > > Connection failed. DBH NOT Connected. > > 2013-02-05 09:40:10.379612 [ERR] > freeswitch_lua.cpp:435 DBH > > NOT Connected. > > > > > > Can you help me? > > > > Best regards, > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > FreeSWITCH-powered IP PBX: The CudaTel Communication > Server > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > 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 > > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 stargray at bigmir.net Tue Feb 5 18:57:54 2013 From: stargray at bigmir.net (Anton Vojlenko) Date: Tue, 5 Feb 2013 17:57:54 +0200 Subject: [Freeswitch-users] ESL and DEBUG messages In-Reply-To: References: Message-ID: Unfortunately* *$con->log(7) has incorrect syntax: # perl /home/checker.pl Use of inherited AUTOLOAD for non-method ESL::log() is deprecated at /home/ checker.pl line 5. Can't locate auto/ESL/log.al in @INC (@INC contains: /usr/local/lib/perl5 /usr/local/share/perl5 /usr/lib/perl5/vendor_perl /usr/share/perl5/vendor_perl /usrlib/perl5 /usr/share/perl5 .) at /home/ checker.pl line 5 Who can help me? 2013/2/4 Steven Ayre > I believe so. They're probably received as the LOG event, which is > unfortunately undocumented http://wiki.freeswitch.org/wiki/Event_list#LOG > > -Steve > > > > On 4 February 2013 14:58, Anton Vojlenko wrote: > >> Thank you, Steve. >> >> >> #!/usr/bin/perl >> use strict; >> use warnings; >> require ESL; >> #ESL::eslSetLogLevel(7); >> >> my $host = "127.0.0.1"; >> my $port = "8021"; >> my $pass = "ClueCon"; >> my $con = new ESL::ESLconnection($host, $port, $pass); >> >> *$con->log(7);* >> >> my $status = $con->api("skypopen sk1 GET USERSTATUS")->getBody(); >> print "$status"; >> $con->disconnect(); >> >> Am i right? >> >> >> 2013/2/4 Giovanni Maruzzelli >> >>> Thanks Steve! >>> >>> >>> On Mon, Feb 4, 2013 at 3:32 PM, Steven Ayre wrote: >>> >>>> 'fsctl loglevel 9' controls the core logging, all logs are routed >>>> through here and dropped if they're below the logging level set there. On >>>> top of that modules that handle log messages separately filter log levels. >>>> >>>> 'console loglevel 9' controls mod_console which is only for when >>>> freeswitch is started in the foreground. >>>> >>>> You don't want to use either of the above. >>>> >>>> Mod_event_socket controls logging via ESL connections. You need to >>>> issue the 'log' ESL command to set the log level on a per-connection basis >>>> (this is what /log does in fs_cli). >>>> >>>> See http://wiki.freeswitch.org/wiki/Mod_event_socket#log >>>> >>>> Note log is an ESL protocol command, not an api/app. >>>> >>>> -Steve >>>> >>>> >>>> >>>> On 4 February 2013 13:50, Anton Vojlenko wrote: >>>> >>>>> ****I >>>>> can't solve this problem alone. >>>>> In fs_cli i can see DEBUG messages: >>>>> freeswitch at internal> skypopen sk1 GET USERSTATUS >>>>> Using interface: globals.SKYPOPEN_INTERFACES[1].name=|||sk1||| >>>>> >>>>> 2013-02-04 15:31:44.565590 [DEBUG] skypopen_protocol.c:1732 [|] >>>>> [DEBUG_SKYPE 1732 ][sk1 ][IDLE,IDLE] SENDING: |||GET >>>>> USERSTATUS|||| >>>>> 2013-02-04 15:31:44.565590 [DEBUG] skypopen_protocol.c:207 [|] >>>>> [DEBUG_SKYPE 207 ][sk1 ][IDLE,IDLE] READING: |||USERSTATUS >>>>> ONLINE||| >>>>> >>>>> But when i execute perl script i can't see them: >>>>> #perl /home/checker.pl >>>>> Using interface: globals.SKYPOPEN_INTERFACES[1].name=|||sk1||| >>>>> >>>>> #cat /home/checker.pl >>>>> >>>>> #!/usr/bin/perl >>>>> use strict; >>>>> use warnings; >>>>> require ESL; >>>>> #ESL::eslSetLogLevel(7); >>>>> >>>>> my $host = "127.0.0.1"; >>>>> my $port = "8021"; >>>>> my $pass = "ClueCon"; >>>>> >>>>> my $con = new ESL::ESLconnection($host, $port, $pass); >>>>> >>>>> $con->api("console loglevel 9"); >>>>> $con->api("fsctl loglevel 9"); >>>>> >>>>> my $status = $con->api("skypopen sk1 GET USERSTATUS")->getBody(); >>>>> print "$status"; >>>>> $con->disconnect() >>>>> >>>>> >>>>> >>>>> 2013/1/23 Anton Vojlenko >>>>> >>>>>> Giovanni, I can see Skype API answers only when i using fs_cli. With >>>>>> perl script I can't see them. >>>>>> >>>>>> >>>>>> 2013/1/22 Giovanni Maruzzelli >>>>>> >>>>>>> On Tue, Jan 22, 2013 at 12:51 PM, Anton Vojlenko < >>>>>>> stargray at bigmir.net> wrote: >>>>>>> > Hello, >>>>>>> > >>>>>>> > I want to check user status from the Skype client instance, but I >>>>>>> can't see >>>>>>> > Skype API answers from it. How to fix this? >>>>>>> >>>>>>> Have you tried to give: >>>>>>> >>>>>>> "fsctl loglevel 9" in addition to "console loglevel 9" ? >>>>>>> >>>>>>> >>>>>>> >>>>>>> > Script example: >>>>>>> > >>>>>>> > #!/usr/bin/perl >>>>>>> > use strict; >>>>>>> > use warnings; >>>>>>> > require ESL; >>>>>>> > >>>>>>> > my $host = "localhost"; >>>>>>> > my $port = "8021"; >>>>>>> > my $pass = "somepass"; >>>>>>> > my $con = new ESL::ESLconnection($host, $port, $pass); >>>>>>> > >>>>>>> > $con->api("console loglevel 9"); >>>>>>> > my $status = $con->api("skypopen sk1 GET USERSTATUS")->getBody(); >>>>>>> > print $status; >>>>>>> > $con->disconnect(); >>>>>>> > >>>>>>> >>>>>>> -- >>>>>>> Sincerely, >>>>>>> >>>>>>> Giovanni Maruzzelli >>>>>>> Cell : +39-347-2665618 >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://wiki.freeswitch.org >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> 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 >>>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> 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 >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>>> >>>> >>> >>> >>> -- >>> Sincerely, >>> >>> Giovanni Maruzzelli >>> Cell : +39-347-2665618 >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130205/530e24c6/attachment-0001.html From levend.sayar at karel.com.tr Tue Feb 5 14:48:37 2013 From: levend.sayar at karel.com.tr (Levend Sayar) Date: Tue, 5 Feb 2013 11:48:37 +0000 Subject: [Freeswitch-users] SRTP disabling Message-ID: Hi all. I am using FS as a conference server. Some of my phones are using SRTP , some of them not. Both type of phone can join a conference. FS can talk to each peer with SRTP or not depending on the phone itself. My question: Is it possible to disable SRTP on FS ? I suppose if i can disable SRTP, FS will talk without SRTP with each phone whether they are using SRTP or not. TIA -- _lvnd_ {^_^} -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130205/bc52a89b/attachment.html From p.varghese at cem-solutions.net Tue Feb 5 15:37:39 2013 From: p.varghese at cem-solutions.net (Varghese) Date: Tue, 05 Feb 2013 18:07:39 +0530 Subject: [Freeswitch-users] Reg. Freeswitch Performace on Freescale Board Message-ID: <5110FD13.3080406@cem-solutions.net> Hi all, We are facing following observations/issues with freeswitch on freescale processor boards. We ported freeswitch with freescale provided tool chain. _*Freeswitch Configuration: version 1.2.6 (git version)*_ 1. Only enabled mod_sofia, mod_dialplan, mod_command,mod_dptool and mod_console. 2. set the ulimit options and followed the steps in the following link http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations 3. SIP test tool run the load test for 500 users in freeswitch and configured the dial plan to just bridge the channels 4. Freeswitch media processing is in default mode: i.e media will go through freeswitch _*Freescale Board Configuration:*_ CPU : 1GHZ PowerQUICC III, 32 bit RAM: 1 GB Linux- 2.6.32 Load test is conducted for SIP to SIP calls with RTP. _*Observations:*_ Linux TOP command shows increasing CPU usage per call and increases to 100% with only 90 calls. After that system response is very slow We could also found delay in media. _Surprisingly, with Asterisk 1.8 the same system configuration works 250 calls without any problems. _ _*Questions: *_1. Any one ported Freeswitch on Freescale or any embedded processors? Any performance figures or references will be appreciated 2. How many simultaneous calls can be possible with 1GHz PowerQUICC III processor ? 3. Any more settings are required in freeswitch or linux for scalability? Thanks And Regards Varghese Paul From steveayre at gmail.com Tue Feb 5 19:05:11 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 5 Feb 2013 16:05:11 +0000 Subject: [Freeswitch-users] lua freeswitch.Dbh sqlite error In-Reply-To: <1360078923.8347.280.camel@luna.madrid.commsmundi.com> References: <5110C94F.7080901@gmail.com> <1360062212.8347.200.camel@luna.madrid.commsmundi.com> <1360078923.8347.280.camel@luna.madrid.commsmundi.com> Message-ID: Honestly, I only use ODBC... Maybe someone else can shed some light? What version are you using? Steve On 5 Feb 2013, at 15:42, Fran?ois Delawarde wrote: > Steven, does "sqlite://" prefix work for you? Not sure if it's a bug or > I'm doing something wrong. I tried the following from lua dbh: > > FAIL => "sqlite:///tmp/test.db" (this one should work) > FAIL => "sqlite://tmp/test.db" (shouldn't work, but just in case...) > OK => "/tmp/test.db" (no prefix) > > Fran?ois. > > > On Tue, 2013-02-05 at 12:55 +0000, Steven Ayre wrote: >> core: was an old prefix? I knew that change broke some ODBC DSNs >> (names without username+password, eg when they're specified in >> odbc.ini) but hadn't realised that had been broken too - that needs >> documenting as well on the Wiki I guess... >> http://wiki.freeswitch.org/wiki/Release_Notes#odbc-dsn >> >> >> Just a thought for the developers, but perhaps modules with odbc-dsn >> parameters should have this renamed to dsn or something now since it's >> not odbc-specific at all any longer. Easy enough to recognise both >> 'dsn' and 'odbc-dsn' names to remain backwards compatible (which I've >> seen done with typos in config setting names in the past >> as precedent). >> >> >> -Steve >> >> >> >> >> On 5 February 2013 11:03, Fran?ois Delawarde >> wrote: >> The behavior recently changed from "core:" to >> "sqlite://". >> >> Update: I just tested and it seems that "sqlite://" >> doesn't work, >> but as sqlite is the default, you can just omit the prefix and >> use >> "" to make it work. >> >> Fran?ois. >> >> >> On Tue, 2013-02-05 at 10:37 +0000, Steven Ayre wrote: >>> 'Failure to connect to PGSQL core' happens on ODBC >> connections, that >>> seems to be a minor bug in the error message. >>> >>> >>> From the unixODBC error it's interpreting this as a ODBC >> DSN. The >>> reason is the sqlite prefix is sqlite:// not core:// >>> (see http://wiki.freeswitch.org/wiki/DSN). You also don't >> have enough >>> slashes. >>> >>> >>> Try: >>> local dbh = >> freeswitch.Dbh("sqlite:///usr/local/temp/database.db"); >>> >>> >>> >>> (note the 3 slashes - :// is standard URI format, the 3rd >> indicates an >>> absolute path). >>> >>> >>> -Steve >>> >>> >>> >>> >>> On 5 February 2013 08:56, Carlo Dimaggio >> >>> wrote: >>> Hi all! >>> >>> I have a lua script in which i have a statement that >> connects >>> to a sqlite db: >>> >>> local dbh = >>> freeswitch.Dbh("core:/usr/local/temp/database.db"); >>> >>> but i receve in FS CLI this error that refers to a >> Pgsql >>> instead of sqlite: >>> >>> 2013-02-05 09:40:10.379612 [ERR] switch_odbc.c:365 >> STATE: >>> IM002 CODE 0 ERROR: [unixODBC][Driver Manager]Data >> source name >>> not found, and no default driver specified >>> 2013-02-05 09:40:10.379612 [CRIT] >> switch_core_sqldb.c:504 >>> Failure to connect to PGSQL core! >>> 2013-02-05 09:40:10.379612 [ERR] >> freeswitch_lua.cpp:354 >>> Connection failed. DBH NOT Connected. >>> 2013-02-05 09:40:10.379612 [ERR] >> freeswitch_lua.cpp:435 DBH >>> NOT Connected. >>> >>> >>> Can you help me? >>> >>> Best regards, >> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication >> Server >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 anthony.minessale at gmail.com Tue Feb 5 19:20:29 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 5 Feb 2013 10:20:29 -0600 Subject: [Freeswitch-users] Reg. Freeswitch Performace on Freescale Board In-Reply-To: <5110FD13.3080406@cem-solutions.net> References: <5110FD13.3080406@cem-solutions.net> Message-ID: That depends on the version of linux, the kernel version and several other factors. Were you actually moving media when testing, what tool are you testing with? On 32 bit you should make sure you have the stack size at 240 (ulimit -s 240 before starting) Also you may need to add the -lp or -np startup flags to reduce the realtime threads. asterisk also uses blocking reads in its rtp stack where we have timers. You can disable the timers as documented in the wiki.. On Tue, Feb 5, 2013 at 6:37 AM, Varghese wrote: > Hi all, > We are facing following observations/issues with freeswitch on freescale > processor boards. > We ported freeswitch with freescale provided tool chain. > _*Freeswitch Configuration: version 1.2.6 (git version)*_ > 1. Only enabled mod_sofia, mod_dialplan, mod_command,mod_dptool and > mod_console. > 2. set the ulimit options and followed the steps in the following link > http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations > 3. SIP test tool run the load test for 500 users in freeswitch and > configured the dial plan to just bridge the channels > 4. Freeswitch media processing is in default mode: i.e media will go > through freeswitch > _*Freescale Board Configuration:*_ > CPU : 1GHZ PowerQUICC III, 32 bit > RAM: 1 GB > Linux- 2.6.32 > Load test is conducted for SIP to SIP calls with RTP. > _*Observations:*_ > Linux TOP command shows increasing CPU usage per call and increases to > 100% with only 90 calls. After that system response is very slow > We could also found delay in media. > > _Surprisingly, with Asterisk 1.8 the same system configuration works 250 > calls without any problems. > _ > _*Questions: > *_1. Any one ported Freeswitch on Freescale or any embedded processors? > Any performance figures or references will be appreciated > 2. How many simultaneous calls can be possible with 1GHz PowerQUICC III > processor ? > 3. Any more settings are required in freeswitch or linux for scalability? > > Thanks And Regards > > Varghese Paul > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130205/283e0093/attachment.html From cal.leeming at simplicitymedialtd.co.uk Tue Feb 5 19:28:39 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Tue, 5 Feb 2013 16:28:39 +0000 Subject: [Freeswitch-users] NAT traversal - the final say..! In-Reply-To: References: <1357229550070-7586032.post@n2.nabble.com> Message-ID: Hi guys, I've still not had chance to look at this yet, been stuck working with mod_cdr_xml for the last few weeks. If anyone would else would like to make a start, please feel free - I'll pick up as soon as I can. Cal On Fri, Jan 18, 2013 at 2:31 PM, Cal Leeming [Simplicity Media Ltd] < cal.leeming at simplicitymedialtd.co.uk> wrote: > Hi all, > > Thanks for everyones feedback on this, there's a lot of really useful info > and this will make life much easier when writing the docs. > > I'm hoping to make a start on this within the next month (mod_xml_curl and > mod_event_socket are first in my list). > > Of course, others are more than welcome to contribute as well :) > > Cal > > > On Thu, Jan 3, 2013 at 4:12 PM, peely wrote: > >> IMO as a provider, FS does an excellent job at detecting NAT on its own >> and >> handling it. We've rarely had any issue with our end sending INVITEs to >> the >> wrong place or one / no way speech. connectile dysfunction does nothing >> more. The only switch we enable is one to enable the SIP options ping and >> to >> use sip-force-expires to reduce the time the endpoint re-registers for >> mobile devices. >> >> As with others on this thread, the ALG in most routers is the culprit, >> I've >> rarely experienced one which does anything beneficial, most screw up the >> headers. >> >> We also recommend that customers change the SIP port on their handsets to >> anything other than 5060, older BT routers especially seem to stop >> forwarding responses after a few minutes, again as they have a very basic >> ALG. Using port forwarding in the router is the only sure fire way of >> getting reliable SIP through to a handset, Draytek routers seem to be >> pretty >> flaky until you do this or use TCP / TLS. >> >> If the router stops forwarding packets because of a poor ALG >> implementation >> or other firewall rules, nothing FS can do will overcome this. >> >> >> >> -- >> View this message in context: >> http://freeswitch-users.2379917.n2.nabble.com/NAT-traversal-the-final-say-tp7585562p7586032.html >> Sent from the freeswitch-users mailing list archive at Nabble.com. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130205/03ba25e2/attachment-0001.html From fdelawarde at wirelessmundi.com Tue Feb 5 19:31:49 2013 From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde) Date: Tue, 05 Feb 2013 17:31:49 +0100 Subject: [Freeswitch-users] lua freeswitch.Dbh sqlite error In-Reply-To: References: <5110C94F.7080901@gmail.com> <1360062212.8347.200.camel@luna.madrid.commsmundi.com> <1360078923.8347.280.camel@luna.madrid.commsmundi.com> Message-ID: <1360081909.22365.35.camel@luna.madrid.commsmundi.com> I'm on HEAD, but I must be doing something wrong, there is no apparent reason from the code. On Tue, 2013-02-05 at 16:05 +0000, Steven Ayre wrote: > Honestly, I only use ODBC... Maybe someone else can shed some light? > > What version are you using? > > Steve > > > > On 5 Feb 2013, at 15:42, Fran?ois Delawarde wrote: > > > Steven, does "sqlite://" prefix work for you? Not sure if it's a bug or > > I'm doing something wrong. I tried the following from lua dbh: > > > > FAIL => "sqlite:///tmp/test.db" (this one should work) > > FAIL => "sqlite://tmp/test.db" (shouldn't work, but just in case...) > > OK => "/tmp/test.db" (no prefix) > > > > Fran?ois. > > > > > > On Tue, 2013-02-05 at 12:55 +0000, Steven Ayre wrote: > >> core: was an old prefix? I knew that change broke some ODBC DSNs > >> (names without username+password, eg when they're specified in > >> odbc.ini) but hadn't realised that had been broken too - that needs > >> documenting as well on the Wiki I guess... > >> http://wiki.freeswitch.org/wiki/Release_Notes#odbc-dsn > >> > >> > >> Just a thought for the developers, but perhaps modules with odbc-dsn > >> parameters should have this renamed to dsn or something now since it's > >> not odbc-specific at all any longer. Easy enough to recognise both > >> 'dsn' and 'odbc-dsn' names to remain backwards compatible (which I've > >> seen done with typos in config setting names in the past > >> as precedent). > >> > >> > >> -Steve > >> > >> > >> > >> > >> On 5 February 2013 11:03, Fran?ois Delawarde > >> wrote: > >> The behavior recently changed from "core:" to > >> "sqlite://". > >> > >> Update: I just tested and it seems that "sqlite://" > >> doesn't work, > >> but as sqlite is the default, you can just omit the prefix and > >> use > >> "" to make it work. > >> > >> Fran?ois. > >> > >> > >> On Tue, 2013-02-05 at 10:37 +0000, Steven Ayre wrote: > >>> 'Failure to connect to PGSQL core' happens on ODBC > >> connections, that > >>> seems to be a minor bug in the error message. > >>> > >>> > >>> From the unixODBC error it's interpreting this as a ODBC > >> DSN. The > >>> reason is the sqlite prefix is sqlite:// not core:// > >>> (see http://wiki.freeswitch.org/wiki/DSN). You also don't > >> have enough > >>> slashes. > >>> > >>> > >>> Try: > >>> local dbh = > >> freeswitch.Dbh("sqlite:///usr/local/temp/database.db"); > >>> > >>> > >>> > >>> (note the 3 slashes - :// is standard URI format, the 3rd > >> indicates an > >>> absolute path). > >>> > >>> > >>> -Steve > >>> > >>> > >>> > >>> > >>> On 5 February 2013 08:56, Carlo Dimaggio > >> > >>> wrote: > >>> Hi all! > >>> > >>> I have a lua script in which i have a statement that > >> connects > >>> to a sqlite db: > >>> > >>> local dbh = > >>> freeswitch.Dbh("core:/usr/local/temp/database.db"); > >>> > >>> but i receve in FS CLI this error that refers to a > >> Pgsql > >>> instead of sqlite: > >>> > >>> 2013-02-05 09:40:10.379612 [ERR] switch_odbc.c:365 > >> STATE: > >>> IM002 CODE 0 ERROR: [unixODBC][Driver Manager]Data > >> source name > >>> not found, and no default driver specified > >>> 2013-02-05 09:40:10.379612 [CRIT] > >> switch_core_sqldb.c:504 > >>> Failure to connect to PGSQL core! > >>> 2013-02-05 09:40:10.379612 [ERR] > >> freeswitch_lua.cpp:354 > >>> Connection failed. DBH NOT Connected. > >>> 2013-02-05 09:40:10.379612 [ERR] > >> freeswitch_lua.cpp:435 DBH > >>> NOT Connected. > >>> > >>> > >>> Can you help me? > >>> > >>> Best regards, > >> _________________________________________________________________________ > >>> Professional FreeSWITCH Consulting Services: > >>> consulting at freeswitch.org > >>> http://www.freeswitchsolutions.com > >>> > >>> FreeSWITCH-powered IP PBX: The CudaTel Communication > >> Server > >>> > >>> > >>> Official FreeSWITCH Sites > >>> http://www.freeswitch.org > >>> http://wiki.freeswitch.org > >>> http://www.cluecon.com > >>> > >>> 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 > >> _________________________________________________________________________ > >>> Professional FreeSWITCH Consulting Services: > >>> consulting at freeswitch.org > >>> http://www.freeswitchsolutions.com > >>> > >>> > >>> > >>> > >>> Official FreeSWITCH Sites > >>> http://www.freeswitch.org > >>> http://wiki.freeswitch.org > >>> http://www.cluecon.com > >>> > >>> 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 > >> > >> > >> > >> _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> 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 > >> > >> > >> > >> _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> 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 > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 steveayre at gmail.com Tue Feb 5 19:42:10 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 5 Feb 2013 16:42:10 +0000 Subject: [Freeswitch-users] lua freeswitch.Dbh sqlite error In-Reply-To: <1360081909.22365.35.camel@luna.madrid.commsmundi.com> References: <5110C94F.7080901@gmail.com> <1360062212.8347.200.camel@luna.madrid.commsmundi.com> <1360078923.8347.280.camel@luna.madrid.commsmundi.com> <1360081909.22365.35.camel@luna.madrid.commsmundi.com> Message-ID: > > FAIL => "sqlite:///tmp/test.db" (this one should work) What error does this give? On 5 February 2013 16:31, Fran?ois Delawarde wrote: > I'm on HEAD, but I must be doing something wrong, there is no apparent > reason from the code. > > On Tue, 2013-02-05 at 16:05 +0000, Steven Ayre wrote: > > Honestly, I only use ODBC... Maybe someone else can shed some light? > > > > What version are you using? > > > > Steve > > > > > > > > On 5 Feb 2013, at 15:42, Fran?ois Delawarde < > fdelawarde at wirelessmundi.com> wrote: > > > > > Steven, does "sqlite://" prefix work for you? Not sure if it's a bug or > > > I'm doing something wrong. I tried the following from lua dbh: > > > > > > FAIL => "sqlite:///tmp/test.db" (this one should work) > > > FAIL => "sqlite://tmp/test.db" (shouldn't work, but just in case...) > > > OK => "/tmp/test.db" (no prefix) > > > > > > Fran?ois. > > > > > > > > > On Tue, 2013-02-05 at 12:55 +0000, Steven Ayre wrote: > > >> core: was an old prefix? I knew that change broke some ODBC DSNs > > >> (names without username+password, eg when they're specified in > > >> odbc.ini) but hadn't realised that had been broken too - that needs > > >> documenting as well on the Wiki I guess... > > >> http://wiki.freeswitch.org/wiki/Release_Notes#odbc-dsn > > >> > > >> > > >> Just a thought for the developers, but perhaps modules with odbc-dsn > > >> parameters should have this renamed to dsn or something now since it's > > >> not odbc-specific at all any longer. Easy enough to recognise both > > >> 'dsn' and 'odbc-dsn' names to remain backwards compatible (which I've > > >> seen done with typos in config setting names in the past > > >> as precedent). > > >> > > >> > > >> -Steve > > >> > > >> > > >> > > >> > > >> On 5 February 2013 11:03, Fran?ois Delawarde > > >> wrote: > > >> The behavior recently changed from "core:" to > > >> "sqlite://". > > >> > > >> Update: I just tested and it seems that "sqlite://" > > >> doesn't work, > > >> but as sqlite is the default, you can just omit the prefix and > > >> use > > >> "" to make it work. > > >> > > >> Fran?ois. > > >> > > >> > > >> On Tue, 2013-02-05 at 10:37 +0000, Steven Ayre wrote: > > >>> 'Failure to connect to PGSQL core' happens on ODBC > > >> connections, that > > >>> seems to be a minor bug in the error message. > > >>> > > >>> > > >>> From the unixODBC error it's interpreting this as a ODBC > > >> DSN. The > > >>> reason is the sqlite prefix is sqlite:// not core:// > > >>> (see http://wiki.freeswitch.org/wiki/DSN). You also don't > > >> have enough > > >>> slashes. > > >>> > > >>> > > >>> Try: > > >>> local dbh = > > >> freeswitch.Dbh("sqlite:///usr/local/temp/database.db"); > > >>> > > >>> > > >>> > > >>> (note the 3 slashes - :// is standard URI format, the 3rd > > >> indicates an > > >>> absolute path). > > >>> > > >>> > > >>> -Steve > > >>> > > >>> > > >>> > > >>> > > >>> On 5 February 2013 08:56, Carlo Dimaggio > > >> > > >>> wrote: > > >>> Hi all! > > >>> > > >>> I have a lua script in which i have a statement that > > >> connects > > >>> to a sqlite db: > > >>> > > >>> local dbh = > > >>> freeswitch.Dbh("core:/usr/local/temp/database.db"); > > >>> > > >>> but i receve in FS CLI this error that refers to a > > >> Pgsql > > >>> instead of sqlite: > > >>> > > >>> 2013-02-05 09:40:10.379612 [ERR] switch_odbc.c:365 > > >> STATE: > > >>> IM002 CODE 0 ERROR: [unixODBC][Driver Manager]Data > > >> source name > > >>> not found, and no default driver specified > > >>> 2013-02-05 09:40:10.379612 [CRIT] > > >> switch_core_sqldb.c:504 > > >>> Failure to connect to PGSQL core! > > >>> 2013-02-05 09:40:10.379612 [ERR] > > >> freeswitch_lua.cpp:354 > > >>> Connection failed. DBH NOT Connected. > > >>> 2013-02-05 09:40:10.379612 [ERR] > > >> freeswitch_lua.cpp:435 DBH > > >>> NOT Connected. > > >>> > > >>> > > >>> Can you help me? > > >>> > > >>> Best regards, > > >> > _________________________________________________________________________ > > >>> Professional FreeSWITCH Consulting Services: > > >>> consulting at freeswitch.org > > >>> http://www.freeswitchsolutions.com > > >>> > > >>> FreeSWITCH-powered IP PBX: The CudaTel Communication > > >> Server > > >>> > > >>> > > >>> Official FreeSWITCH Sites > > >>> http://www.freeswitch.org > > >>> http://wiki.freeswitch.org > > >>> http://www.cluecon.com > > >>> > > >>> 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 > > >> > _________________________________________________________________________ > > >>> Professional FreeSWITCH Consulting Services: > > >>> consulting at freeswitch.org > > >>> http://www.freeswitchsolutions.com > > >>> > > >>> > > >>> > > >>> > > >>> Official FreeSWITCH Sites > > >>> http://www.freeswitch.org > > >>> http://wiki.freeswitch.org > > >>> http://www.cluecon.com > > >>> > > >>> 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 > > >> > > >> > > >> > > >> > _________________________________________________________________________ > > >> Professional FreeSWITCH Consulting Services: > > >> consulting at freeswitch.org > > >> http://www.freeswitchsolutions.com > > >> > > >> > > >> > > >> > > >> Official FreeSWITCH Sites > > >> http://www.freeswitch.org > > >> http://wiki.freeswitch.org > > >> http://www.cluecon.com > > >> > > >> 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 > > >> > > >> > > >> > > >> > _________________________________________________________________________ > > >> Professional FreeSWITCH Consulting Services: > > >> consulting at freeswitch.org > > >> http://www.freeswitchsolutions.com > > >> > > >> > > >> > > >> > > >> Official FreeSWITCH Sites > > >> http://www.freeswitch.org > > >> http://wiki.freeswitch.org > > >> http://www.cluecon.com > > >> > > >> 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 > > > > > > > > > > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > > > > > 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 > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130205/14659c8f/attachment-0001.html From msc at freeswitch.org Tue Feb 5 19:59:17 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 5 Feb 2013 08:59:17 -0800 Subject: [Freeswitch-users] Get recorded file from callcenter via API. In-Reply-To: <510F64BD.6070107@gmail.com> References: <510EAB50.80308@gmail.com> <1FFF97C269757C458224B7C895F35F15215DC1@cantor.std.visionutv.se> <510F4EAD.3040807@gmail.com> <2D54617F-C018-491F-B19F-9FAA02D5412D@visionutveckling.se> <510F64BD.6070107@gmail.com> Message-ID: Load mod_httapi and make the file accessible via Web server. Then you can play the file with playback using this basic syntax: -MC On Sun, Feb 3, 2013 at 11:35 PM, Mimiko wrote: > On 04.02.2013 08:49, Peter Olsson wrote: > > You need to be more specific... What exactly do you want to do? Play the > file as audio inside a call, or grab the actual file from server? > > Get the file from the recordings directory storage to play inline on a > web page for listening what was recorded. > > For now I use direct file access from PHP sending file to client. But > this breaks the security to record storage directory where both, httpd' > user and FS' user have access to same directory. > > -- > Mimiko desu. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130205/e68cb1e3/attachment.html From vbvbrj at gmail.com Tue Feb 5 20:33:20 2013 From: vbvbrj at gmail.com (Mimiko) Date: Tue, 05 Feb 2013 19:33:20 +0200 Subject: [Freeswitch-users] Get recorded file from callcenter via API. In-Reply-To: References: <510EAB50.80308@gmail.com> <1FFF97C269757C458224B7C895F35F15215DC1@cantor.std.visionutv.se> <510F4EAD.3040807@gmail.com> <2D54617F-C018-491F-B19F-9FAA02D5412D@visionutveckling.se> <510F64BD.6070107@gmail.com> Message-ID: <51114260.20207@gmail.com> On 05.02.2013 18:59, Michael Collins wrote: > Load mod_httapi and make the file accessible via Web server. Then you > can play the file with playback using this basic syntax: > > > > -MC Oh? you've got the wrong idea. I don't want to playback a file from remote site to the channel. I want to download a prerecorded file by mod_callcente from FS using remote connection, maybe lua or event socket. May be there is some methods. As I know, there is an incorporated web server in FS to send commands and get status. How to use this to get files? -- Mimiko desu. From fdelawarde at wirelessmundi.com Tue Feb 5 20:54:43 2013 From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde) Date: Tue, 05 Feb 2013 18:54:43 +0100 Subject: [Freeswitch-users] lua freeswitch.Dbh sqlite error In-Reply-To: References: <5110C94F.7080901@gmail.com> <1360062212.8347.200.camel@luna.madrid.commsmundi.com> <1360078923.8347.280.camel@luna.madrid.commsmundi.com> <1360081909.22365.35.camel@luna.madrid.commsmundi.com> Message-ID: <1360086883.25626.4.camel@luna.madrid.commsmundi.com> There you go. ** My script test.lua is the following one liner: local dbh = freeswitch.Dbh("sqlite:///tmp/test.db") freeswitch at internal> lua test.lua -ERR no reply 2013-02-05 18:50:47.230744 [ERR] switch_odbc.c:365 STATE: IM002 CODE 0 ERROR: [unixODBC][Driver Manager]Data source name not found, and no default driver specified 2013-02-05 18:50:47.230744 [CRIT] switch_core_sqldb.c:504 Failure to connect to PGSQL sqlite! 2013-02-05 18:50:47.230744 [ERR] freeswitch_lua.cpp:354 Connection failed. DBH NOT Connected. ** Now if I change test.lua to: local dbh = freeswitch.Dbh("/tmp/test.db") freeswitch at internal> lua test.lua -ERR no reply No ERR or CRIT in this last one. Cheers, Fran?ois. On Tue, 2013-02-05 at 16:42 +0000, Steven Ayre wrote: > FAIL => "sqlite:///tmp/test.db" (this one should work) > > > What error does this give? > > > > On 5 February 2013 16:31, Fran?ois Delawarde > wrote: > I'm on HEAD, but I must be doing something wrong, there is no > apparent > reason from the code. > > On Tue, 2013-02-05 at 16:05 +0000, Steven Ayre wrote: > > Honestly, I only use ODBC... Maybe someone else can shed > some light? > > > > What version are you using? > > > > Steve > > > > > > > > On 5 Feb 2013, at 15:42, Fran?ois Delawarde > wrote: > > > > > Steven, does "sqlite://" prefix work for you? Not sure if > it's a bug or > > > I'm doing something wrong. I tried the following from lua > dbh: > > > > > > FAIL => "sqlite:///tmp/test.db" (this one should work) > > > FAIL => "sqlite://tmp/test.db" (shouldn't work, but just > in case...) > > > OK => "/tmp/test.db" (no prefix) > > > > > > Fran?ois. > > > > > > > > > On Tue, 2013-02-05 at 12:55 +0000, Steven Ayre wrote: > > >> core: was an old prefix? I knew that change broke some > ODBC DSNs > > >> (names without username+password, eg when they're > specified in > > >> odbc.ini) but hadn't realised that had been broken too - > that needs > > >> documenting as well on the Wiki I guess... > > >> http://wiki.freeswitch.org/wiki/Release_Notes#odbc-dsn > > >> > > >> > > >> Just a thought for the developers, but perhaps modules > with odbc-dsn > > >> parameters should have this renamed to dsn or something > now since it's > > >> not odbc-specific at all any longer. Easy enough to > recognise both > > >> 'dsn' and 'odbc-dsn' names to remain backwards compatible > (which I've > > >> seen done with typos in config setting names in the past > > >> as precedent). > > >> > > >> > > >> -Steve > > >> > > >> > > >> > > >> > > >> On 5 February 2013 11:03, Fran?ois Delawarde > > >> wrote: > > >> The behavior recently changed from "core:" > to > > >> "sqlite://". > > >> > > >> Update: I just tested and it seems that > "sqlite://" > > >> doesn't work, > > >> but as sqlite is the default, you can just omit > the prefix and > > >> use > > >> "" to make it work. > > >> > > >> Fran?ois. > > >> > > >> > > >> On Tue, 2013-02-05 at 10:37 +0000, Steven Ayre > wrote: > > >>> 'Failure to connect to PGSQL core' happens on ODBC > > >> connections, that > > >>> seems to be a minor bug in the error message. > > >>> > > >>> > > >>> From the unixODBC error it's interpreting this as a ODBC > > >> DSN. The > > >>> reason is the sqlite prefix is sqlite:// not core:// > > >>> (see http://wiki.freeswitch.org/wiki/DSN). You also > don't > > >> have enough > > >>> slashes. > > >>> > > >>> > > >>> Try: > > >>> local dbh = > > >> > freeswitch.Dbh("sqlite:///usr/local/temp/database.db"); > > >>> > > >>> > > >>> > > >>> (note the 3 slashes - :// is standard URI format, the > 3rd > > >> indicates an > > >>> absolute path). > > >>> > > >>> > > >>> -Steve > > >>> > > >>> > > >>> > > >>> > > >>> On 5 February 2013 08:56, Carlo Dimaggio > > >> > > >>> wrote: > > >>> Hi all! > > >>> > > >>> I have a lua script in which i have a statement > that > > >> connects > > >>> to a sqlite db: > > >>> > > >>> local dbh = > > >>> > freeswitch.Dbh("core:/usr/local/temp/database.db"); > > >>> > > >>> but i receve in FS CLI this error that refers to > a > > >> Pgsql > > >>> instead of sqlite: > > >>> > > >>> 2013-02-05 09:40:10.379612 [ERR] > switch_odbc.c:365 > > >> STATE: > > >>> IM002 CODE 0 ERROR: [unixODBC][Driver > Manager]Data > > >> source name > > >>> not found, and no default driver specified > > >>> 2013-02-05 09:40:10.379612 [CRIT] > > >> switch_core_sqldb.c:504 > > >>> Failure to connect to PGSQL core! > > >>> 2013-02-05 09:40:10.379612 [ERR] > > >> freeswitch_lua.cpp:354 > > >>> Connection failed. DBH NOT Connected. > > >>> 2013-02-05 09:40:10.379612 [ERR] > > >> freeswitch_lua.cpp:435 DBH > > >>> NOT Connected. > > >>> > > >>> > > >>> Can you help me? > > >>> > > >>> Best regards, > > >> > _________________________________________________________________________ > > >>> Professional FreeSWITCH Consulting Services: > > >>> consulting at freeswitch.org > > >>> http://www.freeswitchsolutions.com > > >>> > > >>> FreeSWITCH-powered IP PBX: The CudaTel > Communication > > >> Server > > >>> > > >>> > > >>> Official FreeSWITCH Sites > > >>> http://www.freeswitch.org > > >>> http://wiki.freeswitch.org > > >>> http://www.cluecon.com > > >>> > > >>> 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 > > >> > _________________________________________________________________________ > > >>> Professional FreeSWITCH Consulting Services: > > >>> consulting at freeswitch.org > > >>> http://www.freeswitchsolutions.com > > >>> > > >>> FreeSWITCH-powered IP PBX: The CudaTel Communication > Server > > >>> > > >>> > > >>> Official FreeSWITCH Sites > > >>> http://www.freeswitch.org > > >>> http://wiki.freeswitch.org > > >>> http://www.cluecon.com > > >>> > > >>> 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 > > >> > > >> > > >> > > >> > _________________________________________________________________________ > > >> Professional FreeSWITCH Consulting Services: > > >> consulting at freeswitch.org > > >> http://www.freeswitchsolutions.com > > >> > > >> FreeSWITCH-powered IP PBX: The CudaTel > Communication Server > > >> > > >> > > >> Official FreeSWITCH Sites > > >> http://www.freeswitch.org > > >> http://wiki.freeswitch.org > > >> http://www.cluecon.com > > >> > > >> 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 > > >> > > >> > > >> > > >> > _________________________________________________________________________ > > >> Professional FreeSWITCH Consulting Services: > > >> consulting at freeswitch.org > > >> http://www.freeswitchsolutions.com > > >> > > >> FreeSWITCH-powered IP PBX: The CudaTel Communication > Server > > >> > > >> > > >> Official FreeSWITCH Sites > > >> http://www.freeswitch.org > > >> http://wiki.freeswitch.org > > >> http://www.cluecon.com > > >> > > >> 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 > > > > > > > > > > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org > > > http://www.freeswitchsolutions.com > > > > > > FreeSWITCH-powered IP PBX: The CudaTel Communication > Server > > > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > > > > > 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 > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 msc at freeswitch.org Tue Feb 5 21:27:37 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 5 Feb 2013 10:27:37 -0800 Subject: [Freeswitch-users] Get recorded file from callcenter via API. In-Reply-To: <51114260.20207@gmail.com> References: <510EAB50.80308@gmail.com> <1FFF97C269757C458224B7C895F35F15215DC1@cantor.std.visionutv.se> <510F4EAD.3040807@gmail.com> <2D54617F-C018-491F-B19F-9FAA02D5412D@visionutveckling.se> <510F64BD.6070107@gmail.com> <51114260.20207@gmail.com> Message-ID: If you simply want to download the file then why not just put it into a location that can be accessed with wget? -MC On Tue, Feb 5, 2013 at 9:33 AM, Mimiko wrote: > On 05.02.2013 18:59, Michael Collins wrote: > > Load mod_httapi and make the file accessible via Web server. Then you > > can play the file with playback using this basic syntax: > > > > > > > > -MC > > Oh? you've got the wrong idea. I don't want to playback a file from > remote site to the channel. I want to download a prerecorded file by > mod_callcente from FS using remote connection, maybe lua or event > socket. May be there is some methods. As I know, there is an > incorporated web server in FS to send commands and get status. How to > use this to get files? > > -- > Mimiko desu. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130205/e04bd677/attachment-0001.html From vbvbrj at gmail.com Tue Feb 5 22:00:22 2013 From: vbvbrj at gmail.com (Mimiko) Date: Tue, 05 Feb 2013 21:00:22 +0200 Subject: [Freeswitch-users] Get recorded file from callcenter via API. In-Reply-To: References: <510EAB50.80308@gmail.com> <1FFF97C269757C458224B7C895F35F15215DC1@cantor.std.visionutv.se> <510F4EAD.3040807@gmail.com> <2D54617F-C018-491F-B19F-9FAA02D5412D@visionutveckling.se> <510F64BD.6070107@gmail.com> <51114260.20207@gmail.com> Message-ID: <511156C6.1070701@gmail.com> On 05.02.2013 20:27, Michael Collins wrote: > If you simply want to download the file then why not just put it into a > location that can be accessed with wget? > -MC That is. That location would have to have security permission read access for other users than FS, ie the user under which web server or ftp server or samba will run. Ok. Is a command in CLI which will read a file at given path (assuming it has permission) and type it in window? -- Mimiko desu. From msc at freeswitch.org Tue Feb 5 22:53:46 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 5 Feb 2013 11:53:46 -0800 Subject: [Freeswitch-users] Get recorded file from callcenter via API. In-Reply-To: <511156C6.1070701@gmail.com> References: <510EAB50.80308@gmail.com> <1FFF97C269757C458224B7C895F35F15215DC1@cantor.std.visionutv.se> <510F4EAD.3040807@gmail.com> <2D54617F-C018-491F-B19F-9FAA02D5412D@visionutveckling.se> <510F64BD.6070107@gmail.com> <51114260.20207@gmail.com> <511156C6.1070701@gmail.com> Message-ID: On Tue, Feb 5, 2013 at 11:00 AM, Mimiko wrote: > On 05.02.2013 20:27, Michael Collins wrote: > > If you simply want to download the file then why not just put it into a > > location that can be accessed with wget? > > -MC > > That is. That location would have to have security permission read > access for other users than FS, ie the user under which web server or > ftp server or samba will run. > > Ok. Is a command in CLI which will read a file at given path (assuming > it has permission) and type it in window? > What do you mean "type it in window"? In any case you can use the fs_cli "system" API to send a command to the operating system to be executed. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130205/3cd2db19/attachment.html From vbvbrj at gmail.com Tue Feb 5 23:17:15 2013 From: vbvbrj at gmail.com (Mimiko) Date: Tue, 05 Feb 2013 22:17:15 +0200 Subject: [Freeswitch-users] Get recorded file from callcenter via API. In-Reply-To: References: <510EAB50.80308@gmail.com> <1FFF97C269757C458224B7C895F35F15215DC1@cantor.std.visionutv.se> <510F4EAD.3040807@gmail.com> <2D54617F-C018-491F-B19F-9FAA02D5412D@visionutveckling.se> <510F64BD.6070107@gmail.com> <51114260.20207@gmail.com> <511156C6.1070701@gmail.com> Message-ID: <511168CB.8060905@gmail.com> On 05.02.2013 21:53, Michael Collins wrote: > What do you mean "type it in window"? In any case you can use the fs_cli > "system" API to send a command to the operating system to be executed. > > -MC Well, I will try "system". By the way, is it possible to connect a flash player audio only via fs cli or event socket or other means in php to FS to hear a channel? Ie, A channel is a web browser, and B channel is a phone? -- Mimiko desu. From avi at avimarcus.net Tue Feb 5 23:32:47 2013 From: avi at avimarcus.net (Avi Marcus) Date: Tue, 5 Feb 2013 22:32:47 +0200 Subject: [Freeswitch-users] SPA-2102 Gives 486 instead of Call Waiting In-Reply-To: References: <3010184978485284118@unknownmsgid> <4522496924514533582@unknownmsgid> Message-ID: I think I solved it! The call waiting was turned on, but when I added to provisiong to turn on Call waiting *service *then it started working. I guess I missed that looking at the config of the box. For future reference -- for line 1, similar for line 2: yes yes -Avi Marcus BestFone On Mon, Feb 4, 2013 at 1:57 PM, Avi Marcus wrote: > It's on debug and syslog, level 3 and "full" on line 1. > I seemed to only get this when calling line 1 while it was on the phone. > > http://pastebin.freeswitch.org/20547 > > Should I have gotten more? > > -Avi > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130205/b829e1ee/attachment.html From steveayre at gmail.com Tue Feb 5 23:33:29 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 5 Feb 2013 20:33:29 +0000 Subject: [Freeswitch-users] lua freeswitch.Dbh sqlite error In-Reply-To: <1360086883.25626.4.camel@luna.madrid.commsmundi.com> References: <5110C94F.7080901@gmail.com> <1360062212.8347.200.camel@luna.madrid.commsmundi.com> <1360078923.8347.280.camel@luna.madrid.commsmundi.com> <1360081909.22365.35.camel@luna.madrid.commsmundi.com> <1360086883.25626.4.camel@luna.madrid.commsmundi.com> Message-ID: Ah I've spotted it. There's a bug in the code for detecting the old-style ODBC DSNs. switch_core_sqldb.c:344 } else if ((!(i = strncasecmp(dsn, "odbc://", 7))) || strchr(dsn, ':')) { This will match any DSN containing a ':' which obviously sqlite:// will. Therefore although the sqlite:// prefix is recognised and skipped over, that code path will never get reached. The plain filename works because anything not odbc/pgsql is assumed to be a sqlite filename (switch_core_sqldb.c:366). I've raised this in Jira: http://jira.freeswitch.org/browse/FS-5074 -Steve On 5 February 2013 17:54, Fran?ois Delawarde wrote: > There you go. > > ** My script test.lua is the following one liner: > local dbh = freeswitch.Dbh("sqlite:///tmp/test.db") > > > freeswitch at internal> lua test.lua > -ERR no reply > > 2013-02-05 18:50:47.230744 [ERR] switch_odbc.c:365 STATE: IM002 CODE 0 > ERROR: [unixODBC][Driver Manager]Data source name not found, and no default > driver specified > > 2013-02-05 18:50:47.230744 [CRIT] switch_core_sqldb.c:504 Failure to > connect to PGSQL sqlite! > 2013-02-05 18:50:47.230744 [ERR] freeswitch_lua.cpp:354 Connection failed. > DBH NOT Connected. > > > > > ** Now if I change test.lua to: > local dbh = freeswitch.Dbh("/tmp/test.db") > > > freeswitch at internal> lua test.lua > -ERR no reply > > > No ERR or CRIT in this last one. > > > Cheers, > Fran?ois. > > On Tue, 2013-02-05 at 16:42 +0000, Steven Ayre wrote: > > FAIL => "sqlite:///tmp/test.db" (this one should work) > > > > > > What error does this give? > > > > > > > > On 5 February 2013 16:31, Fran?ois Delawarde > > wrote: > > I'm on HEAD, but I must be doing something wrong, there is no > > apparent > > reason from the code. > > > > On Tue, 2013-02-05 at 16:05 +0000, Steven Ayre wrote: > > > Honestly, I only use ODBC... Maybe someone else can shed > > some light? > > > > > > What version are you using? > > > > > > Steve > > > > > > > > > > > > On 5 Feb 2013, at 15:42, Fran?ois Delawarde > > wrote: > > > > > > > Steven, does "sqlite://" prefix work for you? Not sure if > > it's a bug or > > > > I'm doing something wrong. I tried the following from lua > > dbh: > > > > > > > > FAIL => "sqlite:///tmp/test.db" (this one should work) > > > > FAIL => "sqlite://tmp/test.db" (shouldn't work, but just > > in case...) > > > > OK => "/tmp/test.db" (no prefix) > > > > > > > > Fran?ois. > > > > > > > > > > > > On Tue, 2013-02-05 at 12:55 +0000, Steven Ayre wrote: > > > >> core: was an old prefix? I knew that change broke some > > ODBC DSNs > > > >> (names without username+password, eg when they're > > specified in > > > >> odbc.ini) but hadn't realised that had been broken too - > > that needs > > > >> documenting as well on the Wiki I guess... > > > >> http://wiki.freeswitch.org/wiki/Release_Notes#odbc-dsn > > > >> > > > >> > > > >> Just a thought for the developers, but perhaps modules > > with odbc-dsn > > > >> parameters should have this renamed to dsn or something > > now since it's > > > >> not odbc-specific at all any longer. Easy enough to > > recognise both > > > >> 'dsn' and 'odbc-dsn' names to remain backwards compatible > > (which I've > > > >> seen done with typos in config setting names in the past > > > >> as precedent). > > > >> > > > >> > > > >> -Steve > > > >> > > > >> > > > >> > > > >> > > > >> On 5 February 2013 11:03, Fran?ois Delawarde > > > >> wrote: > > > >> The behavior recently changed from "core:" > > to > > > >> "sqlite://". > > > >> > > > >> Update: I just tested and it seems that > > "sqlite://" > > > >> doesn't work, > > > >> but as sqlite is the default, you can just omit > > the prefix and > > > >> use > > > >> "" to make it work. > > > >> > > > >> Fran?ois. > > > >> > > > >> > > > >> On Tue, 2013-02-05 at 10:37 +0000, Steven Ayre > > wrote: > > > >>> 'Failure to connect to PGSQL core' happens on ODBC > > > >> connections, that > > > >>> seems to be a minor bug in the error message. > > > >>> > > > >>> > > > >>> From the unixODBC error it's interpreting this as a ODBC > > > >> DSN. The > > > >>> reason is the sqlite prefix is sqlite:// not core:// > > > >>> (see http://wiki.freeswitch.org/wiki/DSN). You also > > don't > > > >> have enough > > > >>> slashes. > > > >>> > > > >>> > > > >>> Try: > > > >>> local dbh = > > > >> > > freeswitch.Dbh("sqlite:///usr/local/temp/database.db"); > > > >>> > > > >>> > > > >>> > > > >>> (note the 3 slashes - :// is standard URI format, the > > 3rd > > > >> indicates an > > > >>> absolute path). > > > >>> > > > >>> > > > >>> -Steve > > > >>> > > > >>> > > > >>> > > > >>> > > > >>> On 5 February 2013 08:56, Carlo Dimaggio > > > >> > > > >>> wrote: > > > >>> Hi all! > > > >>> > > > >>> I have a lua script in which i have a statement > > that > > > >> connects > > > >>> to a sqlite db: > > > >>> > > > >>> local dbh = > > > >>> > > freeswitch.Dbh("core:/usr/local/temp/database.db"); > > > >>> > > > >>> but i receve in FS CLI this error that refers to > > a > > > >> Pgsql > > > >>> instead of sqlite: > > > >>> > > > >>> 2013-02-05 09:40:10.379612 [ERR] > > switch_odbc.c:365 > > > >> STATE: > > > >>> IM002 CODE 0 ERROR: [unixODBC][Driver > > Manager]Data > > > >> source name > > > >>> not found, and no default driver specified > > > >>> 2013-02-05 09:40:10.379612 [CRIT] > > > >> switch_core_sqldb.c:504 > > > >>> Failure to connect to PGSQL core! > > > >>> 2013-02-05 09:40:10.379612 [ERR] > > > >> freeswitch_lua.cpp:354 > > > >>> Connection failed. DBH NOT Connected. > > > >>> 2013-02-05 09:40:10.379612 [ERR] > > > >> freeswitch_lua.cpp:435 DBH > > > >>> NOT Connected. > > > >>> > > > >>> > > > >>> Can you help me? > > > >>> > > > >>> Best regards, > > > >> > > > _________________________________________________________________________ > > > >>> Professional FreeSWITCH Consulting Services: > > > >>> consulting at freeswitch.org > > > >>> http://www.freeswitchsolutions.com > > > >>> > > > >>> FreeSWITCH-powered IP PBX: The CudaTel > > Communication > > > >> Server > > > >>> > > > >>> > > > >>> Official FreeSWITCH Sites > > > >>> http://www.freeswitch.org > > > >>> http://wiki.freeswitch.org > > > >>> http://www.cluecon.com > > > >>> > > > >>> 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 > > > >> > > > _________________________________________________________________________ > > > >>> Professional FreeSWITCH Consulting Services: > > > >>> consulting at freeswitch.org > > > >>> http://www.freeswitchsolutions.com > > > >>> > > > >>> FreeSWITCH-powered IP PBX: The CudaTel Communication > > Server > > > >>> > > > >>> > > > >>> Official FreeSWITCH Sites > > > >>> http://www.freeswitch.org > > > >>> http://wiki.freeswitch.org > > > >>> http://www.cluecon.com > > > >>> > > > >>> 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 > > > >> > > > >> > > > >> > > > >> > > > _________________________________________________________________________ > > > >> Professional FreeSWITCH Consulting Services: > > > >> consulting at freeswitch.org > > > >> http://www.freeswitchsolutions.com > > > >> > > > >> FreeSWITCH-powered IP PBX: The CudaTel > > Communication Server > > > >> > > > >> > > > >> Official FreeSWITCH Sites > > > >> http://www.freeswitch.org > > > >> http://wiki.freeswitch.org > > > >> http://www.cluecon.com > > > >> > > > >> 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 > > > >> > > > >> > > > >> > > > >> > > > _________________________________________________________________________ > > > >> Professional FreeSWITCH Consulting Services: > > > >> consulting at freeswitch.org > > > >> http://www.freeswitchsolutions.com > > > >> > > > >> FreeSWITCH-powered IP PBX: The CudaTel Communication > > Server > > > >> > > > >> > > > >> Official FreeSWITCH Sites > > > >> http://www.freeswitch.org > > > >> http://wiki.freeswitch.org > > > >> http://www.cluecon.com > > > >> > > > >> 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 > > > > > > > > > > > > > > > > > > > _________________________________________________________________________ > > > > Professional FreeSWITCH Consulting Services: > > > > consulting at freeswitch.org > > > > http://www.freeswitchsolutions.com > > > > > > > > FreeSWITCH-powered IP PBX: The CudaTel Communication > > Server > > > > > > > > > > > > Official FreeSWITCH Sites > > > > http://www.freeswitch.org > > > > http://wiki.freeswitch.org > > > > http://www.cluecon.com > > > > > > > > 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 > > > > > > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > > > > > 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 > > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > 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 > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130205/050c4d2a/attachment-0001.html From drk at drkngs.net Tue Feb 5 23:45:20 2013 From: drk at drkngs.net (Dave R. Kompel) Date: Tue, 05 Feb 2013 12:45:20 -0800 Subject: [Freeswitch-users] Get recorded file from callcenter via API. In-Reply-To: <511168CB.8060905@gmail.com> Message-ID: <20130205204520.eda103e8@mail.tritonwest.net> You could write your own API using any of the scripting languages that support an API to run a script, however I don't know if the stream passed to an API call will mess up binary data, it is considered a String by most interfaces. --Dave _____ From: Mimiko [mailto:vbvbrj at gmail.com] To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] Sent: Tue, 05 Feb 2013 12:17:15 -0800 Subject: Re: [Freeswitch-users] Get recorded file from callcenter via API. On 05.02.2013 21:53, Michael Collins wrote: > What do you mean "type it in window"? In any case you can use the fs_cli > "system" API to send a command to the operating system to be executed. > > -MC Well, I will try "system". By the way, is it possible to connect a flash player audio only via fs cli or event socket or other means in php to FS to hear a channel? Ie, A channel is a web browser, and B channel is a phone? -- Mimiko desu. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130205/c77aa71d/attachment.html From steveayre at gmail.com Tue Feb 5 23:50:45 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 5 Feb 2013 20:50:45 +0000 Subject: [Freeswitch-users] lua freeswitch.Dbh sqlite error In-Reply-To: References: <5110C94F.7080901@gmail.com> <1360062212.8347.200.camel@luna.madrid.commsmundi.com> <1360078923.8347.280.camel@luna.madrid.commsmundi.com> <1360081909.22365.35.camel@luna.madrid.commsmundi.com> <1360086883.25626.4.camel@luna.madrid.commsmundi.com> Message-ID: Although pretty minor I've also raised a ticket for the PGSQL error message bug: http://jira.freeswitch.org/browse/FS-5075 -Steve On 5 February 2013 20:33, Steven Ayre wrote: > Ah I've spotted it. There's a bug in the code for detecting the old-style > ODBC DSNs. > > switch_core_sqldb.c:344 } else if ((!(i = strncasecmp(dsn, "odbc://", 7))) > || strchr(dsn, ':')) { > > This will match any DSN containing a ':' which obviously sqlite:// will. > Therefore although the sqlite:// prefix is recognised and skipped over, > that code path will never get reached. > > The plain filename works because anything not odbc/pgsql is assumed to be > a sqlite filename (switch_core_sqldb.c:366). > > I've raised this in Jira: http://jira.freeswitch.org/browse/FS-5074 > > -Steve > > > On 5 February 2013 17:54, Fran?ois Delawarde > wrote: > >> There you go. >> >> ** My script test.lua is the following one liner: >> local dbh = freeswitch.Dbh("sqlite:///tmp/test.db") >> >> >> freeswitch at internal> lua test.lua >> -ERR no reply >> >> 2013-02-05 18:50:47.230744 [ERR] switch_odbc.c:365 STATE: IM002 CODE 0 >> ERROR: [unixODBC][Driver Manager]Data source name not found, and no default >> driver specified >> >> 2013-02-05 18:50:47.230744 [CRIT] switch_core_sqldb.c:504 Failure to >> connect to PGSQL sqlite! >> 2013-02-05 18:50:47.230744 [ERR] freeswitch_lua.cpp:354 Connection >> failed. DBH NOT Connected. >> >> >> >> >> ** Now if I change test.lua to: >> local dbh = freeswitch.Dbh("/tmp/test.db") >> >> >> freeswitch at internal> lua test.lua >> -ERR no reply >> >> >> No ERR or CRIT in this last one. >> >> >> Cheers, >> Fran?ois. >> >> On Tue, 2013-02-05 at 16:42 +0000, Steven Ayre wrote: >> > FAIL => "sqlite:///tmp/test.db" (this one should work) >> > >> > >> > What error does this give? >> > >> > >> > >> > On 5 February 2013 16:31, Fran?ois Delawarde >> > wrote: >> > I'm on HEAD, but I must be doing something wrong, there is no >> > apparent >> > reason from the code. >> > >> > On Tue, 2013-02-05 at 16:05 +0000, Steven Ayre wrote: >> > > Honestly, I only use ODBC... Maybe someone else can shed >> > some light? >> > > >> > > What version are you using? >> > > >> > > Steve >> > > >> > > >> > > >> > > On 5 Feb 2013, at 15:42, Fran?ois Delawarde >> > wrote: >> > > >> > > > Steven, does "sqlite://" prefix work for you? Not sure if >> > it's a bug or >> > > > I'm doing something wrong. I tried the following from lua >> > dbh: >> > > > >> > > > FAIL => "sqlite:///tmp/test.db" (this one should work) >> > > > FAIL => "sqlite://tmp/test.db" (shouldn't work, but just >> > in case...) >> > > > OK => "/tmp/test.db" (no prefix) >> > > > >> > > > Fran?ois. >> > > > >> > > > >> > > > On Tue, 2013-02-05 at 12:55 +0000, Steven Ayre wrote: >> > > >> core: was an old prefix? I knew that change broke some >> > ODBC DSNs >> > > >> (names without username+password, eg when they're >> > specified in >> > > >> odbc.ini) but hadn't realised that had been broken too - >> > that needs >> > > >> documenting as well on the Wiki I guess... >> > > >> http://wiki.freeswitch.org/wiki/Release_Notes#odbc-dsn >> > > >> >> > > >> >> > > >> Just a thought for the developers, but perhaps modules >> > with odbc-dsn >> > > >> parameters should have this renamed to dsn or something >> > now since it's >> > > >> not odbc-specific at all any longer. Easy enough to >> > recognise both >> > > >> 'dsn' and 'odbc-dsn' names to remain backwards compatible >> > (which I've >> > > >> seen done with typos in config setting names in the past >> > > >> as precedent). >> > > >> >> > > >> >> > > >> -Steve >> > > >> >> > > >> >> > > >> >> > > >> >> > > >> On 5 February 2013 11:03, Fran?ois Delawarde >> > > >> wrote: >> > > >> The behavior recently changed from "core:" >> > to >> > > >> "sqlite://". >> > > >> >> > > >> Update: I just tested and it seems that >> > "sqlite://" >> > > >> doesn't work, >> > > >> but as sqlite is the default, you can just omit >> > the prefix and >> > > >> use >> > > >> "" to make it work. >> > > >> >> > > >> Fran?ois. >> > > >> >> > > >> >> > > >> On Tue, 2013-02-05 at 10:37 +0000, Steven Ayre >> > wrote: >> > > >>> 'Failure to connect to PGSQL core' happens on ODBC >> > > >> connections, that >> > > >>> seems to be a minor bug in the error message. >> > > >>> >> > > >>> >> > > >>> From the unixODBC error it's interpreting this as a ODBC >> > > >> DSN. The >> > > >>> reason is the sqlite prefix is sqlite:// not core:// >> > > >>> (see http://wiki.freeswitch.org/wiki/DSN). You also >> > don't >> > > >> have enough >> > > >>> slashes. >> > > >>> >> > > >>> >> > > >>> Try: >> > > >>> local dbh = >> > > >> >> > freeswitch.Dbh("sqlite:///usr/local/temp/database.db"); >> > > >>> >> > > >>> >> > > >>> >> > > >>> (note the 3 slashes - :// is standard URI format, the >> > 3rd >> > > >> indicates an >> > > >>> absolute path). >> > > >>> >> > > >>> >> > > >>> -Steve >> > > >>> >> > > >>> >> > > >>> >> > > >>> >> > > >>> On 5 February 2013 08:56, Carlo Dimaggio >> > > >> >> > > >>> wrote: >> > > >>> Hi all! >> > > >>> >> > > >>> I have a lua script in which i have a statement >> > that >> > > >> connects >> > > >>> to a sqlite db: >> > > >>> >> > > >>> local dbh = >> > > >>> >> > freeswitch.Dbh("core:/usr/local/temp/database.db"); >> > > >>> >> > > >>> but i receve in FS CLI this error that refers to >> > a >> > > >> Pgsql >> > > >>> instead of sqlite: >> > > >>> >> > > >>> 2013-02-05 09:40:10.379612 [ERR] >> > switch_odbc.c:365 >> > > >> STATE: >> > > >>> IM002 CODE 0 ERROR: [unixODBC][Driver >> > Manager]Data >> > > >> source name >> > > >>> not found, and no default driver specified >> > > >>> 2013-02-05 09:40:10.379612 [CRIT] >> > > >> switch_core_sqldb.c:504 >> > > >>> Failure to connect to PGSQL core! >> > > >>> 2013-02-05 09:40:10.379612 [ERR] >> > > >> freeswitch_lua.cpp:354 >> > > >>> Connection failed. DBH NOT Connected. >> > > >>> 2013-02-05 09:40:10.379612 [ERR] >> > > >> freeswitch_lua.cpp:435 DBH >> > > >>> NOT Connected. >> > > >>> >> > > >>> >> > > >>> Can you help me? >> > > >>> >> > > >>> Best regards, >> > > >> >> > >> _________________________________________________________________________ >> > > >>> Professional FreeSWITCH Consulting Services: >> > > >>> consulting at freeswitch.org >> > > >>> http://www.freeswitchsolutions.com >> > > >>> >> > > >>> FreeSWITCH-powered IP PBX: The CudaTel >> > Communication >> > > >> Server >> > > >>> >> > > >>> >> > > >>> Official FreeSWITCH Sites >> > > >>> http://www.freeswitch.org >> > > >>> http://wiki.freeswitch.org >> > > >>> http://www.cluecon.com >> > > >>> >> > > >>> 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 >> > > >> >> > >> _________________________________________________________________________ >> > > >>> Professional FreeSWITCH Consulting Services: >> > > >>> consulting at freeswitch.org >> > > >>> http://www.freeswitchsolutions.com >> > > >>> >> > > >>> FreeSWITCH-powered IP PBX: The CudaTel Communication >> > Server >> > > >>> >> > > >>> >> > > >>> Official FreeSWITCH Sites >> > > >>> http://www.freeswitch.org >> > > >>> http://wiki.freeswitch.org >> > > >>> http://www.cluecon.com >> > > >>> >> > > >>> 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 >> > > >> >> > > >> >> > > >> >> > > >> >> > >> _________________________________________________________________________ >> > > >> Professional FreeSWITCH Consulting Services: >> > > >> consulting at freeswitch.org >> > > >> http://www.freeswitchsolutions.com >> > > >> >> > > >> FreeSWITCH-powered IP PBX: The CudaTel >> > Communication Server >> > > >> >> > > >> >> > > >> Official FreeSWITCH Sites >> > > >> http://www.freeswitch.org >> > > >> http://wiki.freeswitch.org >> > > >> http://www.cluecon.com >> > > >> >> > > >> 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 >> > > >> >> > > >> >> > > >> >> > > >> >> > >> _________________________________________________________________________ >> > > >> Professional FreeSWITCH Consulting Services: >> > > >> consulting at freeswitch.org >> > > >> http://www.freeswitchsolutions.com >> > > >> >> > > >> FreeSWITCH-powered IP PBX: The CudaTel Communication >> > Server >> > > >> >> > > >> >> > > >> Official FreeSWITCH Sites >> > > >> http://www.freeswitch.org >> > > >> http://wiki.freeswitch.org >> > > >> http://www.cluecon.com >> > > >> >> > > >> 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 >> > > > >> > > > >> > > > >> > > > >> > >> _________________________________________________________________________ >> > > > Professional FreeSWITCH Consulting Services: >> > > > consulting at freeswitch.org >> > > > http://www.freeswitchsolutions.com >> > > > >> > > > FreeSWITCH-powered IP PBX: The CudaTel Communication >> > Server >> > > > >> > > > >> > > > Official FreeSWITCH Sites >> > > > http://www.freeswitch.org >> > > > http://wiki.freeswitch.org >> > > > http://www.cluecon.com >> > > > >> > > > 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 >> > > >> > > >> > >> _________________________________________________________________________ >> > > Professional FreeSWITCH Consulting Services: >> > > consulting at freeswitch.org >> > > http://www.freeswitchsolutions.com >> > > >> > > >> > > >> > > >> > > Official FreeSWITCH Sites >> > > http://www.freeswitch.org >> > > http://wiki.freeswitch.org >> > > http://www.cluecon.com >> > > >> > > 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 >> > >> > >> > >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > 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 >> > >> > >> > >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > 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 >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130205/87513143/attachment-0001.html From mike at jerris.com Tue Feb 5 23:53:44 2013 From: mike at jerris.com (Michael Jerris) Date: Tue, 5 Feb 2013 14:53:44 -0600 Subject: [Freeswitch-users] lua freeswitch.Dbh sqlite error In-Reply-To: References: <5110C94F.7080901@gmail.com> <1360062212.8347.200.camel@luna.madrid.commsmundi.com> <1360078923.8347.280.camel@luna.madrid.commsmundi.com> <1360081909.22365.35.camel@luna.madrid.commsmundi.com> <1360086883.25626.4.camel@luna.madrid.commsmundi.com> Message-ID: <6C10D0C4-6DAC-449A-843E-AD433898E761@jerris.com> [master 1e6200b] FS-5074: --resolve support sqlite:// for DSNs Please confirm this fixes the issue Thanks Mike On Feb 5, 2013, at 2:33 PM, Steven Ayre wrote: > Ah I've spotted it. There's a bug in the code for detecting the old-style ODBC DSNs. > > switch_core_sqldb.c:344 } else if ((!(i = strncasecmp(dsn, "odbc://", 7))) || strchr(dsn, ':')) { > > This will match any DSN containing a ':' which obviously sqlite:// will. Therefore although the sqlite:// prefix is recognised and skipped over, that code path will never get reached. > > The plain filename works because anything not odbc/pgsql is assumed to be a sqlite filename (switch_core_sqldb.c:366). > > I've raised this in Jira: http://jira.freeswitch.org/browse/FS-5074 > > -Steve > > > On 5 February 2013 17:54, Fran?ois Delawarde wrote: > There you go. > > ** My script test.lua is the following one liner: > local dbh = freeswitch.Dbh("sqlite:///tmp/test.db") > > > freeswitch at internal> lua test.lua > -ERR no reply > > 2013-02-05 18:50:47.230744 [ERR] switch_odbc.c:365 STATE: IM002 CODE 0 ERROR: [unixODBC][Driver Manager]Data source name not found, and no default driver specified > > 2013-02-05 18:50:47.230744 [CRIT] switch_core_sqldb.c:504 Failure to connect to PGSQL sqlite! > 2013-02-05 18:50:47.230744 [ERR] freeswitch_lua.cpp:354 Connection failed. DBH NOT Connected. > > > > > ** Now if I change test.lua to: > local dbh = freeswitch.Dbh("/tmp/test.db") > > > freeswitch at internal> lua test.lua > -ERR no reply > > > No ERR or CRIT in this last one. > > > Cheers, > Fran?ois. > > On Tue, 2013-02-05 at 16:42 +0000, Steven Ayre wrote: > > FAIL => "sqlite:///tmp/test.db" (this one should work) > > > > > > What error does this give? > > > > > > > > On 5 February 2013 16:31, Fran?ois Delawarde > > wrote: > > I'm on HEAD, but I must be doing something wrong, there is no > > apparent > > reason from the code. > > > > On Tue, 2013-02-05 at 16:05 +0000, Steven Ayre wrote: > > > Honestly, I only use ODBC... Maybe someone else can shed > > some light? > > > > > > What version are you using? > > > > > > Steve > > > > > > > > > > > > On 5 Feb 2013, at 15:42, Fran?ois Delawarde > > wrote: > > > > > > > Steven, does "sqlite://" prefix work for you? Not sure if > > it's a bug or > > > > I'm doing something wrong. I tried the following from lua > > dbh: > > > > > > > > FAIL => "sqlite:///tmp/test.db" (this one should work) > > > > FAIL => "sqlite://tmp/test.db" (shouldn't work, but just > > in case...) > > > > OK => "/tmp/test.db" (no prefix) > > > > > > > > Fran?ois. > > > > > > > > > > > > On Tue, 2013-02-05 at 12:55 +0000, Steven Ayre wrote: > > > >> core: was an old prefix? I knew that change broke some > > ODBC DSNs > > > >> (names without username+password, eg when they're > > specified in > > > >> odbc.ini) but hadn't realised that had been broken too - > > that needs > > > >> documenting as well on the Wiki I guess... > > > >> http://wiki.freeswitch.org/wiki/Release_Notes#odbc-dsn > > > >> > > > >> > > > >> Just a thought for the developers, but perhaps modules > > with odbc-dsn > > > >> parameters should have this renamed to dsn or something > > now since it's > > > >> not odbc-specific at all any longer. Easy enough to > > recognise both > > > >> 'dsn' and 'odbc-dsn' names to remain backwards compatible > > (which I've > > > >> seen done with typos in config setting names in the past > > > >> as precedent). > > > >> > > > >> > > > >> -Steve > > > >> > > > >> > > > >> > > > >> > > > >> On 5 February 2013 11:03, Fran?ois Delawarde > > > >> wrote: > > > >> The behavior recently changed from "core:" > > to > > > >> "sqlite://". > > > >> > > > >> Update: I just tested and it seems that > > "sqlite://" > > > >> doesn't work, > > > >> but as sqlite is the default, you can just omit > > the prefix and > > > >> use > > > >> "" to make it work. > > > >> > > > >> Fran?ois. > > > >> > > > >> > > > >> On Tue, 2013-02-05 at 10:37 +0000, Steven Ayre > > wrote: > > > >>> 'Failure to connect to PGSQL core' happens on ODBC > > > >> connections, that > > > >>> seems to be a minor bug in the error message. > > > >>> > > > >>> > > > >>> From the unixODBC error it's interpreting this as a ODBC > > > >> DSN. The > > > >>> reason is the sqlite prefix is sqlite:// not core:// > > > >>> (see http://wiki.freeswitch.org/wiki/DSN). You also > > don't > > > >> have enough > > > >>> slashes. > > > >>> > > > >>> > > > >>> Try: > > > >>> local dbh = > > > >> > > freeswitch.Dbh("sqlite:///usr/local/temp/database.db"); > > > >>> > > > >>> > > > >>> > > > >>> (note the 3 slashes - :// is standard URI format, the > > 3rd > > > >> indicates an > > > >>> absolute path). > > > >>> > > > >>> > > > >>> -Steve > > > >>> > > > >>> > > > >>> > > > >>> > > > >>> On 5 February 2013 08:56, Carlo Dimaggio > > > >> > > > >>> wrote: > > > >>> Hi all! > > > >>> > > > >>> I have a lua script in which i have a statement > > that > > > >> connects > > > >>> to a sqlite db: > > > >>> > > > >>> local dbh = > > > >>> > > freeswitch.Dbh("core:/usr/local/temp/database.db"); > > > >>> > > > >>> but i receve in FS CLI this error that refers to > > a > > > >> Pgsql > > > >>> instead of sqlite: > > > >>> > > > >>> 2013-02-05 09:40:10.379612 [ERR] > > switch_odbc.c:365 > > > >> STATE: > > > >>> IM002 CODE 0 ERROR: [unixODBC][Driver > > Manager]Data > > > >> source name > > > >>> not found, and no default driver specified > > > >>> 2013-02-05 09:40:10.379612 [CRIT] > > > >> switch_core_sqldb.c:504 > > > >>> Failure to connect to PGSQL core! > > > >>> 2013-02-05 09:40:10.379612 [ERR] > > > >> freeswitch_lua.cpp:354 > > > >>> Connection failed. DBH NOT Connected. > > > >>> 2013-02-05 09:40:10.379612 [ERR] > > > >> freeswitch_lua.cpp:435 DBH > > > >>> NOT Connected. > > > >>> > > > >>> > > > >>> Can you help me? > > > >>> > > > >>> Best regards, > > > >> > > _________________________________________________________________________ > > > >>> Professional FreeSWITCH Consulting Services: > > > >>> consulting at freeswitch.org > > > >>> http://www.freeswitchsolutions.com > > > >>> > > > >>> FreeSWITCH-powered IP PBX: The CudaTel > > Communication > > > >> Server > > > >>> > > > >>> > > > >>> Official FreeSWITCH Sites > > > >>> http://www.freeswitch.org > > > >>> http://wiki.freeswitch.org > > > >>> http://www.cluecon.com > > > >>> > > > >>> 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 > > > >> > > _________________________________________________________________________ > > > >>> Professional FreeSWITCH Consulting Services: > > > >>> consulting at freeswitch.org > > > >>> http://www.freeswitchsolutions.com > > > >>> > > > >>> FreeSWITCH-powered IP PBX: The CudaTel Communication > > Server > > > >>> > > > >>> > > > >>> Official FreeSWITCH Sites > > > >>> http://www.freeswitch.org > > > >>> http://wiki.freeswitch.org > > > >>> http://www.cluecon.com > > > >>> > > > >>> 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 > > > >> > > > >> > > > >> > > > >> > > _________________________________________________________________________ > > > >> Professional FreeSWITCH Consulting Services: > > > >> consulting at freeswitch.org > > > >> http://www.freeswitchsolutions.com > > > >> > > > >> FreeSWITCH-powered IP PBX: The CudaTel > > Communication Server > > > >> > > > >> > > > >> Official FreeSWITCH Sites > > > >> http://www.freeswitch.org > > > >> http://wiki.freeswitch.org > > > >> http://www.cluecon.com > > > >> > > > >> 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 > > > >> > > > >> > > > >> > > > >> > > _________________________________________________________________________ > > > >> Professional FreeSWITCH Consulting Services: > > > >> consulting at freeswitch.org > > > >> http://www.freeswitchsolutions.com > > > >> > > > >> FreeSWITCH-powered IP PBX: The CudaTel Communication > > Server > > > >> > > > >> > > > >> Official FreeSWITCH Sites > > > >> http://www.freeswitch.org > > > >> http://wiki.freeswitch.org > > > >> http://www.cluecon.com > > > >> > > > >> 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 > > > > > > > > > > > > > > > > > > _________________________________________________________________________ > > > > Professional FreeSWITCH Consulting Services: > > > > consulting at freeswitch.org > > > > http://www.freeswitchsolutions.com > > > > > > > > FreeSWITCH-powered IP PBX: The CudaTel Communication > > Server > > > > > > > > > > > > Official FreeSWITCH Sites > > > > http://www.freeswitch.org > > > > http://wiki.freeswitch.org > > > > http://www.cluecon.com > > > > > > > > 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 > > > > > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > > > > > 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 > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > 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 > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130205/1cee132c/attachment-0001.html From steveayre at gmail.com Tue Feb 5 23:56:44 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 5 Feb 2013 20:56:44 +0000 Subject: [Freeswitch-users] Get recorded file from callcenter via API. In-Reply-To: <511168CB.8060905@gmail.com> References: <510EAB50.80308@gmail.com> <1FFF97C269757C458224B7C895F35F15215DC1@cantor.std.visionutv.se> <510F4EAD.3040807@gmail.com> <2D54617F-C018-491F-B19F-9FAA02D5412D@visionutveckling.se> <510F64BD.6070107@gmail.com> <51114260.20207@gmail.com> <511156C6.1070701@gmail.com> <511168CB.8060905@gmail.com> Message-ID: See http://wiki.freeswitch.org/wiki/Mod_rtmp and the Flex Client What A and B are don't really matter - all calls go via the core and don't care about what the other endpoint is. Just don't use proxy_media or bypass_media for these calls (being SDP based I doubt that would play nicely with RTMP). -Steve On 5 February 2013 20:17, Mimiko wrote: > On 05.02.2013 21:53, Michael Collins wrote: > > What do you mean "type it in window"? In any case you can use the fs_cli > > "system" API to send a command to the operating system to be executed. > > > > -MC > > Well, I will try "system". By the way, is it possible to connect a flash > player audio only via fs cli or event socket or other means in php to FS > to hear a channel? Ie, A channel is a web browser, and B channel is a > phone? > > -- > Mimiko desu. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130205/46116efe/attachment.html From krice at freeswitch.org Wed Feb 6 01:39:53 2013 From: krice at freeswitch.org (Ken Rice) Date: Tue, 05 Feb 2013 16:39:53 -0600 Subject: [Freeswitch-users] FreeSWITCH git repo undergoing maintenance Message-ID: Hey guys, Yes we know git is hosed up right now... We?re taking the time to do some clean up and upgrades to the GIT server... Once it comes back online there will be an announcement to this. K -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130205/5f2c9f64/attachment.html From itamar at ispbrasil.com.br Wed Feb 6 02:15:10 2013 From: itamar at ispbrasil.com.br (Itamar Reis Peixoto) Date: Tue, 5 Feb 2013 21:15:10 -0200 Subject: [Freeswitch-users] FreeSWITCH git repo undergoing maintenance In-Reply-To: References: Message-ID: On Tue, Feb 5, 2013 at 8:39 PM, Ken Rice wrote: > Hey guys, > > Yes we know git is hosed up right now... We?re taking the time to do some > clean up and upgrades to the GIT server... Once it comes back online there > will be an announcement to this. > what do you think about moving to github and leaving current servers as backup of github ? ------------ Itamar Reis Peixoto From Pascal.Taube at gmx.de Wed Feb 6 02:04:04 2013 From: Pascal.Taube at gmx.de (taube) Date: Tue, 5 Feb 2013 15:04:04 -0800 (PST) Subject: [Freeswitch-users] Freeswitch second internal profile Message-ID: <1360105444715-7587041.post@n2.nabble.com> Hi all, I created a second internal profile to connect with the phones to freeswitch via two networks. The internal profile works. I can call out -in and can contact the extension with fs_cli sofia_contact. At the second profile the phones register and i can contact them but when i try to call them, i get the message that the user is not registered, are the any hints when i create a second internal profile? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-second-internal-profile-tp7587041.html Sent from the freeswitch-users mailing list archive at Nabble.com. From steveayre at gmail.com Wed Feb 6 02:37:58 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 5 Feb 2013 23:37:58 +0000 Subject: [Freeswitch-users] FreeSWITCH git repo undergoing maintenance In-Reply-To: References: Message-ID: Github is awful at handling pull requests. Read Linus Torvalds views on it: https://github.com/torvalds/linux/pull/17 -Steve On 5 February 2013 23:15, Itamar Reis Peixoto wrote: > On Tue, Feb 5, 2013 at 8:39 PM, Ken Rice wrote: > > Hey guys, > > > > Yes we know git is hosed up right now... We?re taking the time to do some > > clean up and upgrades to the GIT server... Once it comes back online > there > > will be an announcement to this. > > > > what do you think about moving to github and leaving current servers > as backup of github ? > > > > ------------ > > Itamar Reis Peixoto > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130205/c27ff33d/attachment.html From steveayre at gmail.com Wed Feb 6 02:39:31 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 5 Feb 2013 23:39:31 +0000 Subject: [Freeswitch-users] FreeSWITCH git repo undergoing maintenance In-Reply-To: References: Message-ID: Plus it's not like github hasn't had their own problems in the past: http://thenextweb.com/insider/2012/12/23/after-scheduled-maintenance-github-confirms-major-service-outage-across-all-its-services/ At least with the devs running their own git server and github as a mirror if there's a problem it's within their control and the mirror means people can still do checkouts if needed. -Steve On 5 February 2013 23:37, Steven Ayre wrote: > Github is awful at handling pull requests. > > Read Linus Torvalds views on it: https://github.com/torvalds/linux/pull/17 > > -Steve > > > > On 5 February 2013 23:15, Itamar Reis Peixoto wrote: > >> On Tue, Feb 5, 2013 at 8:39 PM, Ken Rice wrote: >> > Hey guys, >> > >> > Yes we know git is hosed up right now... We?re taking the time to do >> some >> > clean up and upgrades to the GIT server... Once it comes back online >> there >> > will be an announcement to this. >> > >> >> what do you think about moving to github and leaving current servers >> as backup of github ? >> >> >> >> ------------ >> >> Itamar Reis Peixoto >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130205/d9c9f2f3/attachment.html From krice at freeswitch.org Wed Feb 6 02:44:51 2013 From: krice at freeswitch.org (Ken Rice) Date: Tue, 05 Feb 2013 17:44:51 -0600 Subject: [Freeswitch-users] FreeSWITCH git repo undergoing maintenance In-Reply-To: Message-ID: That will never happen.... What happens when GITHUB decides its no longer going to be free? Or makes some other change we don't agree with? (I'm not saying that this is going to happen, but the possibility is always there) This is the same reason we use Jira, but don't use the hosted versions Atlasian provides. The project controls the servers. This means if for whatever reason we need to change hosting services, etc, we can just move the servers (or in some cases the VMs we have) from provider A to provider B... (we recently did this with very little interruption to the public services) Also, that being said our GIT Servers are back online... K On 2/5/13 5:15 PM, "Itamar Reis Peixoto" wrote: > On Tue, Feb 5, 2013 at 8:39 PM, Ken Rice wrote: >> Hey guys, >> >> Yes we know git is hosed up right now... We?re taking the time to do some >> clean up and upgrades to the GIT server... Once it comes back online there >> will be an announcement to this. >> > > what do you think about moving to github and leaving current servers > as backup of github ? > > > > ------------ > > Itamar Reis Peixoto > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch From krice at freeswitch.org Wed Feb 6 02:47:40 2013 From: krice at freeswitch.org (Ken Rice) Date: Tue, 05 Feb 2013 17:47:40 -0600 Subject: [Freeswitch-users] GIT Is Back.... Upgrades complete Message-ID: Hey Guys, The GIT Server is now upgraded and back online, Special Thanks To Ray ?[intra]lanman? Chandler for taking care of this stuff... K -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130205/99f803b0/attachment.html From william.suffill at gmail.com Wed Feb 6 04:11:29 2013 From: william.suffill at gmail.com (William Suffill) Date: Tue, 5 Feb 2013 20:11:29 -0500 Subject: [Freeswitch-users] FreeSWITCH git repo undergoing maintenance In-Reply-To: References: Message-ID: GitLab is decent for those that care for a similar frontend ala Github but hosted in house. Built on top of Gitolite in RoR. [No vested interest myself just a happy user where appropriate.] Depending on the situation gitolite isn't that bad manually but some people like web management tools. =) Also in the case here Jira provides the issues tracking and Fisheye for the source viewer so no real need. Congrats to getting it fixed up so quickly tho. On Tue, Feb 5, 2013 at 6:44 PM, Ken Rice wrote: > That will never happen.... > > What happens when GITHUB decides its no longer going to be free? Or makes > some other change we don't agree with? (I'm not saying that this is going > to > happen, but the possibility is always there) > > This is the same reason we use Jira, but don't use the hosted versions > Atlasian provides. The project controls the servers. This means if for > whatever reason we need to change hosting services, etc, we can just move > the servers (or in some cases the VMs we have) from provider A to provider > B... (we recently did this with very little interruption to the public > services) > > Also, that being said our GIT Servers are back online... > > K > > > > > On 2/5/13 5:15 PM, "Itamar Reis Peixoto" wrote: > > > On Tue, Feb 5, 2013 at 8:39 PM, Ken Rice wrote: > >> Hey guys, > >> > >> Yes we know git is hosed up right now... We?re taking the time to do > some > >> clean up and upgrades to the GIT server... Once it comes back online > there > >> will be an announcement to this. > >> > > > > what do you think about moving to github and leaving current servers > > as backup of github ? > > > > > > > > ------------ > > > > Itamar Reis Peixoto > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > 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 > > -- > Ken > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > irc.freenode.net #freeswitch > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130205/26f82535/attachment.html From Sirish.MasurMohan at oa.com.au Wed Feb 6 05:08:09 2013 From: Sirish.MasurMohan at oa.com.au (Sirish Masur Mohan) Date: Wed, 6 Feb 2013 13:08:09 +1100 Subject: [Freeswitch-users] FreeSWITCH process dies Message-ID: <965759A53E43FE439E43565A7715E5F0591C0860EE@oa-exchange1.oa.com.au> Hi, In last 2 days, I have observed that FreeSWITCH process, after being up for about a day or so, dies unexpectedly. This has happened twice, on two different servers. There is no indication of failure in freeswitch log. I am using a 32-bit Ubuntu, with FreeSWITCH version being FreeSWITCH Version 1.3.13b. Any clues on how to debug this? Thanks! With regards, Sirish -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130206/634647eb/attachment.html From krice at freeswitch.org Wed Feb 6 05:22:09 2013 From: krice at freeswitch.org (Ken Rice) Date: Tue, 05 Feb 2013 20:22:09 -0600 Subject: [Freeswitch-users] FreeSWITCH process dies In-Reply-To: <965759A53E43FE439E43565A7715E5F0591C0860EE@oa-exchange1.oa.com.au> Message-ID: Enable dropping cores and then get a back trace... On 2/5/13 8:08 PM, "Sirish Masur Mohan" wrote: > Hi, > > In last 2 days, I have observed that FreeSWITCH process, after being up for > about a day or so, dies unexpectedly. This has happened twice, on two > different servers. There is no indication of failure in freeswitch log. > > I am using a 32-bit Ubuntu, with FreeSWITCH version being FreeSWITCH Version > 1.3.13b. > > Any clues on how to debug this? > > Thanks! > > With regards, > Sirish > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130205/e955648d/attachment.html From gabe at gundy.org Wed Feb 6 06:00:15 2013 From: gabe at gundy.org (Gabriel Gunderson) Date: Tue, 5 Feb 2013 20:00:15 -0700 Subject: [Freeswitch-users] Freeswitch second internal profile In-Reply-To: <1360105444715-7587041.post@n2.nabble.com> References: <1360105444715-7587041.post@n2.nabble.com> Message-ID: On Tue, Feb 5, 2013 at 4:04 PM, taube wrote: > At the second profile the phones register and i can contact them but when i > try to call them, i get the message that the user is not registered, are the > any hints when i create a second internal profile? Show us your two dialstrings. GAbe From gabe at gundy.org Wed Feb 6 07:48:01 2013 From: gabe at gundy.org (Gabriel Gunderson) Date: Tue, 5 Feb 2013 21:48:01 -0700 Subject: [Freeswitch-users] FreeSWITCH git repo undergoing maintenance In-Reply-To: References: Message-ID: On Tue, Feb 5, 2013 at 6:11 PM, William Suffill wrote: > GitLab is decent for those that care for a similar frontend ala Github but > hosted in house. Built on top of Gitolite in RoR. Kinda off topic here, but GitLab is dropping Gitolite in v5. They're rolling their own. They say there are too many issues with it. Gabe From t.mahe at b-and-c.net Wed Feb 6 05:19:06 2013 From: t.mahe at b-and-c.net (=?ISO-8859-1?Q?Tristan_Mah=E9?=) Date: Wed, 06 Feb 2013 03:19:06 +0100 Subject: [Freeswitch-users] FreeSWITCH process dies In-Reply-To: <965759A53E43FE439E43565A7715E5F0591C0860EE@oa-exchange1.oa.com.au> References: <965759A53E43FE439E43565A7715E5F0591C0860EE@oa-exchange1.oa.com.au> Message-ID: <5111BD9A.4050703@b-and-c.net> Hi, Turn debug on ( fsctl loglevel 7 ) and check for coredumps on your system ( ulimit -c ). Regards, Gled. On 06/02/2013 03:08, Sirish Masur Mohan wrote: > > Hi, > > In last 2 days, I have observed that FreeSWITCH process, after being > up for about a day or so, dies unexpectedly. This has happened twice, > on two different servers. There is no indication of failure in > freeswitch log. > > I am using a 32-bit Ubuntu, with FreeSWITCH version being FreeSWITCH > Version 1.3.13b. > > Any clues on how to debug this? > > Thanks! > > With regards, > > Sirish > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130206/b7083f47/attachment.html From yyu0378 at gmail.com Wed Feb 6 06:43:47 2013 From: yyu0378 at gmail.com (yyu0378) Date: Wed, 06 Feb 2013 12:43:47 +0900 Subject: [Freeswitch-users] DTMF both inband and RFC2833? Message-ID: <5111D173.5020006@gmail.com> hi: I use peer reg at fs.If peer only send inband DTMF signal,fs passthrough it.But if peer send both inband and RFC2833,then fs can`t passthrough signal.I try add start_dtmf at dialplan,no change.This is bug?Or how I can pass both them?Thank you! From miha at softnet.si Wed Feb 6 11:19:22 2013 From: miha at softnet.si (Miha) Date: Wed, 06 Feb 2013 09:19:22 +0100 Subject: [Freeswitch-users] 407 proxy authorization, 302 redirect Message-ID: <5112120A.4000900@softnet.si> Hi, I set manul-redirect in sip_profiles and when compes 302 FS handles it like it should (I have configured my dialplan for handling manul-redirects). After looking with whireshark I noticed that 407 is not happening when it goes for 302. How to deal whit this issue so that every 302 will be also authorized? thanks! miha From jaasmailing at gmail.com Wed Feb 6 11:47:32 2013 From: jaasmailing at gmail.com (Carlo Dimaggio) Date: Wed, 06 Feb 2013 09:47:32 +0100 Subject: [Freeswitch-users] lua freeswitch.Dbh sqlite error In-Reply-To: References: <5110C94F.7080901@gmail.com> <1360062212.8347.200.camel@luna.madrid.commsmundi.com> <1360078923.8347.280.camel@luna.madrid.commsmundi.com> <1360081909.22365.35.camel@luna.madrid.commsmundi.com> Message-ID: <511218A4.9010107@gmail.com> I have the same behaviour. Without prefix it works, while with sqlite:/// the error is: 2013-02-06 09:46:29.169603 [ERR] switch_odbc.c:365 STATE: IM002 CODE 0 ERROR: [unixODBC][Driver Manager]Data source name not found, and no default driver specified 2013-02-06 09:46:29.169603 [CRIT] switch_core_sqldb.c:504 Failure to connect to PGSQL sqlite! 2013-02-06 09:46:29.169603 [ERR] freeswitch_lua.cpp:354 Connection failed. DBH NOT Connected. Regards, Il 05/02/13 17.42, Steven Ayre ha scritto: > > FAIL => "sqlite:///tmp/test.db" (this one should work) > > > What error does this give? > > > On 5 February 2013 16:31, Fran?ois Delawarde > > > wrote: > > I'm on HEAD, but I must be doing something wrong, there is no apparent > reason from the code. > > On Tue, 2013-02-05 at 16:05 +0000, Steven Ayre wrote: > > Honestly, I only use ODBC... Maybe someone else can shed some light? > > > > What version are you using? > > > > Steve > > > > > > > > On 5 Feb 2013, at 15:42, Fran?ois Delawarde > > wrote: > > > > > Steven, does "sqlite://" prefix work for you? Not sure if it's > a bug or > > > I'm doing something wrong. I tried the following from lua dbh: > > > > > > FAIL => "sqlite:///tmp/test.db" (this one should work) > > > FAIL => "sqlite://tmp/test.db" (shouldn't work, but just in > case...) > > > OK => "/tmp/test.db" (no prefix) > > > > > > Fran?ois. > > > > > > > > > On Tue, 2013-02-05 at 12:55 +0000, Steven Ayre wrote: > > >> core: was an old prefix? I knew that change broke some ODBC DSNs > > >> (names without username+password, eg when they're specified in > > >> odbc.ini) but hadn't realised that had been broken too - that > needs > > >> documenting as well on the Wiki I guess... > > >> http://wiki.freeswitch.org/wiki/Release_Notes#odbc-dsn > > >> > > >> > > >> Just a thought for the developers, but perhaps modules with > odbc-dsn > > >> parameters should have this renamed to dsn or something now > since it's > > >> not odbc-specific at all any longer. Easy enough to recognise > both > > >> 'dsn' and 'odbc-dsn' names to remain backwards compatible > (which I've > > >> seen done with typos in config setting names in the past > > >> as precedent). > > >> > > >> > > >> -Steve > > >> > > >> > > >> > > >> > > >> On 5 February 2013 11:03, Fran?ois Delawarde > > >> > wrote: > > >> The behavior recently changed from "core:" to > > >> "sqlite://". > > >> > > >> Update: I just tested and it seems that "sqlite://" > > >> doesn't work, > > >> but as sqlite is the default, you can just omit the > prefix and > > >> use > > >> "" to make it work. > > >> > > >> Fran?ois. > > >> > > >> > > >> On Tue, 2013-02-05 at 10:37 +0000, Steven Ayre wrote: > > >>> 'Failure to connect to PGSQL core' happens on ODBC > > >> connections, that > > >>> seems to be a minor bug in the error message. > > >>> > > >>> > > >>> From the unixODBC error it's interpreting this as a ODBC > > >> DSN. The > > >>> reason is the sqlite prefix is sqlite:// not core:// > > >>> (see http://wiki.freeswitch.org/wiki/DSN). You also don't > > >> have enough > > >>> slashes. > > >>> > > >>> > > >>> Try: > > >>> local dbh = > > >> freeswitch.Dbh("sqlite:///usr/local/temp/database.db"); > > >>> > > >>> > > >>> > > >>> (note the 3 slashes - :// is standard URI format, the 3rd > > >> indicates an > > >>> absolute path). > > >>> > > >>> > > >>> -Steve > > >>> > > >>> > > >>> > > >>> > > >>> On 5 February 2013 08:56, Carlo Dimaggio > > >> > > > >>> wrote: > > >>> Hi all! > > >>> > > >>> I have a lua script in which i have a statement that > > >> connects > > >>> to a sqlite db: > > >>> > > >>> local dbh = > > >>> freeswitch.Dbh("core:/usr/local/temp/database.db"); > > >>> > > >>> but i receve in FS CLI this error that refers to a > > >> Pgsql > > >>> instead of sqlite: > > >>> > > >>> 2013-02-05 09:40:10.379612 [ERR] switch_odbc.c:365 > > >> STATE: > > >>> IM002 CODE 0 ERROR: [unixODBC][Driver Manager]Data > > >> source name > > >>> not found, and no default driver specified > > >>> 2013-02-05 09:40:10.379612 [CRIT] > > >> switch_core_sqldb.c:504 > > >>> Failure to connect to PGSQL core! > > >>> 2013-02-05 09:40:10.379612 [ERR] > > >> freeswitch_lua.cpp:354 > > >>> Connection failed. DBH NOT Connected. > > >>> 2013-02-05 09:40:10.379612 [ERR] > > >> freeswitch_lua.cpp:435 DBH > > >>> NOT Connected. > > >>> > > >>> > > >>> Can you help me? > > >>> > > >>> Best regards, > > >> > _________________________________________________________________________ > > >>> Professional FreeSWITCH Consulting Services: > > >>> consulting at freeswitch.org > > >>> http://www.freeswitchsolutions.com > > >>> > > >>> FreeSWITCH-powered IP PBX: The CudaTel Communication > > >> Server > > >>> > > >>> > > >>> Official FreeSWITCH Sites > > >>> http://www.freeswitch.org > > >>> http://wiki.freeswitch.org > > >>> http://www.cluecon.com > > >>> > > >>> 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 > > >> > _________________________________________________________________________ > > >>> Professional FreeSWITCH Consulting Services: > > >>> consulting at freeswitch.org > > >>> http://www.freeswitchsolutions.com > > >>> > > >>> > > >>> > > >>> > > >>> Official FreeSWITCH Sites > > >>> http://www.freeswitch.org > > >>> http://wiki.freeswitch.org > > >>> http://www.cluecon.com > > >>> > > >>> 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 > > >> > > >> > > >> > > >> > _________________________________________________________________________ > > >> Professional FreeSWITCH Consulting Services: > > >> consulting at freeswitch.org > > >> http://www.freeswitchsolutions.com > > >> > > >> FreeSWITCH-powered IP PBX: The CudaTel Communication > Server > > >> > > >> > > >> Official FreeSWITCH Sites > > >> http://www.freeswitch.org > > >> http://wiki.freeswitch.org > > >> http://www.cluecon.com > > >> > > >> 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 > > >> > > >> > > >> > > >> > _________________________________________________________________________ > > >> Professional FreeSWITCH Consulting Services: > > >> consulting at freeswitch.org > > >> http://www.freeswitchsolutions.com > > >> > > >> > > >> > > >> > > >> Official FreeSWITCH Sites > > >> http://www.freeswitch.org > > >> http://wiki.freeswitch.org > > >> http://www.cluecon.com > > >> > > >> 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 > > > > > > > > > > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > > > > > 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 > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130206/c6a7058c/attachment-0001.html From p.varghese at cem-solutions.net Wed Feb 6 11:55:52 2013 From: p.varghese at cem-solutions.net (Varghese) Date: Wed, 06 Feb 2013 14:25:52 +0530 Subject: [Freeswitch-users] Reg. Freeswitch Performace on Freescale Board In-Reply-To: References: <5110FD13.3080406@cem-solutions.net> Message-ID: <51121A98.6070109@cem-solutions.net> Hi Anthony, Thanks for the information. Kernel is Linux-2.6.32 patched by Freescale team for the hardware. Further we tested the freeswitch-1.2.6 on Freescale and following are observations. We set the ulimit options as per the wiki link and ulimit ?s 240 1. With switches *-np* and *?lp* the result is same i.e. maximum of 90 calls and top command shows 100% CPU for freeswitch process 2. With RTP Bypass mode we could make 250 calls and CPU usage is 20% 3. With Proxy media mode, and default rtp mode the CPU is very high. Approximately 1 to 1.5% per call and scales up to 90 calls 4. Set the option enable-timer value = false in SIP profile, stun-enabled=false given all IP address in ext-rtp, rtp-timer-name=none/soft etc., still the results are same Q1) Can you explain ?asterisk also uses blocking reads in its rtp stack where we have timers. You can disable the timers as documented in the wiki? which parameter you are referring to and where we need to disable timers. Q2) Do we need to try any other options to increase the performance numbers? F.Y.I, We used Empirix Hammer SIP Call Generator for testing SIP calls. Thanks Varghese Paul Anthony Minessale wrote: > That depends on the version of linux, the kernel version and several > other factors. > > Were you actually moving media when testing, what tool are you testing > with? > > On 32 bit you should make sure you have the stack size at 240 (ulimit > -s 240 before starting) > Also you may need to add the -lp or -np startup flags to reduce the > realtime threads. > > asterisk also uses blocking reads in its rtp stack where we have > timers. You can disable the timers as documented in the wiki.. > > > > > On Tue, Feb 5, 2013 at 6:37 AM, Varghese > wrote: > > Hi all, > We are facing following observations/issues with freeswitch on > freescale > processor boards. > We ported freeswitch with freescale provided tool chain. > _*Freeswitch Configuration: version 1.2.6 (git version)*_ > 1. Only enabled mod_sofia, mod_dialplan, mod_command,mod_dptool and > mod_console. > 2. set the ulimit options and followed the steps in the following link > http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations > 3. SIP test tool run the load test for 500 users in freeswitch and > configured the dial plan to just bridge the channels > 4. Freeswitch media processing is in default mode: i.e media will go > through freeswitch > _*Freescale Board Configuration:*_ > CPU : 1GHZ PowerQUICC III, 32 bit > RAM: 1 GB > Linux- 2.6.32 > Load test is conducted for SIP to SIP calls with RTP. > _*Observations:*_ > Linux TOP command shows increasing CPU usage per call and increases to > 100% with only 90 calls. After that system response is very slow > We could also found delay in media. > > _Surprisingly, with Asterisk 1.8 the same system configuration > works 250 > calls without any problems. > _ > _*Questions: > *_1. Any one ported Freeswitch on Freescale or any embedded > processors? > Any performance figures or references will be appreciated > 2. How many simultaneous calls can be possible with 1GHz > PowerQUICC III > processor ? > 3. Any more settings are required in freeswitch or linux for > scalability? > > Thanks And Regards > > Varghese Paul > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > googletalk:conf+888 at conference.freeswitch.org > > pstn:+19193869900 > ------------------------------------------------------------------------ > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 steveayre at gmail.com Wed Feb 6 12:40:41 2013 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 6 Feb 2013 09:40:41 +0000 Subject: [Freeswitch-users] lua freeswitch.Dbh sqlite error In-Reply-To: <511218A4.9010107@gmail.com> References: <5110C94F.7080901@gmail.com> <1360062212.8347.200.camel@luna.madrid.commsmundi.com> <1360078923.8347.280.camel@luna.madrid.commsmundi.com> <1360081909.22365.35.camel@luna.madrid.commsmundi.com> <511218A4.9010107@gmail.com> Message-ID: Hi Carlo, It was a bug in FS which has been fixed in the master branch (1.3). Try upgrading to verify that fixes your error. At some point that'll trickle down into the v1.2.stable branch too. -Steve On 6 February 2013 08:47, Carlo Dimaggio wrote: > I have the same behaviour. > Without prefix it works, while with sqlite:/// the error is: > > 2013-02-06 09:46:29.169603 [ERR] switch_odbc.c:365 STATE: IM002 CODE 0 > ERROR: [unixODBC][Driver Manager]Data source name not found, and no default > driver specified > > 2013-02-06 09:46:29.169603 [CRIT] switch_core_sqldb.c:504 Failure to > connect to PGSQL sqlite! > 2013-02-06 09:46:29.169603 [ERR] freeswitch_lua.cpp:354 Connection > failed. DBH NOT Connected. > > > Regards, > > Il 05/02/13 17.42, Steven Ayre ha scritto: > > FAIL => "sqlite:///tmp/test.db" (this one should work) > > > What error does this give? > > > On 5 February 2013 16:31, Fran?ois Delawarde > wrote: > >> I'm on HEAD, but I must be doing something wrong, there is no apparent >> reason from the code. >> >> On Tue, 2013-02-05 at 16:05 +0000, Steven Ayre wrote: >> > Honestly, I only use ODBC... Maybe someone else can shed some light? >> > >> > What version are you using? >> > >> > Steve >> > >> > >> > >> > On 5 Feb 2013, at 15:42, Fran?ois Delawarde < >> fdelawarde at wirelessmundi.com> wrote: >> > >> > > Steven, does "sqlite://" prefix work for you? Not sure if it's a bug >> or >> > > I'm doing something wrong. I tried the following from lua dbh: >> > > >> > > FAIL => "sqlite:///tmp/test.db" (this one should work) >> > > FAIL => "sqlite://tmp/test.db" (shouldn't work, but just in case...) >> > > OK => "/tmp/test.db" (no prefix) >> > > >> > > Fran?ois. >> > > >> > > >> > > On Tue, 2013-02-05 at 12:55 +0000, Steven Ayre wrote: >> > >> core: was an old prefix? I knew that change broke some ODBC DSNs >> > >> (names without username+password, eg when they're specified in >> > >> odbc.ini) but hadn't realised that had been broken too - that needs >> > >> documenting as well on the Wiki I guess... >> > >> http://wiki.freeswitch.org/wiki/Release_Notes#odbc-dsn >> > >> >> > >> >> > >> Just a thought for the developers, but perhaps modules with odbc-dsn >> > >> parameters should have this renamed to dsn or something now since >> it's >> > >> not odbc-specific at all any longer. Easy enough to recognise both >> > >> 'dsn' and 'odbc-dsn' names to remain backwards compatible (which I've >> > >> seen done with typos in config setting names in the past >> > >> as precedent). >> > >> >> > >> >> > >> -Steve >> > >> >> > >> >> > >> >> > >> >> > >> On 5 February 2013 11:03, Fran?ois Delawarde >> > >> wrote: >> > >> The behavior recently changed from "core:" to >> > >> "sqlite://". >> > >> >> > >> Update: I just tested and it seems that "sqlite://" >> > >> doesn't work, >> > >> but as sqlite is the default, you can just omit the prefix and >> > >> use >> > >> "" to make it work. >> > >> >> > >> Fran?ois. >> > >> >> > >> >> > >> On Tue, 2013-02-05 at 10:37 +0000, Steven Ayre wrote: >> > >>> 'Failure to connect to PGSQL core' happens on ODBC >> > >> connections, that >> > >>> seems to be a minor bug in the error message. >> > >>> >> > >>> >> > >>> From the unixODBC error it's interpreting this as a ODBC >> > >> DSN. The >> > >>> reason is the sqlite prefix is sqlite:// not core:// >> > >>> (see http://wiki.freeswitch.org/wiki/DSN). You also don't >> > >> have enough >> > >>> slashes. >> > >>> >> > >>> >> > >>> Try: >> > >>> local dbh = >> > >> freeswitch.Dbh("sqlite:///usr/local/temp/database.db"); >> > >>> >> > >>> >> > >>> >> > >>> (note the 3 slashes - :// is standard URI format, the 3rd >> > >> indicates an >> > >>> absolute path). >> > >>> >> > >>> >> > >>> -Steve >> > >>> >> > >>> >> > >>> >> > >>> >> > >>> On 5 February 2013 08:56, Carlo Dimaggio >> > >> >> > >>> wrote: >> > >>> Hi all! >> > >>> >> > >>> I have a lua script in which i have a statement that >> > >> connects >> > >>> to a sqlite db: >> > >>> >> > >>> local dbh = >> > >>> freeswitch.Dbh("core:/usr/local/temp/database.db"); >> > >>> >> > >>> but i receve in FS CLI this error that refers to a >> > >> Pgsql >> > >>> instead of sqlite: >> > >>> >> > >>> 2013-02-05 09:40:10.379612 [ERR] switch_odbc.c:365 >> > >> STATE: >> > >>> IM002 CODE 0 ERROR: [unixODBC][Driver Manager]Data >> > >> source name >> > >>> not found, and no default driver specified >> > >>> 2013-02-05 09:40:10.379612 [CRIT] >> > >> switch_core_sqldb.c:504 >> > >>> Failure to connect to PGSQL core! >> > >>> 2013-02-05 09:40:10.379612 [ERR] >> > >> freeswitch_lua.cpp:354 >> > >>> Connection failed. DBH NOT Connected. >> > >>> 2013-02-05 09:40:10.379612 [ERR] >> > >> freeswitch_lua.cpp:435 DBH >> > >>> NOT Connected. >> > >>> >> > >>> >> > >>> Can you help me? >> > >>> >> > >>> Best regards, >> > >> >> _________________________________________________________________________ >> > >>> Professional FreeSWITCH Consulting Services: >> > >>> consulting at freeswitch.org >> > >>> http://www.freeswitchsolutions.com >> > >>> >> > >>> FreeSWITCH-powered IP PBX: The CudaTel Communication >> > >> Server >> > >>> >> > >>> >> > >>> Official FreeSWITCH Sites >> > >>> http://www.freeswitch.org >> > >>> http://wiki.freeswitch.org >> > >>> http://www.cluecon.com >> > >>> >> > >>> 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 >> > >> >> _________________________________________________________________________ >> > >>> Professional FreeSWITCH Consulting Services: >> > >>> consulting at freeswitch.org >> > >>> http://www.freeswitchsolutions.com >> > >>> >> > >>> >> > >>> >> > >>> >> > >>> Official FreeSWITCH Sites >> > >>> http://www.freeswitch.org >> > >>> http://wiki.freeswitch.org >> > >>> http://www.cluecon.com >> > >>> >> > >>> 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 >> > >> >> > >> >> > >> >> > >> >> _________________________________________________________________________ >> > >> Professional FreeSWITCH Consulting Services: >> > >> consulting at freeswitch.org >> > >> http://www.freeswitchsolutions.com >> > >> >> > >> >> > >> >> > >> >> > >> Official FreeSWITCH Sites >> > >> http://www.freeswitch.org >> > >> http://wiki.freeswitch.org >> > >> http://www.cluecon.com >> > >> >> > >> 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 >> > >> >> > >> >> > >> >> > >> >> _________________________________________________________________________ >> > >> Professional FreeSWITCH Consulting Services: >> > >> consulting at freeswitch.org >> > >> http://www.freeswitchsolutions.com >> > >> >> > >> >> > >> >> > >> >> > >> Official FreeSWITCH Sites >> > >> http://www.freeswitch.org >> > >> http://wiki.freeswitch.org >> > >> http://www.cluecon.com >> > >> >> > >> 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 >> > > >> > > >> > > >> > > >> _________________________________________________________________________ >> > > Professional FreeSWITCH Consulting Services: >> > > consulting at freeswitch.org >> > > http://www.freeswitchsolutions.com >> > > >> > > >> > > >> > > >> > > Official FreeSWITCH Sites >> > > http://www.freeswitch.org >> > > http://wiki.freeswitch.org >> > > http://www.cluecon.com >> > > >> > > 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 >> > >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > 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 >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130206/83fa806f/attachment-0001.html From steveayre at gmail.com Wed Feb 6 12:52:01 2013 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 6 Feb 2013 09:52:01 +0000 Subject: [Freeswitch-users] FreeSWITCH process dies In-Reply-To: <5111BD9A.4050703@b-and-c.net> References: <965759A53E43FE439E43565A7715E5F0591C0860EE@oa-exchange1.oa.com.au> <5111BD9A.4050703@b-and-c.net> Message-ID: I'd recommend you examine the 'dmesg' command output. Look for lines that indicate a segfault or general protection fault occurred, eg: [16203366.364647] freeswitch[949]: segfault at 7f170000001a ip 7f171a86c34b sp 4c32ce20 error 4 in libc-2.7.so[7f171a7f6000+14a000] By default coredumps (if enabled) get written to the current working directory of freeswitch as the user it's running as - so it needs to be writeable by that user. They'll generally either be called core or core.PID. To get a coredump start freeswitch with the -core option. This is equivalent to starting freeswitch with 'ulimit -c unlimited' in effect (on systems supporting setrlimit). Once you have a coredump you can collect a backtrace and attach that to the Jira ticket. That shows what FS was doing when the crash occurred. http://wiki.freeswitch.org/wiki/Reporting_Bugs#Creating_A_Backtrace_With_gdb_.28Linux.2FUnix.29 Before raising a Jira upgrade to the latest version of the master branch and try to reproduce the crash there, just in case you're hitting a bug that's already been fixed. If it's fixed there and not in the latest version of the v1.2.stable branch that would also be useful to know. -Steve On 6 February 2013 02:19, Tristan Mah? wrote: > Hi, > > Turn debug on ( fsctl loglevel 7 ) and check for coredumps on your system > ( ulimit -c ). > > Regards, > > Gled. > > > On 06/02/2013 03:08, Sirish Masur Mohan wrote: > > Hi,**** > > ** ** > > In last 2 days, I have observed that FreeSWITCH process, after being up > for about a day or so, dies unexpectedly. This has happened twice, on two > different servers. There is no indication of failure in freeswitch log.*** > * > > ** ** > > I am using a 32-bit Ubuntu, with FreeSWITCH version being FreeSWITCH > Version 1.3.13b.**** > > ** ** > > Any clues on how to debug this? **** > > ** ** > > Thanks!**** > > ** ** > > With regards,**** > > Sirish**** > > ** ** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130206/16ad3448/attachment.html From jaasmailing at gmail.com Wed Feb 6 16:09:27 2013 From: jaasmailing at gmail.com (Carlo Dimaggio) Date: Wed, 06 Feb 2013 14:09:27 +0100 Subject: [Freeswitch-users] lua freeswitch.Dbh sqlite error In-Reply-To: References: <5110C94F.7080901@gmail.com> <1360062212.8347.200.camel@luna.madrid.commsmundi.com> <1360078923.8347.280.camel@luna.madrid.commsmundi.com> <1360081909.22365.35.camel@luna.madrid.commsmundi.com> <511218A4.9010107@gmail.com> Message-ID: <51125607.3070801@gmail.com> I'm triying to download the latest git (as I usually do) but I have this error... do you know what could be the problem? <14>:git clone http://git.freeswitch.org/freeswitch.git Initialized empty Git repository in /usr/src/freeswitch/.git/ error: The requested URL returned error: 500 while accessing http://git.freeswitch.org/freeswitch.git/info/refs fatal: HTTP request failed Il 06/02/13 10.40, Steven Ayre ha scritto: > Hi Carlo, > > It was a bug in FS which has been fixed in the master branch (1.3). > Try upgrading to verify that fixes your error. At some point that'll > trickle down into the v1.2.stable branch too. > > -Steve > > > > On 6 February 2013 08:47, Carlo Dimaggio > wrote: > > I have the same behaviour. > Without prefix it works, while with sqlite:/// the error is: > > 2013-02-06 09:46:29.169603 [ERR] switch_odbc.c:365 STATE: IM002 > CODE 0 ERROR: [unixODBC][Driver Manager]Data source name not > found, and no default driver specified > > 2013-02-06 09:46:29.169603 [CRIT] switch_core_sqldb.c:504 Failure > to connect to PGSQL sqlite! > 2013-02-06 09:46:29.169603 [ERR] freeswitch_lua.cpp:354 Connection > failed. DBH NOT Connected. > > > Regards, > > Il 05/02/13 17.42, Steven Ayre ha scritto: >> >> FAIL => "sqlite:///tmp/test.db" (this one should work) >> >> >> What error does this give? >> >> >> On 5 February 2013 16:31, Fran?ois Delawarde >> > > wrote: >> >> I'm on HEAD, but I must be doing something wrong, there is no >> apparent >> reason from the code. >> >> On Tue, 2013-02-05 at 16:05 +0000, Steven Ayre wrote: >> > Honestly, I only use ODBC... Maybe someone else can shed >> some light? >> > >> > What version are you using? >> > >> > Steve >> > >> > >> > >> > On 5 Feb 2013, at 15:42, Fran?ois Delawarde >> > > wrote: >> > >> > > Steven, does "sqlite://" prefix work for you? Not sure if >> it's a bug or >> > > I'm doing something wrong. I tried the following from lua >> dbh: >> > > >> > > FAIL => "sqlite:///tmp/test.db" (this one should work) >> > > FAIL => "sqlite://tmp/test.db" (shouldn't work, but just >> in case...) >> > > OK => "/tmp/test.db" (no prefix) >> > > >> > > Fran?ois. >> > > >> > > >> > > On Tue, 2013-02-05 at 12:55 +0000, Steven Ayre wrote: >> > >> core: was an old prefix? I knew that change broke some >> ODBC DSNs >> > >> (names without username+password, eg when they're >> specified in >> > >> odbc.ini) but hadn't realised that had been broken too - >> that needs >> > >> documenting as well on the Wiki I guess... >> > >> http://wiki.freeswitch.org/wiki/Release_Notes#odbc-dsn >> > >> >> > >> >> > >> Just a thought for the developers, but perhaps modules >> with odbc-dsn >> > >> parameters should have this renamed to dsn or something >> now since it's >> > >> not odbc-specific at all any longer. Easy enough to >> recognise both >> > >> 'dsn' and 'odbc-dsn' names to remain backwards >> compatible (which I've >> > >> seen done with typos in config setting names in the past >> > >> as precedent). >> > >> >> > >> >> > >> -Steve >> > >> >> > >> >> > >> >> > >> >> > >> On 5 February 2013 11:03, Fran?ois Delawarde >> > >> > > wrote: >> > >> The behavior recently changed from "core:" to >> > >> "sqlite://". >> > >> >> > >> Update: I just tested and it seems that >> "sqlite://" >> > >> doesn't work, >> > >> but as sqlite is the default, you can just omit >> the prefix and >> > >> use >> > >> "" to make it work. >> > >> >> > >> Fran?ois. >> > >> >> > >> >> > >> On Tue, 2013-02-05 at 10:37 +0000, Steven Ayre wrote: >> > >>> 'Failure to connect to PGSQL core' happens on ODBC >> > >> connections, that >> > >>> seems to be a minor bug in the error message. >> > >>> >> > >>> >> > >>> From the unixODBC error it's interpreting this as a ODBC >> > >> DSN. The >> > >>> reason is the sqlite prefix is sqlite:// not core:// >> > >>> (see http://wiki.freeswitch.org/wiki/DSN). You also don't >> > >> have enough >> > >>> slashes. >> > >>> >> > >>> >> > >>> Try: >> > >>> local dbh = >> > >> freeswitch.Dbh("sqlite:///usr/local/temp/database.db"); >> > >>> >> > >>> >> > >>> >> > >>> (note the 3 slashes - :// is standard URI format, the 3rd >> > >> indicates an >> > >>> absolute path). >> > >>> >> > >>> >> > >>> -Steve >> > >>> >> > >>> >> > >>> >> > >>> >> > >>> On 5 February 2013 08:56, Carlo Dimaggio >> > >> > > >> > >>> wrote: >> > >>> Hi all! >> > >>> >> > >>> I have a lua script in which i have a statement that >> > >> connects >> > >>> to a sqlite db: >> > >>> >> > >>> local dbh = >> > >>> freeswitch.Dbh("core:/usr/local/temp/database.db"); >> > >>> >> > >>> but i receve in FS CLI this error that refers to a >> > >> Pgsql >> > >>> instead of sqlite: >> > >>> >> > >>> 2013-02-05 09:40:10.379612 [ERR] switch_odbc.c:365 >> > >> STATE: >> > >>> IM002 CODE 0 ERROR: [unixODBC][Driver Manager]Data >> > >> source name >> > >>> not found, and no default driver specified >> > >>> 2013-02-05 09:40:10.379612 [CRIT] >> > >> switch_core_sqldb.c:504 >> > >>> Failure to connect to PGSQL core! >> > >>> 2013-02-05 09:40:10.379612 [ERR] >> > >> freeswitch_lua.cpp:354 >> > >>> Connection failed. DBH NOT Connected. >> > >>> 2013-02-05 09:40:10.379612 [ERR] >> > >> freeswitch_lua.cpp:435 DBH >> > >>> NOT Connected. >> > >>> >> > >>> >> > >>> Can you help me? >> > >>> >> > >>> Best regards, >> > >> >> _________________________________________________________________________ >> > >>> Professional FreeSWITCH Consulting Services: >> > >>> consulting at freeswitch.org >> >> > >>> http://www.freeswitchsolutions.com >> > >>> >> > >>> FreeSWITCH-powered IP PBX: The CudaTel Communication >> > >> Server >> > >>> >> > >>> >> > >>> Official FreeSWITCH Sites >> > >>> http://www.freeswitch.org >> > >>> http://wiki.freeswitch.org >> > >>> http://www.cluecon.com >> > >>> >> > >>> 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 >> > >> >> _________________________________________________________________________ >> > >>> Professional FreeSWITCH Consulting Services: >> > >>> consulting at freeswitch.org >> >> > >>> http://www.freeswitchsolutions.com >> > >>> >> > >>> >> > >>> >> > >>> >> > >>> Official FreeSWITCH Sites >> > >>> http://www.freeswitch.org >> > >>> http://wiki.freeswitch.org >> > >>> http://www.cluecon.com >> > >>> >> > >>> 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 >> > >> >> > >> >> > >> >> > >> >> _________________________________________________________________________ >> > >> Professional FreeSWITCH Consulting Services: >> > >> consulting at freeswitch.org >> > >> http://www.freeswitchsolutions.com >> > >> >> > >> FreeSWITCH-powered IP PBX: The CudaTel >> Communication Server >> > >> >> > >> >> > >> Official FreeSWITCH Sites >> > >> http://www.freeswitch.org >> > >> http://wiki.freeswitch.org >> > >> http://www.cluecon.com >> > >> >> > >> 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 >> > >> >> > >> >> > >> >> > >> >> _________________________________________________________________________ >> > >> Professional FreeSWITCH Consulting Services: >> > >> consulting at freeswitch.org >> > >> http://www.freeswitchsolutions.com >> > >> >> > >> >> > >> >> > >> >> > >> Official FreeSWITCH Sites >> > >> http://www.freeswitch.org >> > >> http://wiki.freeswitch.org >> > >> http://www.cluecon.com >> > >> >> > >> 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 >> > > >> > > >> > > >> > > >> _________________________________________________________________________ >> > > Professional FreeSWITCH Consulting Services: >> > > consulting at freeswitch.org >> > > http://www.freeswitchsolutions.com >> > > >> > > >> > > >> > > >> > > Official FreeSWITCH Sites >> > > http://www.freeswitch.org >> > > http://wiki.freeswitch.org >> > > http://www.cluecon.com >> > > >> > > 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 >> > >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > 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 >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130206/6344c65b/attachment-0001.html From steveayre at gmail.com Wed Feb 6 16:40:31 2013 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 6 Feb 2013 13:40:31 +0000 Subject: [Freeswitch-users] lua freeswitch.Dbh sqlite error In-Reply-To: <51125607.3070801@gmail.com> References: <5110C94F.7080901@gmail.com> <1360062212.8347.200.camel@luna.madrid.commsmundi.com> <1360078923.8347.280.camel@luna.madrid.commsmundi.com> <1360081909.22365.35.camel@luna.madrid.commsmundi.com> <511218A4.9010107@gmail.com> <51125607.3070801@gmail.com> Message-ID: It's git:// not http:// -Steve On 6 February 2013 13:09, Carlo Dimaggio wrote: > I'm triying to download the latest git (as I usually do) but I have this > error... do you know what could be the problem? > > > <14>:git clone http://git.freeswitch.org/freeswitch.git > Initialized empty Git repository in /usr/src/freeswitch/.git/ > error: The requested URL returned error: 500 while accessing > http://git.freeswitch.org/freeswitch.git/info/refs > > fatal: HTTP request failed > > > > > > Il 06/02/13 10.40, Steven Ayre ha scritto: > > Hi Carlo, > > It was a bug in FS which has been fixed in the master branch (1.3). Try > upgrading to verify that fixes your error. At some point that'll trickle > down into the v1.2.stable branch too. > > -Steve > > > > On 6 February 2013 08:47, Carlo Dimaggio wrote: > >> I have the same behaviour. >> Without prefix it works, while with sqlite:/// the error is: >> >> 2013-02-06 09:46:29.169603 [ERR] switch_odbc.c:365 STATE: IM002 CODE 0 >> ERROR: [unixODBC][Driver Manager]Data source name not found, and no default >> driver specified >> >> 2013-02-06 09:46:29.169603 [CRIT] switch_core_sqldb.c:504 Failure to >> connect to PGSQL sqlite! >> 2013-02-06 09:46:29.169603 [ERR] freeswitch_lua.cpp:354 Connection >> failed. DBH NOT Connected. >> >> >> Regards, >> >> Il 05/02/13 17.42, Steven Ayre ha scritto: >> >> FAIL => "sqlite:///tmp/test.db" (this one should work) >> >> >> What error does this give? >> >> >> On 5 February 2013 16:31, Fran?ois Delawarde < >> fdelawarde at wirelessmundi.com> wrote: >> >>> I'm on HEAD, but I must be doing something wrong, there is no apparent >>> reason from the code. >>> >>> On Tue, 2013-02-05 at 16:05 +0000, Steven Ayre wrote: >>> > Honestly, I only use ODBC... Maybe someone else can shed some light? >>> > >>> > What version are you using? >>> > >>> > Steve >>> > >>> > >>> > >>> > On 5 Feb 2013, at 15:42, Fran?ois Delawarde < >>> fdelawarde at wirelessmundi.com> wrote: >>> > >>> > > Steven, does "sqlite://" prefix work for you? Not sure if it's a bug >>> or >>> > > I'm doing something wrong. I tried the following from lua dbh: >>> > > >>> > > FAIL => "sqlite:///tmp/test.db" (this one should work) >>> > > FAIL => "sqlite://tmp/test.db" (shouldn't work, but just in case...) >>> > > OK => "/tmp/test.db" (no prefix) >>> > > >>> > > Fran?ois. >>> > > >>> > > >>> > > On Tue, 2013-02-05 at 12:55 +0000, Steven Ayre wrote: >>> > >> core: was an old prefix? I knew that change broke some ODBC DSNs >>> > >> (names without username+password, eg when they're specified in >>> > >> odbc.ini) but hadn't realised that had been broken too - that needs >>> > >> documenting as well on the Wiki I guess... >>> > >> http://wiki.freeswitch.org/wiki/Release_Notes#odbc-dsn >>> > >> >>> > >> >>> > >> Just a thought for the developers, but perhaps modules with odbc-dsn >>> > >> parameters should have this renamed to dsn or something now since >>> it's >>> > >> not odbc-specific at all any longer. Easy enough to recognise both >>> > >> 'dsn' and 'odbc-dsn' names to remain backwards compatible (which >>> I've >>> > >> seen done with typos in config setting names in the past >>> > >> as precedent). >>> > >> >>> > >> >>> > >> -Steve >>> > >> >>> > >> >>> > >> >>> > >> >>> > >> On 5 February 2013 11:03, Fran?ois Delawarde >>> > >> wrote: >>> > >> The behavior recently changed from "core:" to >>> > >> "sqlite://". >>> > >> >>> > >> Update: I just tested and it seems that "sqlite://" >>> > >> doesn't work, >>> > >> but as sqlite is the default, you can just omit the prefix >>> and >>> > >> use >>> > >> "" to make it work. >>> > >> >>> > >> Fran?ois. >>> > >> >>> > >> >>> > >> On Tue, 2013-02-05 at 10:37 +0000, Steven Ayre wrote: >>> > >>> 'Failure to connect to PGSQL core' happens on ODBC >>> > >> connections, that >>> > >>> seems to be a minor bug in the error message. >>> > >>> >>> > >>> >>> > >>> From the unixODBC error it's interpreting this as a ODBC >>> > >> DSN. The >>> > >>> reason is the sqlite prefix is sqlite:// not core:// >>> > >>> (see http://wiki.freeswitch.org/wiki/DSN). You also don't >>> > >> have enough >>> > >>> slashes. >>> > >>> >>> > >>> >>> > >>> Try: >>> > >>> local dbh = >>> > >> freeswitch.Dbh("sqlite:///usr/local/temp/database.db"); >>> > >>> >>> > >>> >>> > >>> >>> > >>> (note the 3 slashes - :// is standard URI format, the 3rd >>> > >> indicates an >>> > >>> absolute path). >>> > >>> >>> > >>> >>> > >>> -Steve >>> > >>> >>> > >>> >>> > >>> >>> > >>> >>> > >>> On 5 February 2013 08:56, Carlo Dimaggio >>> > >> >>> > >>> wrote: >>> > >>> Hi all! >>> > >>> >>> > >>> I have a lua script in which i have a statement that >>> > >> connects >>> > >>> to a sqlite db: >>> > >>> >>> > >>> local dbh = >>> > >>> freeswitch.Dbh("core:/usr/local/temp/database.db"); >>> > >>> >>> > >>> but i receve in FS CLI this error that refers to a >>> > >> Pgsql >>> > >>> instead of sqlite: >>> > >>> >>> > >>> 2013-02-05 09:40:10.379612 [ERR] switch_odbc.c:365 >>> > >> STATE: >>> > >>> IM002 CODE 0 ERROR: [unixODBC][Driver Manager]Data >>> > >> source name >>> > >>> not found, and no default driver specified >>> > >>> 2013-02-05 09:40:10.379612 [CRIT] >>> > >> switch_core_sqldb.c:504 >>> > >>> Failure to connect to PGSQL core! >>> > >>> 2013-02-05 09:40:10.379612 [ERR] >>> > >> freeswitch_lua.cpp:354 >>> > >>> Connection failed. DBH NOT Connected. >>> > >>> 2013-02-05 09:40:10.379612 [ERR] >>> > >> freeswitch_lua.cpp:435 DBH >>> > >>> NOT Connected. >>> > >>> >>> > >>> >>> > >>> Can you help me? >>> > >>> >>> > >>> Best regards, >>> > >> >>> _________________________________________________________________________ >>> > >>> Professional FreeSWITCH Consulting Services: >>> > >>> consulting at freeswitch.org >>> > >>> http://www.freeswitchsolutions.com >>> > >>> >>> > >>> FreeSWITCH-powered IP PBX: The CudaTel Communication >>> > >> Server >>> > >>> >>> > >>> >>> > >>> Official FreeSWITCH Sites >>> > >>> http://www.freeswitch.org >>> > >>> http://wiki.freeswitch.org >>> > >>> http://www.cluecon.com >>> > >>> >>> > >>> 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 >>> > >> >>> _________________________________________________________________________ >>> > >>> Professional FreeSWITCH Consulting Services: >>> > >>> consulting at freeswitch.org >>> > >>> http://www.freeswitchsolutions.com >>> > >>> >>> > >>> >>> > >>> >>> > >>> >>> > >>> Official FreeSWITCH Sites >>> > >>> http://www.freeswitch.org >>> > >>> http://wiki.freeswitch.org >>> > >>> http://www.cluecon.com >>> > >>> >>> > >>> 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 >>> > >> >>> > >> >>> > >> >>> > >> >>> _________________________________________________________________________ >>> > >> Professional FreeSWITCH Consulting Services: >>> > >> consulting at freeswitch.org >>> > >> http://www.freeswitchsolutions.com >>> > >> >>> > >> >>> > >> >>> > >> >>> > >> Official FreeSWITCH Sites >>> > >> http://www.freeswitch.org >>> > >> http://wiki.freeswitch.org >>> > >> http://www.cluecon.com >>> > >> >>> > >> 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 >>> > >> >>> > >> >>> > >> >>> > >> >>> _________________________________________________________________________ >>> > >> Professional FreeSWITCH Consulting Services: >>> > >> consulting at freeswitch.org >>> > >> http://www.freeswitchsolutions.com >>> > >> >>> > >> >>> > >> >>> > >> >>> > >> Official FreeSWITCH Sites >>> > >> http://www.freeswitch.org >>> > >> http://wiki.freeswitch.org >>> > >> http://www.cluecon.com >>> > >> >>> > >> 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 >>> > > >>> > > >>> > > >>> > > >>> _________________________________________________________________________ >>> > > Professional FreeSWITCH Consulting Services: >>> > > consulting at freeswitch.org >>> > > http://www.freeswitchsolutions.com >>> > > >>> > > >>> > > >>> > > >>> > > Official FreeSWITCH Sites >>> > > http://www.freeswitch.org >>> > > http://wiki.freeswitch.org >>> > > http://www.cluecon.com >>> > > >>> > > 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 >>> > >>> > >>> _________________________________________________________________________ >>> > Professional FreeSWITCH Consulting Services: >>> > consulting at freeswitch.org >>> > http://www.freeswitchsolutions.com >>> > >>> > >>> > >>> > >>> > Official FreeSWITCH Sites >>> > http://www.freeswitch.org >>> > http://wiki.freeswitch.org >>> > http://www.cluecon.com >>> > >>> > 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 >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >> >> >> >> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com >> >> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130206/2f7c501d/attachment-0001.html From jaasmailing at gmail.com Wed Feb 6 17:01:23 2013 From: jaasmailing at gmail.com (Carlo Dimaggio) Date: Wed, 06 Feb 2013 15:01:23 +0100 Subject: [Freeswitch-users] lua freeswitch.Dbh sqlite error In-Reply-To: References: <5110C94F.7080901@gmail.com> <1360062212.8347.200.camel@luna.madrid.commsmundi.com> <1360078923.8347.280.camel@luna.madrid.commsmundi.com> <1360081909.22365.35.camel@luna.madrid.commsmundi.com> <511218A4.9010107@gmail.com> <51125607.3070801@gmail.com> Message-ID: <51126233.7030706@gmail.com> No, if you go to http://git.freeswitch.org/git/freeswitch/, you can verify it is possible to connect through http. I think there is a problem in the server... Carlo p.s. I have no outside access to git port. Il 06/02/13 14.40, Steven Ayre ha scritto: > It's git:// not http:// > > -Steve > > > On 6 February 2013 13:09, Carlo Dimaggio > wrote: > > I'm triying to download the latest git (as I usually do) but I > have this error... do you know what could be the problem? > > > <14>:git clone http://git.freeswitch.org/freeswitch.git > Initialized empty Git repository in /usr/src/freeswitch/.git/ > error: The requested URL returned error: 500 while accessing > http://git.freeswitch.org/freeswitch.git/info/refs > > fatal: HTTP request failed > > > > > > Il 06/02/13 10.40, Steven Ayre ha scritto: >> Hi Carlo, >> >> It was a bug in FS which has been fixed in the master branch >> (1.3). Try upgrading to verify that fixes your error. At some >> point that'll trickle down into the v1.2.stable branch too. >> >> -Steve >> >> >> >> On 6 February 2013 08:47, Carlo Dimaggio > > wrote: >> >> I have the same behaviour. >> Without prefix it works, while with sqlite:/// the >> error is: >> >> 2013-02-06 09:46:29.169603 [ERR] switch_odbc.c:365 STATE: >> IM002 CODE 0 ERROR: [unixODBC][Driver Manager]Data source >> name not found, and no default driver specified >> >> 2013-02-06 09:46:29.169603 [CRIT] switch_core_sqldb.c:504 >> Failure to connect to PGSQL sqlite! >> 2013-02-06 09:46:29.169603 [ERR] freeswitch_lua.cpp:354 >> Connection failed. DBH NOT Connected. >> >> >> Regards, >> >> Il 05/02/13 17.42, Steven Ayre ha scritto: >>> >>> FAIL => "sqlite:///tmp/test.db" (this one should work) >>> >>> >>> What error does this give? >>> >>> >>> On 5 February 2013 16:31, Fran?ois Delawarde >>> >> > wrote: >>> >>> I'm on HEAD, but I must be doing something wrong, there >>> is no apparent >>> reason from the code. >>> >>> On Tue, 2013-02-05 at 16:05 +0000, Steven Ayre wrote: >>> > Honestly, I only use ODBC... Maybe someone else can >>> shed some light? >>> > >>> > What version are you using? >>> > >>> > Steve >>> > >>> > >>> > >>> > On 5 Feb 2013, at 15:42, Fran?ois Delawarde >>> >> > wrote: >>> > >>> > > Steven, does "sqlite://" prefix work for you? Not >>> sure if it's a bug or >>> > > I'm doing something wrong. I tried the following >>> from lua dbh: >>> > > >>> > > FAIL => "sqlite:///tmp/test.db" (this one should work) >>> > > FAIL => "sqlite://tmp/test.db" (shouldn't work, but >>> just in case...) >>> > > OK => "/tmp/test.db" (no prefix) >>> > > >>> > > Fran?ois. >>> > > >>> > > >>> > > On Tue, 2013-02-05 at 12:55 +0000, Steven Ayre wrote: >>> > >> core: was an old prefix? I knew that change broke >>> some ODBC DSNs >>> > >> (names without username+password, eg when they're >>> specified in >>> > >> odbc.ini) but hadn't realised that had been broken >>> too - that needs >>> > >> documenting as well on the Wiki I guess... >>> > >> http://wiki.freeswitch.org/wiki/Release_Notes#odbc-dsn >>> > >> >>> > >> >>> > >> Just a thought for the developers, but perhaps >>> modules with odbc-dsn >>> > >> parameters should have this renamed to dsn or >>> something now since it's >>> > >> not odbc-specific at all any longer. Easy enough to >>> recognise both >>> > >> 'dsn' and 'odbc-dsn' names to remain backwards >>> compatible (which I've >>> > >> seen done with typos in config setting names in the >>> past >>> > >> as precedent). >>> > >> >>> > >> >>> > >> -Steve >>> > >> >>> > >> >>> > >> >>> > >> >>> > >> On 5 February 2013 11:03, Fran?ois Delawarde >>> > >> >> > wrote: >>> > >> The behavior recently changed from >>> "core:" to >>> > >> "sqlite://". >>> > >> >>> > >> Update: I just tested and it seems that >>> "sqlite://" >>> > >> doesn't work, >>> > >> but as sqlite is the default, you can just >>> omit the prefix and >>> > >> use >>> > >> "" to make it work. >>> > >> >>> > >> Fran?ois. >>> > >> >>> > >> >>> > >> On Tue, 2013-02-05 at 10:37 +0000, Steven >>> Ayre wrote: >>> > >>> 'Failure to connect to PGSQL core' happens on ODBC >>> > >> connections, that >>> > >>> seems to be a minor bug in the error message. >>> > >>> >>> > >>> >>> > >>> From the unixODBC error it's interpreting this as >>> a ODBC >>> > >> DSN. The >>> > >>> reason is the sqlite prefix is sqlite:// not core:// >>> > >>> (see http://wiki.freeswitch.org/wiki/DSN). You >>> also don't >>> > >> have enough >>> > >>> slashes. >>> > >>> >>> > >>> >>> > >>> Try: >>> > >>> local dbh = >>> > >> >>> freeswitch.Dbh("sqlite:///usr/local/temp/database.db"); >>> > >>> >>> > >>> >>> > >>> >>> > >>> (note the 3 slashes - :// is standard URI format, >>> the 3rd >>> > >> indicates an >>> > >>> absolute path). >>> > >>> >>> > >>> >>> > >>> -Steve >>> > >>> >>> > >>> >>> > >>> >>> > >>> >>> > >>> On 5 February 2013 08:56, Carlo Dimaggio >>> > >> >> > >>> > >>> wrote: >>> > >>> Hi all! >>> > >>> >>> > >>> I have a lua script in which i have a >>> statement that >>> > >> connects >>> > >>> to a sqlite db: >>> > >>> >>> > >>> local dbh = >>> > >>> freeswitch.Dbh("core:/usr/local/temp/database.db"); >>> > >>> >>> > >>> but i receve in FS CLI this error that refers to a >>> > >> Pgsql >>> > >>> instead of sqlite: >>> > >>> >>> > >>> 2013-02-05 09:40:10.379612 [ERR] switch_odbc.c:365 >>> > >> STATE: >>> > >>> IM002 CODE 0 ERROR: [unixODBC][Driver Manager]Data >>> > >> source name >>> > >>> not found, and no default driver specified >>> > >>> 2013-02-05 09:40:10.379612 [CRIT] >>> > >> switch_core_sqldb.c:504 >>> > >>> Failure to connect to PGSQL core! >>> > >>> 2013-02-05 09:40:10.379612 [ERR] >>> > >> freeswitch_lua.cpp:354 >>> > >>> Connection failed. DBH NOT Connected. >>> > >>> 2013-02-05 09:40:10.379612 [ERR] >>> > >> freeswitch_lua.cpp:435 DBH >>> > >>> NOT Connected. >>> > >>> >>> > >>> >>> > >>> Can you help me? >>> > >>> >>> > >>> Best regards, >>> > >> >>> _________________________________________________________________________ >>> > >>> Professional FreeSWITCH Consulting Services: >>> > >>> consulting at freeswitch.org >>> >>> > >>> http://www.freeswitchsolutions.com >>> > >>> >>> > >>> FreeSWITCH-powered IP PBX: The CudaTel Communication >>> > >> Server >>> > >>> >>> > >>> >>> > >>> Official FreeSWITCH Sites >>> > >>> http://www.freeswitch.org >>> > >>> http://wiki.freeswitch.org >>> > >>> http://www.cluecon.com >>> > >>> >>> > >>> 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 >>> > >> >>> _________________________________________________________________________ >>> > >>> Professional FreeSWITCH Consulting Services: >>> > >>> consulting at freeswitch.org >>> >>> > >>> http://www.freeswitchsolutions.com >>> > >>> >>> > >>> FreeSWITCH-powered IP PBX: The CudaTel >>> Communication Server >>> > >>> >>> > >>> >>> > >>> Official FreeSWITCH Sites >>> > >>> http://www.freeswitch.org >>> > >>> http://wiki.freeswitch.org >>> > >>> http://www.cluecon.com >>> > >>> >>> > >>> 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 >>> > >> >>> > >> >>> > >> >>> > >> >>> _________________________________________________________________________ >>> > >> Professional FreeSWITCH Consulting Services: >>> > >> consulting at freeswitch.org >>> >>> > >> http://www.freeswitchsolutions.com >>> > >> >>> > >> FreeSWITCH-powered IP PBX: The CudaTel >>> Communication Server >>> > >> >>> > >> >>> > >> Official FreeSWITCH Sites >>> > >> http://www.freeswitch.org >>> > >> http://wiki.freeswitch.org >>> > >> http://www.cluecon.com >>> > >> >>> > >> 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 >>> > >> >>> > >> >>> > >> >>> > >> >>> _________________________________________________________________________ >>> > >> Professional FreeSWITCH Consulting Services: >>> > >> consulting at freeswitch.org >>> >>> > >> http://www.freeswitchsolutions.com >>> > >> >>> > >> FreeSWITCH-powered IP PBX: The CudaTel >>> Communication Server >>> > >> >>> > >> >>> > >> Official FreeSWITCH Sites >>> > >> http://www.freeswitch.org >>> > >> http://wiki.freeswitch.org >>> > >> http://www.cluecon.com >>> > >> >>> > >> 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 >>> > > >>> > > >>> > > >>> > > >>> _________________________________________________________________________ >>> > > Professional FreeSWITCH Consulting Services: >>> > > consulting at freeswitch.org >>> >>> > > http://www.freeswitchsolutions.com >>> > > >>> > > FreeSWITCH-powered IP PBX: The CudaTel Communication >>> Server >>> > > >>> > > >>> > > Official FreeSWITCH Sites >>> > > http://www.freeswitch.org >>> > > http://wiki.freeswitch.org >>> > > http://www.cluecon.com >>> > > >>> > > 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 >>> > >>> > >>> _________________________________________________________________________ >>> > Professional FreeSWITCH Consulting Services: >>> > consulting at freeswitch.org >>> >>> > http://www.freeswitchsolutions.com >>> > >>> > FreeSWITCH-powered IP PBX: The CudaTel Communication >>> Server >>> > >>> > >>> > Official FreeSWITCH Sites >>> > http://www.freeswitch.org >>> > http://wiki.freeswitch.org >>> > http://www.cluecon.com >>> > >>> > 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 >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130206/b3a672b1/attachment-0001.html From steveayre at gmail.com Wed Feb 6 17:28:38 2013 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 6 Feb 2013 14:28:38 +0000 Subject: [Freeswitch-users] lua freeswitch.Dbh sqlite error In-Reply-To: <51126233.7030706@gmail.com> References: <5110C94F.7080901@gmail.com> <1360062212.8347.200.camel@luna.madrid.commsmundi.com> <1360078923.8347.280.camel@luna.madrid.commsmundi.com> <1360081909.22365.35.camel@luna.madrid.commsmundi.com> <511218A4.9010107@gmail.com> <51125607.3070801@gmail.com> <51126233.7030706@gmail.com> Message-ID: That is a browsable interface to the repository that coincidentally runs on the same domain name. The git checkouts are done via the git protocol not http: http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#Recommended:_Git The advice if the git port is blocked by your firewall/proxy is to checkout from the github mirror over https: http://wiki.freeswitch.org/wiki/Git_Tips#Git_https_via_github_and_behind_proxy -Steve On 6 February 2013 14:01, Carlo Dimaggio wrote: > No, if you go to http://git.freeswitch.org/git/freeswitch/, you can > verify it is possible to connect through http. > > I think there is a problem in the server... > > Carlo > > p.s. I have no outside access to git port. > > > Il 06/02/13 14.40, Steven Ayre ha scritto: > > It's git:// not http:// > > -Steve > > > On 6 February 2013 13:09, Carlo Dimaggio wrote: > >> I'm triying to download the latest git (as I usually do) but I have >> this error... do you know what could be the problem? >> >> >> <14>:git clone http://git.freeswitch.org/freeswitch.git >> Initialized empty Git repository in /usr/src/freeswitch/.git/ >> error: The requested URL returned error: 500 while accessing >> http://git.freeswitch.org/freeswitch.git/info/refs >> >> fatal: HTTP request failed >> >> >> >> >> >> Il 06/02/13 10.40, Steven Ayre ha scritto: >> >> Hi Carlo, >> >> It was a bug in FS which has been fixed in the master branch (1.3). Try >> upgrading to verify that fixes your error. At some point that'll trickle >> down into the v1.2.stable branch too. >> >> -Steve >> >> >> >> On 6 February 2013 08:47, Carlo Dimaggio wrote: >> >>> I have the same behaviour. >>> Without prefix it works, while with sqlite:/// the error is: >>> >>> 2013-02-06 09:46:29.169603 [ERR] switch_odbc.c:365 STATE: IM002 CODE 0 >>> ERROR: [unixODBC][Driver Manager]Data source name not found, and no default >>> driver specified >>> >>> 2013-02-06 09:46:29.169603 [CRIT] switch_core_sqldb.c:504 Failure to >>> connect to PGSQL sqlite! >>> 2013-02-06 09:46:29.169603 [ERR] freeswitch_lua.cpp:354 Connection >>> failed. DBH NOT Connected. >>> >>> >>> Regards, >>> >>> Il 05/02/13 17.42, Steven Ayre ha scritto: >>> >>> FAIL => "sqlite:///tmp/test.db" (this one should work) >>> >>> >>> What error does this give? >>> >>> >>> On 5 February 2013 16:31, Fran?ois Delawarde < >>> fdelawarde at wirelessmundi.com> wrote: >>> >>>> I'm on HEAD, but I must be doing something wrong, there is no apparent >>>> reason from the code. >>>> >>>> On Tue, 2013-02-05 at 16:05 +0000, Steven Ayre wrote: >>>> > Honestly, I only use ODBC... Maybe someone else can shed some light? >>>> > >>>> > What version are you using? >>>> > >>>> > Steve >>>> > >>>> > >>>> > >>>> > On 5 Feb 2013, at 15:42, Fran?ois Delawarde < >>>> fdelawarde at wirelessmundi.com> wrote: >>>> > >>>> > > Steven, does "sqlite://" prefix work for you? Not sure if it's a >>>> bug or >>>> > > I'm doing something wrong. I tried the following from lua dbh: >>>> > > >>>> > > FAIL => "sqlite:///tmp/test.db" (this one should work) >>>> > > FAIL => "sqlite://tmp/test.db" (shouldn't work, but just in case...) >>>> > > OK => "/tmp/test.db" (no prefix) >>>> > > >>>> > > Fran?ois. >>>> > > >>>> > > >>>> > > On Tue, 2013-02-05 at 12:55 +0000, Steven Ayre wrote: >>>> > >> core: was an old prefix? I knew that change broke some ODBC DSNs >>>> > >> (names without username+password, eg when they're specified in >>>> > >> odbc.ini) but hadn't realised that had been broken too - that needs >>>> > >> documenting as well on the Wiki I guess... >>>> > >> http://wiki.freeswitch.org/wiki/Release_Notes#odbc-dsn >>>> > >> >>>> > >> >>>> > >> Just a thought for the developers, but perhaps modules with >>>> odbc-dsn >>>> > >> parameters should have this renamed to dsn or something now since >>>> it's >>>> > >> not odbc-specific at all any longer. Easy enough to recognise both >>>> > >> 'dsn' and 'odbc-dsn' names to remain backwards compatible (which >>>> I've >>>> > >> seen done with typos in config setting names in the past >>>> > >> as precedent). >>>> > >> >>>> > >> >>>> > >> -Steve >>>> > >> >>>> > >> >>>> > >> >>>> > >> >>>> > >> On 5 February 2013 11:03, Fran?ois Delawarde >>>> > >> wrote: >>>> > >> The behavior recently changed from "core:" to >>>> > >> "sqlite://". >>>> > >> >>>> > >> Update: I just tested and it seems that "sqlite://" >>>> > >> doesn't work, >>>> > >> but as sqlite is the default, you can just omit the prefix >>>> and >>>> > >> use >>>> > >> "" to make it work. >>>> > >> >>>> > >> Fran?ois. >>>> > >> >>>> > >> >>>> > >> On Tue, 2013-02-05 at 10:37 +0000, Steven Ayre wrote: >>>> > >>> 'Failure to connect to PGSQL core' happens on ODBC >>>> > >> connections, that >>>> > >>> seems to be a minor bug in the error message. >>>> > >>> >>>> > >>> >>>> > >>> From the unixODBC error it's interpreting this as a ODBC >>>> > >> DSN. The >>>> > >>> reason is the sqlite prefix is sqlite:// not core:// >>>> > >>> (see http://wiki.freeswitch.org/wiki/DSN). You also don't >>>> > >> have enough >>>> > >>> slashes. >>>> > >>> >>>> > >>> >>>> > >>> Try: >>>> > >>> local dbh = >>>> > >> freeswitch.Dbh("sqlite:///usr/local/temp/database.db"); >>>> > >>> >>>> > >>> >>>> > >>> >>>> > >>> (note the 3 slashes - :// is standard URI format, the 3rd >>>> > >> indicates an >>>> > >>> absolute path). >>>> > >>> >>>> > >>> >>>> > >>> -Steve >>>> > >>> >>>> > >>> >>>> > >>> >>>> > >>> >>>> > >>> On 5 February 2013 08:56, Carlo Dimaggio >>>> > >> >>>> > >>> wrote: >>>> > >>> Hi all! >>>> > >>> >>>> > >>> I have a lua script in which i have a statement that >>>> > >> connects >>>> > >>> to a sqlite db: >>>> > >>> >>>> > >>> local dbh = >>>> > >>> freeswitch.Dbh("core:/usr/local/temp/database.db"); >>>> > >>> >>>> > >>> but i receve in FS CLI this error that refers to a >>>> > >> Pgsql >>>> > >>> instead of sqlite: >>>> > >>> >>>> > >>> 2013-02-05 09:40:10.379612 [ERR] switch_odbc.c:365 >>>> > >> STATE: >>>> > >>> IM002 CODE 0 ERROR: [unixODBC][Driver Manager]Data >>>> > >> source name >>>> > >>> not found, and no default driver specified >>>> > >>> 2013-02-05 09:40:10.379612 [CRIT] >>>> > >> switch_core_sqldb.c:504 >>>> > >>> Failure to connect to PGSQL core! >>>> > >>> 2013-02-05 09:40:10.379612 [ERR] >>>> > >> freeswitch_lua.cpp:354 >>>> > >>> Connection failed. DBH NOT Connected. >>>> > >>> 2013-02-05 09:40:10.379612 [ERR] >>>> > >> freeswitch_lua.cpp:435 DBH >>>> > >>> NOT Connected. >>>> > >>> >>>> > >>> >>>> > >>> Can you help me? >>>> > >>> >>>> > >>> Best regards, >>>> > >> >>>> _________________________________________________________________________ >>>> > >>> Professional FreeSWITCH Consulting Services: >>>> > >>> consulting at freeswitch.org >>>> > >>> http://www.freeswitchsolutions.com >>>> > >>> >>>> > >>> FreeSWITCH-powered IP PBX: The CudaTel Communication >>>> > >> Server >>>> > >>> >>>> > >>> >>>> > >>> Official FreeSWITCH Sites >>>> > >>> http://www.freeswitch.org >>>> > >>> http://wiki.freeswitch.org >>>> > >>> http://www.cluecon.com >>>> > >>> >>>> > >>> 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 >>>> > >> >>>> _________________________________________________________________________ >>>> > >>> Professional FreeSWITCH Consulting Services: >>>> > >>> consulting at freeswitch.org >>>> > >>> http://www.freeswitchsolutions.com >>>> > >>> >>>> > >>> >>>> > >>> >>>> > >>> >>>> > >>> Official FreeSWITCH Sites >>>> > >>> http://www.freeswitch.org >>>> > >>> http://wiki.freeswitch.org >>>> > >>> http://www.cluecon.com >>>> > >>> >>>> > >>> 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 >>>> > >> >>>> > >> >>>> > >> >>>> > >> >>>> _________________________________________________________________________ >>>> > >> Professional FreeSWITCH Consulting Services: >>>> > >> consulting at freeswitch.org >>>> > >> http://www.freeswitchsolutions.com >>>> > >> >>>> > >> >>>> > >> >>>> > >> >>>> > >> Official FreeSWITCH Sites >>>> > >> http://www.freeswitch.org >>>> > >> http://wiki.freeswitch.org >>>> > >> http://www.cluecon.com >>>> > >> >>>> > >> 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 >>>> > >> >>>> > >> >>>> > >> >>>> > >> >>>> _________________________________________________________________________ >>>> > >> Professional FreeSWITCH Consulting Services: >>>> > >> consulting at freeswitch.org >>>> > >> http://www.freeswitchsolutions.com >>>> > >> >>>> > >> >>>> > >> >>>> > >> >>>> > >> Official FreeSWITCH Sites >>>> > >> http://www.freeswitch.org >>>> > >> http://wiki.freeswitch.org >>>> > >> http://www.cluecon.com >>>> > >> >>>> > >> 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 >>>> > > >>>> > > >>>> > > >>>> > > >>>> _________________________________________________________________________ >>>> > > Professional FreeSWITCH Consulting Services: >>>> > > consulting at freeswitch.org >>>> > > http://www.freeswitchsolutions.com >>>> > > >>>> > > >>>> > > >>>> > > >>>> > > Official FreeSWITCH Sites >>>> > > http://www.freeswitch.org >>>> > > http://wiki.freeswitch.org >>>> > > http://www.cluecon.com >>>> > > >>>> > > 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 >>>> > >>>> > >>>> _________________________________________________________________________ >>>> > Professional FreeSWITCH Consulting Services: >>>> > consulting at freeswitch.org >>>> > http://www.freeswitchsolutions.com >>>> > >>>> > >>>> > >>>> > >>>> > Official FreeSWITCH Sites >>>> > http://www.freeswitch.org >>>> > http://wiki.freeswitch.org >>>> > http://www.cluecon.com >>>> > >>>> > 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 >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >>> >>> >>> >>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com >>> >>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >> >> >> >> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com >> >> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130206/75371fc2/attachment-0001.html From a.venugopan at mundio.com Wed Feb 6 17:29:46 2013 From: a.venugopan at mundio.com (Archana Venugopan) Date: Wed, 6 Feb 2013 14:29:46 +0000 Subject: [Freeswitch-users] call queue In-Reply-To: References: <592A9CF93E12394E8472A6CC66E66BF2357883@Mail-Kilo.squay.com> Message-ID: <592A9CF93E12394E8472A6CC66E66BF2357D7D@Mail-Kilo.squay.com> Hi, I have seen the below entries in callcenter wiki. Sorry to post a simple query here. Incase if I want to ring all phone sequentially according to the below information, I can give sequentially-by-agent-order. Should I need to give this in dialplan xml or in autoload_config? Also can you please provide me how the syntax looks too. Many thanks. Queues Queues can only be configured in the XML configuration. They are only loaded once. Distribution Strategy String Description ring-all Rings all agents simultaneously. longest-idle-agent Rings the agent who has been idle the longest. round-robin Rings the agent in position but remember last tried agent. top-down Rings the agent in order position starting from 1 for every member. agent-with-least-talk-time Rings the agent with least talk time. agent-with-fewest-calls Rings the agent with fewest calls. sequentially-by-agent-order Rings agents sequentially by tier & order. random Rings agents in random order Regards, Archana From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi Marcus Sent: 05 February 2013 12:15 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] call queue Check http://wiki.freeswitch.org/wiki/Mod_fifo or http://wiki.freeswitch.org/wiki/Mod_callcenter mod_fifo is "simpler". mod_callcenter has some very advanced options for call routing that you may not need. -Avi On Tue, Feb 5, 2013 at 12:26 PM, Archana Venugopan > wrote: Hi, Can you please tell me how can I enable call queue in freeswitch? Regards, Archana _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130206/323e7a23/attachment.html From vbvbrj at gmail.com Wed Feb 6 18:02:53 2013 From: vbvbrj at gmail.com (Mimiko) Date: Wed, 06 Feb 2013 17:02:53 +0200 Subject: [Freeswitch-users] call queue In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF2357D7D@Mail-Kilo.squay.com> References: <592A9CF93E12394E8472A6CC66E66BF2357883@Mail-Kilo.squay.com> <592A9CF93E12394E8472A6CC66E66BF2357D7D@Mail-Kilo.squay.com> Message-ID: <5112709D.5060708@gmail.com> On 06.02.2013 16:29, Archana Venugopan wrote: > Hi, > > I have seen the below entries in callcenter wiki. Sorry to post a simple > query here. Incase if I want to ring all phone sequentially according to > the below information, I can give sequentially-by-agent-order. > > Should I need to give this in dialplan xml or in autoload_config? Also > can you please provide me how the syntax looks too. Many thanks. > > *Queues * > > Queues can only be configured in the XML configuration. They are only > loaded once. Yes, callcenter queues are defined in callcenter.conf.xml. Queues parameters like order and distribution are set per queue. It is a global for full FS process work. Althought, agents adding and tiering can be done dinamically from dialplan. To set agents sequentially, you have to chose sorting order. I prefer longest-idle-agent, as this loads all agents equally. In general this is good, as a member does not care which agent will answer. -- Mimiko desu. From jaasmailing at gmail.com Wed Feb 6 18:20:27 2013 From: jaasmailing at gmail.com (Carlo Dimaggio) Date: Wed, 06 Feb 2013 16:20:27 +0100 Subject: [Freeswitch-users] lua freeswitch.Dbh sqlite error In-Reply-To: References: <5110C94F.7080901@gmail.com> <1360062212.8347.200.camel@luna.madrid.commsmundi.com> <1360078923.8347.280.camel@luna.madrid.commsmundi.com> <1360081909.22365.35.camel@luna.madrid.commsmundi.com> <511218A4.9010107@gmail.com> Message-ID: <511274BB.2060907@gmail.com> Hi Steve, now the sqlite:/// works: freeswitch at internal> lua test.lua -ERR no reply 2013-02-06 16:19:02.169603 [DEBUG] freeswitch_lua.cpp:352 DBH handle 0x7f1a7406c240 Connected. 2013-02-06 16:19:02.169603 [DEBUG] freeswitch_lua.cpp:370 DBH handle (nil) released. Should you merge into v1.2.stable branch? I have other production systems in v.1.2 with sqlite. Best regards Il 06/02/13 10.40, Steven Ayre ha scritto: > Hi Carlo, > > It was a bug in FS which has been fixed in the master branch (1.3). > Try upgrading to verify that fixes your error. At some point that'll > trickle down into the v1.2.stable branch too. > > -Steve > > > > On 6 February 2013 08:47, Carlo Dimaggio > wrote: > > I have the same behaviour. > Without prefix it works, while with sqlite:/// the error is: > > 2013-02-06 09:46:29.169603 [ERR] switch_odbc.c:365 STATE: IM002 > CODE 0 ERROR: [unixODBC][Driver Manager]Data source name not > found, and no default driver specified > > 2013-02-06 09:46:29.169603 [CRIT] switch_core_sqldb.c:504 Failure > to connect to PGSQL sqlite! > 2013-02-06 09:46:29.169603 [ERR] freeswitch_lua.cpp:354 Connection > failed. DBH NOT Connected. > > > Regards, > > Il 05/02/13 17.42, Steven Ayre ha scritto: >> >> FAIL => "sqlite:///tmp/test.db" (this one should work) >> >> >> What error does this give? >> >> >> On 5 February 2013 16:31, Fran?ois Delawarde >> > > wrote: >> >> I'm on HEAD, but I must be doing something wrong, there is no >> apparent >> reason from the code. >> >> On Tue, 2013-02-05 at 16:05 +0000, Steven Ayre wrote: >> > Honestly, I only use ODBC... Maybe someone else can shed >> some light? >> > >> > What version are you using? >> > >> > Steve >> > >> > >> > >> > On 5 Feb 2013, at 15:42, Fran?ois Delawarde >> > > wrote: >> > >> > > Steven, does "sqlite://" prefix work for you? Not sure if >> it's a bug or >> > > I'm doing something wrong. I tried the following from lua >> dbh: >> > > >> > > FAIL => "sqlite:///tmp/test.db" (this one should work) >> > > FAIL => "sqlite://tmp/test.db" (shouldn't work, but just >> in case...) >> > > OK => "/tmp/test.db" (no prefix) >> > > >> > > Fran?ois. >> > > >> > > >> > > On Tue, 2013-02-05 at 12:55 +0000, Steven Ayre wrote: >> > >> core: was an old prefix? I knew that change broke some >> ODBC DSNs >> > >> (names without username+password, eg when they're >> specified in >> > >> odbc.ini) but hadn't realised that had been broken too - >> that needs >> > >> documenting as well on the Wiki I guess... >> > >> http://wiki.freeswitch.org/wiki/Release_Notes#odbc-dsn >> > >> >> > >> >> > >> Just a thought for the developers, but perhaps modules >> with odbc-dsn >> > >> parameters should have this renamed to dsn or something >> now since it's >> > >> not odbc-specific at all any longer. Easy enough to >> recognise both >> > >> 'dsn' and 'odbc-dsn' names to remain backwards >> compatible (which I've >> > >> seen done with typos in config setting names in the past >> > >> as precedent). >> > >> >> > >> >> > >> -Steve >> > >> >> > >> >> > >> >> > >> >> > >> On 5 February 2013 11:03, Fran?ois Delawarde >> > >> > > wrote: >> > >> The behavior recently changed from "core:" to >> > >> "sqlite://". >> > >> >> > >> Update: I just tested and it seems that >> "sqlite://" >> > >> doesn't work, >> > >> but as sqlite is the default, you can just omit >> the prefix and >> > >> use >> > >> "" to make it work. >> > >> >> > >> Fran?ois. >> > >> >> > >> >> > >> On Tue, 2013-02-05 at 10:37 +0000, Steven Ayre wrote: >> > >>> 'Failure to connect to PGSQL core' happens on ODBC >> > >> connections, that >> > >>> seems to be a minor bug in the error message. >> > >>> >> > >>> >> > >>> From the unixODBC error it's interpreting this as a ODBC >> > >> DSN. The >> > >>> reason is the sqlite prefix is sqlite:// not core:// >> > >>> (see http://wiki.freeswitch.org/wiki/DSN). You also don't >> > >> have enough >> > >>> slashes. >> > >>> >> > >>> >> > >>> Try: >> > >>> local dbh = >> > >> freeswitch.Dbh("sqlite:///usr/local/temp/database.db"); >> > >>> >> > >>> >> > >>> >> > >>> (note the 3 slashes - :// is standard URI format, the 3rd >> > >> indicates an >> > >>> absolute path). >> > >>> >> > >>> >> > >>> -Steve >> > >>> >> > >>> >> > >>> >> > >>> >> > >>> On 5 February 2013 08:56, Carlo Dimaggio >> > >> > > >> > >>> wrote: >> > >>> Hi all! >> > >>> >> > >>> I have a lua script in which i have a statement that >> > >> connects >> > >>> to a sqlite db: >> > >>> >> > >>> local dbh = >> > >>> freeswitch.Dbh("core:/usr/local/temp/database.db"); >> > >>> >> > >>> but i receve in FS CLI this error that refers to a >> > >> Pgsql >> > >>> instead of sqlite: >> > >>> >> > >>> 2013-02-05 09:40:10.379612 [ERR] switch_odbc.c:365 >> > >> STATE: >> > >>> IM002 CODE 0 ERROR: [unixODBC][Driver Manager]Data >> > >> source name >> > >>> not found, and no default driver specified >> > >>> 2013-02-05 09:40:10.379612 [CRIT] >> > >> switch_core_sqldb.c:504 >> > >>> Failure to connect to PGSQL core! >> > >>> 2013-02-05 09:40:10.379612 [ERR] >> > >> freeswitch_lua.cpp:354 >> > >>> Connection failed. DBH NOT Connected. >> > >>> 2013-02-05 09:40:10.379612 [ERR] >> > >> freeswitch_lua.cpp:435 DBH >> > >>> NOT Connected. >> > >>> >> > >>> >> > >>> Can you help me? >> > >>> >> > >>> Best regards, >> > >> >> _________________________________________________________________________ >> > >>> Professional FreeSWITCH Consulting Services: >> > >>> consulting at freeswitch.org >> >> > >>> http://www.freeswitchsolutions.com >> > >>> >> > >>> FreeSWITCH-powered IP PBX: The CudaTel Communication >> > >> Server >> > >>> >> > >>> >> > >>> Official FreeSWITCH Sites >> > >>> http://www.freeswitch.org >> > >>> http://wiki.freeswitch.org >> > >>> http://www.cluecon.com >> > >>> >> > >>> 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 >> > >> >> _________________________________________________________________________ >> > >>> Professional FreeSWITCH Consulting Services: >> > >>> consulting at freeswitch.org >> >> > >>> http://www.freeswitchsolutions.com >> > >>> >> > >>> >> > >>> >> > >>> >> > >>> Official FreeSWITCH Sites >> > >>> http://www.freeswitch.org >> > >>> http://wiki.freeswitch.org >> > >>> http://www.cluecon.com >> > >>> >> > >>> 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 >> > >> >> > >> >> > >> >> > >> >> _________________________________________________________________________ >> > >> Professional FreeSWITCH Consulting Services: >> > >> consulting at freeswitch.org >> > >> http://www.freeswitchsolutions.com >> > >> >> > >> FreeSWITCH-powered IP PBX: The CudaTel >> Communication Server >> > >> >> > >> >> > >> Official FreeSWITCH Sites >> > >> http://www.freeswitch.org >> > >> http://wiki.freeswitch.org >> > >> http://www.cluecon.com >> > >> >> > >> 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 >> > >> >> > >> >> > >> >> > >> >> _________________________________________________________________________ >> > >> Professional FreeSWITCH Consulting Services: >> > >> consulting at freeswitch.org >> > >> http://www.freeswitchsolutions.com >> > >> >> > >> >> > >> >> > >> >> > >> Official FreeSWITCH Sites >> > >> http://www.freeswitch.org >> > >> http://wiki.freeswitch.org >> > >> http://www.cluecon.com >> > >> >> > >> 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 >> > > >> > > >> > > >> > > >> _________________________________________________________________________ >> > > Professional FreeSWITCH Consulting Services: >> > > consulting at freeswitch.org >> > > http://www.freeswitchsolutions.com >> > > >> > > >> > > >> > > >> > > Official FreeSWITCH Sites >> > > http://www.freeswitch.org >> > > http://wiki.freeswitch.org >> > > http://www.cluecon.com >> > > >> > > 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 >> > >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > 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 >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130206/5650bed9/attachment-0001.html From sertys at gmail.com Wed Feb 6 18:55:57 2013 From: sertys at gmail.com (Daniel Ivanov) Date: Wed, 6 Feb 2013 16:55:57 +0100 Subject: [Freeswitch-users] SRTP disabling In-Reply-To: References: Message-ID: Of course you can. Just set the secure_media var to false and you will be srtp-free in sip. On Feb 5, 2013 6:06 PM, "Levend Sayar" wrote: > ** > Hi all. > > I am using FS as a conference server. Some of my phones are using SRTP , > some of them not. Both type of phone can > join a conference. FS can talk to each peer with SRTP or not depending on > the phone itself. > > My question: > > Is it possible to disable SRTP on FS ? > > I suppose if i can disable SRTP, FS will talk without SRTP with each phone > whether they are using SRTP or not. > > TIA > > > > -- > > _lvnd_ > {^_^} > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130206/205be16c/attachment.html From kris at kriskinc.com Wed Feb 6 19:07:39 2013 From: kris at kriskinc.com (Kristian Kielhofner) Date: Wed, 6 Feb 2013 11:07:39 -0500 Subject: [Freeswitch-users] Somewhat OT: Specific SIP packets (and other traffic) can cause ethernet controller reset Message-ID: While not strictly FreeSWITCH related this issue could certainly affect some of you: http://blog.krisk.org/2013/02/packets-of-death.html -- Kristian Kielhofner From levend.sayar at karel.com.tr Wed Feb 6 19:09:07 2013 From: levend.sayar at karel.com.tr (Levend Sayar) Date: Wed, 6 Feb 2013 16:09:07 +0000 Subject: [Freeswitch-users] SRTP disabling In-Reply-To: References: , Message-ID: Thanks Daniel for the reply. I tried But did not work. Upon your reply I also tried But did not work either. I am doing something wrong ? ?yi ?al??malar. _lvnd_ {^_^} KAREL`UCAP On 6 ?ub 2013, at 18:00, "Daniel Ivanov" > wrote: Of course you can. Just set the secure_media var to false and you will be srtp-free in sip. On Feb 5, 2013 6:06 PM, "Levend Sayar" > wrote: Hi all. I am using FS as a conference server. Some of my phones are using SRTP , some of them not. Both type of phone can join a conference. FS can talk to each peer with SRTP or not depending on the phone itself. My question: Is it possible to disable SRTP on FS ? I suppose if i can disable SRTP, FS will talk without SRTP with each phone whether they are using SRTP or not. TIA -- _lvnd_ {^_^} _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130206/fb64385e/attachment.html From msc at freeswitch.org Wed Feb 6 19:55:47 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 6 Feb 2013 08:55:47 -0800 Subject: [Freeswitch-users] Somewhat OT: Specific SIP packets (and other traffic) can cause ethernet controller reset In-Reply-To: References: Message-ID: Kris, Thanks for sharing. I think I'll put this up on freeswitch.org as well, just to get the word out. -MC On Wed, Feb 6, 2013 at 8:07 AM, Kristian Kielhofner wrote: > While not strictly FreeSWITCH related this issue could certainly > affect some of you: > > http://blog.krisk.org/2013/02/packets-of-death.html > > -- > Kristian Kielhofner > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130206/3f8c8d5c/attachment.html From mario_fs at mgtech.com Wed Feb 6 20:04:41 2013 From: mario_fs at mgtech.com (Mario G) Date: Wed, 6 Feb 2013 09:04:41 -0800 Subject: [Freeswitch-users] Can local_stream MOH have a time limit/max duration? Message-ID: <219BCB63-3A5A-419F-B374-BDD45FDE3FAB@mgtech.com> Could not find anything in the wiki and this may not be possible but... I have early media with instructions, is has voice, ring, voice, music. I want to change the ring to music but I could not find a way to limit the time the music plays. See the second example, I would like to limit the first moh to a maximum time. If not possible, is this a "bounty" enhancement? Thanks for any help. Mario G Currently: Would Like (add something to limit the first moh): -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130206/12bcfd23/attachment-0001.html From mehroz.ashraf85 at gmail.com Wed Feb 6 20:09:47 2013 From: mehroz.ashraf85 at gmail.com (mehroz) Date: Wed, 6 Feb 2013 09:09:47 -0800 (PST) Subject: [Freeswitch-users] Understanding Codecs preferences Message-ID: <1360170587108-7587071.post@n2.nabble.com> Hi, I need to assign seperate codec list for each directory e,g PCMU, h264 for 1000.xml PCMA , h263 for 2000.xml I see, codec preference is somthing that wil help me. But cant get this on head! Can any one help me step wise what do i need to do to acquire this scenario. By the way, ill be using mod_xml_curl to load codecs for each directory. I believe i can convert the scenario to fir with dymanic direcotry parameters with xml_curl once i get the way to do it with static XML files. If i am wrong, please do help. Thanks -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Understanding-Codecs-preferences-tp7587071.html Sent from the freeswitch-users mailing list archive at Nabble.com. From msc at freeswitch.org Wed Feb 6 20:23:12 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 6 Feb 2013 09:23:12 -0800 Subject: [Freeswitch-users] Freeswitch second internal profile In-Reply-To: <1360105444715-7587041.post@n2.nabble.com> References: <1360105444715-7587041.post@n2.nabble.com> Message-ID: You may have an older configuration. Look in the freeswitch source directory under conf/vanilla/directory/default.xml. There's a specific entry that you want to find: Make sure that the conf/directory/default.xml file in your production configuration has the above line in it. Most likely the issue is that the dial-string param in your config file does not have the "*/" after "sofia_contact(" and therefore FreeSWITCH does not know which profile to look in for the registration. (The "*/" tells sofia_contact to look at registrations in all SIP profiles on the system.) Let us know how it goes. -Michael On Tue, Feb 5, 2013 at 3:04 PM, taube wrote: > Hi all, > > I created a second internal profile to connect with the phones to > freeswitch > via two networks. > > The internal profile works. I can call out -in and can contact the > extension > with fs_cli sofia_contact. > > At the second profile the phones register and i can contact them but when i > try to call them, i get the message that the user is not registered, are > the > any hints when i create a second internal profile? > > > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-second-internal-profile-tp7587041.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130206/f628fab9/attachment.html From mahesh.katta at flexydial.com Wed Feb 6 20:59:03 2013 From: mahesh.katta at flexydial.com (mahesh katta) Date: Wed, 6 Feb 2013 23:29:03 +0530 Subject: [Freeswitch-users] Passthrough Calling Message-ID: Hi List, I am new to this freeswitch, Any body know regarding "passthrough". for example i get call from first ip and passthrough to second ip using media proxy. like: IP====>FS Server====>IP I have GATEWAY: xxx.xxx.xxx.xxx Thanking you. Best Regards, Mahesh Katta -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130206/90fc7870/attachment.html From william.suffill at gmail.com Wed Feb 6 21:54:42 2013 From: william.suffill at gmail.com (William Suffill) Date: Wed, 6 Feb 2013 13:54:42 -0500 Subject: [Freeswitch-users] Somewhat OT: Specific SIP packets (and other traffic) can cause ethernet controller reset In-Reply-To: References: Message-ID: +1. Quite interesting read. Appreciate the HEX warning too. =) On Wed, Feb 6, 2013 at 11:55 AM, Michael Collins wrote: > Kris, > > Thanks for sharing. I think I'll put this up on freeswitch.org as well, > just to get the word out. > > -MC > > > On Wed, Feb 6, 2013 at 8:07 AM, Kristian Kielhofner wrote: > >> While not strictly FreeSWITCH related this issue could certainly >> affect some of you: >> >> http://blog.krisk.org/2013/02/packets-of-death.html >> >> -- >> Kristian Kielhofner >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130206/a9167cb1/attachment.html From sdevoy at bizfocused.com Wed Feb 6 22:42:57 2013 From: sdevoy at bizfocused.com (Sean Devoy) Date: Wed, 6 Feb 2013 14:42:57 -0500 Subject: [Freeswitch-users] Somewhat OT: Specific SIP packets (and other traffic) can cause ethernet controller reset In-Reply-To: References: Message-ID: <0f2601ce04a2$29fe7c80$7dfb7580$@bizfocused.com> Kristian, Excellent persistence and analysis. Not many people would have had the patients and analytic skills to track that down let alone create a patch. Well done. Thanks too for documenting it for us. Having said that, Dude you need a girlfriend! Sean -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Kristian Kielhofner Sent: Wednesday, February 06, 2013 11:08 AM To: FreeSWITCH Users Help Subject: [Freeswitch-users] Somewhat OT: Specific SIP packets (and other traffic) can cause ethernet controller reset While not strictly FreeSWITCH related this issue could certainly affect some of you: http://blog.krisk.org/2013/02/packets-of-death.html -- Kristian Kielhofner _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 kris at kriskinc.com Wed Feb 6 23:12:35 2013 From: kris at kriskinc.com (Kristian Kielhofner) Date: Wed, 6 Feb 2013 15:12:35 -0500 Subject: [Freeswitch-users] Somewhat OT: Specific SIP packets (and other traffic) can cause ethernet controller reset In-Reply-To: References: Message-ID: It seems this has garnered quite a bit of interest. I've updated the post with a link to instructions and different versions of the packet. You can use them to find out if you're affected. On Wed, Feb 6, 2013 at 11:07 AM, Kristian Kielhofner wrote: > While not strictly FreeSWITCH related this issue could certainly > affect some of you: > > http://blog.krisk.org/2013/02/packets-of-death.html > > -- > Kristian Kielhofner -- Kristian Kielhofner From william.suffill at gmail.com Wed Feb 6 23:57:45 2013 From: william.suffill at gmail.com (William Suffill) Date: Wed, 6 Feb 2013 15:57:45 -0500 Subject: [Freeswitch-users] FreeSWITCH git repo undergoing maintenance In-Reply-To: References: Message-ID: Just read that today as well but so far from my usage I haven't had any issue with it. I did see a slight clarification on Twitter. *"*Gitolite does much more than Gitlab needs." So I guess theory is replacing it completely with just what they need and can integrate more closely instead of something written before GitLab existed would be better. I don't have anything bad to say about Gitolite from my usage of it even before GitLab. -- W - On Tue, Feb 5, 2013 at 11:48 PM, Gabriel Gunderson wrote: > On Tue, Feb 5, 2013 at 6:11 PM, William Suffill > wrote: > > GitLab is decent for those that care for a similar frontend ala Github > but > > hosted in house. Built on top of Gitolite in RoR. > > Kinda off topic here, but GitLab is dropping Gitolite in v5. They're > rolling their own. They say there are too many issues with it. > > > Gabe > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130206/81fe3a1b/attachment-0001.html From krice at freeswitch.org Wed Feb 6 23:58:39 2013 From: krice at freeswitch.org (Ken Rice) Date: Wed, 06 Feb 2013 14:58:39 -0600 Subject: [Freeswitch-users] Somewhat OT: Specific SIP packets (and other traffic) can cause ethernet controller reset In-Reply-To: Message-ID: Hey Kris, I posted you to reddit.... http://www.reddit.com/r/technology/comments/180h55/packets_of_death_voip_guy _finds_bug_in_intel/ Maybe we can get some traction on that On 2/6/13 2:12 PM, "Kristian Kielhofner" wrote: > It seems this has garnered quite a bit of interest. I've updated the > post with a link to instructions and different versions of the packet. > You can use them to find out if you're affected. > > On Wed, Feb 6, 2013 at 11:07 AM, Kristian Kielhofner > wrote: >> While not strictly FreeSWITCH related this issue could certainly >> affect some of you: >> >> http://blog.krisk.org/2013/02/packets-of-death.html >> >> -- >> Kristian Kielhofner > > -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch From sertys at gmail.com Thu Feb 7 01:42:23 2013 From: sertys at gmail.com (Daniel Ivanov) Date: Wed, 6 Feb 2013 23:42:23 +0100 Subject: [Freeswitch-users] SRTP disabling In-Reply-To: References: Message-ID: Of course you can. Just set the secure_media var to false and you will be srtp-free in sip. On Feb 5, 2013 6:06 PM, "Levend Sayar" wrote: ** Hi all. I am using FS as a conference server. Some of my phones are using SRTP , some of them not. Both type of phone can join a conference. FS can talk to each peer with SRTP or not depending on the phone itself. My question: Is it possible to disable SRTP on FS ? I suppose if i can disable SRTP, FS will talk without SRTP with each phone whether they are using SRTP or not. TIA -- _lvnd_ {^_^} _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130206/58604a33/attachment.html From msc at freeswitch.org Thu Feb 7 02:31:27 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 6 Feb 2013 15:31:27 -0800 Subject: [Freeswitch-users] Passthrough Calling In-Reply-To: References: Message-ID: What is the issue you are having? -MC On Wed, Feb 6, 2013 at 9:59 AM, mahesh katta wrote: > Hi List, > > I am new to this freeswitch, Any body know regarding "passthrough". for > example i get call from first ip and passthrough to second ip using media > proxy. > like: IP====>FS Server====>IP > > I have GATEWAY: xxx.xxx.xxx.xxx > > > Thanking you. > > Best Regards, > > Mahesh Katta > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130206/21e024a4/attachment.html From grcamauer at gmail.com Thu Feb 7 03:30:44 2013 From: grcamauer at gmail.com (Guillermo Ruiz Camauer) Date: Thu, 7 Feb 2013 01:30:44 +0100 Subject: [Freeswitch-users] Somewhat OT: Specific SIP packets (and other traffic) can cause ethernet controller reset In-Reply-To: References: Message-ID: <1737766105479867232@unknownmsgid> This is on Slashdot also Gulermo Sent from my iPhone On 06/02/2013, at 22:03, Ken Rice wrote: > Hey Kris, > > I posted you to reddit.... > > http://www.reddit.com/r/technology/comments/180h55/packets_of_death_voip_guy > _finds_bug_in_intel/ > > Maybe we can get some traction on that > > > On 2/6/13 2:12 PM, "Kristian Kielhofner" wrote: > >> It seems this has garnered quite a bit of interest. I've updated the >> post with a link to instructions and different versions of the packet. >> You can use them to find out if you're affected. >> >> On Wed, Feb 6, 2013 at 11:07 AM, Kristian Kielhofner >> wrote: >>> While not strictly FreeSWITCH related this issue could certainly >>> affect some of you: >>> >>> http://blog.krisk.org/2013/02/packets-of-death.html >>> >>> -- >>> Kristian Kielhofner >> >> > > -- > Ken > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > irc.freenode.net #freeswitch > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 gabe at gundy.org Thu Feb 7 03:43:22 2013 From: gabe at gundy.org (Gabriel Gunderson) Date: Wed, 6 Feb 2013 17:43:22 -0700 Subject: [Freeswitch-users] FreeSWITCH git repo undergoing maintenance In-Reply-To: References: Message-ID: On Wed, Feb 6, 2013 at 1:57 PM, William Suffill wrote: > Just read that today as well but so far from my usage I haven't had any > issue with it. I did see a slight clarification on Twitter. > "Gitolite does much more than Gitlab needs." Thanks for the clarification; that sounds right. Gabe From jaykris at gmail.com Thu Feb 7 03:44:55 2013 From: jaykris at gmail.com (JP) Date: Wed, 6 Feb 2013 16:44:55 -0800 Subject: [Freeswitch-users] mod-curl POST Message-ID: How do I POST some XML data to a URL using mod_curl? I want to call it from a Lua script. I cannot use the usual "key=value" format in the URL for my scenario. Thanks, JP -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130206/2ee0e781/attachment.html From rhuddleston at gmail.com Thu Feb 7 03:48:33 2013 From: rhuddleston at gmail.com (Robert-GMAIL) Date: Wed, 6 Feb 2013 19:48:33 -0500 Subject: [Freeswitch-users] Somewhat OT: Specific SIP packets (and other traffic) can cause ethernet controller reset In-Reply-To: <1737766105479867232@unknownmsgid> References: <1737766105479867232@unknownmsgid> Message-ID: Awesome read.. Shocked about no o/s needed... Hopefully Intel will start a recall. Sent from my iPhone 5 On Feb 6, 2013, at 7:30 PM, Guillermo Ruiz Camauer wrote: > This is on Slashdot also > > Gulermo > > Sent from my iPhone > > On 06/02/2013, at 22:03, Ken Rice wrote: > >> Hey Kris, >> >> I posted you to reddit.... >> >> http://www.reddit.com/r/technology/comments/180h55/packets_of_death_voip_guy >> _finds_bug_in_intel/ >> >> Maybe we can get some traction on that >> >> >> On 2/6/13 2:12 PM, "Kristian Kielhofner" wrote: >> >>> It seems this has garnered quite a bit of interest. I've updated the >>> post with a link to instructions and different versions of the packet. >>> You can use them to find out if you're affected. >>> >>> On Wed, Feb 6, 2013 at 11:07 AM, Kristian Kielhofner >>> wrote: >>>> While not strictly FreeSWITCH related this issue could certainly >>>> affect some of you: >>>> >>>> http://blog.krisk.org/2013/02/packets-of-death.html >>>> >>>> -- >>>> Kristian Kielhofner >> >> -- >> Ken >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> irc.freenode.net #freeswitch >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 cal.leeming at simplicitymedialtd.co.uk Thu Feb 7 03:56:28 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Thu, 7 Feb 2013 00:56:28 +0000 Subject: [Freeswitch-users] mod-curl POST In-Reply-To: References: Message-ID: The docs says this; curl url [headers|json] [get|head|post [url_encode_data]] So, I assume you'd just do. curl http://example.com/ post xml=encodedxmlhere Taken from here; http://wiki.freeswitch.org/wiki/Mod_curl Hope this helps, apologies if I misunderstood the question Cal On Thu, Feb 7, 2013 at 12:44 AM, JP wrote: > How do I POST some XML data to a URL using mod_curl? I want to call it > from a Lua script. I cannot use the usual "key=value" format in the URL for > my scenario. > > Thanks, > JP > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130207/7445b231/attachment-0001.html From schoch+freeswitch.org at xwin32.com Thu Feb 7 04:58:02 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Wed, 6 Feb 2013 17:58:02 -0800 Subject: [Freeswitch-users] FAX problems In-Reply-To: References: <20130119191943.ca09ab6d@mail.tritonwest.net> <5102AF93.2000201@coppice.org> Message-ID: On Fri, Feb 1, 2013 at 4:53 AM, shouldbe q931 wrote: > As a replacement for the faxback plugin. > > how about using hylafax+ with t38modem ? http://hylafax.sourceforge.net/ > > or the commercial "fork" from iFax ? > http://www.ifax.com/products/hylafax/fax-over-ip.html > I am happy to report that I have (partial) success! I used HylaFSP (commercial HylaFAX Windows client) to send a FAX to HylaFAX running on the FS machine, connecting to /dev/FS0. The FAX wasn't received 100%, because it didn't go through the T.38 gateway, and the Caller-ID didn't get set, so that's what I'd like to fix next. I already have this rule in the outgoing dialplan: I set the $fax_call variable (and the $outbound_caller_id_number variable) in the directory entry that's used by the ATA that the analog FAX machine uses. What I want to do now is set the same variables on channels opened by the /dev/FS0 modem. Where should I put these variables? -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130206/d1153e66/attachment.html From mike at jerris.com Thu Feb 7 05:43:06 2013 From: mike at jerris.com (Michael Jerris) Date: Wed, 6 Feb 2013 20:43:06 -0600 Subject: [Freeswitch-users] Somewhat OT: Specific SIP packets (and other traffic) can cause ethernet controller reset In-Reply-To: <1737766105479867232@unknownmsgid> References: <1737766105479867232@unknownmsgid> Message-ID: Yep, promoted on /., reddit, Hacker News, etc. Kris you broke everything! On Feb 6, 2013, at 6:30 PM, Guillermo Ruiz Camauer wrote: > This is on Slashdot also > > Gulermo > > Sent from my iPhone > > On 06/02/2013, at 22:03, Ken Rice wrote: > >> Hey Kris, >> >> I posted you to reddit.... >> >> http://www.reddit.com/r/technology/comments/180h55/packets_of_death_voip_guy >> _finds_bug_in_intel/ >> >> Maybe we can get some traction on that >> >> >> On 2/6/13 2:12 PM, "Kristian Kielhofner" wrote: >> >>> It seems this has garnered quite a bit of interest. I've updated the >>> post with a link to instructions and different versions of the packet. >>> You can use them to find out if you're affected. >>> >>> On Wed, Feb 6, 2013 at 11:07 AM, Kristian Kielhofner >>> wrote: >>>> While not strictly FreeSWITCH related this issue could certainly >>>> affect some of you: >>>> >>>> http://blog.krisk.org/2013/02/packets-of-death.html >>>> >>>> -- >>>> Kristian Kielhofner >>> >>> From jaykris at gmail.com Thu Feb 7 06:19:23 2013 From: jaykris at gmail.com (JP) Date: Wed, 6 Feb 2013 19:19:23 -0800 Subject: [Freeswitch-users] mod-curl POST In-Reply-To: References: Message-ID: Thanks for the response. I want the XML data to go as the body of the POST and not as part of the URL. -JP On Wed, Feb 6, 2013 at 4:56 PM, Cal Leeming [Simplicity Media Ltd] < cal.leeming at simplicitymedialtd.co.uk> wrote: > The docs says this; > > curl url [headers|json] [get|head|post [url_encode_data]] > > So, I assume you'd just do. > > curl http://example.com/ post xml=encodedxmlhere > > Taken from here; > http://wiki.freeswitch.org/wiki/Mod_curl > > Hope this helps, apologies if I misunderstood the question > > Cal > > > > On Thu, Feb 7, 2013 at 12:44 AM, JP wrote: > >> How do I POST some XML data to a URL using mod_curl? I want to call it >> from a Lua script. I cannot use the usual "key=value" format in the URL for >> my scenario. >> >> Thanks, >> JP >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130206/77739304/attachment.html From rupa at rupa.com Thu Feb 7 06:21:42 2013 From: rupa at rupa.com (Rupa Schomaker) Date: Wed, 6 Feb 2013 21:21:42 -0600 Subject: [Freeswitch-users] mod-curl POST In-Reply-To: References: Message-ID: On Wed, Feb 6, 2013 at 6:56 PM, Cal Leeming [Simplicity Media Ltd] < cal.leeming at simplicitymedialtd.co.uk> wrote: > The docs says this; > > curl url [headers|json] [get|head|post [url_encode_data]] > The [headers|json] is all about the response format. Just to make sure that is clear. get/head/post should be self-evident url_encode_data is supposed to be the url encoded version of what you want to send to the server. It is encoded on purpose (so that we don't have to worry about embedded spaces or other special chars). So, you really just need to do: curl http://example.com/ post encodedxmlhere > So, I assume you'd just do. > > curl http://example.com/ post xml=encodedxmlhere > > This will result in www-url-encoded with key/value pair with xml as key and the encoded xml as the value. This may be what the OP wants, but I think what he wants is just XML. Note that the content-type will still be application/x-www-form-urlencoded which is "wrong". I've written xml service endpoints that behave differently based on the content-type. Depending on how your service is written this may be ok or not. There is an open ticket to address the content-type issue. http://jira.freeswitch.org/browse/FS-4714 > Taken from here; > http://wiki.freeswitch.org/wiki/Mod_curl > > Hope this helps, apologies if I misunderstood the question > > Cal > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130206/5ec94434/attachment.html From gvvsubhashkumar at gmail.com Thu Feb 7 08:33:40 2013 From: gvvsubhashkumar at gmail.com (Subhash) Date: Thu, 7 Feb 2013 11:03:40 +0530 Subject: [Freeswitch-users] Log Rollover issue In-Reply-To: References: Message-ID: Any Update On this? Thanks, Subhash. On Mon, Jan 7, 2013 at 6:31 PM, Subhash wrote: > I used the latest binaries(Windows Installer),version info is given below > and observered that when it is not able to rename the log file it is keep > on writing the logs in freeswitch.log file which is crossing the limt set > in logconf.xml file. > > The error seen in the freeswitch log file is > > [CRIT] mod_logfile.c:164 Error renaming log from C:/Program > Files/FreeSWITCH/log/freeswitch.log.4 to C:/Program > Files/FreeSWITCH/log/freeswitch.log.5 [No error] > > And the version info is > > Freswitch version 1.3.10 git b1b89e7 2012-12-14 22:53:48z > > > > Thanks, > Subhash. > > > On Thu, Nov 22, 2012 at 12:34 AM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> Are you deleting the old ones? >> It was probably already out of disk space when it was refusing to rename >> the files. >> >> Looks like you have 40 of them, you may need to auto delete some. >> >> You are not providing much other data to go on so all I can do is guess. >> The system call to move the file is failing which could be due to >> permissions or disk full or whatever. >> >> If you want you can update to latest and I added a call to >> strerror(errno) so you can see the exact reason. >> >> >> >> >> On Tue, Nov 20, 2012 at 10:23 PM, Nageshwara Rao Moova < >> mnrao2001 at gmail.com> wrote: >> >>> Hi Anthony, >>> >>> I doubt any permission issue as the failure happened not initially but >>> after a while. The user we are using has sufficient permissions for doing >>> rwx operations. Below is the log snippet from freeswitch log. >>> >>> If it?s any help, the freeswitch log had entries like this every few >>> seconds: >>> >>> 2012-10-26 11:53:16.653473 [CRIT] mod_logfile.c:164 Error renaming log >>> from /usr/local/freeswitch/log/freeswitch.log.40 to >>> /usr/local/freeswitch/log/freeswitch.log.41 >>> >>> ?? >>> >>> 2012-10-27 00:19:30.313497 [CRIT] mod_logfile.c:164 Error renaming log >>> from /usr/local/freeswitch/log/freeswitch.log.40 to >>> /usr/local/freeswitch/log/freeswitch.log.41 >>> >>> >>> Until it ran out of disk space and crashed. No reason why it couldn?t >>> rename the file, though. >>> >>> >>> >>> >>> On Tue, Nov 20, 2012 at 1:50 AM, Anthony Minessale < >>> anthony.minessale at gmail.com> wrote: >>> >>>> When it says it cannot, is that because of file permissions? Dos FS >>>> have permission to write to the directory? >>>> Can you reproduce this with logs and if so can you report it to Jira >>>> http://jira.freeswitch.org >>>> >>>> >>>> >>>> On Mon, Nov 19, 2012 at 6:45 AM, Nageshwara Rao Moova < >>>> mnrao2001 at gmail.com> wrote: >>>> >>>>> Hi all, >>>>> >>>>> I have modified my default logconf file for rollover to be restricted >>>>> to 100. But sometimes freeswitch is unable to rename rollover say ?cannot >>>>> rename freeswitch.log.41? and fails. But the fail follows with serious >>>>> issue of writing all the logs to "freeswitch.log" and ends up filling the >>>>> disk space. >>>>> >>>>> I have not change the default file size i.e 10MB. >>>>> >>>>> How does 10MB size overridden by freeswitch? >>>>> >>>>> -- >>>>> regards & thanks >>>>> -- >>>>> m nageshwara rao >>>>> 99891 86280 >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> 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 >>>>> >>>>> >>>> >>>> >>>> -- >>>> Anthony Minessale II >>>> >>>> FreeSWITCH http://www.freeswitch.org/ >>>> ClueCon http://www.cluecon.com/ >>>> Twitter: http://twitter.com/FreeSWITCH_wire >>>> >>>> AIM: anthm >>>> MSN:anthony_minessale at hotmail.com >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>> IRC: irc.freenode.net #freeswitch >>>> >>>> FreeSWITCH Developer Conference >>>> sip:888 at conference.freeswitch.org >>>> googletalk:conf+888 at conference.freeswitch.org >>>> pstn:+19193869900 >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>>> >>>> >>> >>> >>> -- >>> regards & thanks >>> -- >>> m nageshwara rao >>> 99891 86280 >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130207/e5a13556/attachment-0001.html From krice at freeswitch.org Thu Feb 7 08:41:21 2013 From: krice at freeswitch.org (Ken Rice) Date: Wed, 06 Feb 2013 23:41:21 -0600 Subject: [Freeswitch-users] Log Rollover issue In-Reply-To: Message-ID: Is there a ticket in jira opened on this issue? On 2/6/13 11:33 PM, "Subhash" wrote: > Any Update On this? > > Thanks, > Subhash. > > On Mon, Jan 7, 2013 at 6:31 PM, Subhash wrote: >> I used the latest binaries(Windows Installer),version info is given below and >> observered that when it is not able to rename the log file?it is keep on >> writing the logs in freeswitch.log file which is crossing the limt set in >> logconf.xml file. >> ? >> The error seen in the freeswitch log file is >> ? >> [CRIT] mod_logfile.c:164 Error renaming log from C:/Program >> Files/FreeSWITCH/log/freeswitch.log.4 to C:/Program >> Files/FreeSWITCH/log/freeswitch.log.5 [No error] >> ? >> And the version info is >> ? >> Freswitch version 1.3.10 git b1b89e7 2012-12-14 22:53:48z >> >> >> >> Thanks, >> Subhash. >> >> >> On Thu, Nov 22, 2012 at 12:34 AM, Anthony Minessale >> wrote: >>> Are you deleting the old ones? >>> It was probably already out of disk space when it was refusing to rename the >>> files. >>> >>> Looks like you have 40 of them, you may need to auto delete some. >>> >>> You are not providing much other data to go on so all I can do is guess. >>> The system call to move the file is failing which could be due to >>> permissions or disk full or whatever. >>> >>> If you want you can update to latest and I added a call to strerror(errno) >>> so you can see the exact reason. >>> >>> >>> >>> >>> On Tue, Nov 20, 2012 at 10:23 PM, Nageshwara Rao Moova >>> wrote: >>>> Hi Anthony, >>>> >>>> I doubt any permission issue as the failure happened not initially but >>>> after a while. The user we are using has sufficient permissions for doing >>>> rwx operations. Below is the log snippet from freeswitch log. >>>> >>>> >>>> If it?s any help, the freeswitch log had entries like this every few >>>> seconds: >>>> >>>> 2012-10-26 11:53:16.653473 [CRIT] mod_logfile.c:164 Error renaming log from >>>> /usr/local/freeswitch/log/freeswitch.log.40 to >>>> /usr/local/freeswitch/log/freeswitch.log.41 >>>> >>>> ?? >>>> >>>> 2012-10-27 00:19:30.313497 [CRIT] mod_logfile.c:164 Error renaming log from >>>> /usr/local/freeswitch/log/freeswitch.log.40 to >>>> /usr/local/freeswitch/log/freeswitch.log.41 >>>> >>>> >>>> >>>> Until it ran out of disk space and crashed. No reason why it couldn?t >>>> rename the file, though. >>>> >>>> >>>> >>>> >>>> >>>> On Tue, Nov 20, 2012 at 1:50 AM, Anthony Minessale >>>> wrote: >>>>> When it says it cannot, is that because of file permissions? Dos FS have >>>>> permission to write to the directory? >>>>> Can you reproduce this with logs and if so can you report it to Jira >>>>> http://jira.freeswitch.org >>>>> >>>>> >>>>> >>>>> On Mon, Nov 19, 2012 at 6:45 AM, Nageshwara Rao Moova >>>>> wrote: >>>>>> Hi all, >>>>>> >>>>>> I have modified my default logconf file for rollover to be restricted to >>>>>> 100. But sometimes freeswitch is unable to rename rollover say ?cannot >>>>>> rename freeswitch.log.41? and fails. But the fail follows with serious >>>>>> issue of writing all the logs to "freeswitch.log" and ends up filling the >>>>>> disk space. >>>>>> >>>>>> I have not change the default file size i.e 10MB. >>>>>> >>>>>> How does 10MB size overridden by freeswitch? -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130206/9fd4b649/attachment.html From gavin.henry at gmail.com Thu Feb 7 11:26:31 2013 From: gavin.henry at gmail.com (Gavin Henry) Date: Thu, 7 Feb 2013 08:26:31 +0000 Subject: [Freeswitch-users] Homer/sipcapture Message-ID: Hi all, What's the performance of this like? We have OpenSIPS and FreeSWITCH boxes and not sure what size we need for the homer storage box/s Any experience to share? Thanks. -- Gavin Henry Sent with Sparrow (http://www.sparrowmailapp.com/?sig) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130207/449efc7a/attachment.html From steveayre at gmail.com Thu Feb 7 12:09:05 2013 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 7 Feb 2013 09:09:05 +0000 Subject: [Freeswitch-users] SRTP disabling In-Reply-To: References: Message-ID: It's also going to rely on the phone actually offering RTP/AVP as well as SRTP/AVP in their SDP - without that there'd be nowhere to send insecure RTP. -Steve On 6 February 2013 16:09, Levend Sayar wrote: > Thanks Daniel for the reply. > > I tried > > > > But did not work. Upon your reply I also tried > > > > But did not work either. I am doing something wrong ? > > ?yi ?al??malar. > > _lvnd_ > {^_^} > > KAREL`UCAP > > > On 6 ?ub 2013, at 18:00, "Daniel Ivanov" wrote: > > Of course you can. Just set the secure_media var to false and you will > be srtp-free in sip. > On Feb 5, 2013 6:06 PM, "Levend Sayar" wrote: > >> ** >> Hi all. >> >> I am using FS as a conference server. Some of my phones are using SRTP , >> some of them not. Both type of phone can >> join a conference. FS can talk to each peer with SRTP or not depending on >> the phone itself. >> >> My question: >> >> Is it possible to disable SRTP on FS ? >> >> I suppose if i can disable SRTP, FS will talk without SRTP with each >> phone whether they are using SRTP or not. >> >> TIA >> >> >> -- >> >> _lvnd_ >> {^_^} >> >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130207/bd012710/attachment.html From william at xofap.com Thu Feb 7 13:37:06 2013 From: william at xofap.com (William Alianto) Date: Thu, 07 Feb 2013 17:37:06 +0700 Subject: [Freeswitch-users] Force G729 codec for E1 connection Message-ID: <511383D2.9090806@xofap.com> Hi, I just finished a freeswitch server that act as E1 gateway for PBX system. I noticed that all the call from the gateway use PCMU codec before transcoded to G729 in my main SIP server. My current situation is as following : [PBX] --- E1 ---> [Gateway] --- SIP/PCMU ---> [SIP Server] --- G729 ---> [ITSP] What I intended to do is as following : [PBX] --- E1 ---> [Gateway] --- SIP/G729 ---> [SIP Server] --- G729 ---> [ITSP] Is there anyway to send the media from E1 gateway as G729 instead of PCMU? I need to preserve bandwidth for future expansion. Regards, William From levend.sayar at karel.com.tr Thu Feb 7 14:01:35 2013 From: levend.sayar at karel.com.tr (Levend Sayar) Date: Thu, 7 Feb 2013 11:01:35 +0000 Subject: [Freeswitch-users] SRTP disabling In-Reply-To: References: , Message-ID: Thanx Steven. Caller makes the offer for SDP but callee chooses whatever it wants. So caller can offer SRTP but callee can prefer not to talk encrypted. In our case I want FS to choose non secure media. Phone will offer SRTP on the conference call but FS must prefer RTP, not SRTP. ?yi ?al??malar. _lvnd_ {^_^} KAREL`UCAP On 7 ?ub 2013, at 11:13, "Steven Ayre" > wrote: It's also going to rely on the phone actually offering RTP/AVP as well as SRTP/AVP in their SDP - without that there'd be nowhere to send insecure RTP. -Steve On 6 February 2013 16:09, Levend Sayar > wrote: Thanks Daniel for the reply. I tried But did not work. Upon your reply I also tried But did not work either. I am doing something wrong ? ?yi ?al??malar. _lvnd_ {^_^} KAREL`UCAP On 6 ?ub 2013, at 18:00, "Daniel Ivanov" > wrote: Of course you can. Just set the secure_media var to false and you will be srtp-free in sip. On Feb 5, 2013 6:06 PM, "Levend Sayar" > wrote: Hi all. I am using FS as a conference server. Some of my phones are using SRTP , some of them not. Both type of phone can join a conference. FS can talk to each peer with SRTP or not depending on the phone itself. My question: Is it possible to disable SRTP on FS ? I suppose if i can disable SRTP, FS will talk without SRTP with each phone whether they are using SRTP or not. TIA -- _lvnd_ {^_^} _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130207/073c2614/attachment.html From david.villasmil.work at gmail.com Thu Feb 7 14:34:20 2013 From: david.villasmil.work at gmail.com (David Villasmil) Date: Thu, 7 Feb 2013 12:34:20 +0100 Subject: [Freeswitch-users] Force G729 codec for E1 connection In-Reply-To: <511383D2.9090806@xofap.com> References: <511383D2.9090806@xofap.com> Message-ID: Hello William, E1 doesn't work with G.729, it is intrinsically PCM. Unless of course it is non-structured and it is used as an IP link... David On Thu, Feb 7, 2013 at 11:37 AM, William Alianto wrote: > Hi, > > I just finished a freeswitch server that act as E1 gateway for PBX > system. I noticed that all the call from the gateway use PCMU codec > before transcoded to G729 in my main SIP server. > > My current situation is as following : > > [PBX] --- E1 ---> [Gateway] --- SIP/PCMU ---> [SIP Server] --- G729 > ---> [ITSP] > > What I intended to do is as following : > > [PBX] --- E1 ---> [Gateway] --- SIP/G729 ---> [SIP Server] --- G729 > ---> [ITSP] > > Is there anyway to send the media from E1 gateway as G729 instead of > PCMU? I need to preserve bandwidth for future expansion. > > Regards, > > William > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130207/d07e87a4/attachment.html From ben122uk at gmail.com Thu Feb 7 14:44:03 2013 From: ben122uk at gmail.com (Ben) Date: Thu, 7 Feb 2013 11:44:03 -0000 Subject: [Freeswitch-users] Use local IPv4 for specific LAN IP address range Message-ID: <01b001ce0528$6e4eece0$4aecc6a0$@gmail.com> Greetings all, A question on NATing in Freeswitch - I have my internal SIP profile to use a static public IP for the ext-sip-ip and ext-rtp-ip. The Freeswitch server is on a local LAN IP, for example 192.168.0.10. External SIP clients can access the server via the external IP, as there's a static NAT on the router to map the public IP to the private IP. I have another SIP client on a different LAN, but behind the same firewall, for example, 10.0.0.10. This client is allowed to talk to the server on its private ip of 192.168.0.10, by allowing inter-vlan traffic to pass through. My challenge is, how do I get Freeswitch to recognise this internal SIP client as being allowed to use the internal IP for SIP/RTP communications, and therefore present the client with internal addresses in the SIP header for SIP/RTP? Due to statically setting ext-sip-ip and ext-rtp-ip, Freeswitch tells the client to go via the public IP. The firewall has a policy to not accept traffic to the public IP if sourced from an internal device. I have explored the NATing options in Freeswitch, but can't find anything that fits my scenario. It seems to be more towards how to enable NAT in certain situations, not disable it. I have also looked at re-writing the SDP, but this re-writes the incoming SDP from the clients. My intention was to re-write the outgoing SDP from the Freeswitch to include the internal IP address as the RTP connect in address. I don't have much Asterisk experience, but I seem to remember a nat.conf file where you could specify subnets that were treated differently. These could be classed as internal subnets, and would be given the private IP in the SDP from asterisk, everything got the public IP. I'm also aware that I could create another SIP profile easily with different NATing/IP options, but I want to find out if there's anything I can do with the above first. Many thanks for the help! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130207/a410cecf/attachment-0001.html From steveayre at gmail.com Thu Feb 7 15:10:05 2013 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 7 Feb 2013 12:10:05 +0000 Subject: [Freeswitch-users] Force G729 codec for E1 connection In-Reply-To: <511383D2.9090806@xofap.com> References: <511383D2.9090806@xofap.com> Message-ID: E1 uses G711... I'm assuming by saying that FS is acting as the gateway you're using something like mod_freetdm? If so FS can transcode so that it uses G729 for the SIP side of the bridge. Look at the commercial mod_com_g729 (licenses available for $10/channel on www.freeswitch.org). http://wiki.freeswitch.org/wiki/Mod_com_g729 You just need to have that loaded instead of mod_g729, and tweak your codec preferences accordingly. http://wiki.freeswitch.org/wiki/Codec_Negotiation -Steve On 7 February 2013 10:37, William Alianto wrote: > Hi, > > I just finished a freeswitch server that act as E1 gateway for PBX > system. I noticed that all the call from the gateway use PCMU codec > before transcoded to G729 in my main SIP server. > > My current situation is as following : > > [PBX] --- E1 ---> [Gateway] --- SIP/PCMU ---> [SIP Server] --- G729 > ---> [ITSP] > > What I intended to do is as following : > > [PBX] --- E1 ---> [Gateway] --- SIP/G729 ---> [SIP Server] --- G729 > ---> [ITSP] > > Is there anyway to send the media from E1 gateway as G729 instead of > PCMU? I need to preserve bandwidth for future expansion. > > Regards, > > William > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130207/6818dee4/attachment.html From steveayre at gmail.com Thu Feb 7 16:07:29 2013 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 7 Feb 2013 13:07:29 +0000 Subject: [Freeswitch-users] Use local IPv4 for specific LAN IP address range In-Reply-To: <01b001ce0528$6e4eece0$4aecc6a0$@gmail.com> References: <01b001ce0528$6e4eece0$4aecc6a0$@gmail.com> Message-ID: One simple option would be to have a 2nd profile listening on another port (eg 5080) that does not use ext-*-ip and connect to that internally. -Steve On 7 February 2013 11:44, Ben wrote: > Greetings all,**** > > ** ** > > A question on NATing in Freeswitch ? I have my internal SIP profile to use > a static public IP for the ext-sip-ip and ext-rtp-ip. The Freeswitch > server is on a local LAN IP, for example 192.168.0.10. External SIP > clients can access the server via the external IP, as there?s a static NAT > on the router to map the public IP to the private IP.**** > > ** ** > > I have another SIP client on a different LAN, but behind the same > firewall, for example, 10.0.0.10. This client is allowed to talk to the > server on its private ip of 192.168.0.10, by allowing inter-vlan traffic to > pass through. My challenge is, how do I get Freeswitch to recognise this > internal SIP client as being allowed to use the internal IP for SIP/RTP > communications, and therefore present the client with internal addresses in > the SIP header for SIP/RTP? Due to statically setting ext-sip-ip and > ext-rtp-ip, Freeswitch tells the client to go via the public IP. The > firewall has a policy to not accept traffic to the public IP if sourced > from an internal device.**** > > ** ** > > I have explored the NATing options in Freeswitch, but can?t find anything > that fits my scenario. It seems to be more towards how to enable NAT in > certain situations, not disable it. I have also looked at re-writing the > SDP, but this re-writes the incoming SDP from the clients. My intention > was to re-write the outgoing SDP from the Freeswitch to include the > internal IP address as the RTP connect in address.**** > > ** ** > > I don?t have much Asterisk experience, but I seem to remember a nat.conf > file where you could specify subnets that were treated differently. These > could be classed as internal subnets, and would be given the private IP in > the SDP from asterisk, everything got the public IP.**** > > ** ** > > I?m also aware that I could create another SIP profile easily with > different NATing/IP options, but I want to find out if there?s anything I > can do with the above first.**** > > ** ** > > Many thanks for the help!**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130207/4acec0af/attachment.html From steveayre at gmail.com Thu Feb 7 16:09:47 2013 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 7 Feb 2013 13:09:47 +0000 Subject: [Freeswitch-users] SRTP disabling In-Reply-To: References: Message-ID: What I mean is you'll see two separate m=audio lines within the callee's SDP, one for 'RTP/AVP' and one for 'SRTP/AVP'. If there is no m=audio line for RTP/AVP the caller won't know of a port that's expecting RTP. So if the callee only sends SRTP/AVP the caller can't send RTP. Can you show us the SDP being sent by the phone? -Steve On 7 February 2013 11:01, Levend Sayar wrote: > Thanx Steven. > > Caller makes the offer for SDP but callee chooses whatever it wants. So > caller can offer SRTP but callee can prefer not to talk encrypted. In our > case I want FS to choose non secure media. > Phone will offer SRTP on the conference call but FS must prefer RTP, not > SRTP. > > > ?yi ?al??malar. > > _lvnd_ > {^_^} > > KAREL`UCAP > > > On 7 ?ub 2013, at 11:13, "Steven Ayre" wrote: > > It's also going to rely on the phone actually offering RTP/AVP as well > as SRTP/AVP in their SDP - without that there'd be nowhere to send insecure > RTP. > > -Steve > > > > On 6 February 2013 16:09, Levend Sayar wrote: > >> Thanks Daniel for the reply. >> >> I tried >> >> >> >> But did not work. Upon your reply I also tried >> >> >> >> But did not work either. I am doing something wrong ? >> >> ?yi ?al??malar. >> >> _lvnd_ >> {^_^} >> >> KAREL`UCAP >> >> >> On 6 ?ub 2013, at 18:00, "Daniel Ivanov" wrote: >> >> Of course you can. Just set the secure_media var to false and you will >> be srtp-free in sip. >> On Feb 5, 2013 6:06 PM, "Levend Sayar" wrote: >> >>> ** >>> Hi all. >>> >>> I am using FS as a conference server. Some of my phones are using SRTP , >>> some of them not. Both type of phone can >>> join a conference. FS can talk to each peer with SRTP or not depending >>> on the phone itself. >>> >>> My question: >>> >>> Is it possible to disable SRTP on FS ? >>> >>> I suppose if i can disable SRTP, FS will talk without SRTP with each >>> phone whether they are using SRTP or not. >>> >>> TIA >>> >>> >>> -- >>> >>> _lvnd_ >>> {^_^} >>> >>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130207/8b4e2976/attachment-0001.html From levend.sayar at karel.com.tr Thu Feb 7 16:26:01 2013 From: levend.sayar at karel.com.tr (Levend Sayar) Date: Thu, 7 Feb 2013 13:26:01 +0000 Subject: [Freeswitch-users] SRTP disabling In-Reply-To: References: Message-ID: Below is the SDP offer sent by the phone. v=0 o=- 20185 20185 IN IP4 192.168.173.69 s=SDP data c=IN IP4 192.168.173.69 t=0 0 m=audio 11780 RTP/SAVP 0 8 18 9 101 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:MTM2MjVhMGI1NDZjYmRjADU5NWVjNGVkNTNlYzA1 a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:YmExYmZhNQAzN2ZjNDgzYTRkNGU2ZjFiN2Q0MmE3 a=crypto:3 F8_128_HMAC_SHA1_80 inline:N2Q2NTRiYQAxZjA3MWY3ZjI1YTI5NjIyM2FjODYw a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:9 G722/8000 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=ptime:20 a=sendrecv And below is the SDP answer sent by FS v=0 o=FreeSWITCH 1360230601 1360230602 IN IP4 192.168.169.114 s=FreeSWITCH c=IN IP4 192.168.169.114 t=0 0 m=audio 12532 RTP/SAVP 9 101 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:l8v0R64H7CP0vEx9j0Ycdbob8bgMCpLDppWGy7Dy -- ?yi ?al??malar. _lvnd_ {^_^} KAREL`UCAP On Thu, 2013-02-07 at 13:09 +0000, Steven Ayre wrote: What I mean is you'll see two separate m=audio lines within the callee's SDP, one for 'RTP/AVP' and one for 'SRTP/AVP'. If there is no m=audio line for RTP/AVP the caller won't know of a port that's expecting RTP. So if the callee only sends SRTP/AVP the caller can't send RTP. Can you show us the SDP being sent by the phone? -Steve On 7 February 2013 11:01, Levend Sayar > wrote: Thanx Steven. Caller makes the offer for SDP but callee chooses whatever it wants. So caller can offer SRTP but callee can prefer not to talk encrypted. In our case I want FS to choose non secure media. Phone will offer SRTP on the conference call but FS must prefer RTP, not SRTP. ?yi ?al??malar. _lvnd_ {^_^} KAREL`UCAP On 7 ?ub 2013, at 11:13, "Steven Ayre" > wrote: It's also going to rely on the phone actually offering RTP/AVP as well as SRTP/AVP in their SDP - without that there'd be nowhere to send insecure RTP. -Steve On 6 February 2013 16:09, Levend Sayar > wrote: Thanks Daniel for the reply. I tried But did not work. Upon your reply I also tried But did not work either. I am doing something wrong ? ?yi ?al??malar. _lvnd_ {^_^} KAREL`UCAP On 6 ?ub 2013, at 18:00, "Daniel Ivanov" > wrote: Of course you can. Just set the secure_media var to false and you will be srtp-free in sip. On Feb 5, 2013 6:06 PM, "Levend Sayar" > wrote: Hi all. I am using FS as a conference server. Some of my phones are using SRTP , some of them not. Both type of phone can join a conference. FS can talk to each peer with SRTP or not depending on the phone itself. My question: Is it possible to disable SRTP on FS ? I suppose if i can disable SRTP, FS will talk without SRTP with each phone whether they are using SRTP or not. TIA -- _lvnd_ {^_^} _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 plain text document attachment (ATT00001) _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130207/7bf06ac6/attachment-0001.html From ben122uk at gmail.com Thu Feb 7 16:32:50 2013 From: ben122uk at gmail.com (Ben) Date: Thu, 7 Feb 2013 13:32:50 -0000 Subject: [Freeswitch-users] Use local IPv4 for specific LAN IP address range In-Reply-To: References: <01b001ce0528$6e4eece0$4aecc6a0$@gmail.com> Message-ID: <01ea01ce0537$a120e1f0$e362a5d0$@gmail.com> Hi Steve, thanks for getting back to me. Yes that's certainly my secondary option. However, if there's a chance that my scenario below could work then I'd like to explore it. As a bit more information, the internal SIP client is a SIP call generator that health checks the server periodically by putting through a call. I'd like it to put through a call to the live profile in parallel with the external clients so that it's testing the live profile, and not a separate test profile. If there's a situation where the live profile fails, but the test profile is still functional, the call generator would never know the live profile is down if it was calling the test profile. Unless you think this isn't likely to happen, and it's more likely the Sofia module would go down, bringing all profiles down with it? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven Ayre Sent: 07 February 2013 13:07 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Use local IPv4 for specific LAN IP address range One simple option would be to have a 2nd profile listening on another port (eg 5080) that does not use ext-*-ip and connect to that internally. -Steve On 7 February 2013 11:44, Ben wrote: Greetings all, A question on NATing in Freeswitch - I have my internal SIP profile to use a static public IP for the ext-sip-ip and ext-rtp-ip. The Freeswitch server is on a local LAN IP, for example 192.168.0.10. External SIP clients can access the server via the external IP, as there's a static NAT on the router to map the public IP to the private IP. I have another SIP client on a different LAN, but behind the same firewall, for example, 10.0.0.10. This client is allowed to talk to the server on its private ip of 192.168.0.10, by allowing inter-vlan traffic to pass through. My challenge is, how do I get Freeswitch to recognise this internal SIP client as being allowed to use the internal IP for SIP/RTP communications, and therefore present the client with internal addresses in the SIP header for SIP/RTP? Due to statically setting ext-sip-ip and ext-rtp-ip, Freeswitch tells the client to go via the public IP. The firewall has a policy to not accept traffic to the public IP if sourced from an internal device. I have explored the NATing options in Freeswitch, but can't find anything that fits my scenario. It seems to be more towards how to enable NAT in certain situations, not disable it. I have also looked at re-writing the SDP, but this re-writes the incoming SDP from the clients. My intention was to re-write the outgoing SDP from the Freeswitch to include the internal IP address as the RTP connect in address. I don't have much Asterisk experience, but I seem to remember a nat.conf file where you could specify subnets that were treated differently. These could be classed as internal subnets, and would be given the private IP in the SDP from asterisk, everything got the public IP. I'm also aware that I could create another SIP profile easily with different NATing/IP options, but I want to find out if there's anything I can do with the above first. Many thanks for the help! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130207/12d441d2/attachment.html From steveayre at gmail.com Thu Feb 7 16:37:34 2013 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 7 Feb 2013 13:37:34 +0000 Subject: [Freeswitch-users] SRTP disabling In-Reply-To: References: Message-ID: m=audio 11780 RTP/SAVP 0 8 18 9 101 RTP/SAVP means SRTP is mandatory. You need to reconfigure the phone. If the phone sends RTP/AVP then that means plain RTP, and RTP/AVP with a a=crypto attribute means SRTP is optional. -Steve On 7 February 2013 13:26, Levend Sayar wrote: > ** > Below is the SDP offer sent by the phone. > > v=0 > o=- 20185 20185 IN IP4 192.168.173.69 > s=SDP data > c=IN IP4 192.168.173.69 > t=0 0 > m=audio 11780 RTP/SAVP 0 8 18 9 101 > a=crypto:1 AES_CM_128_HMAC_SHA1_80 > inline:MTM2MjVhMGI1NDZjYmRjADU5NWVjNGVkNTNlYzA1 > a=crypto:2 AES_CM_128_HMAC_SHA1_32 > inline:YmExYmZhNQAzN2ZjNDgzYTRkNGU2ZjFiN2Q0MmE3 > a=crypto:3 F8_128_HMAC_SHA1_80 > inline:N2Q2NTRiYQAxZjA3MWY3ZjI1YTI5NjIyM2FjODYw > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:9 G722/8000 > a=fmtp:101 0-15 > a=rtpmap:101 telephone-event/8000 > a=ptime:20 > a=sendrecv > > > > And below is the SDP answer sent by FS > > v=0 > o=FreeSWITCH 1360230601 1360230602 IN IP4 192.168.169.114 > s=FreeSWITCH > c=IN IP4 192.168.169.114 > t=0 0 > m=audio 12532 RTP/SAVP 9 101 > a=rtpmap:9 G722/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=crypto:1 AES_CM_128_HMAC_SHA1_80 > inline:l8v0R64H7CP0vEx9j0Ycdbob8bgMCpLDppWGy7Dy > > > > > -- > ?yi ?al??malar. > > _lvnd_ > {^_^} > > KAREL`UCAP > > > > On Thu, 2013-02-07 at 13:09 +0000, Steven Ayre wrote: > > What I mean is you'll see two separate m=audio lines within the callee's > SDP, one for 'RTP/AVP' and one for 'SRTP/AVP'. If there is no m=audio line > for RTP/AVP the caller won't know of a port that's expecting RTP. So if the > callee only sends SRTP/AVP the caller can't send RTP. > > > > Can you show us the SDP being sent by the phone? > > > > -Steve > > > > > > > On 7 February 2013 11:01, Levend Sayar > wrote: > > Thanx Steven. > > > > Caller makes the offer for SDP but callee chooses whatever it wants. So > caller can offer SRTP but callee can prefer not to talk encrypted. In our > case I want FS to choose non secure media. > > Phone will offer SRTP on the conference call but FS must prefer RTP, not > SRTP. > > > > ?yi ?al??malar. > > > > _lvnd_ > > {^_^} > > > > KAREL`UCAP > > > > > On 7 ?ub 2013, at 11:13, "Steven Ayre" wrote: > > > It's also going to rely on the phone actually offering RTP/AVP as well > as SRTP/AVP in their SDP - without that there'd be nowhere to send insecure > RTP. > > > > -Steve > > > > > > On 6 February 2013 16:09, Levend Sayar > wrote: > > Thanks Daniel for the reply. > > > > I tried > > > > > > > > But did not work. Upon your reply I also tried > > > > > > > > But did not work either. I am doing something wrong ? > > ?yi ?al??malar. > > > > _lvnd_ > > {^_^} > > > > KAREL`UCAP > > > > > On 6 ?ub 2013, at 18:00, "Daniel Ivanov" wrote: > > > > Of course you can. Just set the secure_media var to false and you > will be srtp-free in sip. > > On Feb 5, 2013 6:06 PM, "Levend Sayar" > wrote: > > Hi all. > > I am using FS as a conference server. Some of my phones are using SRTP , > some of them not. Both type of phone can > join a conference. FS can talk to each peer with SRTP or not depending on > the phone itself. > > My question: > > Is it possible to disable SRTP on FS ? > > I suppose if i can disable SRTP, FS will talk without SRTP with each phone > whether they are using SRTP or not. > > TIA > > > -- > > _lvnd_ > {^_^} > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > plain text document attachment (ATT00001) > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130207/b7bbf26d/attachment-0001.html From levend.sayar at karel.com.tr Thu Feb 7 16:54:27 2013 From: levend.sayar at karel.com.tr (Levend Sayar) Date: Thu, 7 Feb 2013 13:54:27 +0000 Subject: [Freeswitch-users] SRTP disabling In-Reply-To: References: Message-ID: But the very same phone calls another phone and talk with RTP, not SRTP if the peer does not accept SRTP Here is the SDP offer by the same phone v=0 o=- 20186 20186 IN IP4 192.168.173.69 s=SDP data c=IN IP4 192.168.173.69 t=0 0 m=audio 11782 RTP/SAVP 0 8 18 9 101 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:NzFmYjdiMjk1OTY2ODQwYzExZjM0ZmE2NGM0YWMw a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:M2MxMTE2OWFjOGY2ZjEwADEzZmZkNzAxNjRlMzFm a=crypto:3 F8_128_HMAC_SHA1_80 inline:NjkzZDg2Mjk0ZTkxMjg1YzdmYjFiNjRlMmFhNGFm a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:9 G722/8000 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=ptime:20 a=sendrecv And here is the SDP answer sent by the other phone v=0 o=- 20029 20029 IN IP4 192.168.173.65 s=SDP data c=IN IP4 192.168.173.65 t=0 0 m=audio 11794 RTP/SAVP 0 101 a=rtpmap:0 PCMU/8000 a=sendrecv a=ptime:20 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 -- ?yi ?al??malar. _lvnd_ {^_^} KAREL`UCAP On Thu, 2013-02-07 at 13:37 +0000, Steven Ayre wrote: m=audio 11780 RTP/SAVP 0 8 18 9 101 RTP/SAVP means SRTP is mandatory. You need to reconfigure the phone. If the phone sends RTP/AVP then that means plain RTP, and RTP/AVP with a a=crypto attribute means SRTP is optional. -Steve On 7 February 2013 13:26, Levend Sayar > wrote: Below is the SDP offer sent by the phone. v=0 o=- 20185 20185 IN IP4 192.168.173.69 s=SDP data c=IN IP4 192.168.173.69 t=0 0 m=audio 11780 RTP/SAVP 0 8 18 9 101 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:MTM2MjVhMGI1NDZjYmRjADU5NWVjNGVkNTNlYzA1 a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:YmExYmZhNQAzN2ZjNDgzYTRkNGU2ZjFiN2Q0MmE3 a=crypto:3 F8_128_HMAC_SHA1_80 inline:N2Q2NTRiYQAxZjA3MWY3ZjI1YTI5NjIyM2FjODYw a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:9 G722/8000 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=ptime:20 a=sendrecv And below is the SDP answer sent by FS v=0 o=FreeSWITCH 1360230601 1360230602 IN IP4 192.168.169.114 s=FreeSWITCH c=IN IP4 192.168.169.114 t=0 0 m=audio 12532 RTP/SAVP 9 101 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:l8v0R64H7CP0vEx9j0Ycdbob8bgMCpLDppWGy7Dy -- ?yi ?al??malar. _lvnd_ {^_^} KAREL`UCAP On Thu, 2013-02-07 at 13:09 +0000, Steven Ayre wrote: What I mean is you'll see two separate m=audio lines within the callee's SDP, one for 'RTP/AVP' and one for 'SRTP/AVP'. If there is no m=audio line for RTP/AVP the caller won't know of a port that's expecting RTP. So if the callee only sends SRTP/AVP the caller can't send RTP. Can you show us the SDP being sent by the phone? -Steve On 7 February 2013 11:01, Levend Sayar > wrote: Thanx Steven. Caller makes the offer for SDP but callee chooses whatever it wants. So caller can offer SRTP but callee can prefer not to talk encrypted. In our case I want FS to choose non secure media. Phone will offer SRTP on the conference call but FS must prefer RTP, not SRTP. ?yi ?al??malar. _lvnd_ {^_^} KAREL`UCAP On 7 ?ub 2013, at 11:13, "Steven Ayre" > wrote: It's also going to rely on the phone actually offering RTP/AVP as well as SRTP/AVP in their SDP - without that there'd be nowhere to send insecure RTP. -Steve On 6 February 2013 16:09, Levend Sayar > wrote: Thanks Daniel for the reply. I tried But did not work. Upon your reply I also tried But did not work either. I am doing something wrong ? ?yi ?al??malar. _lvnd_ {^_^} KAREL`UCAP On 6 ?ub 2013, at 18:00, "Daniel Ivanov" > wrote: Of course you can. Just set the secure_media var to false and you will be srtp-free in sip. On Feb 5, 2013 6:06 PM, "Levend Sayar" > wrote: Hi all. I am using FS as a conference server. Some of my phones are using SRTP , some of them not. Both type of phone can join a conference. FS can talk to each peer with SRTP or not depending on the phone itself. My question: Is it possible to disable SRTP on FS ? I suppose if i can disable SRTP, FS will talk without SRTP with each phone whether they are using SRTP or not. TIA -- _lvnd_ {^_^} _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 plain text document attachment (ATT00001) _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 plain text document attachment (ATT00001) _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130207/52ecd076/attachment-0001.html From a.venugopan at mundio.com Thu Feb 7 17:42:50 2013 From: a.venugopan at mundio.com (Archana Venugopan) Date: Thu, 7 Feb 2013 14:42:50 +0000 Subject: [Freeswitch-users] call_queu Message-ID: <592A9CF93E12394E8472A6CC66E66BF2357FF1@Mail-Kilo.squay.com> Hi, I have very basic doubt in call queue and sorry for that query. If I just put the below in dialplan xml and dial say 250 and there is other one who is also dialling 250 will the call be in queue. In /usr/local/freeswich/conf/autoload_configs, default callcenter.conf.xml file is also present. Please let me know if i have to change anything else too to make call in queue ? Regards, Archana -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130207/8a70488b/attachment.html From jaykris at gmail.com Thu Feb 7 20:26:30 2013 From: jaykris at gmail.com (JP) Date: Thu, 7 Feb 2013 09:26:30 -0800 Subject: [Freeswitch-users] mod-curl POST In-Reply-To: References: Message-ID: Yes what I am looking for is to POST xml data in the body and not as name/value pairs. Is this possible? Thanks, -JP On Wed, Feb 6, 2013 at 7:21 PM, Rupa Schomaker wrote: > > > > On Wed, Feb 6, 2013 at 6:56 PM, Cal Leeming [Simplicity Media Ltd] < > cal.leeming at simplicitymedialtd.co.uk> wrote: > >> The docs says this; >> >> curl url [headers|json] [get|head|post [url_encode_data]] >> > > The [headers|json] is all about the response format. Just to make sure > that is clear. > > get/head/post should be self-evident > > url_encode_data is supposed to be the url encoded version of what you want > to send to the server. It is encoded on purpose (so that we don't have to > worry about embedded spaces or other special chars). > > So, you really just need to do: > > curl http://example.com/ post encodedxmlhere > > >> So, I assume you'd just do. >> >> curl http://example.com/ post xml=encodedxmlhere >> >> > This will result in www-url-encoded with key/value pair with xml as key > and the encoded xml as the value. This may be what the OP wants, but I > think what he wants is just XML. > > > Note that the content-type will still be application/x-www-form-urlencoded > which is "wrong". I've written xml service endpoints that behave > differently based on the content-type. Depending on how your service is > written this may be ok or not. There is an open ticket to address the > content-type issue. > > http://jira.freeswitch.org/browse/FS-4714 > > > >> Taken from here; >> http://wiki.freeswitch.org/wiki/Mod_curl >> >> Hope this helps, apologies if I misunderstood the question >> >> Cal >> >> > -- > -Rupa > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130207/00be21ff/attachment.html From cal.leeming at simplicitymedialtd.co.uk Thu Feb 7 21:14:50 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Thu, 7 Feb 2013 18:14:50 +0000 Subject: [Freeswitch-users] mod-curl POST In-Reply-To: References: Message-ID: Hi JP, >From what I can tell, this is not possible. Some options are; * Submit a ticket to JIRA asking for this feature * If you have control over the end point, add support for non XML POST requests. * If you have no control over the end point, create a proxy script which transforms the POST request into the necessary format and passes it along Hope this helps Cal On Thu, Feb 7, 2013 at 5:26 PM, JP wrote: > Yes what I am looking for is to POST xml data in the body and not as > name/value pairs. Is this possible? > > Thanks, > -JP > > On Wed, Feb 6, 2013 at 7:21 PM, Rupa Schomaker wrote: > >> >> >> >> On Wed, Feb 6, 2013 at 6:56 PM, Cal Leeming [Simplicity Media Ltd] < >> cal.leeming at simplicitymedialtd.co.uk> wrote: >> >>> The docs says this; >>> >>> curl url [headers|json] [get|head|post [url_encode_data]] >>> >> >> The [headers|json] is all about the response format. Just to make sure >> that is clear. >> >> get/head/post should be self-evident >> >> url_encode_data is supposed to be the url encoded version of what you >> want to send to the server. It is encoded on purpose (so that we don't >> have to worry about embedded spaces or other special chars). >> >> So, you really just need to do: >> >> curl http://example.com/ post encodedxmlhere >> >> >>> So, I assume you'd just do. >>> >>> curl http://example.com/ post xml=encodedxmlhere >>> >>> >> This will result in www-url-encoded with key/value pair with xml as key >> and the encoded xml as the value. This may be what the OP wants, but I >> think what he wants is just XML. >> >> >> Note that the content-type will still be application/x-www-form-urlencoded >> which is "wrong". I've written xml service endpoints that behave >> differently based on the content-type. Depending on how your service is >> written this may be ok or not. There is an open ticket to address the >> content-type issue. >> >> http://jira.freeswitch.org/browse/FS-4714 >> >> >> >>> Taken from here; >>> http://wiki.freeswitch.org/wiki/Mod_curl >>> >>> Hope this helps, apologies if I misunderstood the question >>> >>> Cal >>> >>> >> -- >> -Rupa >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130207/d1055ef1/attachment-0001.html From vbvbrj at gmail.com Thu Feb 7 22:36:01 2013 From: vbvbrj at gmail.com (Mimiko) Date: Thu, 07 Feb 2013 21:36:01 +0200 Subject: [Freeswitch-users] VoIP gateway with PRI Message-ID: <51140221.4040107@gmail.com> Hello. I want to propose for purchasing a VoIP gateway to connect some PRI lines with FS. Basic operation has to be routing like this: One line PRI will be connected to phone provider. Some calls to certain numbers thru this connection must be routed to FS via IP. Other destination numbers must be routed to second PRI connection for legacy PBX for analog phones. So the VoIP gateway must have two PRI (T1/E1) connections and two LAN connection for failover. After a while this VoIP gateway will be as a tehnology converter from IP to PRI. So LAN connectors will be used for connection to FS in a failover manner. And two PRI connectors will be used to connect to legacy PBX which have two PRI inputs to handle 60 concurrent connection. I've spotted three models: 1) Patton SN4951/4E60V2GS/EUI 4T1/E1/PRI 60 VoIP Channels G.SHDSL for $5,000 2) Audiocodes Mediant 1000 M1K-D2 2T1/E1 for $3,800 3) Sangoma Vega 400 2 T1/E1 Digital Gateway for $4,437 Which of this is a better working solution with failover and extended call routing plans? Thansk. -- Mimiko desu. From jaykris at gmail.com Thu Feb 7 22:41:31 2013 From: jaykris at gmail.com (JP) Date: Thu, 7 Feb 2013 11:41:31 -0800 Subject: [Freeswitch-users] mod-curl POST In-Reply-To: References: Message-ID: Thanks Cal, Let me weigh my options and figure out what is the best approach for me. Regards, JP On Thu, Feb 7, 2013 at 10:14 AM, Cal Leeming [Simplicity Media Ltd] < cal.leeming at simplicitymedialtd.co.uk> wrote: > Hi JP, > > From what I can tell, this is not possible. > > Some options are; > > * Submit a ticket to JIRA asking for this feature > * If you have control over the end point, add support for non XML POST > requests. > * If you have no control over the end point, create a proxy script which > transforms the POST request into the necessary format and passes it along > > Hope this helps > > Cal > > On Thu, Feb 7, 2013 at 5:26 PM, JP wrote: > >> Yes what I am looking for is to POST xml data in the body and not as >> name/value pairs. Is this possible? >> >> Thanks, >> -JP >> >> On Wed, Feb 6, 2013 at 7:21 PM, Rupa Schomaker wrote: >> >>> >>> >>> >>> On Wed, Feb 6, 2013 at 6:56 PM, Cal Leeming [Simplicity Media Ltd] < >>> cal.leeming at simplicitymedialtd.co.uk> wrote: >>> >>>> The docs says this; >>>> >>>> curl url [headers|json] [get|head|post [url_encode_data]] >>>> >>> >>> The [headers|json] is all about the response format. Just to make sure >>> that is clear. >>> >>> get/head/post should be self-evident >>> >>> url_encode_data is supposed to be the url encoded version of what you >>> want to send to the server. It is encoded on purpose (so that we don't >>> have to worry about embedded spaces or other special chars). >>> >>> So, you really just need to do: >>> >>> curl http://example.com/ post encodedxmlhere >>> >>> >>>> So, I assume you'd just do. >>>> >>>> curl http://example.com/ post xml=encodedxmlhere >>>> >>>> >>> This will result in www-url-encoded with key/value pair with xml as key >>> and the encoded xml as the value. This may be what the OP wants, but I >>> think what he wants is just XML. >>> >>> >>> Note that the content-type will still be application/x-www-form-urlencoded >>> which is "wrong". I've written xml service endpoints that behave >>> differently based on the content-type. Depending on how your service is >>> written this may be ok or not. There is an open ticket to address the >>> content-type issue. >>> >>> http://jira.freeswitch.org/browse/FS-4714 >>> >>> >>> >>>> Taken from here; >>>> http://wiki.freeswitch.org/wiki/Mod_curl >>>> >>>> Hope this helps, apologies if I misunderstood the question >>>> >>>> Cal >>>> >>>> >>> -- >>> -Rupa >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130207/6d75104b/attachment.html From yehavi.bourvine at gmail.com Thu Feb 7 23:05:51 2013 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Thu, 7 Feb 2013 22:05:51 +0200 Subject: [Freeswitch-users] VoIP gateway with PRI In-Reply-To: <51140221.4040107@gmail.com> References: <51140221.4040107@gmail.com> Message-ID: Hi, I have used only the Audiocodes ones (had also Cisco, but Audiocodes is way better than it). I did not try what you want, but I think it can do it. Since I moved from Cisco to Audiocodes I am happy. Do you want to route calls from one PRI to the other without having any SIP involved for these calls? If this is the case the I'll have to ask my Audiocodes contacts about it. The Mediant-1,000 can use two ethernet ports with spanning tree for failover. I did not try it, and you should make sure that the software version you get supports this. __Yehavi: 2013/2/7 Mimiko > Hello. > > I want to propose for purchasing a VoIP gateway to connect some PRI > lines with FS. Basic operation has to be routing like this: > > One line PRI will be connected to phone provider. Some calls to certain > numbers thru this connection must be routed to FS via IP. Other > destination numbers must be routed to second PRI connection for legacy > PBX for analog phones. So the VoIP gateway must have two PRI (T1/E1) > connections and two LAN connection for failover. > > After a while this VoIP gateway will be as a tehnology converter from IP > to PRI. So LAN connectors will be used for connection to FS in a > failover manner. And two PRI connectors will be used to connect to > legacy PBX which have two PRI inputs to handle 60 concurrent connection. > > I've spotted three models: > 1) Patton SN4951/4E60V2GS/EUI 4T1/E1/PRI 60 VoIP Channels G.SHDSL for > $5,000 > > 2) Audiocodes Mediant 1000 M1K-D2 2T1/E1 for $3,800 > > 3) Sangoma Vega 400 2 T1/E1 Digital Gateway for $4,437 > > Which of this is a better working solution with failover and extended > call routing plans? > > Thansk. > > -- > Mimiko desu. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130207/9c7a606d/attachment.html From carlos.voiperu at gmail.com Thu Feb 7 23:27:56 2013 From: carlos.voiperu at gmail.com (Carlos Rojas) Date: Thu, 7 Feb 2013 15:27:56 -0500 Subject: [Freeswitch-users] Real certificate in freeswitch Message-ID: Hi Everybody I am new with freeswitch, I was trying, put in freeswithc my real certificate, in my freeswitch, but doesn't work. looks like freeswitch accepts the certificate, but when the phone tried register doesn't work. Anybody has configured a real certificate with freeswitch? I hope somebody could help me. Kind Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130207/3ed2f1bd/attachment.html From msc at freeswitch.org Thu Feb 7 23:37:39 2013 From: msc at freeswitch.org (Michael Collins) Date: Thu, 7 Feb 2013 12:37:39 -0800 Subject: [Freeswitch-users] Homer/sipcapture In-Reply-To: References: Message-ID: IIRC HOMER by default stores 7 days of data. Alexandr says that you can store millions of calls with a GB of storage space. Hopefully he can come on board to offer some more specific information. -MC On Thu, Feb 7, 2013 at 12:26 AM, Gavin Henry wrote: > Hi all, > > What's the performance of this like? We have OpenSIPS and FreeSWITCH boxes > and not sure what size we need for the homer storage box/s > > Any experience to share? > > Thanks. > > -- > Gavin Henry > Sent with Sparrow > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130207/1bf06427/attachment-0001.html From steveayre at gmail.com Thu Feb 7 23:39:32 2013 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 7 Feb 2013 20:39:32 +0000 Subject: [Freeswitch-users] VoIP gateway with PRI In-Reply-To: <51140221.4040107@gmail.com> References: <51140221.4040107@gmail.com> Message-ID: As another option, you can also get PRI cards that you install in the FS server itself that might prove cheaper (if you have the space). You could quite easily handle both PRI-PRI and PRI-SIP calls in the FS dialplan. For example: http://sangoma.com/products/telecom_boards/digital.html Google Shopper says the 60 channel A102 is about a quarter of the price of the options you're currently looking at. http://wiki.freeswitch.org/wiki/FreeTDM (check the card is supported before buying) -Steve On 7 February 2013 19:36, Mimiko wrote: > Hello. > > I want to propose for purchasing a VoIP gateway to connect some PRI > lines with FS. Basic operation has to be routing like this: > > One line PRI will be connected to phone provider. Some calls to certain > numbers thru this connection must be routed to FS via IP. Other > destination numbers must be routed to second PRI connection for legacy > PBX for analog phones. So the VoIP gateway must have two PRI (T1/E1) > connections and two LAN connection for failover. > > After a while this VoIP gateway will be as a tehnology converter from IP > to PRI. So LAN connectors will be used for connection to FS in a > failover manner. And two PRI connectors will be used to connect to > legacy PBX which have two PRI inputs to handle 60 concurrent connection. > > I've spotted three models: > 1) Patton SN4951/4E60V2GS/EUI 4T1/E1/PRI 60 VoIP Channels G.SHDSL for > $5,000 > > 2) Audiocodes Mediant 1000 M1K-D2 2T1/E1 for $3,800 > > 3) Sangoma Vega 400 2 T1/E1 Digital Gateway for $4,437 > > Which of this is a better working solution with failover and extended > call routing plans? > > Thansk. > > -- > Mimiko desu. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130207/63ad9bd9/attachment.html From steveayre at gmail.com Thu Feb 7 23:42:37 2013 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 7 Feb 2013 20:42:37 +0000 Subject: [Freeswitch-users] VoIP gateway with PRI In-Reply-To: References: <51140221.4040107@gmail.com> Message-ID: And since the PRI is within FS itself you don't need to worry about LAN failover between the PRI gateway and FS at all. On my FS servers I use bonded interfaces to get failover for LAN/WAN: http://www.kernel.org/doc/Documentation/networking/bonding.txt -Steve On 7 February 2013 20:39, Steven Ayre wrote: > As another option, you can also get PRI cards that you install in the FS > server itself that might prove cheaper (if you have the space). You could > quite easily handle both PRI-PRI and PRI-SIP calls in the FS dialplan. > > For example: > http://sangoma.com/products/telecom_boards/digital.html > > Google Shopper says the 60 channel A102 is about a quarter of the price of > the options you're currently looking at. > > http://wiki.freeswitch.org/wiki/FreeTDM > (check the card is supported before buying) > > -Steve > > > > On 7 February 2013 19:36, Mimiko wrote: > >> Hello. >> >> I want to propose for purchasing a VoIP gateway to connect some PRI >> lines with FS. Basic operation has to be routing like this: >> >> One line PRI will be connected to phone provider. Some calls to certain >> numbers thru this connection must be routed to FS via IP. Other >> destination numbers must be routed to second PRI connection for legacy >> PBX for analog phones. So the VoIP gateway must have two PRI (T1/E1) >> connections and two LAN connection for failover. >> >> After a while this VoIP gateway will be as a tehnology converter from IP >> to PRI. So LAN connectors will be used for connection to FS in a >> failover manner. And two PRI connectors will be used to connect to >> legacy PBX which have two PRI inputs to handle 60 concurrent connection. >> >> I've spotted three models: >> 1) Patton SN4951/4E60V2GS/EUI 4T1/E1/PRI 60 VoIP Channels G.SHDSL for >> $5,000 >> >> 2) Audiocodes Mediant 1000 M1K-D2 2T1/E1 for $3,800 >> >> 3) Sangoma Vega 400 2 T1/E1 Digital Gateway for $4,437 >> >> Which of this is a better working solution with failover and extended >> call routing plans? >> >> Thansk. >> >> -- >> Mimiko desu. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130207/e014c295/attachment.html From msc at freeswitch.org Fri Feb 8 02:42:24 2013 From: msc at freeswitch.org (Michael Collins) Date: Thu, 7 Feb 2013 15:42:24 -0800 Subject: [Freeswitch-users] Can local_stream MOH have a time limit/max duration? In-Reply-To: <219BCB63-3A5A-419F-B374-BDD45FDE3FAB@mgtech.com> References: <219BCB63-3A5A-419F-B374-BDD45FDE3FAB@mgtech.com> Message-ID: The ringback will play until the call is answered or until the call_timeout value is met and the call moves on in the dialplan. You can probably just use call_timeout to tell the call to move on after x number of seconds. -MC On Wed, Feb 6, 2013 at 9:04 AM, Mario G wrote: > Could not find anything in the wiki and this may not be possible but... I > have early media with instructions, is has voice, ring, voice, music. I > want to change the ring to music but I could not find a way to limit the > time the music plays. See the second example, I would like to limit the > first moh to a maximum time. If not possible, is this a "bounty" > enhancement? Thanks for any help. > Mario G > > > Currently: > "ringback=file_string://${lua_ring1}!tone_stream://${us-ring};loops=3!file_string://${lua_ring2}!local_stream://moh" > /> > > Would Like (add something to limit the first moh): > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130207/925b63c8/attachment.html From jaybinks at gmail.com Fri Feb 8 02:56:00 2013 From: jaybinks at gmail.com (jay binks) Date: Fri, 8 Feb 2013 09:56:00 +1000 Subject: [Freeswitch-users] Homer/sipcapture In-Reply-To: References: Message-ID: I think the real question there is, how much extra cpu does the homer hep code ( in freeswitch ) require especially compared to tcpdump / libpcap solutions ? On 8 February 2013 06:37, Michael Collins wrote: > IIRC HOMER by default stores 7 days of data. Alexandr says that you can > store millions of calls with a GB of storage space. Hopefully he can come > on board to offer some more specific information. > > -MC > > On Thu, Feb 7, 2013 at 12:26 AM, Gavin Henry wrote: > >> Hi all, >> >> What's the performance of this like? We have OpenSIPS and FreeSWITCH >> boxes and not sure what size we need for the homer storage box/s >> >> Any experience to share? >> >> Thanks. >> >> -- >> Gavin Henry >> Sent with Sparrow >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Sincerely Jay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130208/d09998c7/attachment-0001.html From msc at freeswitch.org Fri Feb 8 03:23:29 2013 From: msc at freeswitch.org (Michael Collins) Date: Thu, 7 Feb 2013 16:23:29 -0800 Subject: [Freeswitch-users] Homer/sipcapture In-Reply-To: References: Message-ID: I'm not sure we have that quantified but I believe that it's pretty low. It does, though increase network traffic so be on the lookout for that... -MC On Thu, Feb 7, 2013 at 3:56 PM, jay binks wrote: > I think the real question there is, how much extra cpu does the homer hep > code ( in freeswitch ) require > especially compared to tcpdump / libpcap solutions ? > > > On 8 February 2013 06:37, Michael Collins wrote: > >> IIRC HOMER by default stores 7 days of data. Alexandr says that you can >> store millions of calls with a GB of storage space. Hopefully he can come >> on board to offer some more specific information. >> >> -MC >> >> On Thu, Feb 7, 2013 at 12:26 AM, Gavin Henry wrote: >> >>> Hi all, >>> >>> What's the performance of this like? We have OpenSIPS and FreeSWITCH >>> boxes and not sure what size we need for the homer storage box/s >>> >>> Any experience to share? >>> >>> Thanks. >>> >>> -- >>> Gavin Henry >>> Sent with Sparrow >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > > -- > Sincerely > > Jay > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130207/09222ac5/attachment.html From rupa at rupa.com Fri Feb 8 04:49:48 2013 From: rupa at rupa.com (Rupa Schomaker) Date: Thu, 7 Feb 2013 19:49:48 -0600 Subject: [Freeswitch-users] mod-curl POST In-Reply-To: References: Message-ID: Umm.. I just explained how to do it. The only piece missing is the setting of the content-type header, but if that isn't going to be a problem you odn't have to do anything special. Just remember to urlencode the (full) contents of the XML. And try it out. Use wireshark to see what is sent over the wire to the server. On Thu, Feb 7, 2013 at 11:26 AM, JP wrote: > Yes what I am looking for is to POST xml data in the body and not as > name/value pairs. Is this possible? > > Thanks, > -JP > > On Wed, Feb 6, 2013 at 7:21 PM, Rupa Schomaker wrote: > >> >> >> >> On Wed, Feb 6, 2013 at 6:56 PM, Cal Leeming [Simplicity Media Ltd] < >> cal.leeming at simplicitymedialtd.co.uk> wrote: >> >>> The docs says this; >>> >>> curl url [headers|json] [get|head|post [url_encode_data]] >>> >> >> The [headers|json] is all about the response format. Just to make sure >> that is clear. >> >> get/head/post should be self-evident >> >> url_encode_data is supposed to be the url encoded version of what you >> want to send to the server. It is encoded on purpose (so that we don't >> have to worry about embedded spaces or other special chars). >> >> So, you really just need to do: >> >> curl http://example.com/ post encodedxmlhere >> >> >>> So, I assume you'd just do. >>> >>> curl http://example.com/ post xml=encodedxmlhere >>> >>> >> This will result in www-url-encoded with key/value pair with xml as key >> and the encoded xml as the value. This may be what the OP wants, but I >> think what he wants is just XML. >> >> >> Note that the content-type will still be application/x-www-form-urlencoded >> which is "wrong". I've written xml service endpoints that behave >> differently based on the content-type. Depending on how your service is >> written this may be ok or not. There is an open ticket to address the >> content-type issue. >> >> http://jira.freeswitch.org/browse/FS-4714 >> >> >> >>> Taken from here; >>> http://wiki.freeswitch.org/wiki/Mod_curl >>> >>> Hope this helps, apologies if I misunderstood the question >>> >>> Cal >>> >>> >> -- >> -Rupa >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130207/7a556689/attachment.html From william at xofap.com Fri Feb 8 06:27:22 2013 From: william at xofap.com (William Alianto) Date: Fri, 08 Feb 2013 10:27:22 +0700 Subject: [Freeswitch-users] Force G729 codec for E1 connection In-Reply-To: References: <511383D2.9090806@xofap.com> Message-ID: <5114709A.8020704@xofap.com> Hi Steve, I actually has installed a transcoding card along with the E1 card. I was hoping to do something like media proxy at the gateway. Is that possible? Regards, William On 07/02/2013 19:10, Steven Ayre wrote: > E1 uses G711... I'm assuming by saying that FS is acting as the > gateway you're using something like mod_freetdm? > > If so FS can transcode so that it uses G729 for the SIP side of the > bridge. > > Look at the commercial mod_com_g729 (licenses available for > $10/channel on www.freeswitch.org ). > http://wiki.freeswitch.org/wiki/Mod_com_g729 > > You just need to have that loaded instead of mod_g729, and tweak your > codec preferences accordingly. > http://wiki.freeswitch.org/wiki/Codec_Negotiation > > -Steve > > > On 7 February 2013 10:37, William Alianto > wrote: > > Hi, > > I just finished a freeswitch server that act as E1 gateway for PBX > system. I noticed that all the call from the gateway use PCMU codec > before transcoded to G729 in my main SIP server. > > My current situation is as following : > > [PBX] --- E1 ---> [Gateway] --- SIP/PCMU ---> [SIP Server] --- G729 > ---> [ITSP] > > What I intended to do is as following : > > [PBX] --- E1 ---> [Gateway] --- SIP/G729 ---> [SIP Server] --- G729 > ---> [ITSP] > > Is there anyway to send the media from E1 gateway as G729 instead of > PCMU? I need to preserve bandwidth for future expansion. > > Regards, > > William > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130208/914af343/attachment-0001.html From victor.chukalovskiy at gmail.com Fri Feb 8 07:24:29 2013 From: victor.chukalovskiy at gmail.com (Victor Chukalovskiy) Date: Thu, 07 Feb 2013 23:24:29 -0500 Subject: [Freeswitch-users] [ERR] switch_limit.c:86 Unset limit backendlist! Message-ID: <51147DFD.6040304@gmail.com> Hello, I'm getting this error occasionally in the log: [ERR] switch_limit.c:86 Unset limit backendlist! Seems to happen only after I change the value of CPS limit on the working system. For example, after I change *4* to 3 below and send some more calls: Could someone please explain when exactly does this error occur? How to reset it other than restarting FS instance? What I found is that I'm not the only one who got annoyed, and that loglevel was reduced in later versions: http://jira.freeswitch.org/browse/FS-4792?page=com.atlassian.streams.streams-jira-plugin:activity-stream-issue-tab However, I want to figure out how to avoid this error in the first place rather than just update and suppress it. Any help is appreciated. Thank you, Victor -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130207/b79dfbf1/attachment.html From vbvbrj at gmail.com Fri Feb 8 09:11:21 2013 From: vbvbrj at gmail.com (Mimiko) Date: Fri, 08 Feb 2013 08:11:21 +0200 Subject: [Freeswitch-users] VoIP gateway with PRI In-Reply-To: References: <51140221.4040107@gmail.com> Message-ID: <51149709.1000908@gmail.com> On 07.02.2013 22:05, Yehavi Bourvine wrote: > Do you want to route calls from one PRI to the other without having any > SIP involved for these calls? If this is the case the I'll have to ask > my Audiocodes contacts about it. Yes, certain calls will be routed only between PRI interfaces, as the calls may be free of charge and loading VoIP sessions with those calls is unnecessary. > As another option, you can also get PRI cards that you install in the FS > server itself that might prove cheaper (if you have the space). You > could quite easily handle both PRI-PRI and PRI-SIP calls in the FS dialplan. > Some times ago I had this discussion on IRC about what is better to use in a failover environment: internal PRI cards or external VoIP gateways. So in case of connection two or more servers to a legacy PBX with only two PRI inputs, an external gateway is a better solution, despite internal cards a cheaper. Also if you have a PRI phone provider, in case of an FS server failing you have to manually switch the cable. But I will look to this links. -- Mimiko desu. From eagle.antonio at gmail.com Fri Feb 8 11:05:58 2013 From: eagle.antonio at gmail.com (Antonio Teixeira) Date: Fri, 08 Feb 2013 08:05:58 +0000 Subject: [Freeswitch-users] Sorry To Bump but .. Message-ID: <5114B1E6.7090402@gmail.com> Can anyone take a look at this ? http://jira.freeswitch.org/browse/FS-5069 I have performed dozens of make's the last few days and still i have been unable to get this to work... Thanks For Your Time Antonio From vbvbrj at gmail.com Fri Feb 8 11:14:43 2013 From: vbvbrj at gmail.com (Mimiko) Date: Fri, 08 Feb 2013 10:14:43 +0200 Subject: [Freeswitch-users] Specifying file permission for audio records. In-Reply-To: References: <50DF3BAB.4040209@gmail.com> Message-ID: <5114B3F3.4010309@gmail.com> On 30.12.2012 02:30, Gabriel Gunderson wrote: > On Sat, Dec 29, 2012 at 11:51 AM, Mimiko wrote: >> I want audio files recorded from callcenter to be available to a web >> server. Both webserver and FS runs on same server. The storage directory >> for holding recorded audio files is on the same server. FS run in >> unprivileged mode, and the directory for recorded audio files have >> permission 775. Webserver runs unprivileged under another user. > > Something like this? > > http://en.wikipedia.org/wiki/Setuid#setuid_and_setgid_on_directories > Well I looked and tried finally. It not quite what I need. I need the everyone permission to be inherited. Not the group ID, but permission set on directory. For example directory have this permission: 777. When FS writes a file to that directory, permission of that file became 750. And is not inherited from parent. -- Mimiko desu. From shaheryarkh at gmail.com Fri Feb 8 11:53:18 2013 From: shaheryarkh at gmail.com (Muhammad Shahzad) Date: Fri, 8 Feb 2013 09:53:18 +0100 Subject: [Freeswitch-users] Sorry To Bump but .. In-Reply-To: <5114B1E6.7090402@gmail.com> References: <5114B1E6.7090402@gmail.com> Message-ID: I haven't used centos since version 5.4. However following should fix libstdc++ problem that you seem to have with building libedit. yum groupinstall 'development tools' yum groupinstall 'development libraries' It would be better to use a freshly installed machine if possible for you. After doing group install check if following libs were successfully installed (only for double check), yum install libxml2-devel libncurses-devel postgresql-devel sqlite-devel curl-devel openssl-devel Note: some lib names might be different. Try with or without lib prefix. Thank you. On 8 Feb 2013 09:09, "Antonio Teixeira" wrote: > > Can anyone take a look at this ? > > http://jira.freeswitch.org/browse/FS-5069 > > I have performed dozens of make's the last few days and still i have > been unable to get this to work... > > Thanks For Your Time > Antonio > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130208/b648a665/attachment.html From steveayre at gmail.com Fri Feb 8 12:08:23 2013 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 8 Feb 2013 09:08:23 +0000 Subject: [Freeswitch-users] [ERR] switch_limit.c:86 Unset limit backendlist! In-Reply-To: <51147DFD.6040304@gmail.com> References: <51147DFD.6040304@gmail.com> Message-ID: Looks like it's an assert-type error. limit_state_handler() is called without the limit_backend variable being set, which is an internal variable for storing limit information used to internally track limit information during the call. The odd thing is that variable should always be set prior to limit_state_handler being registered. I'm guessing it's some sort of race condition... -Steve On 8 February 2013 04:24, Victor Chukalovskiy wrote: > Hello, > > I'm getting this error occasionally in the log: > [ERR] switch_limit.c:86 Unset limit backendlist! > > Seems to happen only after I change the value of CPS limit on the working > system. For example, after I change *4* to 3 below and send some more > calls: > > > Could someone please explain when exactly does this error occur? How to > reset it other than restarting FS instance? > > What I found is that I'm not the only one who got annoyed, and that > loglevel was reduced in later versions: > > http://jira.freeswitch.org/browse/FS-4792?page=com.atlassian.streams.streams-jira-plugin:activity-stream-issue-tab > > However, I want to figure out how to avoid this error in the first place > rather than just update and suppress it. Any help is appreciated. > > Thank you, > Victor > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130208/c6bb7740/attachment.html From eagle.antonio at gmail.com Fri Feb 8 13:33:39 2013 From: eagle.antonio at gmail.com (Antonio Teixeira) Date: Fri, 08 Feb 2013 10:33:39 +0000 Subject: [Freeswitch-users] Sorry To Bump but .. In-Reply-To: References: <5114B1E6.7090402@gmail.com> Message-ID: <5114D483.5060506@gmail.com> Hello Muhammad. Still no go , http://pastebin.freeswitch.org/20558 Tried yum groupinstall 'development tools' <- Works ... And also yum install libxml2-devel libncurses-devel postgresql-devel sqlite-devel curl-devel openssl-devel <- Works And also yum install libedit-devel <- just in case With every make attempt a clean build dir ( i mean rm-rf clean ..) Going to make a new VM ... :( Thanks for the help Antonio On 2/8/13 8:53 AM, Muhammad Shahzad wrote: > > I haven't used centos since version 5.4. However following should fix > libstdc++ problem that you seem to have with building libedit. > > yum groupinstall 'development tools' > yum groupinstall 'development libraries' > > It would be better to use a freshly installed machine if possible for > you. After doing group install check if following libs were > successfully installed (only for double check), > > yum install libxml2-devel libncurses-devel postgresql-devel > sqlite-devel curl-devel openssl-devel > > Note: some lib names might be different. Try with or without lib prefix. > > Thank you. > > > On 8 Feb 2013 09:09, "Antonio Teixeira" > wrote: > > > > Can anyone take a look at this ? > > > > http://jira.freeswitch.org/browse/FS-5069 > > > > I have performed dozens of make's the last few days and still i have > > been unable to get this to work... > > > > Thanks For Your Time > > Antonio > > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130208/b377013e/attachment-0001.html From sertys at gmail.com Fri Feb 8 13:41:22 2013 From: sertys at gmail.com (Daniel Ivanov) Date: Fri, 8 Feb 2013 11:41:22 +0100 Subject: [Freeswitch-users] Real certificate in freeswitch In-Reply-To: References: Message-ID: But of course we have. Freeswitch accepts certificate bundles. So be sure to include your whole certificate chain in a concatenated file. Like ca.crt + intermediate (if any) + your.crt. and to be sure you've got the right combo use some of the online ssl checkers and point the to your ip/ domain and port. They just make a regular tcp session and show you the presented certificate chain. Good luck. On Feb 7, 2013 9:40 PM, "Carlos Rojas" wrote: > Hi Everybody > > I am new with freeswitch, I was trying, put in freeswithc my real > certificate, in my freeswitch, but doesn't work. > > looks like freeswitch accepts the certificate, but when the phone tried > register doesn't work. > > Anybody has configured a real certificate with freeswitch? > > > I hope somebody could help me. > > > Kind Regards > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130208/0ee3130b/attachment.html From steveayre at gmail.com Fri Feb 8 14:00:27 2013 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 8 Feb 2013 11:00:27 +0000 Subject: [Freeswitch-users] Real certificate in freeswitch In-Reply-To: References: Message-ID: <24F5E90E-1B95-403E-AC3B-4D66BAB72FED@gmail.com> If you're talking about SIP/TLS and a server certificate on FS then the phone will also need the correct CA certificates installed which may not be the case by default, and the correct time set so that it can verify them. Steve On 8 Feb 2013, at 10:41, Daniel Ivanov wrote: > But of course we have. Freeswitch accepts certificate bundles. So be sure to include your whole certificate chain in a concatenated file. Like ca.crt + intermediate (if any) + your.crt. and to be sure you've got the right combo use some of the online ssl checkers and point the to your ip/ domain and port. They just make a regular tcp session and show you the presented certificate chain. > > Good luck. > > On Feb 7, 2013 9:40 PM, "Carlos Rojas" wrote: >> Hi Everybody >> >> I am new with freeswitch, I was trying, put in freeswithc my real certificate, in my freeswitch, but doesn't work. >> >> looks like freeswitch accepts the certificate, but when the phone tried register doesn't work. >> >> Anybody has configured a real certificate with freeswitch? >> >> >> I hope somebody could help me. >> >> >> Kind Regards >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130208/46e471da/attachment.html From shaheryarkh at gmail.com Fri Feb 8 14:55:48 2013 From: shaheryarkh at gmail.com (Muhammad Shahzad) Date: Fri, 8 Feb 2013 12:55:48 +0100 Subject: [Freeswitch-users] Sorry To Bump but .. In-Reply-To: <5114D483.5060506@gmail.com> References: <5114B1E6.7090402@gmail.com> <5114D483.5060506@gmail.com> Message-ID: can't login to paste bin. Anyhow, let me try centos 6.3 myself and see if i can reproduce the issue. Thank you. On Fri, Feb 8, 2013 at 11:33 AM, Antonio Teixeira wrote: > Hello Muhammad. > > Still no go , > > > http://pastebin.freeswitch.org/20558 > > Tried > yum groupinstall 'development tools' <- Works ... > And also > yum install libxml2-devel libncurses-devel postgresql-devel sqlite-devel > curl-devel openssl-devel <- Works > And also > yum install libedit-devel <- just in case > > With every make attempt a clean build dir ( i mean rm-rf clean ..) > > Going to make a new VM ... :( > > Thanks for the help > Antonio > > > On 2/8/13 8:53 AM, Muhammad Shahzad wrote: > > I haven't used centos since version 5.4. However following should fix > libstdc++ problem that you seem to have with building libedit. > > yum groupinstall 'development tools' > yum groupinstall 'development libraries' > > It would be better to use a freshly installed machine if possible for you. > After doing group install check if following libs were successfully > installed (only for double check), > > yum install libxml2-devel libncurses-devel postgresql-devel sqlite-devel > curl-devel openssl-devel > > Note: some lib names might be different. Try with or without lib prefix. > > Thank you. > > > On 8 Feb 2013 09:09, "Antonio Teixeira" wrote: > > > > Can anyone take a look at this ? > > > > http://jira.freeswitch.org/browse/FS-5069 > > > > I have performed dozens of make's the last few days and still i have > > been unable to get this to work... > > > > Thanks For Your Time > > Antonio > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +49 176 99 83 10 85 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130208/4f10a627/attachment-0001.html From POlsson at enghouse.com Fri Feb 8 15:06:10 2013 From: POlsson at enghouse.com (Peter Olsson) Date: Fri, 8 Feb 2013 12:06:10 +0000 Subject: [Freeswitch-users] Sorry To Bump but .. Message-ID: <1FFF97C269757C458224B7C895F35F1521B0ED@cantor.std.visionutv.se> Install ncurses-devel package. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Antonio Teixeira Skickat: den 8 februari 2013 11:34 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] Sorry To Bump but .. Hello Muhammad. Still no go , http://pastebin.freeswitch.org/20558 Tried yum groupinstall 'development tools' <- Works ... And also yum install libxml2-devel libncurses-devel postgresql-devel sqlite-devel curl-devel openssl-devel <- Works And also yum install libedit-devel <- just in case With every make attempt a clean build dir ( i mean rm-rf clean ..) Going to make a new VM ... :( Thanks for the help Antonio On 2/8/13 8:53 AM, Muhammad Shahzad wrote: I haven't used centos since version 5.4. However following should fix libstdc++ problem that you seem to have with building libedit. yum groupinstall 'development tools' yum groupinstall 'development libraries' It would be better to use a freshly installed machine if possible for you. After doing group install check if following libs were successfully installed (only for double check), yum install libxml2-devel libncurses-devel postgresql-devel sqlite-devel curl-devel openssl-devel Note: some lib names might be different. Try with or without lib prefix. Thank you. On 8 Feb 2013 09:09, "Antonio Teixeira" > wrote: > > Can anyone take a look at this ? > > http://jira.freeswitch.org/browse/FS-5069 > > I have performed dozens of make's the last few days and still i have > been unable to get this to work... > > Thanks For Your Time > Antonio > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 !DSPAM:5114d19532766387414707! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130208/9b4a143a/attachment.html From eagle.antonio at gmail.com Fri Feb 8 15:16:41 2013 From: eagle.antonio at gmail.com (Antonio Teixeira) Date: Fri, 08 Feb 2013 12:16:41 +0000 Subject: [Freeswitch-users] Sorry To Bump but .. In-Reply-To: <1FFF97C269757C458224B7C895F35F1521B0ED@cantor.std.visionutv.se> References: <1FFF97C269757C458224B7C895F35F1521B0ED@cantor.std.visionutv.se> Message-ID: <5114ECA9.2040700@gmail.com> Hi Peter Package ncurses-devel-5.7-3.20090208.el6.x86_64 already installed and latest version Thanks A/t On 2/8/13 12:06 PM, Peter Olsson wrote: > > Install ncurses-devel package. > > /Peter > > *Fr?n:*freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] *F?r *Antonio > Teixeira > *Skickat:* den 8 februari 2013 11:34 > *Till:* freeswitch-users at lists.freeswitch.org > *?mne:* Re: [Freeswitch-users] Sorry To Bump but .. > > Hello Muhammad. > > Still no go , > > > http://pastebin.freeswitch.org/20558 > > Tried > yum groupinstall 'development tools' <- Works ... > And also > yum install libxml2-devel libncurses-devel postgresql-devel > sqlite-devel curl-devel openssl-devel <- Works > And also > yum install libedit-devel <- just in case > > With every make attempt a clean build dir ( i mean rm-rf clean ..) > > Going to make a new VM ... :( > > Thanks for the help > Antonio > > On 2/8/13 8:53 AM, Muhammad Shahzad wrote: > > I haven't used centos since version 5.4. However following should > fix libstdc++ problem that you seem to have with building libedit. > > yum groupinstall 'development tools' > yum groupinstall 'development libraries' > > It would be better to use a freshly installed machine if possible > for you. After doing group install check if following libs were > successfully installed (only for double check), > > yum install libxml2-devel libncurses-devel postgresql-devel > sqlite-devel curl-devel openssl-devel > > Note: some lib names might be different. Try with or without lib > prefix. > > Thank you. > > On 8 Feb 2013 09:09, "Antonio Teixeira" > wrote: > > > > Can anyone take a look at this ? > > > > http://jira.freeswitch.org/browse/FS-5069 > > > > I have performed dozens of make's the last few days and still i have > > been unable to get this to work... > > > > Thanks For Your Time > > Antonio > > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > 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 > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > 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 > > > !DSPAM:5114d19532766387414707! > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130208/2660f293/attachment-0001.html From p.varghese at cem-solutions.net Fri Feb 8 15:21:52 2013 From: p.varghese at cem-solutions.net (Varghese) Date: Fri, 08 Feb 2013 17:51:52 +0530 Subject: [Freeswitch-users] Reg. Freeswitch Performace on Freescale Board In-Reply-To: <51121A98.6070109@cem-solutions.net> References: <5110FD13.3080406@cem-solutions.net> <51121A98.6070109@cem-solutions.net> Message-ID: <5114EDE0.6020400@cem-solutions.net> Hi all, Any update on this issue. Any other options we need to try with freeswitch. E1 PRI with freetdm freeswitch in freescale and cpu load is going to 100 with 15 calls. Thanks And Regards Varghese Paul > Hi Anthony, > > Thanks for the information. > > Kernel is Linux-2.6.32 patched by Freescale team for the hardware. > > Further we tested the freeswitch-1.2.6 on Freescale and following are > observations. > > We set the ulimit options as per the wiki link and ulimit ?s 240 > > 1. With switches *-np* and *?lp* the result is same i.e. maximum of 90 > calls and top command shows 100% CPU for freeswitch process > > 2. With RTP Bypass mode we could make 250 calls and CPU usage is 20% > > 3. With Proxy media mode, and default rtp mode the CPU is very high. > Approximately 1 to 1.5% per call and scales up to 90 calls > > 4. Set the option enable-timer value = false in SIP profile, > stun-enabled=false given all IP address in ext-rtp, > rtp-timer-name=none/soft etc., still the results are same > > Q1) Can you explain ?asterisk also uses blocking reads in its rtp > stack where we have timers. You can disable the timers as documented > in the wiki? which parameter you are referring to and where we need to > disable timers. > > Q2) Do we need to try any other options to increase the performance > numbers? > > F.Y.I, We used Empirix Hammer SIP Call Generator for testing SIP calls. > > Thanks > Varghese Paul > > Anthony Minessale wrote: >> That depends on the version of linux, the kernel version and several >> other factors. >> >> Were you actually moving media when testing, what tool are you >> testing with? >> >> On 32 bit you should make sure you have the stack size at 240 (ulimit >> -s 240 before starting) >> Also you may need to add the -lp or -np startup flags to reduce the >> realtime threads. >> >> asterisk also uses blocking reads in its rtp stack where we have >> timers. You can disable the timers as documented in the wiki.. >> >> >> >> >> On Tue, Feb 5, 2013 at 6:37 AM, Varghese >> > >> wrote: >> >> Hi all, >> We are facing following observations/issues with freeswitch on >> freescale >> processor boards. >> We ported freeswitch with freescale provided tool chain. >> _*Freeswitch Configuration: version 1.2.6 (git version)*_ >> 1. Only enabled mod_sofia, mod_dialplan, mod_command,mod_dptool and >> mod_console. >> 2. set the ulimit options and followed the steps in the following >> link >> >> http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations >> 3. SIP test tool run the load test for 500 users in freeswitch and >> configured the dial plan to just bridge the channels >> 4. Freeswitch media processing is in default mode: i.e media will go >> through freeswitch >> _*Freescale Board Configuration:*_ >> CPU : 1GHZ PowerQUICC III, 32 bit >> RAM: 1 GB >> Linux- 2.6.32 >> Load test is conducted for SIP to SIP calls with RTP. >> _*Observations:*_ >> Linux TOP command shows increasing CPU usage per call and >> increases to >> 100% with only 90 calls. After that system response is very slow >> We could also found delay in media. >> >> _Surprisingly, with Asterisk 1.8 the same system configuration >> works 250 >> calls without any problems. >> _ >> _*Questions: >> *_1. Any one ported Freeswitch on Freescale or any embedded >> processors? >> Any performance figures or references will be appreciated >> 2. How many simultaneous calls can be possible with 1GHz >> PowerQUICC III >> processor ? >> 3. Any more settings are required in freeswitch or linux for >> scalability? >> >> Thanks And Regards >> >> Varghese Paul >> >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> >> googletalk:conf+888 at conference.freeswitch.org >> >> pstn:+19193869900 >> ------------------------------------------------------------------------ >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 POlsson at enghouse.com Fri Feb 8 15:29:50 2013 From: POlsson at enghouse.com (Peter Olsson) Date: Fri, 8 Feb 2013 12:29:50 +0000 Subject: [Freeswitch-users] Sorry To Bump but .. Message-ID: <1FFF97C269757C458224B7C895F35F1521B1AA@cantor.std.visionutv.se> Try to bootstrap.sh and configure again - just in case. I don't know really, I've never had any problems building on CentOS 6. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Antonio Teixeira Skickat: den 8 februari 2013 13:17 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] Sorry To Bump but .. Hi Peter Package ncurses-devel-5.7-3.20090208.el6.x86_64 already installed and latest version Thanks A/t On 2/8/13 12:06 PM, Peter Olsson wrote: Install ncurses-devel package. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Antonio Teixeira Skickat: den 8 februari 2013 11:34 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] Sorry To Bump but .. Hello Muhammad. Still no go , http://pastebin.freeswitch.org/20558 Tried yum groupinstall 'development tools' <- Works ... And also yum install libxml2-devel libncurses-devel postgresql-devel sqlite-devel curl-devel openssl-devel <- Works And also yum install libedit-devel <- just in case With every make attempt a clean build dir ( i mean rm-rf clean ..) Going to make a new VM ... :( Thanks for the help Antonio On 2/8/13 8:53 AM, Muhammad Shahzad wrote: I haven't used centos since version 5.4. However following should fix libstdc++ problem that you seem to have with building libedit. yum groupinstall 'development tools' yum groupinstall 'development libraries' It would be better to use a freshly installed machine if possible for you. After doing group install check if following libs were successfully installed (only for double check), yum install libxml2-devel libncurses-devel postgresql-devel sqlite-devel curl-devel openssl-devel Note: some lib names might be different. Try with or without lib prefix. Thank you. On 8 Feb 2013 09:09, "Antonio Teixeira" > wrote: > > Can anyone take a look at this ? > > http://jira.freeswitch.org/browse/FS-5069 > > I have performed dozens of make's the last few days and still i have > been unable to get this to work... > > Thanks For Your Time > Antonio > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 !DSPAM:5114e94632761618517767! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130208/4646991e/attachment.html From garmt.noname at gmail.com Fri Feb 8 16:01:46 2013 From: garmt.noname at gmail.com (grmt) Date: Fri, 8 Feb 2013 14:01:46 +0100 Subject: [Freeswitch-users] Sorry To Bump but .. In-Reply-To: <1FFF97C269757C458224B7C895F35F1521B1AA@cantor.std.visionutv.se> References: <1FFF97C269757C458224B7C895F35F1521B1AA@cantor.std.visionutv.se> Message-ID: <00de01ce05fc$742089b0$5c619d10$@gmail.com> You may have problems because of your LOCALE setting. Use en_US.utf8 From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peter Olsson Sent: Friday, February 08, 2013 13:30 To: 'FreeSWITCH Users Help' Subject: Re: [Freeswitch-users] Sorry To Bump but .. Try to bootstrap.sh and configure again ? just in case. I don?t know really, I?ve never had any problems building on CentOS 6. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Antonio Teixeira Skickat: den 8 februari 2013 13:17 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] Sorry To Bump but .. Hi Peter Package ncurses-devel-5.7-3.20090208.el6.x86_64 already installed and latest version Thanks A/t On 2/8/13 12:06 PM, Peter Olsson wrote: Install ncurses-devel package. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Antonio Teixeira Skickat: den 8 februari 2013 11:34 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] Sorry To Bump but .. Hello Muhammad. Still no go , http://pastebin.freeswitch.org/20558 Tried yum groupinstall 'development tools' <- Works ... And also yum install libxml2-devel libncurses-devel postgresql-devel sqlite-devel curl-devel openssl-devel <- Works And also yum install libedit-devel <- just in case With every make attempt a clean build dir ( i mean rm-rf clean ..) Going to make a new VM ... :( Thanks for the help Antonio On 2/8/13 8:53 AM, Muhammad Shahzad wrote: I haven't used centos since version 5.4. However following should fix libstdc++ problem that you seem to have with building libedit. yum groupinstall 'development tools' yum groupinstall 'development libraries' It would be better to use a freshly installed machine if possible for you. After doing group install check if following libs were successfully installed (only for double check), yum install libxml2-devel libncurses-devel postgresql-devel sqlite-devel curl-devel openssl-devel Note: some lib names might be different. Try with or without lib prefix. Thank you. On 8 Feb 2013 09:09, "Antonio Teixeira" wrote: > > Can anyone take a look at this ? > > http://jira.freeswitch.org/browse/FS-5069 > > I have performed dozens of make's the last few days and still i have > been unable to get this to work... > > Thanks For Your Time > Antonio > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 !DSPAM:5114e94632761618517767! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130208/6bc7b6be/attachment-0001.html From cal.leeming at simplicitymedialtd.co.uk Fri Feb 8 16:06:50 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Fri, 8 Feb 2013 13:06:50 +0000 Subject: [Freeswitch-users] mod-curl POST In-Reply-To: References: Message-ID: Sorry Rupa, I didn't read your email properly before hand and didn't realise there was already a ticket for this issue. Thanks Cal On Fri, Feb 8, 2013 at 1:49 AM, Rupa Schomaker wrote: > Umm.. I just explained how to do it. The only piece missing is the > setting of the content-type header, but if that isn't going to be a problem > you odn't have to do anything special. > > Just remember to urlencode the (full) contents of the XML. > > And try it out. Use wireshark to see what is sent over the wire to the > server. > > > On Thu, Feb 7, 2013 at 11:26 AM, JP wrote: > >> Yes what I am looking for is to POST xml data in the body and not as >> name/value pairs. Is this possible? >> >> Thanks, >> -JP >> >> On Wed, Feb 6, 2013 at 7:21 PM, Rupa Schomaker wrote: >> >>> >>> >>> >>> On Wed, Feb 6, 2013 at 6:56 PM, Cal Leeming [Simplicity Media Ltd] < >>> cal.leeming at simplicitymedialtd.co.uk> wrote: >>> >>>> The docs says this; >>>> >>>> curl url [headers|json] [get|head|post [url_encode_data]] >>>> >>> >>> The [headers|json] is all about the response format. Just to make sure >>> that is clear. >>> >>> get/head/post should be self-evident >>> >>> url_encode_data is supposed to be the url encoded version of what you >>> want to send to the server. It is encoded on purpose (so that we don't >>> have to worry about embedded spaces or other special chars). >>> >>> So, you really just need to do: >>> >>> curl http://example.com/ post encodedxmlhere >>> >>> >>>> So, I assume you'd just do. >>>> >>>> curl http://example.com/ post xml=encodedxmlhere >>>> >>>> >>> This will result in www-url-encoded with key/value pair with xml as key >>> and the encoded xml as the value. This may be what the OP wants, but I >>> think what he wants is just XML. >>> >>> >>> Note that the content-type will still be application/x-www-form-urlencoded >>> which is "wrong". I've written xml service endpoints that behave >>> differently based on the content-type. Depending on how your service is >>> written this may be ok or not. There is an open ticket to address the >>> content-type issue. >>> >>> http://jira.freeswitch.org/browse/FS-4714 >>> >>> >>> >>>> Taken from here; >>>> http://wiki.freeswitch.org/wiki/Mod_curl >>>> >>>> Hope this helps, apologies if I misunderstood the question >>>> >>>> Cal >>>> >>>> >>> -- >>> -Rupa >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > > -- > -Rupa > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130208/b739782b/attachment.html From shaheryarkh at gmail.com Fri Feb 8 16:07:38 2013 From: shaheryarkh at gmail.com (Muhammad Shahzad) Date: Fri, 8 Feb 2013 14:07:38 +0100 Subject: [Freeswitch-users] Sorry To Bump but .. In-Reply-To: <1FFF97C269757C458224B7C895F35F1521B1AA@cantor.std.visionutv.se> References: <1FFF97C269757C458224B7C895F35F1521B1AA@cantor.std.visionutv.se> Message-ID: OK, i have successfully build FS on CentOS 6.3, the only changes i observed are, 1. They have changed "development libraries" group to "additional development", so i had to do, yum groupinstall "additional development" 2. I have to manually install zlib-devel and libjpeg-devel packages, which i think should already come with development groups ("development tools" and/or "additional development"). 3. This may not be relevant but I always do "yum update" on every fresh CentOS installation. I download latest copy of FS from GIT and build with default settings (./bootstrap; ./configure; make all). Thank you. On Fri, Feb 8, 2013 at 1:29 PM, Peter Olsson wrote: > Try to bootstrap.sh and configure again ? just in case. I don?t know > really, I?ve never had any problems building on CentOS 6.**** > > ** ** > > /Peter**** > > ** ** > > ** ** > > *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *F?r *Antonio Teixeira > *Skickat:* den 8 februari 2013 13:17 > > *Till:* freeswitch-users at lists.freeswitch.org > *?mne:* Re: [Freeswitch-users] Sorry To Bump but ..**** > > ** ** > > Hi Peter > > Package ncurses-devel-5.7-3.20090208.el6.x86_64 already installed and > latest version > > Thanks > A/t > On 2/8/13 12:06 PM, Peter Olsson wrote:**** > > Install ncurses-devel package.**** > > **** > > /Peter**** > > **** > > *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [ > mailto:freeswitch-users-bounces at lists.freeswitch.org] > *F?r *Antonio Teixeira > *Skickat:* den 8 februari 2013 11:34 > *Till:* freeswitch-users at lists.freeswitch.org > *?mne:* Re: [Freeswitch-users] Sorry To Bump but ..**** > > **** > > Hello Muhammad. > > Still no go , > > > http://pastebin.freeswitch.org/20558 > > Tried > yum groupinstall 'development tools' <- Works ... > And also > yum install libxml2-devel libncurses-devel postgresql-devel sqlite-devel > curl-devel openssl-devel <- Works > And also > yum install libedit-devel <- just in case > > With every make attempt a clean build dir ( i mean rm-rf clean ..) > > Going to make a new VM ... :( > > Thanks for the help > Antonio > > On 2/8/13 8:53 AM, Muhammad Shahzad wrote:**** > > I haven't used centos since version 5.4. However following should fix > libstdc++ problem that you seem to have with building libedit.**** > > yum groupinstall 'development tools' > yum groupinstall 'development libraries'**** > > It would be better to use a freshly installed machine if possible for you. > After doing group install check if following libs were successfully > installed (only for double check),**** > > yum install libxml2-devel libncurses-devel postgresql-devel sqlite-devel > curl-devel openssl-devel**** > > Note: some lib names might be different. Try with or without lib prefix.** > ** > > Thank you. > > > **** > > On 8 Feb 2013 09:09, "Antonio Teixeira" wrote: > > > > Can anyone take a look at this ? > > > > http://jira.freeswitch.org/browse/FS-5069 > > > > I have performed dozens of make's the last few days and still i have > > been unable to get this to work... > > > > Thanks For Your Time > > Antonio > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > 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**** > > > > > > **** > > _________________________________________________________________________**** > > Professional FreeSWITCH Consulting Services:**** > > consulting at freeswitch.org**** > > http://www.freeswitchsolutions.com**** > > **** > > **** > > **** > > **** > > Official FreeSWITCH Sites**** > > http://www.freeswitch.org**** > > http://wiki.freeswitch.org**** > > http://www.cluecon.com**** > > **** > > 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**** > > > > **** > > > > > **** > > _________________________________________________________________________**** > > Professional FreeSWITCH Consulting Services:**** > > consulting at freeswitch.org**** > > http://www.freeswitchsolutions.com**** > > ** ** > > **** > > **** > > ** ** > > Official FreeSWITCH Sites**** > > http://www.freeswitch.org**** > > http://wiki.freeswitch.org**** > > http://www.cluecon.com**** > > ** ** > > 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**** > > > !DSPAM:5114e94632761618517767! **** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +49 176 99 83 10 85 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130208/c17dcf8f/attachment-0001.html From vipkilla at gmail.com Fri Feb 8 16:48:27 2013 From: vipkilla at gmail.com (Vik Killa) Date: Fri, 8 Feb 2013 08:48:27 -0500 Subject: [Freeswitch-users] Can local_stream MOH have a time limit/max duration? In-Reply-To: References: <219BCB63-3A5A-419F-B374-BDD45FDE3FAB@mgtech.com> Message-ID: I don't quite understand what you are trying to do here. mod_local_stream plays audio in a loop constantly with chime-ins (announcements) set to play periodically. If you need something to play from the beginning for a certain amount of time, you need to use something other than mod_local_stream. > On Wed, Feb 6, 2013 at 9:04 AM, Mario G wrote: >> >> Could not find anything in the wiki and this may not be possible but... I >> have early media with instructions, is has voice, ring, voice, music. I want >> to change the ring to music but I could not find a way to limit the time the >> music plays. See the second example, I would like to limit the first moh to >> a maximum time. If not possible, is this a "bounty" enhancement? Thanks for >> any help. >> Mario G >> >> >> Currently: >> > data="ringback=file_string://${lua_ring1}!tone_stream://${us-ring};loops=3!file_string://${lua_ring2}!local_stream://moh"/> >> >> Would Like (add something to limit the first moh): >> > data="ringback=file_string://${lua_ring1}!local_stream://moh!file_string://${lua_ring2}!local_stream://moh"/> >> >> >> ______________________________________________________________________ From dar at helia.ca Fri Feb 8 12:24:23 2013 From: dar at helia.ca (Dar Zuch) Date: Fri, 8 Feb 2013 02:24:23 -0700 Subject: [Freeswitch-users] mod_managed on OSX 10.8 fails to load on FreeSWITCH 1.3.13 Message-ID: I'm getting started on mod_managed with Freeswitch. I'm using OSX for dev but will be deploying on debian. I'm using FreeSWITCH version: FreeSWITCH Version 1.3.13b+git~20130205T233143Z~f5151691c7 (git f515169 2013-02-05 23:31:43Z) When I type: mono -V I get: Mono JIT compiler version 2.10.9 (tarball Mon May 7 20:25:51 EDT 2012) Copyright (C) 2002-2011 Novell, Inc, Xamarin, Inc and Contributors. www.mono-project.com TLS: normal SIGSEGV: normal Notification: kqueue Architecture: x86 Disabled: none Misc: debugger softdebug LLVM: yes(2.9svn-mono) GC: Included Boehm (with typed GC) I can start FreeSWITCH successfully but when when I do a: load mod_managed I get: -ERR [module load file routine returned an error] 2013-02-08 01:39:09.905374 [CRIT] switch_loadable_module.c:1330 Error Loading module /usr/local/freeswitch/mod/mod_managed.so **dlopen(/usr/local/freeswitch/mod/mod_managed.so, 6): Symbol not found: _mono_assembly_get_image Referenced from: /usr/local/freeswitch/mod/mod_managed.so Expected in: flat namespace in /usr/local/freeswitch/mod/mod_managed.so** both mod_managed.la and mod_managed.so exist at /usr/local/freeswitch/mod/ I'm not sure what this means or where to start looking. Thanks for the help. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130208/4fa57c33/attachment.html From krice at freeswitch.org Fri Feb 8 17:59:15 2013 From: krice at freeswitch.org (Ken Rice) Date: Fri, 8 Feb 2013 08:59:15 -0600 Subject: [Freeswitch-users] Sorry To Bump but .. In-Reply-To: References: <5114B1E6.7090402@gmail.com> <5114D483.5060506@gmail.com> Message-ID: you failed the pastebin antispam test... the login information is in the popup window asking you for the login Ken Sent from my iPad On Feb 8, 2013, at 5:55, Muhammad Shahzad wrote: > can't login to paste bin. Anyhow, let me try centos 6.3 myself and see if i can reproduce the issue. > > Thank you. > > > On Fri, Feb 8, 2013 at 11:33 AM, Antonio Teixeira wrote: >> Hello Muhammad. >> >> Still no go , >> >> >> http://pastebin.freeswitch.org/20558 >> >> Tried >> yum groupinstall 'development tools' <- Works ... >> And also >> yum install libxml2-devel libncurses-devel postgresql-devel sqlite-devel curl-devel openssl-devel <- Works >> And also >> yum install libedit-devel <- just in case >> >> With every make attempt a clean build dir ( i mean rm-rf clean ..) >> >> Going to make a new VM ... :( >> >> Thanks for the help >> Antonio >> >> >> On 2/8/13 8:53 AM, Muhammad Shahzad wrote: >>> I haven't used centos since version 5.4. However following should fix libstdc++ problem that you seem to have with building libedit. >>> >>> yum groupinstall 'development tools' >>> yum groupinstall 'development libraries' >>> >>> It would be better to use a freshly installed machine if possible for you. After doing group install check if following libs were successfully installed (only for double check), >>> >>> yum install libxml2-devel libncurses-devel postgresql-devel sqlite-devel curl-devel openssl-devel >>> >>> Note: some lib names might be different. Try with or without lib prefix. >>> >>> Thank you. >>> >>> >>> On 8 Feb 2013 09:09, "Antonio Teixeira" wrote: >>> > >>> > Can anyone take a look at this ? >>> > >>> > http://jira.freeswitch.org/browse/FS-5069 >>> > >>> > I have performed dozens of make's the last few days and still i have >>> > been unable to get this to work... >>> > >>> > Thanks For Your Time >>> > Antonio >>> > >>> > >>> > >>> > _________________________________________________________________________ >>> > Professional FreeSWITCH Consulting Services: >>> > consulting at freeswitch.org >>> > http://www.freeswitchsolutions.com >>> > >>> > >>> > >>> > >>> > Official FreeSWITCH Sites >>> > http://www.freeswitch.org >>> > http://wiki.freeswitch.org >>> > http://www.cluecon.com >>> > >>> > 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 >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 > > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +49 176 99 83 10 85 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130208/c872f8e1/attachment.html From krice at freeswitch.org Fri Feb 8 18:03:42 2013 From: krice at freeswitch.org (Ken Rice) Date: Fri, 8 Feb 2013 09:03:42 -0600 Subject: [Freeswitch-users] Sorry To Bump but .. In-Reply-To: References: <1FFF97C269757C458224B7C895F35F1521B1AA@cantor.std.visionutv.se> Message-ID: see http://wiki.freeswitch.org/wiki/Installation_Guide#CentOS this is a list of required rpms, everything else on centos 6 should install by dependancy from there. once you have that list of rpms installed, start from clean source tree and bootstrap, configure and make install as per the instrucions on that same page on the wiki... if you are still not building on centos 6 you have some other error in you build enviornment... centos 5 and 6 are heavily used for running and devloping freeswith Ken Sent from my iPad On Feb 8, 2013, at 7:07, Muhammad Shahzad wrote: > OK, i have successfully build FS on CentOS 6.3, the only changes i observed are, > > 1. They have changed "development libraries" group to "additional development", so i had to do, > > yum groupinstall "additional development" > > 2. I have to manually install zlib-devel and libjpeg-devel packages, which i think should already come with development groups ("development tools" and/or "additional development"). > > 3. This may not be relevant but I always do "yum update" on every fresh CentOS installation. > > I download latest copy of FS from GIT and build with default settings (./bootstrap; ./configure; make all). > > Thank you. > > > On Fri, Feb 8, 2013 at 1:29 PM, Peter Olsson wrote: >> Try to bootstrap.sh and configure again ? just in case. I don?t know really, I?ve never had any problems building on CentOS 6. >> >> >> >> /Peter >> >> >> >> >> >> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Antonio Teixeira >> Skickat: den 8 februari 2013 13:17 >> >> >> Till: freeswitch-users at lists.freeswitch.org >> ?mne: Re: [Freeswitch-users] Sorry To Bump but .. >> >> >> Hi Peter >> >> Package ncurses-devel-5.7-3.20090208.el6.x86_64 already installed and latest version >> >> Thanks >> A/t >> On 2/8/13 12:06 PM, Peter Olsson wrote: >> >> Install ncurses-devel package. >> >> >> >> /Peter >> >> >> >> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Antonio Teixeira >> Skickat: den 8 februari 2013 11:34 >> Till: freeswitch-users at lists.freeswitch.org >> ?mne: Re: [Freeswitch-users] Sorry To Bump but .. >> >> >> >> Hello Muhammad. >> >> Still no go , >> >> >> http://pastebin.freeswitch.org/20558 >> >> Tried >> yum groupinstall 'development tools' <- Works ... >> And also >> yum install libxml2-devel libncurses-devel postgresql-devel sqlite-devel curl-devel openssl-devel <- Works >> And also >> yum install libedit-devel <- just in case >> >> With every make attempt a clean build dir ( i mean rm-rf clean ..) >> >> Going to make a new VM ... :( >> >> Thanks for the help >> Antonio >> >> On 2/8/13 8:53 AM, Muhammad Shahzad wrote: >> >> I haven't used centos since version 5.4. However following should fix libstdc++ problem that you seem to have with building libedit. >> >> yum groupinstall 'development tools' >> yum groupinstall 'development libraries' >> >> It would be better to use a freshly installed machine if possible for you. After doing group install check if following libs were successfully installed (only for double check), >> >> yum install libxml2-devel libncurses-devel postgresql-devel sqlite-devel curl-devel openssl-devel >> >> Note: some lib names might be different. Try with or without lib prefix. >> >> Thank you. >> >> >> >> On 8 Feb 2013 09:09, "Antonio Teixeira" wrote: >> > >> > Can anyone take a look at this ? >> > >> > http://jira.freeswitch.org/browse/FS-5069 >> > >> > I have performed dozens of make's the last few days and still i have >> > been unable to get this to work... >> > >> > Thanks For Your Time >> > Antonio >> > >> > >> > >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > 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 >> >> >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> >> >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> !DSPAM:5114e94632761618517767! >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 > > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +49 176 99 83 10 85 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130208/7ab87993/attachment-0001.html From eagle.antonio at gmail.com Fri Feb 8 18:04:48 2013 From: eagle.antonio at gmail.com (Antonio Teixeira) Date: Fri, 08 Feb 2013 15:04:48 +0000 Subject: [Freeswitch-users] Sorry To Bump but .. In-Reply-To: References: <1FFF97C269757C458224B7C895F35F1521B1AA@cantor.std.visionutv.se> Message-ID: <51151410.8030801@gmail.com> Hello Guys Thanks for all the help i was able to nail it : 1) To be Able To Compile : yum groupinstall "Development tools" yum groupinstall "Additional Development" 2) Plus All the Dependencis in The Installation Page plus zlib-devel , libjpeg-devel 3) I was able to compile everything both the Normal version and the Postgres Version. 4) FS Fails With An Error invalid connection option "application_name" when running with Postgres , just remove the application_name option. Everything looks fine i will try it again on a clean machine to see how it goes and try to clean the excessive dependencies and try to ask someone from staff to update the installation page. Thank you all that provided your input to try and solve this. Antonio On 2/8/13 1:07 PM, Muhammad Shahzad wrote: > OK, i have successfully build FS on CentOS 6.3, the only changes i > observed are, > > 1. They have changed "development libraries" group to "additional > development", so i had to do, > > yum groupinstall "additional development" > > 2. I have to manually install zlib-devel and libjpeg-devel packages, > which i think should already come with development groups > ("development tools" and/or "additional development"). > > 3. This may not be relevant but I always do "yum update" on every > fresh CentOS installation. > > I download latest copy of FS from GIT and build with default settings > (./bootstrap; ./configure; make all). > > Thank you. > > > On Fri, Feb 8, 2013 at 1:29 PM, Peter Olsson > wrote: > > Try to bootstrap.sh and configure again -- just in case. I don't > know really, I've never had any problems building on CentOS 6. > > /Peter > > *Fr?n:*freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] *F?r > *Antonio Teixeira > *Skickat:* den 8 februari 2013 13:17 > > > *Till:* freeswitch-users at lists.freeswitch.org > > *?mne:* Re: [Freeswitch-users] Sorry To Bump but .. > > Hi Peter > > Package ncurses-devel-5.7-3.20090208.el6.x86_64 already installed > and latest version > > Thanks > A/t > On 2/8/13 12:06 PM, Peter Olsson wrote: > > Install ncurses-devel package. > > /Peter > > *Fr?n:*freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] *F?r > *Antonio Teixeira > *Skickat:* den 8 februari 2013 11:34 > *Till:* freeswitch-users at lists.freeswitch.org > > *?mne:* Re: [Freeswitch-users] Sorry To Bump but .. > > Hello Muhammad. > > Still no go , > > > http://pastebin.freeswitch.org/20558 > > Tried > yum groupinstall 'development tools' <- Works ... > And also > yum install libxml2-devel libncurses-devel postgresql-devel > sqlite-devel curl-devel openssl-devel <- Works > And also > yum install libedit-devel <- just in case > > With every make attempt a clean build dir ( i mean rm-rf clean ..) > > Going to make a new VM ... :( > > Thanks for the help > Antonio > > On 2/8/13 8:53 AM, Muhammad Shahzad wrote: > > I haven't used centos since version 5.4. However following > should fix libstdc++ problem that you seem to have with > building libedit. > > yum groupinstall 'development tools' > yum groupinstall 'development libraries' > > It would be better to use a freshly installed machine if > possible for you. After doing group install check if > following libs were successfully installed (only for > double check), > > yum install libxml2-devel libncurses-devel > postgresql-devel sqlite-devel curl-devel openssl-devel > > Note: some lib names might be different. Try with or > without lib prefix. > > Thank you. > > > On 8 Feb 2013 09:09, "Antonio Teixeira" > > > wrote: > > > > Can anyone take a look at this ? > > > > http://jira.freeswitch.org/browse/FS-5069 > > > > I have performed dozens of make's the last few days and > still i have > > been unable to get this to work... > > > > Thanks For Your Time > > Antonio > > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > 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 > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > 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 > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > 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 > > > !DSPAM:5114e94632761618517767! > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +49 176 99 83 10 85 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130208/6e7c15cf/attachment-0001.html From eagle.antonio at gmail.com Fri Feb 8 18:13:45 2013 From: eagle.antonio at gmail.com (Antonio Teixeira) Date: Fri, 08 Feb 2013 15:13:45 +0000 Subject: [Freeswitch-users] Sorry To Bump but .. In-Reply-To: References: <1FFF97C269757C458224B7C895F35F1521B1AA@cantor.std.visionutv.se> Message-ID: <51151629.8000000@gmail.com> Ken , i have been trying to setup FS on CentOS6.3 for a week and believe me i tried everything on that page and was only able to get it working with Muhammad advise. I will dig a little further on this to see how i can get a simpler working solution. Thanks On 2/8/13 3:03 PM, Ken Rice wrote: > see http://wiki.freeswitch.org/wiki/Installation_Guide#CentOS this is > a list of required rpms, everything else on centos 6 should install by > dependancy from there. once you have that list of rpms installed, > start from clean source tree and bootstrap, configure and make install > as per the instrucions on that same page on the wiki... if you are > still not building on centos 6 you have some other error in you build > enviornment... centos 5 and 6 are heavily used for running and > devloping freeswith > > Ken > Sent from my iPad > > On Feb 8, 2013, at 7:07, Muhammad Shahzad > wrote: > >> OK, i have successfully build FS on CentOS 6.3, the only changes i >> observed are, >> >> 1. They have changed "development libraries" group to "additional >> development", so i had to do, >> >> yum groupinstall "additional development" >> >> 2. I have to manually install zlib-devel and libjpeg-devel packages, >> which i think should already come with development groups >> ("development tools" and/or "additional development"). >> >> 3. This may not be relevant but I always do "yum update" on every >> fresh CentOS installation. >> >> I download latest copy of FS from GIT and build with default settings >> (./bootstrap; ./configure; make all). >> >> Thank you. >> >> >> On Fri, Feb 8, 2013 at 1:29 PM, Peter Olsson > > wrote: >> >> Try to bootstrap.sh and configure again -- just in case. I don't >> know really, I've never had any problems building on CentOS 6. >> >> /Peter >> >> *Fr?n:*freeswitch-users-bounces at lists.freeswitch.org >> >> [mailto:freeswitch-users-bounces at lists.freeswitch.org >> ] *F?r >> *Antonio Teixeira >> *Skickat:* den 8 februari 2013 13:17 >> >> >> *Till:* freeswitch-users at lists.freeswitch.org >> >> *?mne:* Re: [Freeswitch-users] Sorry To Bump but .. >> >> Hi Peter >> >> Package ncurses-devel-5.7-3.20090208.el6.x86_64 already installed >> and latest version >> >> Thanks >> A/t >> On 2/8/13 12:06 PM, Peter Olsson wrote: >> >> Install ncurses-devel package. >> >> /Peter >> >> *Fr?n:*freeswitch-users-bounces at lists.freeswitch.org >> >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] *F?r >> *Antonio Teixeira >> *Skickat:* den 8 februari 2013 11:34 >> *Till:* freeswitch-users at lists.freeswitch.org >> >> *?mne:* Re: [Freeswitch-users] Sorry To Bump but .. >> >> Hello Muhammad. >> >> Still no go , >> >> >> http://pastebin.freeswitch.org/20558 >> >> Tried >> yum groupinstall 'development tools' <- Works ... >> And also >> yum install libxml2-devel libncurses-devel postgresql-devel >> sqlite-devel curl-devel openssl-devel <- Works >> And also >> yum install libedit-devel <- just in case >> >> With every make attempt a clean build dir ( i mean rm-rf >> clean ..) >> >> Going to make a new VM ... :( >> >> Thanks for the help >> Antonio >> >> On 2/8/13 8:53 AM, Muhammad Shahzad wrote: >> >> I haven't used centos since version 5.4. However >> following should fix libstdc++ problem that you seem to >> have with building libedit. >> >> yum groupinstall 'development tools' >> yum groupinstall 'development libraries' >> >> It would be better to use a freshly installed machine if >> possible for you. After doing group install check if >> following libs were successfully installed (only for >> double check), >> >> yum install libxml2-devel libncurses-devel >> postgresql-devel sqlite-devel curl-devel openssl-devel >> >> Note: some lib names might be different. Try with or >> without lib prefix. >> >> Thank you. >> >> >> On 8 Feb 2013 09:09, "Antonio Teixeira" >> > > wrote: >> > >> > Can anyone take a look at this ? >> > >> > http://jira.freeswitch.org/browse/FS-5069 >> > >> > I have performed dozens of make's the last few days and >> still i have >> > been unable to get this to work... >> > >> > Thanks For Your Time >> > Antonio >> > >> > >> > >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > 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 >> >> >> >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://wiki.freeswitch.org >> >> http://www.cluecon.com >> >> >> >> 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 >> >> >> >> >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://wiki.freeswitch.org >> >> http://www.cluecon.com >> >> >> >> 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 >> >> >> !DSPAM:5114e94632761618517767! >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> >> >> >> -- >> Muhammad Shahzad >> ----------------------------------- >> CISCO Rich Media Communication Specialist (CRMCS) >> CISCO Certified Network Associate (CCNA) >> Cell: +49 176 99 83 10 85 >> MSN: shari_786pk at hotmail.com >> Email: shaheryarkh at googlemail.com >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130208/8f824851/attachment-0001.html From sertys at gmail.com Fri Feb 8 18:35:58 2013 From: sertys at gmail.com (Daniel Ivanov) Date: Fri, 8 Feb 2013 16:35:58 +0100 Subject: [Freeswitch-users] Real certificate in freeswitch In-Reply-To: References: Message-ID: But of course we have. Freeswitch accepts certificate bundles. So be sure to include your whole certificate chain in a concatenated file. Like ca.crt + intermediate (if any) + your.crt. and to be sure you've got the right combo use some of the online ssl checkers and point the to your ip/ domain and port. They just make a regular tcp session and show you the presented certificate chain. Good luck. On Feb 7, 2013 9:40 PM, "Carlos Rojas" wrote: Hi Everybody I am new with freeswitch, I was trying, put in freeswithc my real certificate, in my freeswitch, but doesn't work. looks like freeswitch accepts the certificate, but when the phone tried register doesn't work. Anybody has configured a real certificate with freeswitch? I hope somebody could help me. Kind Regards _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130208/e9777f6a/attachment.html From carlos.voiperu at gmail.com Fri Feb 8 18:41:43 2013 From: carlos.voiperu at gmail.com (Carlos Rojas) Date: Fri, 8 Feb 2013 10:41:43 -0500 Subject: [Freeswitch-users] Real certificate in freeswitch In-Reply-To: References: Message-ID: Hi Thanks a lot, I will try it, Thank you for you help Carlos Rojas On Fri, Feb 8, 2013 at 5:41 AM, Daniel Ivanov wrote: > But of course we have. Freeswitch accepts certificate bundles. So be sure > to include your whole certificate chain in a concatenated file. Like ca.crt > + intermediate (if any) + your.crt. and to be sure you've got the right > combo use some of the online ssl checkers and point the to your ip/ domain > and port. They just make a regular tcp session and show you the presented > certificate chain. > > Good luck. > On Feb 7, 2013 9:40 PM, "Carlos Rojas" wrote: > >> Hi Everybody >> >> I am new with freeswitch, I was trying, put in freeswithc my real >> certificate, in my freeswitch, but doesn't work. >> >> looks like freeswitch accepts the certificate, but when the phone tried >> register doesn't work. >> >> Anybody has configured a real certificate with freeswitch? >> >> >> I hope somebody could help me. >> >> >> Kind Regards >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130208/b67d276f/attachment.html From carlos.voiperu at gmail.com Fri Feb 8 18:44:18 2013 From: carlos.voiperu at gmail.com (Carlos Rojas) Date: Fri, 8 Feb 2013 10:44:18 -0500 Subject: [Freeswitch-users] Real certificate in freeswitch In-Reply-To: <24F5E90E-1B95-403E-AC3B-4D66BAB72FED@gmail.com> References: <24F5E90E-1B95-403E-AC3B-4D66BAB72FED@gmail.com> Message-ID: Yes I am trying with sip and TLS, I've got several problems, I'll try it, again, Thank you for you help On Fri, Feb 8, 2013 at 6:00 AM, Steven Ayre wrote: > If you're talking about SIP/TLS and a server certificate on FS then the > phone will also need the correct CA certificates installed which may not be > the case by default, and the correct time set so that it can verify them. > > Steve > > > > On 8 Feb 2013, at 10:41, Daniel Ivanov wrote: > > But of course we have. Freeswitch accepts certificate bundles. So be sure > to include your whole certificate chain in a concatenated file. Like ca.crt > + intermediate (if any) + your.crt. and to be sure you've got the right > combo use some of the online ssl checkers and point the to your ip/ domain > and port. They just make a regular tcp session and show you the presented > certificate chain. > > Good luck. > On Feb 7, 2013 9:40 PM, "Carlos Rojas" wrote: > >> Hi Everybody >> >> I am new with freeswitch, I was trying, put in freeswithc my real >> certificate, in my freeswitch, but doesn't work. >> >> looks like freeswitch accepts the certificate, but when the phone tried >> register doesn't work. >> >> Anybody has configured a real certificate with freeswitch? >> >> >> I hope somebody could help me. >> >> >> Kind Regards >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130208/6502204a/attachment.html From steveayre at gmail.com Fri Feb 8 19:32:31 2013 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 8 Feb 2013 16:32:31 +0000 Subject: [Freeswitch-users] Sorry To Bump but .. In-Reply-To: References: <5114B1E6.7090402@gmail.com> <5114D483.5060506@gmail.com> Message-ID: Read the message on the login popup. On 8 February 2013 11:55, Muhammad Shahzad wrote: > can't login to paste bin. Anyhow, let me try centos 6.3 myself and see if > i can reproduce the issue. > > Thank you. > > > On Fri, Feb 8, 2013 at 11:33 AM, Antonio Teixeira > wrote: > >> Hello Muhammad. >> >> Still no go , >> >> >> http://pastebin.freeswitch.org/20558 >> >> Tried >> yum groupinstall 'development tools' <- Works ... >> And also >> yum install libxml2-devel libncurses-devel postgresql-devel sqlite-devel >> curl-devel openssl-devel <- Works >> And also >> yum install libedit-devel <- just in case >> >> With every make attempt a clean build dir ( i mean rm-rf clean ..) >> >> Going to make a new VM ... :( >> >> Thanks for the help >> Antonio >> >> >> On 2/8/13 8:53 AM, Muhammad Shahzad wrote: >> >> I haven't used centos since version 5.4. However following should fix >> libstdc++ problem that you seem to have with building libedit. >> >> yum groupinstall 'development tools' >> yum groupinstall 'development libraries' >> >> It would be better to use a freshly installed machine if possible for >> you. After doing group install check if following libs were successfully >> installed (only for double check), >> >> yum install libxml2-devel libncurses-devel postgresql-devel sqlite-devel >> curl-devel openssl-devel >> >> Note: some lib names might be different. Try with or without lib prefix. >> >> Thank you. >> >> >> On 8 Feb 2013 09:09, "Antonio Teixeira" wrote: >> > >> > Can anyone take a look at this ? >> > >> > http://jira.freeswitch.org/browse/FS-5069 >> > >> > I have performed dozens of make's the last few days and still i have >> > been unable to get this to work... >> > >> > Thanks For Your Time >> > Antonio >> > >> > >> > >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > 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 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >> >> >> >> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com >> >> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +49 176 99 83 10 85 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130208/5ab2528b/attachment-0001.html From steveayre at gmail.com Fri Feb 8 19:34:54 2013 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 8 Feb 2013 16:34:54 +0000 Subject: [Freeswitch-users] Real certificate in freeswitch In-Reply-To: References: <24F5E90E-1B95-403E-AC3B-4D66BAB72FED@gmail.com> Message-ID: What phone is it, out of interest? Also try testing from a softphone to see if it works from elsewhere - it could be the phone at fault, not FS. Also see if the FS log shows anything useful. -Steve On 8 February 2013 15:44, Carlos Rojas wrote: > Yes > > I am trying with sip and TLS, I've got several problems, I'll try it, > again, > > > Thank you for you help > > On Fri, Feb 8, 2013 at 6:00 AM, Steven Ayre wrote: > >> If you're talking about SIP/TLS and a server certificate on FS then the >> phone will also need the correct CA certificates installed which may not be >> the case by default, and the correct time set so that it can verify them. >> >> Steve >> >> >> >> On 8 Feb 2013, at 10:41, Daniel Ivanov wrote: >> >> But of course we have. Freeswitch accepts certificate bundles. So be sure >> to include your whole certificate chain in a concatenated file. Like ca.crt >> + intermediate (if any) + your.crt. and to be sure you've got the right >> combo use some of the online ssl checkers and point the to your ip/ domain >> and port. They just make a regular tcp session and show you the presented >> certificate chain. >> >> Good luck. >> On Feb 7, 2013 9:40 PM, "Carlos Rojas" wrote: >> >>> Hi Everybody >>> >>> I am new with freeswitch, I was trying, put in freeswithc my real >>> certificate, in my freeswitch, but doesn't work. >>> >>> looks like freeswitch accepts the certificate, but when the phone tried >>> register doesn't work. >>> >>> Anybody has configured a real certificate with freeswitch? >>> >>> >>> I hope somebody could help me. >>> >>> >>> Kind Regards >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130208/4adf39b5/attachment.html From kris at kriskinc.com Fri Feb 8 19:50:08 2013 From: kris at kriskinc.com (Kristian Kielhofner) Date: Fri, 8 Feb 2013 11:50:08 -0500 Subject: [Freeswitch-users] Homer/sipcapture In-Reply-To: References: Message-ID: Jay, We did some pretty extensive performance testing with the built in Homer/HEP stuff in Kamailio and determined the call setup penalty was negligible. Not sure what that means for FreeSWITCH but it can be done well. Knowing FreeSWITCH it probably is. The bandwidth, however, is no joke. Our ops guy routinely complains about how much bandwidth we're using just for SIP capture. On Thu, Feb 7, 2013 at 6:56 PM, jay binks wrote: > I think the real question there is, how much extra cpu does the homer hep > code ( in freeswitch ) require > especially compared to tcpdump / libpcap solutions ? > -- Kristian Kielhofner From mario_fs at mgtech.com Fri Feb 8 19:52:12 2013 From: mario_fs at mgtech.com (Mario G) Date: Fri, 8 Feb 2013 08:52:12 -0800 Subject: [Freeswitch-users] Can local_stream MOH have a time limit/max duration? In-Reply-To: References: <219BCB63-3A5A-419F-B374-BDD45FDE3FAB@mgtech.com> Message-ID: I do use call_timeout, see below. I want to use moh twice in the ringback with an intervening message. So this means the first moh must be limited to a specific time. Here is what the caller hears for the ringback now: 1. Custom greeting to caller. 2. Ring for 45 seconds <--- I want to replace this with music instead, but limit moh duration to 45 seconds just here. 3. Custom 2nd greeting telling caller to please wait while call is forwarded. 4. moh for 20 seconds (csall_timeout completes if no answer) call times out 5. Custom please leave a message This has been working fine using the ringback below (I use LUA to build the customized greetings based on caller ID), it would just be nice to replace #2 with music but it needs to end so the ringback sequence can continue. Mario G On Feb 7, 2013, at 3:42 PM, Michael Collins wrote: > The ringback will play until the call is answered or until the call_timeout value is met and the call moves on in the dialplan. You can probably just use call_timeout to tell the call to move on after x number of seconds. > -MC > > On Wed, Feb 6, 2013 at 9:04 AM, Mario G wrote: > Could not find anything in the wiki and this may not be possible but... I have early media with instructions, is has voice, ring, voice, music. I want to change the ring to music but I could not find a way to limit the time the music plays. See the second example, I would like to limit the first moh to a maximum time. If not possible, is this a "bounty" enhancement? Thanks for any help. > Mario G > > > Currently: > > > Would Like (add something to limit the first moh): > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130208/a7447ca8/attachment-0001.html From mario_fs at mgtech.com Fri Feb 8 19:55:20 2013 From: mario_fs at mgtech.com (Mario G) Date: Fri, 8 Feb 2013 08:55:20 -0800 Subject: [Freeswitch-users] Can local_stream MOH have a time limit/max duration? In-Reply-To: References: <219BCB63-3A5A-419F-B374-BDD45FDE3FAB@mgtech.com> Message-ID: I looked into chime-ins, but it only chimes in at fixed intervals, and it was not flexible enough for true customization. I could not find anything else on the wiki that would play moh (or any other file) for a specific time. See my previous post for a details of how it works and what I wanted to do. Thanks. Mario G On Feb 8, 2013, at 5:48 AM, Vik Killa wrote: > I don't quite understand what you are trying to do here. > mod_local_stream plays audio in a loop constantly with chime-ins > (announcements) set to play periodically. If you need something to > play from the beginning for a certain amount of time, you need to use > something other than mod_local_stream. > > >> On Wed, Feb 6, 2013 at 9:04 AM, Mario G wrote: >>> >>> Could not find anything in the wiki and this may not be possible but... I >>> have early media with instructions, is has voice, ring, voice, music. I want >>> to change the ring to music but I could not find a way to limit the time the >>> music plays. See the second example, I would like to limit the first moh to >>> a maximum time. If not possible, is this a "bounty" enhancement? Thanks for >>> any help. >>> Mario G >>> >>> >>> Currently: >>> >> data="ringback=file_string://${lua_ring1}!tone_stream://${us-ring};loops=3!file_string://${lua_ring2}!local_stream://moh"/> >>> >>> Would Like (add something to limit the first moh): >>> >> data="ringback=file_string://${lua_ring1}!local_stream://moh!file_string://${lua_ring2}!local_stream://moh"/> >>> >>> >>> ______________________________________________________________________ > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 jpyle at fidelityvoice.com Fri Feb 8 20:06:40 2013 From: jpyle at fidelityvoice.com (Jeff Pyle) Date: Fri, 8 Feb 2013 12:06:40 -0500 Subject: [Freeswitch-users] Reg. Freeswitch Performace on Freescale Board In-Reply-To: <5114EDE0.6020400@cem-solutions.net> References: <5110FD13.3080406@cem-solutions.net> <51121A98.6070109@cem-solutions.net> <5114EDE0.6020400@cem-solutions.net> Message-ID: Hello, I believe the timers are disabled by setting in the sofia profile(s). The default configurations use a value of "soft". I have similar performance concerns bridging G711 SIP calls between two sofia profiles on an AMD Geode processor. With full media it occupies roughly 3.0% CPU per call. It scales to 25 calls with room for other basic system processes. Maybe 26 or 27 if we really pushed it and maintained the cps extremely low. Proxy media mode gives about a 15% performance increase in our case. Bypass mode was not tested because it isn't appropriate for our application. - Jeff On Fri, Feb 8, 2013 at 7:21 AM, Varghese wrote: > Hi all, > > Any update on this issue. Any other options we need to try with freeswitch. > > E1 PRI with freetdm freeswitch in freescale and cpu load is going to 100 > with 15 calls. > > Thanks And Regards > > Varghese Paul > > Hi Anthony, > > > > Thanks for the information. > > > > Kernel is Linux-2.6.32 patched by Freescale team for the hardware. > > > > Further we tested the freeswitch-1.2.6 on Freescale and following are > > observations. > > > > We set the ulimit options as per the wiki link and ulimit ?s 240 > > > > 1. With switches *-np* and *?lp* the result is same i.e. maximum of 90 > > calls and top command shows 100% CPU for freeswitch process > > > > 2. With RTP Bypass mode we could make 250 calls and CPU usage is 20% > > > > 3. With Proxy media mode, and default rtp mode the CPU is very high. > > Approximately 1 to 1.5% per call and scales up to 90 calls > > > > 4. Set the option enable-timer value = false in SIP profile, > > stun-enabled=false given all IP address in ext-rtp, > > rtp-timer-name=none/soft etc., still the results are same > > > > Q1) Can you explain ?asterisk also uses blocking reads in its rtp > > stack where we have timers. You can disable the timers as documented > > in the wiki? which parameter you are referring to and where we need to > > disable timers. > > > > Q2) Do we need to try any other options to increase the performance > > numbers? > > > > F.Y.I, We used Empirix Hammer SIP Call Generator for testing SIP calls. > > > > Thanks > > Varghese Paul > > > > Anthony Minessale wrote: > >> That depends on the version of linux, the kernel version and several > >> other factors. > >> > >> Were you actually moving media when testing, what tool are you > >> testing with? > >> > >> On 32 bit you should make sure you have the stack size at 240 (ulimit > >> -s 240 before starting) > >> Also you may need to add the -lp or -np startup flags to reduce the > >> realtime threads. > >> > >> asterisk also uses blocking reads in its rtp stack where we have > >> timers. You can disable the timers as documented in the wiki.. > >> > >> > >> > >> > >> On Tue, Feb 5, 2013 at 6:37 AM, Varghese > >> > > >> wrote: > >> > >> Hi all, > >> We are facing following observations/issues with freeswitch on > >> freescale > >> processor boards. > >> We ported freeswitch with freescale provided tool chain. > >> _*Freeswitch Configuration: version 1.2.6 (git version)*_ > >> 1. Only enabled mod_sofia, mod_dialplan, mod_command,mod_dptool and > >> mod_console. > >> 2. set the ulimit options and followed the steps in the following > >> link > >> > >> http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations > >> 3. SIP test tool run the load test for 500 users in freeswitch and > >> configured the dial plan to just bridge the channels > >> 4. Freeswitch media processing is in default mode: i.e media will go > >> through freeswitch > >> _*Freescale Board Configuration:*_ > >> CPU : 1GHZ PowerQUICC III, 32 bit > >> RAM: 1 GB > >> Linux- 2.6.32 > >> Load test is conducted for SIP to SIP calls with RTP. > >> _*Observations:*_ > >> Linux TOP command shows increasing CPU usage per call and > >> increases to > >> 100% with only 90 calls. After that system response is very slow > >> We could also found delay in media. > >> > >> _Surprisingly, with Asterisk 1.8 the same system configuration > >> works 250 > >> calls without any problems. > >> _ > >> _*Questions: > >> *_1. Any one ported Freeswitch on Freescale or any embedded > >> processors? > >> Any performance figures or references will be appreciated > >> 2. How many simultaneous calls can be possible with 1GHz > >> PowerQUICC III > >> processor ? > >> 3. Any more settings are required in freeswitch or linux for > >> scalability? > >> > >> Thanks And Regards > >> > >> Varghese Paul > >> > >> > >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130208/5e8416cc/attachment.html From shaheryarkh at gmail.com Fri Feb 8 20:10:45 2013 From: shaheryarkh at gmail.com (Muhammad Shahzad) Date: Fri, 8 Feb 2013 18:10:45 +0100 Subject: [Freeswitch-users] Sorry To Bump but .. In-Reply-To: References: <5114B1E6.7090402@gmail.com> <5114D483.5060506@gmail.com> Message-ID: Thanks Ken and Steven, I didn't actually read the login box, and tried to login with my freeswitch wiki log in account. Thank you. On Fri, Feb 8, 2013 at 5:32 PM, Steven Ayre wrote: > Read the message on the login popup. > > > On 8 February 2013 11:55, Muhammad Shahzad wrote: > >> can't login to paste bin. Anyhow, let me try centos 6.3 myself and see if >> i can reproduce the issue. >> >> Thank you. >> >> >> On Fri, Feb 8, 2013 at 11:33 AM, Antonio Teixeira < >> eagle.antonio at gmail.com> wrote: >> >>> Hello Muhammad. >>> >>> Still no go , >>> >>> >>> http://pastebin.freeswitch.org/20558 >>> >>> Tried >>> yum groupinstall 'development tools' <- Works ... >>> And also >>> yum install libxml2-devel libncurses-devel postgresql-devel sqlite-devel >>> curl-devel openssl-devel <- Works >>> And also >>> yum install libedit-devel <- just in case >>> >>> With every make attempt a clean build dir ( i mean rm-rf clean ..) >>> >>> Going to make a new VM ... :( >>> >>> Thanks for the help >>> Antonio >>> >>> >>> On 2/8/13 8:53 AM, Muhammad Shahzad wrote: >>> >>> I haven't used centos since version 5.4. However following should fix >>> libstdc++ problem that you seem to have with building libedit. >>> >>> yum groupinstall 'development tools' >>> yum groupinstall 'development libraries' >>> >>> It would be better to use a freshly installed machine if possible for >>> you. After doing group install check if following libs were successfully >>> installed (only for double check), >>> >>> yum install libxml2-devel libncurses-devel postgresql-devel sqlite-devel >>> curl-devel openssl-devel >>> >>> Note: some lib names might be different. Try with or without lib prefix. >>> >>> Thank you. >>> >>> >>> On 8 Feb 2013 09:09, "Antonio Teixeira" >>> wrote: >>> > >>> > Can anyone take a look at this ? >>> > >>> > http://jira.freeswitch.org/browse/FS-5069 >>> > >>> > I have performed dozens of make's the last few days and still i have >>> > been unable to get this to work... >>> > >>> > Thanks For Your Time >>> > Antonio >>> > >>> > >>> > >>> > >>> _________________________________________________________________________ >>> > Professional FreeSWITCH Consulting Services: >>> > consulting at freeswitch.org >>> > http://www.freeswitchsolutions.com >>> > >>> > >>> > >>> > >>> > Official FreeSWITCH Sites >>> > http://www.freeswitch.org >>> > http://wiki.freeswitch.org >>> > http://www.cluecon.com >>> > >>> > 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 >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >>> >>> >>> >>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com >>> >>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> >> -- >> Muhammad Shahzad >> ----------------------------------- >> CISCO Rich Media Communication Specialist (CRMCS) >> CISCO Certified Network Associate (CCNA) >> Cell: +49 176 99 83 10 85 >> MSN: shari_786pk at hotmail.com >> Email: shaheryarkh at googlemail.com >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +49 176 99 83 10 85 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130208/a3a346c7/attachment-0001.html From anthony.minessale at gmail.com Fri Feb 8 20:19:10 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 8 Feb 2013 11:19:10 -0600 Subject: [Freeswitch-users] Reg. Freeswitch Performace on Freescale Board In-Reply-To: References: <5110FD13.3080406@cem-solutions.net> <51121A98.6070109@cem-solutions.net> <5114EDE0.6020400@cem-solutions.net> Message-ID: The only thing that would make proxy mode work any better than not using it would be the fact that it also disables the timers. When there are no timers unless you are actually sending RTP to the box it will not move. That will increase performance because its not doing anything. On Fri, Feb 8, 2013 at 11:06 AM, Jeff Pyle wrote: > Hello, > > I believe the timers are disabled by setting > > in the sofia profile(s). The default configurations use a value of "soft". > > I have similar performance concerns bridging G711 SIP calls between two > sofia profiles on an AMD Geode processor. With full media it occupies > roughly 3.0% CPU per call. It scales to 25 calls with room for other basic > system processes. Maybe 26 or 27 if we really pushed it and maintained the > cps extremely low. Proxy media mode gives about a 15% performance increase > in our case. Bypass mode was not tested because it isn't appropriate for > our application. > > > - Jeff > > > On Fri, Feb 8, 2013 at 7:21 AM, Varghese wrote: > >> Hi all, >> >> Any update on this issue. Any other options we need to try with >> freeswitch. >> >> E1 PRI with freetdm freeswitch in freescale and cpu load is going to 100 >> with 15 calls. >> >> Thanks And Regards >> >> Varghese Paul >> > Hi Anthony, >> > >> > Thanks for the information. >> > >> > Kernel is Linux-2.6.32 patched by Freescale team for the hardware. >> > >> > Further we tested the freeswitch-1.2.6 on Freescale and following are >> > observations. >> > >> > We set the ulimit options as per the wiki link and ulimit ?s 240 >> > >> > 1. With switches *-np* and *?lp* the result is same i.e. maximum of 90 >> > calls and top command shows 100% CPU for freeswitch process >> > >> > 2. With RTP Bypass mode we could make 250 calls and CPU usage is 20% >> > >> > 3. With Proxy media mode, and default rtp mode the CPU is very high. >> > Approximately 1 to 1.5% per call and scales up to 90 calls >> > >> > 4. Set the option enable-timer value = false in SIP profile, >> > stun-enabled=false given all IP address in ext-rtp, >> > rtp-timer-name=none/soft etc., still the results are same >> > >> > Q1) Can you explain ?asterisk also uses blocking reads in its rtp >> > stack where we have timers. You can disable the timers as documented >> > in the wiki? which parameter you are referring to and where we need to >> > disable timers. >> > >> > Q2) Do we need to try any other options to increase the performance >> > numbers? >> > >> > F.Y.I, We used Empirix Hammer SIP Call Generator for testing SIP calls. >> > >> > Thanks >> > Varghese Paul >> > >> > Anthony Minessale wrote: >> >> That depends on the version of linux, the kernel version and several >> >> other factors. >> >> >> >> Were you actually moving media when testing, what tool are you >> >> testing with? >> >> >> >> On 32 bit you should make sure you have the stack size at 240 (ulimit >> >> -s 240 before starting) >> >> Also you may need to add the -lp or -np startup flags to reduce the >> >> realtime threads. >> >> >> >> asterisk also uses blocking reads in its rtp stack where we have >> >> timers. You can disable the timers as documented in the wiki.. >> >> >> >> >> >> >> >> >> >> On Tue, Feb 5, 2013 at 6:37 AM, Varghese >> >> > >> >> wrote: >> >> >> >> Hi all, >> >> We are facing following observations/issues with freeswitch on >> >> freescale >> >> processor boards. >> >> We ported freeswitch with freescale provided tool chain. >> >> _*Freeswitch Configuration: version 1.2.6 (git version)*_ >> >> 1. Only enabled mod_sofia, mod_dialplan, mod_command,mod_dptool and >> >> mod_console. >> >> 2. set the ulimit options and followed the steps in the following >> >> link >> >> >> >> http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations >> >> 3. SIP test tool run the load test for 500 users in freeswitch and >> >> configured the dial plan to just bridge the channels >> >> 4. Freeswitch media processing is in default mode: i.e media will >> go >> >> through freeswitch >> >> _*Freescale Board Configuration:*_ >> >> CPU : 1GHZ PowerQUICC III, 32 bit >> >> RAM: 1 GB >> >> Linux- 2.6.32 >> >> Load test is conducted for SIP to SIP calls with RTP. >> >> _*Observations:*_ >> >> Linux TOP command shows increasing CPU usage per call and >> >> increases to >> >> 100% with only 90 calls. After that system response is very slow >> >> We could also found delay in media. >> >> >> >> _Surprisingly, with Asterisk 1.8 the same system configuration >> >> works 250 >> >> calls without any problems. >> >> _ >> >> _*Questions: >> >> *_1. Any one ported Freeswitch on Freescale or any embedded >> >> processors? >> >> Any performance figures or references will be appreciated >> >> 2. How many simultaneous calls can be possible with 1GHz >> >> PowerQUICC III >> >> processor ? >> >> 3. Any more settings are required in freeswitch or linux for >> >> scalability? >> >> >> >> Thanks And Regards >> >> >> >> Varghese Paul >> >> >> >> >> >> >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130208/a0ab6bd3/attachment.html From carlos.voiperu at gmail.com Fri Feb 8 21:35:06 2013 From: carlos.voiperu at gmail.com (Carlos Rojas) Date: Fri, 8 Feb 2013 13:35:06 -0500 Subject: [Freeswitch-users] Real certificate in freeswitch In-Reply-To: References: <24F5E90E-1B95-403E-AC3B-4D66BAB72FED@gmail.com> Message-ID: I am working with yealink phones, but when I select, the option "only trust certificate", freeswtch refuse the register, if it unselected, the yealink phone it works. On Fri, Feb 8, 2013 at 11:34 AM, Steven Ayre wrote: > What phone is it, out of interest? > > Also try testing from a softphone to see if it works from elsewhere - it > could be the phone at fault, not FS. > > Also see if the FS log shows anything useful. > > -Steve > > > > On 8 February 2013 15:44, Carlos Rojas wrote: > >> Yes >> >> I am trying with sip and TLS, I've got several problems, I'll try it, >> again, >> >> >> Thank you for you help >> >> On Fri, Feb 8, 2013 at 6:00 AM, Steven Ayre wrote: >> >>> If you're talking about SIP/TLS and a server certificate on FS then the >>> phone will also need the correct CA certificates installed which may not be >>> the case by default, and the correct time set so that it can verify them. >>> >>> Steve >>> >>> >>> >>> On 8 Feb 2013, at 10:41, Daniel Ivanov wrote: >>> >>> But of course we have. Freeswitch accepts certificate bundles. So be >>> sure to include your whole certificate chain in a concatenated file. Like >>> ca.crt + intermediate (if any) + your.crt. and to be sure you've got the >>> right combo use some of the online ssl checkers and point the to your ip/ >>> domain and port. They just make a regular tcp session and show you the >>> presented certificate chain. >>> >>> Good luck. >>> On Feb 7, 2013 9:40 PM, "Carlos Rojas" wrote: >>> >>>> Hi Everybody >>>> >>>> I am new with freeswitch, I was trying, put in freeswithc my real >>>> certificate, in my freeswitch, but doesn't work. >>>> >>>> looks like freeswitch accepts the certificate, but when the phone tried >>>> register doesn't work. >>>> >>>> Anybody has configured a real certificate with freeswitch? >>>> >>>> >>>> I hope somebody could help me. >>>> >>>> >>>> Kind Regards >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>>> >>>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130208/7c32bdf0/attachment-0001.html From steveayre at gmail.com Fri Feb 8 22:05:31 2013 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 8 Feb 2013 19:05:31 +0000 Subject: [Freeswitch-users] Real certificate in freeswitch In-Reply-To: References: <24F5E90E-1B95-403E-AC3B-4D66BAB72FED@gmail.com> Message-ID: You would need to upload the CA certificate on that page, there are no trusted certificates by default. BTW if you plan to encrypt your RTP too that's a separate setting. http://wiki.freeswitch.org/wiki/Interop_List#Yealink_TLS_Configuration -Steve On 8 February 2013 18:35, Carlos Rojas wrote: > I am working with yealink phones, but when I select, the option "only > trust certificate", freeswtch refuse the register, if it unselected, the > yealink phone it works. > > > > > > On Fri, Feb 8, 2013 at 11:34 AM, Steven Ayre wrote: > >> What phone is it, out of interest? >> >> Also try testing from a softphone to see if it works from elsewhere - it >> could be the phone at fault, not FS. >> >> Also see if the FS log shows anything useful. >> >> -Steve >> >> >> >> On 8 February 2013 15:44, Carlos Rojas wrote: >> >>> Yes >>> >>> I am trying with sip and TLS, I've got several problems, I'll try it, >>> again, >>> >>> >>> Thank you for you help >>> >>> On Fri, Feb 8, 2013 at 6:00 AM, Steven Ayre wrote: >>> >>>> If you're talking about SIP/TLS and a server certificate on FS then the >>>> phone will also need the correct CA certificates installed which may not be >>>> the case by default, and the correct time set so that it can verify them. >>>> >>>> Steve >>>> >>>> >>>> >>>> On 8 Feb 2013, at 10:41, Daniel Ivanov wrote: >>>> >>>> But of course we have. Freeswitch accepts certificate bundles. So be >>>> sure to include your whole certificate chain in a concatenated file. Like >>>> ca.crt + intermediate (if any) + your.crt. and to be sure you've got the >>>> right combo use some of the online ssl checkers and point the to your ip/ >>>> domain and port. They just make a regular tcp session and show you the >>>> presented certificate chain. >>>> >>>> Good luck. >>>> On Feb 7, 2013 9:40 PM, "Carlos Rojas" >>>> wrote: >>>> >>>>> Hi Everybody >>>>> >>>>> I am new with freeswitch, I was trying, put in freeswithc my real >>>>> certificate, in my freeswitch, but doesn't work. >>>>> >>>>> looks like freeswitch accepts the certificate, but when the phone >>>>> tried register doesn't work. >>>>> >>>>> Anybody has configured a real certificate with freeswitch? >>>>> >>>>> >>>>> I hope somebody could help me. >>>>> >>>>> >>>>> Kind Regards >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> 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 >>>>> >>>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130208/6994b859/attachment.html From victor.chukalovskiy at gmail.com Fri Feb 8 22:43:23 2013 From: victor.chukalovskiy at gmail.com (Victor Chukalovskiy) Date: Fri, 08 Feb 2013 14:43:23 -0500 Subject: [Freeswitch-users] [ERR] switch_limit.c:86 Unset limit backendlist! In-Reply-To: References: <51147DFD.6040304@gmail.com> Message-ID: <5115555B.8020001@gmail.com> Hi Steve, Thanks for the input. Do you think /reload mod_hash/ will reset whatever might be the cause? -Victor On 02/08/2013 04:08 AM, Steven Ayre wrote: > Looks like it's an assert-type error. > > limit_state_handler() is called without the limit_backend variable > being set, which is an internal variable for storing limit information > used to internally track limit information during the call. The odd > thing is that variable should always be set prior to > limit_state_handler being registered. > > I'm guessing it's some sort of race condition... > > -Steve > > > > > On 8 February 2013 04:24, Victor Chukalovskiy > > > wrote: > > Hello, > > I'm getting this error occasionally in the log: > [ERR] switch_limit.c:86 Unset limit backendlist! > > Seems to happen only after I change the value of CPS limit on the > working system. For example, after I change *4* to 3 below and > send some more calls: > > > Could someone please explain when exactly does this error occur? > How to reset it other than restarting FS instance? > > What I found is that I'm not the only one who got annoyed, and > that loglevel was reduced in later versions: > http://jira.freeswitch.org/browse/FS-4792?page=com.atlassian.streams.streams-jira-plugin:activity-stream-issue-tab > > However, I want to figure out how to avoid this error in the first > place rather than just update and suppress it. Any help is > appreciated. > > Thank you, > Victor > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130208/a3865e14/attachment-0001.html From jaykris at gmail.com Fri Feb 8 23:07:11 2013 From: jaykris at gmail.com (JP) Date: Fri, 8 Feb 2013 12:07:11 -0800 Subject: [Freeswitch-users] mod-curl POST In-Reply-To: References: Message-ID: Thanks Rupa. I guess I need to re-read your email to understand better. For now I have asked the vendor controlling the end point to switch to GET. I will try out what you suggested when I get some free cycles and update this thread. Regards, JP On Thu, Feb 7, 2013 at 5:49 PM, Rupa Schomaker wrote: > Umm.. I just explained how to do it. The only piece missing is the > setting of the content-type header, but if that isn't going to be a problem > you odn't have to do anything special. > > Just remember to urlencode the (full) contents of the XML. > > And try it out. Use wireshark to see what is sent over the wire to the > server. > > > On Thu, Feb 7, 2013 at 11:26 AM, JP wrote: > >> Yes what I am looking for is to POST xml data in the body and not as >> name/value pairs. Is this possible? >> >> Thanks, >> -JP >> >> On Wed, Feb 6, 2013 at 7:21 PM, Rupa Schomaker wrote: >> >>> >>> >>> >>> On Wed, Feb 6, 2013 at 6:56 PM, Cal Leeming [Simplicity Media Ltd] < >>> cal.leeming at simplicitymedialtd.co.uk> wrote: >>> >>>> The docs says this; >>>> >>>> curl url [headers|json] [get|head|post [url_encode_data]] >>>> >>> >>> The [headers|json] is all about the response format. Just to make sure >>> that is clear. >>> >>> get/head/post should be self-evident >>> >>> url_encode_data is supposed to be the url encoded version of what you >>> want to send to the server. It is encoded on purpose (so that we don't >>> have to worry about embedded spaces or other special chars). >>> >>> So, you really just need to do: >>> >>> curl http://example.com/ post encodedxmlhere >>> >>> >>>> So, I assume you'd just do. >>>> >>>> curl http://example.com/ post xml=encodedxmlhere >>>> >>>> >>> This will result in www-url-encoded with key/value pair with xml as key >>> and the encoded xml as the value. This may be what the OP wants, but I >>> think what he wants is just XML. >>> >>> >>> Note that the content-type will still be application/x-www-form-urlencoded >>> which is "wrong". I've written xml service endpoints that behave >>> differently based on the content-type. Depending on how your service is >>> written this may be ok or not. There is an open ticket to address the >>> content-type issue. >>> >>> http://jira.freeswitch.org/browse/FS-4714 >>> >>> >>> >>>> Taken from here; >>>> http://wiki.freeswitch.org/wiki/Mod_curl >>>> >>>> Hope this helps, apologies if I misunderstood the question >>>> >>>> Cal >>>> >>>> >>> -- >>> -Rupa >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > > -- > -Rupa > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130208/9c0e64d2/attachment.html From kris at kriskinc.com Fri Feb 8 23:50:00 2013 From: kris at kriskinc.com (Kristian Kielhofner) Date: Fri, 8 Feb 2013 15:50:00 -0500 Subject: [Freeswitch-users] Somewhat OT: Specific SIP packets (and other traffic) can cause ethernet controller reset In-Reply-To: References: Message-ID: Update with response to the statement from Intel: http://blog.krisk.org/2013/02/packets-of-death-update.html On Wed, Feb 6, 2013 at 11:07 AM, Kristian Kielhofner wrote: > While not strictly FreeSWITCH related this issue could certainly > affect some of you: > > http://blog.krisk.org/2013/02/packets-of-death.html > > -- > Kristian Kielhofner -- Kristian Kielhofner From krice at freeswitch.org Sat Feb 9 00:37:33 2013 From: krice at freeswitch.org (Ken Rice) Date: Fri, 08 Feb 2013 15:37:33 -0600 Subject: [Freeswitch-users] Ummmmmm why is there no one on the free for all? Message-ID: GET IN HERE -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130208/af11d3e5/attachment.html From cal.leeming at simplicitymedialtd.co.uk Sat Feb 9 00:50:53 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Fri, 8 Feb 2013 21:50:53 +0000 Subject: [Freeswitch-users] Ummmmmm why is there no one on the free for all? In-Reply-To: References: Message-ID: I wish I had more time to join in on the fun... 15 hours/day work atm :( Cal On Fri, Feb 8, 2013 at 9:37 PM, Ken Rice wrote: > GET IN HERE > -- > Ken > *http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > *irc.freenode.net #freeswitch > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130208/e7b8520f/attachment.html From drk at drkngs.net Sat Feb 9 01:58:01 2013 From: drk at drkngs.net (Dave R. Kompel) Date: Fri, 08 Feb 2013 14:58:01 -0800 Subject: [Freeswitch-users] =?iso-8859-1?q?Ummmmmm_why_is_there_no_one_on_?= =?iso-8859-1?q?the_free_for=09all=3F?= In-Reply-To: Message-ID: <20130208225801.70c357e4@mail.tritonwest.net> I just got back from doing "work" shit. _____ From: Cal Leeming [Simplicity Media Ltd] [mailto:cal.leeming at simplicitymedialtd.co.uk] To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] Sent: Fri, 08 Feb 2013 13:50:53 -0800 Subject: Re: [Freeswitch-users] Ummmmmm why is there no one on the free for all? I wish I had more time to join in on the fun... 15 hours/day work atm :( Cal On Fri, Feb 8, 2013 at 9:37 PM, Ken Rice wrote: GET IN HERE -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130208/7a059ba9/attachment-0001.html From curriegrad2004 at gmail.com Sat Feb 9 02:08:57 2013 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Fri, 8 Feb 2013 15:08:57 -0800 Subject: [Freeswitch-users] Ummmmmm why is there no one on the free for all? In-Reply-To: <20130208225801.70c357e4@mail.tritonwest.net> References: <20130208225801.70c357e4@mail.tritonwest.net> Message-ID: Now it's empty... On Fri, Feb 8, 2013 at 2:58 PM, Dave R. Kompel wrote: > I just got back from doing "work" shit. > > ________________________________ > From: Cal Leeming [Simplicity Media Ltd] > [mailto:cal.leeming at simplicitymedialtd.co.uk] > To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] > Sent: Fri, 08 Feb 2013 13:50:53 -0800 > Subject: Re: [Freeswitch-users] Ummmmmm why is there no one on the free for > all? > > > I wish I had more time to join in on the fun... 15 hours/day work atm :( > > Cal > > On Fri, Feb 8, 2013 at 9:37 PM, Ken Rice wrote: >> >> GET IN HERE >> -- >> Ken >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> irc.freenode.net #freeswitch >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 jaykris at gmail.com Sat Feb 9 03:03:54 2013 From: jaykris at gmail.com (JP) Date: Fri, 8 Feb 2013 16:03:54 -0800 Subject: [Freeswitch-users] DTMF in IVR menu Message-ID: Hi, I am observing that when I use IVR menu, the main menu greeting does not get stopped as soon as I enter the dtmf digit. It plays for a couple of seconds longer and then stops and then the dtmf is processed. Is there any tuning parameters that I can tweak to make it stop sooner. Thanks, JP -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130208/9ee1f381/attachment.html From jaybinks at gmail.com Sat Feb 9 03:15:31 2013 From: jaybinks at gmail.com (jay binks) Date: Sat, 9 Feb 2013 10:15:31 +1000 Subject: [Freeswitch-users] Ummmmmm why is there no one on the free for all? In-Reply-To: References: Message-ID: because its 7:30 am on a saturday :) On 9 February 2013 07:37, Ken Rice wrote: > GET IN HERE > -- > Ken > *http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > *irc.freenode.net #freeswitch > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Sincerely Jay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130209/7616dc15/attachment.html From 8f27e956 at gmail.com Sat Feb 9 04:43:44 2013 From: 8f27e956 at gmail.com (S. Scott) Date: Fri, 8 Feb 2013 20:43:44 -0500 Subject: [Freeswitch-users] Ummmmmm why is there no one on the free for all? In-Reply-To: References: Message-ID: <8798545613865220138@unknownmsgid> Lol! ????? iThing: Big thumbs & little keys. Please excuse typo, spelling and grammar errors ? Thought of the Day ? "With all this manure, there must be a pony in here somewhere.? On 2013-02-08, at 19:19, jay binks wrote: because its 7:30 am on a saturday :) On 9 February 2013 07:37, Ken Rice wrote: > GET IN HERE > -- > Ken > *http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > *irc.freenode.net #freeswitch > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Sincerely Jay _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130208/defb74fe/attachment.html From msc at freeswitch.org Sat Feb 9 06:59:38 2013 From: msc at freeswitch.org (Michael Collins) Date: Fri, 8 Feb 2013 19:59:38 -0800 Subject: [Freeswitch-users] Somewhat OT: Specific SIP packets (and other traffic) can cause ethernet controller reset In-Reply-To: References: Message-ID: Awesome! Thanks for following up and congrats on getting slashdotted. -MC On Fri, Feb 8, 2013 at 12:50 PM, Kristian Kielhofner wrote: > Update with response to the statement from Intel: > > http://blog.krisk.org/2013/02/packets-of-death-update.html > > On Wed, Feb 6, 2013 at 11:07 AM, Kristian Kielhofner > wrote: > > While not strictly FreeSWITCH related this issue could certainly > > affect some of you: > > > > http://blog.krisk.org/2013/02/packets-of-death.html > > > > -- > > Kristian Kielhofner > > > > -- > Kristian Kielhofner > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130208/c6a2fac7/attachment.html From slickwilly2000 at gmx.de Fri Feb 8 18:18:26 2013 From: slickwilly2000 at gmx.de (Michael Huber) Date: Fri, 8 Feb 2013 16:18:26 +0100 Subject: [Freeswitch-users] Different music for Hold and Transfer (internal SIP-functions) Message-ID: I am using some regular sip-phones and the internal sip-functions for HOLD and TRANSFER. I do NOT use the applications ?transfer/att_xfer? and ?soft_hold?. The phone is a ?Siemens OpenStage 60 G?, but this fact should not be important. My problem is now that I can?t have different music for transfer and hold. If I would use the freeswitch-applications ?transfer/att_xfer? and ?soft_hold? then I would be able to set the variables ?hold_music? and ?temp_hold_music?. Any idea to achieve this? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130208/640d9bd4/attachment-0001.html From rupa at rupa.com Sat Feb 9 07:28:58 2013 From: rupa at rupa.com (Rupa Schomaker) Date: Fri, 8 Feb 2013 22:28:58 -0600 Subject: [Freeswitch-users] mod-curl POST In-Reply-To: References: Message-ID: mod_curl will send a POST if you ask for it to (use post, not get in the call). You have to be precise when describing the issue. You don't want it to send as a form post, but rather just to post the 'raw' data. you can do this today. The ticket just gives you control of the header that sets the content-type. The receiver on the other end can choose to ignore the content type if it knows it should always just get raw XML. Use wireshark (or equiv) to verify what is being sent... On Fri, Feb 8, 2013 at 2:07 PM, JP wrote: > Thanks Rupa. I guess I need to re-read your email to understand better. > For now I have asked the vendor controlling the end point to switch to GET. > I will try out what you suggested when I get some free cycles and update > this thread. > > Regards, > JP > > > On Thu, Feb 7, 2013 at 5:49 PM, Rupa Schomaker wrote: > >> Umm.. I just explained how to do it. The only piece missing is the >> setting of the content-type header, but if that isn't going to be a problem >> you odn't have to do anything special. >> >> Just remember to urlencode the (full) contents of the XML. >> >> And try it out. Use wireshark to see what is sent over the wire to the >> server. >> >> >> On Thu, Feb 7, 2013 at 11:26 AM, JP wrote: >> >>> Yes what I am looking for is to POST xml data in the body and not as >>> name/value pairs. Is this possible? >>> >>> Thanks, >>> -JP >>> >>> On Wed, Feb 6, 2013 at 7:21 PM, Rupa Schomaker wrote: >>> >>>> >>>> >>>> >>>> On Wed, Feb 6, 2013 at 6:56 PM, Cal Leeming [Simplicity Media Ltd] < >>>> cal.leeming at simplicitymedialtd.co.uk> wrote: >>>> >>>>> The docs says this; >>>>> >>>>> curl url [headers|json] [get|head|post [url_encode_data]] >>>>> >>>> >>>> The [headers|json] is all about the response format. Just to make sure >>>> that is clear. >>>> >>>> get/head/post should be self-evident >>>> >>>> url_encode_data is supposed to be the url encoded version of what you >>>> want to send to the server. It is encoded on purpose (so that we don't >>>> have to worry about embedded spaces or other special chars). >>>> >>>> So, you really just need to do: >>>> >>>> curl http://example.com/ post encodedxmlhere >>>> >>>> >>>>> So, I assume you'd just do. >>>>> >>>>> curl http://example.com/ post xml=encodedxmlhere >>>>> >>>>> >>>> This will result in www-url-encoded with key/value pair with xml as key >>>> and the encoded xml as the value. This may be what the OP wants, but I >>>> think what he wants is just XML. >>>> >>>> >>>> Note that the content-type will still be application/x-www-form-urlencoded >>>> which is "wrong". I've written xml service endpoints that behave >>>> differently based on the content-type. Depending on how your service is >>>> written this may be ok or not. There is an open ticket to address the >>>> content-type issue. >>>> >>>> http://jira.freeswitch.org/browse/FS-4714 >>>> >>>> >>>> >>>>> Taken from here; >>>>> http://wiki.freeswitch.org/wiki/Mod_curl >>>>> >>>>> Hope this helps, apologies if I misunderstood the question >>>>> >>>>> Cal >>>>> >>>>> >>>> -- >>>> -Rupa >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> >> -- >> -Rupa >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130208/dbadc5bd/attachment.html From martyn at magiccow.co.uk Sat Feb 9 12:01:34 2013 From: martyn at magiccow.co.uk (Martyn Davies) Date: Sat, 9 Feb 2013 09:01:34 +0000 Subject: [Freeswitch-users] Conference moderator via the ESL API? Message-ID: Is there a way to switch the moderator flag on/off for members of an active conference, and can this be done from the API? Also, can the moderator flag be switched on at "conference dial..."? The only way I can see to control mod status seems to be to define two profiles in the config. Regards, Martyn -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130209/0ada832b/attachment.html From vbvbrj at gmail.com Sat Feb 9 19:29:45 2013 From: vbvbrj at gmail.com (Mimiko Mimiko) Date: Sat, 9 Feb 2013 18:29:45 +0200 Subject: [Freeswitch-users] Running FS with "daemon" group. Message-ID: Hello. On this page: http://wiki.freeswitch.org/wiki/Freeswitch_init I see that FS is started under "daemon" group. Is it safe to start it that way, and is it safe if other services will be started under the same group? About "daemon" group debian says that is better that each service is started in its own uid/guid. But that way services cannot share same directory for storing and reading common files. Especially that FS creates files with 750 permission. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130209/de8bad58/attachment.html From egable+freeswitch at gmail.com Sat Feb 9 20:00:30 2013 From: egable+freeswitch at gmail.com (Eliot Gable) Date: Sat, 9 Feb 2013 12:00:30 -0500 Subject: [Freeswitch-users] High Availability Cluster Module for FreeSWITCH Message-ID: For the past two years, I have been working on and off on a high availability cluster module for FreeSWITCH. If you would like to see HA built into FreeSWITCH directly, instead of dealing with Pacemaker + Corosync or other 3rd party systems, please check out the module, voice your support, provide feedback and suggestions, and consider supporting it. Some advantages over a 3rd party solution like Pacemaker include: - Sub-second failure detection and response (meaning it can start recovering calls less than a second after a failure occurs) - Failure detection is built into FreeSWITCH so it can detect conditions which 3rd party modules cannot detect - One shared configuration file between all nodes (no per-node IP configuration required) - No fixed number of nodes - Graceful migration of calls with a maintenance mode (requires DNS) These are just some of the benefits of building a FreeSWITCH-specific HA system. For additional details about the project, please visit: http://wiki.freeswitch.org/wiki/Mod_ha_cluster If you would like to see this project come to fruition, please consider financially supporting it through Kickstarter: http://www.kickstarter.com/projects/1175121367/high-availability-cluster-module-for-freeswitch Thanks in advance for anyone who contributes, either with just a shout-out, with feedback and/or suggestions, and especially to those who pledge some financial support. -- Eliot Gable From stargray at bigmir.net Sat Feb 9 20:15:50 2013 From: stargray at bigmir.net (Anton Vojlenko) Date: Sat, 9 Feb 2013 19:15:50 +0200 Subject: [Freeswitch-users] ESL and DEBUG messages In-Reply-To: References: Message-ID: Hello! Giovanni, can you recommend me how to check skype client availability? I need this because skypopen module sometimes lost connection with skype client. 2013/2/5 Anton Vojlenko > Unfortunately* *$con->log(7) has incorrect syntax: > # perl /home/checker.pl > Use of inherited AUTOLOAD for non-method ESL::log() is deprecated at /home/ > checker.pl line 5. > Can't locate auto/ESL/log.al in @INC (@INC contains: /usr/local/lib/perl5 > /usr/local/share/perl5 /usr/lib/perl5/vendor_perl > /usr/share/perl5/vendor_perl /usrlib/perl5 /usr/share/perl5 .) at /home/ > checker.pl line 5 > > Who can help me? > > > 2013/2/4 Steven Ayre > >> I believe so. They're probably received as the LOG event, which is >> unfortunately undocumented http://wiki.freeswitch.org/wiki/Event_list#LOG >> >> -Steve >> >> >> >> On 4 February 2013 14:58, Anton Vojlenko wrote: >> >>> Thank you, Steve. >>> >>> >>> #!/usr/bin/perl >>> use strict; >>> use warnings; >>> require ESL; >>> #ESL::eslSetLogLevel(7); >>> >>> my $host = "127.0.0.1"; >>> my $port = "8021"; >>> my $pass = "ClueCon"; >>> my $con = new ESL::ESLconnection($host, $port, $pass); >>> >>> *$con->log(7);* >>> >>> my $status = $con->api("skypopen sk1 GET USERSTATUS")->getBody(); >>> print "$status"; >>> $con->disconnect(); >>> >>> Am i right? >>> >>> >>> 2013/2/4 Giovanni Maruzzelli >>> >>>> Thanks Steve! >>>> >>>> >>>> On Mon, Feb 4, 2013 at 3:32 PM, Steven Ayre wrote: >>>> >>>>> 'fsctl loglevel 9' controls the core logging, all logs are routed >>>>> through here and dropped if they're below the logging level set there. On >>>>> top of that modules that handle log messages separately filter log levels. >>>>> >>>>> 'console loglevel 9' controls mod_console which is only for when >>>>> freeswitch is started in the foreground. >>>>> >>>>> You don't want to use either of the above. >>>>> >>>>> Mod_event_socket controls logging via ESL connections. You need to >>>>> issue the 'log' ESL command to set the log level on a per-connection basis >>>>> (this is what /log does in fs_cli). >>>>> >>>>> See http://wiki.freeswitch.org/wiki/Mod_event_socket#log >>>>> >>>>> Note log is an ESL protocol command, not an api/app. >>>>> >>>>> -Steve >>>>> >>>>> >>>>> >>>>> On 4 February 2013 13:50, Anton Vojlenko wrote: >>>>> >>>>>> ****I >>>>>> can't solve this problem alone. >>>>>> In fs_cli i can see DEBUG messages: >>>>>> freeswitch at internal> skypopen sk1 GET USERSTATUS >>>>>> Using interface: globals.SKYPOPEN_INTERFACES[1].name=|||sk1||| >>>>>> >>>>>> 2013-02-04 15:31:44.565590 [DEBUG] skypopen_protocol.c:1732 [|] >>>>>> [DEBUG_SKYPE 1732 ][sk1 ][IDLE,IDLE] SENDING: |||GET >>>>>> USERSTATUS|||| >>>>>> 2013-02-04 15:31:44.565590 [DEBUG] skypopen_protocol.c:207 [|] >>>>>> [DEBUG_SKYPE 207 ][sk1 ][IDLE,IDLE] READING: |||USERSTATUS >>>>>> ONLINE||| >>>>>> >>>>>> But when i execute perl script i can't see them: >>>>>> #perl /home/checker.pl >>>>>> Using interface: globals.SKYPOPEN_INTERFACES[1].name=|||sk1||| >>>>>> >>>>>> #cat /home/checker.pl >>>>>> >>>>>> #!/usr/bin/perl >>>>>> use strict; >>>>>> use warnings; >>>>>> require ESL; >>>>>> #ESL::eslSetLogLevel(7); >>>>>> >>>>>> my $host = "127.0.0.1"; >>>>>> my $port = "8021"; >>>>>> my $pass = "ClueCon"; >>>>>> >>>>>> my $con = new ESL::ESLconnection($host, $port, $pass); >>>>>> >>>>>> $con->api("console loglevel 9"); >>>>>> $con->api("fsctl loglevel 9"); >>>>>> >>>>>> my $status = $con->api("skypopen sk1 GET USERSTATUS")->getBody(); >>>>>> print "$status"; >>>>>> $con->disconnect() >>>>>> >>>>>> >>>>>> >>>>>> 2013/1/23 Anton Vojlenko >>>>>> >>>>>>> Giovanni, I can see Skype API answers only when i using fs_cli. With >>>>>>> perl script I can't see them. >>>>>>> >>>>>>> >>>>>>> 2013/1/22 Giovanni Maruzzelli >>>>>>> >>>>>>>> On Tue, Jan 22, 2013 at 12:51 PM, Anton Vojlenko < >>>>>>>> stargray at bigmir.net> wrote: >>>>>>>> > Hello, >>>>>>>> > >>>>>>>> > I want to check user status from the Skype client instance, but I >>>>>>>> can't see >>>>>>>> > Skype API answers from it. How to fix this? >>>>>>>> >>>>>>>> Have you tried to give: >>>>>>>> >>>>>>>> "fsctl loglevel 9" in addition to "console loglevel 9" ? >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> > Script example: >>>>>>>> > >>>>>>>> > #!/usr/bin/perl >>>>>>>> > use strict; >>>>>>>> > use warnings; >>>>>>>> > require ESL; >>>>>>>> > >>>>>>>> > my $host = "localhost"; >>>>>>>> > my $port = "8021"; >>>>>>>> > my $pass = "somepass"; >>>>>>>> > my $con = new ESL::ESLconnection($host, $port, $pass); >>>>>>>> > >>>>>>>> > $con->api("console loglevel 9"); >>>>>>>> > my $status = $con->api("skypopen sk1 GET USERSTATUS")->getBody(); >>>>>>>> > print $status; >>>>>>>> > $con->disconnect(); >>>>>>>> > >>>>>>>> >>>>>>>> -- >>>>>>>> Sincerely, >>>>>>>> >>>>>>>> Giovanni Maruzzelli >>>>>>>> Cell : +39-347-2665618 >>>>>>>> >>>>>>>> >>>>>>>> _________________________________________________________________________ >>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>> consulting at freeswitch.org >>>>>>>> http://www.freeswitchsolutions.com >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> Official FreeSWITCH Sites >>>>>>>> http://www.freeswitch.org >>>>>>>> http://wiki.freeswitch.org >>>>>>>> http://www.cluecon.com >>>>>>>> >>>>>>>> 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 >>>>>>>> >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> 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 >>>>>> >>>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> 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 >>>>> >>>>> >>>> >>>> >>>> -- >>>> Sincerely, >>>> >>>> Giovanni Maruzzelli >>>> Cell : +39-347-2665618 >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130209/d0cd7260/attachment-0001.html From jsun at junsun.net Sun Feb 10 01:52:15 2013 From: jsun at junsun.net (Jun Sun) Date: Sat, 09 Feb 2013 14:52:15 -0800 Subject: [Freeswitch-users] how to end conference after initiator hangs up (dialout style) Message-ID: <5116D31F.8030807@junsun.net> I have set up a conferencing call node using dialout style. That is, I use the following commands to bridge everyone into a conference: conference $myRoom dial sofia/internal/$myNumber at x.x.x.x:5060 $myNumber conference_test"; conference $myRoom dial sofia/internal/$theirNumber1 at x.x.x.x:5060 $myNumber conference_test"; conference $myRoom dial sofia/internal/$theirNumber2 at x.x.x.x:5060 $myNumber conference_test"; ... Right now things are working fine except for one thing. I like to end the conference (i.e., all legs are disconnected) when the initiator ($myNumber) hangs up. But it is not doing that (understandably so!). How can I achieve that? I searched around and it seems most conf call controls only apply to dial-in style conference calls. Thanks for any pointers! Jun From ga at steadfasttelecom.com Sun Feb 10 02:49:56 2013 From: ga at steadfasttelecom.com (Gilad Abada) Date: Sat, 9 Feb 2013 18:49:56 -0500 Subject: [Freeswitch-users] High Availability Cluster Module for FreeSWITCH In-Reply-To: References: Message-ID: <-2120316468757263517@unknownmsgid> Hey Eliot Sounds exciting. Which DB will you be using Postgres? I will try to get my company to pitch in a few dollars. Sent from my mobile device. On Feb 9, 2013, at 12:11 PM, Eliot Gable wrote: > For the past two years, I have been working on and off on a high > availability cluster module for FreeSWITCH. If you would like to see > HA built into FreeSWITCH directly, instead of dealing with Pacemaker + > Corosync or other 3rd party systems, please check out the module, > voice your support, provide feedback and suggestions, and consider > supporting it. > > Some advantages over a 3rd party solution like Pacemaker include: > > - Sub-second failure detection and response (meaning it can start > recovering calls less than a second after a failure occurs) > - Failure detection is built into FreeSWITCH so it can detect > conditions which 3rd party modules cannot detect > - One shared configuration file between all nodes (no per-node IP > configuration required) > - No fixed number of nodes > - Graceful migration of calls with a maintenance mode (requires DNS) > > These are just some of the benefits of building a FreeSWITCH-specific > HA system. For additional details about the project, please visit: > > http://wiki.freeswitch.org/wiki/Mod_ha_cluster > > > If you would like to see this project come to fruition, please > consider financially supporting it through Kickstarter: > > http://www.kickstarter.com/projects/1175121367/high-availability-cluster-module-for-freeswitch > > > > Thanks in advance for anyone who contributes, either with just a > shout-out, with feedback and/or suggestions, and especially to those > who pledge some financial support. > > -- > Eliot Gable > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 egable+freeswitch at gmail.com Sun Feb 10 07:00:59 2013 From: egable+freeswitch at gmail.com (Eliot Gable) Date: Sat, 9 Feb 2013 23:00:59 -0500 Subject: [Freeswitch-users] High Availability Cluster Module for FreeSWITCH In-Reply-To: <-2120316468757263517@unknownmsgid> References: <-2120316468757263517@unknownmsgid> Message-ID: The beauty of the module is that it will not require any 3rd party software to operate. I am designing it to run entirely in-memory using multi-cast events to communicate between cluster nodes. No external database will be required to enable the automatic fail over and call recovery. If you have your customers / clients using DNS to point traffic to your system and you run multiple master nodes with DNS performing load balancing, you can also make use of the maintenance mode I have planned which will use a separate IP address and update your DNS records to remove the master which you put into maintenance mode and put in the maintenance IP which is assigned to the slave node which is being promoted as master. This allows calls to drain off the master while new calls come up on the slave, and then you can gracefully shut down the master when all calls are finished. Once you are done with maintenance, you simply bring it back online and it comes up as a slave or a master as needed (according to your configured master / slave ratio). The IP which it was using as a master node becomes the new designated maintenance IP. This is the only case where a 3rd party software package is required, and it is an optional feature. On Sat, Feb 9, 2013 at 6:49 PM, Gilad Abada wrote: > Hey Eliot > > Sounds exciting. Which DB will you be using Postgres? I will try to > get my company to pitch in a few dollars. > > Sent from my mobile device. > > On Feb 9, 2013, at 12:11 PM, Eliot Gable wrote: > >> For the past two years, I have been working on and off on a high >> availability cluster module for FreeSWITCH. If you would like to see >> HA built into FreeSWITCH directly, instead of dealing with Pacemaker + >> Corosync or other 3rd party systems, please check out the module, >> voice your support, provide feedback and suggestions, and consider >> supporting it. >> >> Some advantages over a 3rd party solution like Pacemaker include: >> >> - Sub-second failure detection and response (meaning it can start >> recovering calls less than a second after a failure occurs) >> - Failure detection is built into FreeSWITCH so it can detect >> conditions which 3rd party modules cannot detect >> - One shared configuration file between all nodes (no per-node IP >> configuration required) >> - No fixed number of nodes >> - Graceful migration of calls with a maintenance mode (requires DNS) >> >> These are just some of the benefits of building a FreeSWITCH-specific >> HA system. For additional details about the project, please visit: >> >> http://wiki.freeswitch.org/wiki/Mod_ha_cluster >> >> >> If you would like to see this project come to fruition, please >> consider financially supporting it through Kickstarter: >> >> http://www.kickstarter.com/projects/1175121367/high-availability-cluster-module-for-freeswitch >> >> >> >> Thanks in advance for anyone who contributes, either with just a >> shout-out, with feedback and/or suggestions, and especially to those >> who pledge some financial support. >> From yura407 at gmail.com Sun Feb 10 08:10:39 2013 From: yura407 at gmail.com (Yuriy Ostapchuk) Date: Sun, 10 Feb 2013 00:10:39 -0500 Subject: [Freeswitch-users] USSD support for mod_gsmopen Message-ID: Hi, I've submitted a patch that implements USSD messaging support. (useful for checking and refilling account balance and other provider-specific usage) Here is the JIRA ticket: http://jira.freeswitch.org/browse/FS-5078 If this feature is considered useful, I would appreciate if it's integrated in git. Thanks, Yuriy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130210/784da45c/attachment.html From yehavi.bourvine at gmail.com Sun Feb 10 12:21:25 2013 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Sun, 10 Feb 2013 11:21:25 +0200 Subject: [Freeswitch-users] VoIP gateway with PRI In-Reply-To: <51140221.4040107@gmail.com> References: <51140221.4040107@gmail.com> Message-ID: Hi, I just checked with our Audiocodes support, and you can do what you want. In order to route a call from one PRI to the other, you insert an entry in two tables: Tel->IP: the destination is localhost. IP->Tel - the source is either any or localhost. Regards, __Yehavi: 2013/2/7 Mimiko > Hello. > > I want to propose for purchasing a VoIP gateway to connect some PRI > lines with FS. Basic operation has to be routing like this: > > One line PRI will be connected to phone provider. Some calls to certain > numbers thru this connection must be routed to FS via IP. Other > destination numbers must be routed to second PRI connection for legacy > PBX for analog phones. So the VoIP gateway must have two PRI (T1/E1) > connections and two LAN connection for failover. > > After a while this VoIP gateway will be as a tehnology converter from IP > to PRI. So LAN connectors will be used for connection to FS in a > failover manner. And two PRI connectors will be used to connect to > legacy PBX which have two PRI inputs to handle 60 concurrent connection. > > I've spotted three models: > 1) Patton SN4951/4E60V2GS/EUI 4T1/E1/PRI 60 VoIP Channels G.SHDSL for > $5,000 > > 2) Audiocodes Mediant 1000 M1K-D2 2T1/E1 for $3,800 > > 3) Sangoma Vega 400 2 T1/E1 Digital Gateway for $4,437 > > Which of this is a better working solution with failover and extended > call routing plans? > > Thansk. > > -- > Mimiko desu. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130210/d8e5b8e5/attachment.html From gmaruzz at gmail.com Sun Feb 10 14:36:28 2013 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Sun, 10 Feb 2013 12:36:28 +0100 Subject: [Freeswitch-users] USSD support for mod_gsmopen In-Reply-To: References: Message-ID: On Sun, Feb 10, 2013 at 6:10 AM, Yuriy Ostapchuk wrot > Hi, > I've submitted a patch that implements USSD messaging support. > (useful for checking and refilling account balance and other > provider-specific usage) > > > Here is the JIRA ticket: > http://jira.freeswitch.org/browse/FS-5078 > > If this feature is considered useful, I would appreciate if it's > integrated in git. > Thanks Yuriy, I'll have a look into it asap next week. Thanks again, yes is considered useful and grateful received. -giovanni -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130210/b181f0c3/attachment.html From gozdal at gmail.com Sun Feb 10 14:54:12 2013 From: gozdal at gmail.com (Marcin Gozdalik) Date: Sun, 10 Feb 2013 12:54:12 +0100 Subject: [Freeswitch-users] High Availability Cluster Module for FreeSWITCH In-Reply-To: References: <-2120316468757263517@unknownmsgid> Message-ID: I wonder how do you handle network partition? Simply "knowing" which nodes are up in a distributed system is hard enough and there is a reason why solutions like pacemaker/corosync have years of development behind them. W dniu niedziela, 10 lutego 2013 u?ytkownik Eliot Gable napisa?: > The beauty of the module is that it will not require any 3rd party > software to operate. I am designing it to run entirely in-memory using > multi-cast events to communicate between cluster nodes. No external > database will be required to enable the automatic fail over and call > recovery. > > If you have your customers / clients using DNS to point traffic to > your system and you run multiple master nodes with DNS performing load > balancing, you can also make use of the maintenance mode I have > planned which will use a separate IP address and update your DNS > records to remove the master which you put into maintenance mode and > put in the maintenance IP which is assigned to the slave node which is > being promoted as master. This allows calls to drain off the master > while new calls come up on the slave, and then you can gracefully shut > down the master when all calls are finished. Once you are done with > maintenance, you simply bring it back online and it comes up as a > slave or a master as needed (according to your configured master / > slave ratio). The IP which it was using as a master node becomes the > new designated maintenance IP. This is the only case where a 3rd party > software package is required, and it is an optional feature. > > > On Sat, Feb 9, 2013 at 6:49 PM, Gilad Abada > > wrote: > > Hey Eliot > > > > Sounds exciting. Which DB will you be using Postgres? I will try to > > get my company to pitch in a few dollars. > > > > Sent from my mobile device. > > > > On Feb 9, 2013, at 12:11 PM, Eliot Gable > > wrote: > > > >> For the past two years, I have been working on and off on a high > >> availability cluster module for FreeSWITCH. If you would like to see > >> HA built into FreeSWITCH directly, instead of dealing with Pacemaker + > >> Corosync or other 3rd party systems, please check out the module, > >> voice your support, provide feedback and suggestions, and consider > >> supporting it. > >> > >> Some advantages over a 3rd party solution like Pacemaker include: > >> > >> - Sub-second failure detection and response (meaning it can start > >> recovering calls less than a second after a failure occurs) > >> - Failure detection is built into FreeSWITCH so it can detect > >> conditions which 3rd party modules cannot detect > >> - One shared configuration file between all nodes (no per-node IP > >> configuration required) > >> - No fixed number of nodes > >> - Graceful migration of calls with a maintenance mode (requires DNS) > >> > >> These are just some of the benefits of building a FreeSWITCH-specific > >> HA system. For additional details about the project, please visit: > >> > >> http://wiki.freeswitch.org/wiki/Mod_ha_cluster > >> > >> > >> If you would like to see this project come to fruition, please > >> consider financially supporting it through Kickstarter: > >> > >> > http://www.kickstarter.com/projects/1175121367/high-availability-cluster-module-for-freeswitch > >> > >> > >> > >> Thanks in advance for anyone who contributes, either with just a > >> shout-out, with feedback and/or suggestions, and especially to those > >> who pledge some financial support. > >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -- Marcin Gozdalik -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130210/da8477de/attachment-0001.html From egable+freeswitch at gmail.com Sun Feb 10 18:27:31 2013 From: egable+freeswitch at gmail.com (Eliot Gable) Date: Sun, 10 Feb 2013 10:27:31 -0500 Subject: [Freeswitch-users] High Availability Cluster Module for FreeSWITCH In-Reply-To: References: <-2120316468757263517@unknownmsgid> Message-ID: You use multiple NICs in the systems and send heartbeats out all of them. There is no other way to do it. Two NICs are required, and 3 are recommended. On Sun, Feb 10, 2013 at 6:54 AM, Marcin Gozdalik wrote: > I wonder how do you handle network partition? Simply "knowing" which nodes > are up in a distributed system is hard enough and there is a reason why > solutions like pacemaker/corosync have years of development behind them. > > W dniu niedziela, 10 lutego 2013 u?ytkownik Eliot Gable napisa?: >> >> The beauty of the module is that it will not require any 3rd party >> software to operate. I am designing it to run entirely in-memory using >> multi-cast events to communicate between cluster nodes. No external >> database will be required to enable the automatic fail over and call >> recovery. >> >> If you have your customers / clients using DNS to point traffic to >> your system and you run multiple master nodes with DNS performing load >> balancing, you can also make use of the maintenance mode I have >> planned which will use a separate IP address and update your DNS >> records to remove the master which you put into maintenance mode and >> put in the maintenance IP which is assigned to the slave node which is >> being promoted as master. This allows calls to drain off the master >> while new calls come up on the slave, and then you can gracefully shut >> down the master when all calls are finished. Once you are done with >> maintenance, you simply bring it back online and it comes up as a >> slave or a master as needed (according to your configured master / >> slave ratio). The IP which it was using as a master node becomes the >> new designated maintenance IP. This is the only case where a 3rd party >> software package is required, and it is an optional feature. >> >> >> On Sat, Feb 9, 2013 at 6:49 PM, Gilad Abada >> wrote: >> > Hey Eliot >> > >> > Sounds exciting. Which DB will you be using Postgres? I will try to >> > get my company to pitch in a few dollars. >> > >> > Sent from my mobile device. >> > >> > On Feb 9, 2013, at 12:11 PM, Eliot Gable >> > wrote: >> > >> >> For the past two years, I have been working on and off on a high >> >> availability cluster module for FreeSWITCH. If you would like to see >> >> HA built into FreeSWITCH directly, instead of dealing with Pacemaker + >> >> Corosync or other 3rd party systems, please check out the module, >> >> voice your support, provide feedback and suggestions, and consider >> >> supporting it. >> >> >> >> Some advantages over a 3rd party solution like Pacemaker include: >> >> >> >> - Sub-second failure detection and response (meaning it can start >> >> recovering calls less than a second after a failure occurs) >> >> - Failure detection is built into FreeSWITCH so it can detect >> >> conditions which 3rd party modules cannot detect >> >> - One shared configuration file between all nodes (no per-node IP >> >> configuration required) >> >> - No fixed number of nodes >> >> - Graceful migration of calls with a maintenance mode (requires DNS) >> >> >> >> These are just some of the benefits of building a FreeSWITCH-specific >> >> HA system. For additional details about the project, please visit: >> >> >> >> http://wiki.freeswitch.org/wiki/Mod_ha_cluster >> >> >> >> >> >> If you would like to see this project come to fruition, please >> >> consider financially supporting it through Kickstarter: >> >> >> >> >> >> http://www.kickstarter.com/projects/1175121367/high-availability-cluster-module-for-freeswitch >> >> >> >> >> >> >> >> Thanks in advance for anyone who contributes, either with just a >> >> shout-out, with feedback and/or suggestions, and especially to those >> >> who pledge some financial support. >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 > > > > -- > Marcin Gozdalik > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -- Eliot Gable "We do not inherit the Earth from our ancestors: we borrow it from our children." ~David Brower "I decided the words were too conservative for me. We're not borrowing from our children, we're stealing from them--and it's not even considered to be a crime." ~David Brower "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to live; not live to eat.) ~Marcus Tullius Cicero From egable+freeswitch at gmail.com Sun Feb 10 18:51:43 2013 From: egable+freeswitch at gmail.com (Eliot Gable) Date: Sun, 10 Feb 2013 10:51:43 -0500 Subject: [Freeswitch-users] High Availability Cluster Module for FreeSWITCH In-Reply-To: References: <-2120316468757263517@unknownmsgid> Message-ID: On Sun, Feb 10, 2013 at 10:27 AM, Eliot Gable wrote: > You use multiple NICs in the systems and send heartbeats out all of > them. There is no other way to do it. Two NICs are required, and 3 are > recommended. Just to clarify a little, this is something that needs to be handled mostly by the person / organization deploying the HA module. The HA module supports multiple heartbeat NICs. The entity deploying the module needs to design the physical layer of their network to ensure a full network partition can never occur. You do that in general by deploying 2 or more physical networks. This means you need redundant power, battery backups, multiple physical switches, etc. You need to design and deploy your physical network to ensure that no matter what fails (power, wiring, switch ports, switches, NICs, etc), the systems using the module always have an alternative method of communicating with each other. Typically, two physical networks are sufficient for most users. However, carrier deployments who want to offer 99.999% or better uptime might want to go with three physical networks. This means placing three NICs in each system (one on each network) and configuring mod_ha_cluster to send and receive messages on all three NICs. The module keeps several seconds of cached messages in a hash table to eliminate / ignore duplicate messages. The first one received will be used and it's message ID is stored in the hash table. If another copy arrives before that entry is pruned from the cache, the additional copy is ignored. In addition, you can deploy multiple clusters distributed geographically to ensure that if one entire cluster goes offline, your services do not fail entirely. There is nothing magical or "hard" per-se about detecting and preventing a network split from screwing with the cluster. The hardest part is simply educating people on how to design and deploy the physical network and configure the module so it can do the detection. Also, please do not confuse the difficulty of writing a general purpose HA system like Pacemaker with the relative simplicity of writing one for a single specific application. When you write something general purpose like Pacemaker, your task is vastly more difficult due to the multitude of software configurations you need to support. For mod_ha_cluster, there is a very specific design in mind. Namely, you set a master-to-slave ratio and bring nodes online and it is designed to behave a specific way and bring those nodes online in a specific, well-defined way. There is really only "one way" that this module supports bringing up the cluster. This vastly simplifies the code and the entire project. From martyn at magiccow.co.uk Sun Feb 10 19:27:52 2013 From: martyn at magiccow.co.uk (Martyn Davies) Date: Sun, 10 Feb 2013 16:27:52 +0000 Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 80, Issue 70 In-Reply-To: References: Message-ID: Hi Jun, There is a member-flag called endconf which does what you want. However, I'm not sure how to specify that as part of "conference dial"... Regards, Martyn On 10 February 2013 11:54, wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users at lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request at lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner at lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. how to end conference after initiator hangs up (dialout > style) (Jun Sun) > 2. Re: High Availability Cluster Module for FreeSWITCH (Gilad Abada) > 3. Re: High Availability Cluster Module for FreeSWITCH (Eliot Gable) > 4. USSD support for mod_gsmopen (Yuriy Ostapchuk) > 5. Re: VoIP gateway with PRI (Yehavi Bourvine) > 6. Re: USSD support for mod_gsmopen (Giovanni Maruzzelli) > 7. Re: High Availability Cluster Module for FreeSWITCH > (Marcin Gozdalik) > > > ---------- Forwarded message ---------- > From: Jun Sun > To: freeswitch-users at lists.freeswitch.org > Cc: > Date: Sat, 09 Feb 2013 14:52:15 -0800 > Subject: [Freeswitch-users] how to end conference after initiator hangs up > (dialout style) > > I have set up a conferencing call node using dialout style. That is, I use > the following commands to bridge everyone into a conference: > > conference $myRoom dial sofia/internal/$myNumber at x.x.**x.x:5060 $myNumber > conference_test"; > conference $myRoom dial sofia/internal/$theirNumber1@**x.x.x.x:5060 > $myNumber conference_test"; > conference $myRoom dial sofia/internal/$theirNumber2@**x.x.x.x:5060 > $myNumber conference_test"; > ... > > Right now things are working fine except for one thing. I like to end the > conference (i.e., all legs are disconnected) when the initiator ($myNumber) > hangs up. But it is not doing that (understandably so!). > > How can I achieve that? > > I searched around and it seems most conf call controls only apply to > dial-in style conference calls. > > Thanks for any pointers! > > Jun > > > > > ---------- Forwarded message ---------- > From: Gilad Abada > To: FreeSWITCH Users Help > Cc: > Date: Sat, 9 Feb 2013 18:49:56 -0500 > Subject: Re: [Freeswitch-users] High Availability Cluster Module for > FreeSWITCH > Hey Eliot > > Sounds exciting. Which DB will you be using Postgres? I will try to > get my company to pitch in a few dollars. > > Sent from my mobile device. > > On Feb 9, 2013, at 12:11 PM, Eliot Gable > wrote: > > > For the past two years, I have been working on and off on a high > > availability cluster module for FreeSWITCH. If you would like to see > > HA built into FreeSWITCH directly, instead of dealing with Pacemaker + > > Corosync or other 3rd party systems, please check out the module, > > voice your support, provide feedback and suggestions, and consider > > supporting it. > > > > Some advantages over a 3rd party solution like Pacemaker include: > > > > - Sub-second failure detection and response (meaning it can start > > recovering calls less than a second after a failure occurs) > > - Failure detection is built into FreeSWITCH so it can detect > > conditions which 3rd party modules cannot detect > > - One shared configuration file between all nodes (no per-node IP > > configuration required) > > - No fixed number of nodes > > - Graceful migration of calls with a maintenance mode (requires DNS) > > > > These are just some of the benefits of building a FreeSWITCH-specific > > HA system. For additional details about the project, please visit: > > > > http://wiki.freeswitch.org/wiki/Mod_ha_cluster > > > > > > If you would like to see this project come to fruition, please > > consider financially supporting it through Kickstarter: > > > > > http://www.kickstarter.com/projects/1175121367/high-availability-cluster-module-for-freeswitch > > > > > > > > Thanks in advance for anyone who contributes, either with just a > > shout-out, with feedback and/or suggestions, and especially to those > > who pledge some financial support. > > > > -- > > Eliot Gable > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > 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 > > > > > ---------- Forwarded message ---------- > From: Eliot Gable > To: FreeSWITCH Users Help > Cc: > Date: Sat, 9 Feb 2013 23:00:59 -0500 > Subject: Re: [Freeswitch-users] High Availability Cluster Module for > FreeSWITCH > The beauty of the module is that it will not require any 3rd party > software to operate. I am designing it to run entirely in-memory using > multi-cast events to communicate between cluster nodes. No external > database will be required to enable the automatic fail over and call > recovery. > > If you have your customers / clients using DNS to point traffic to > your system and you run multiple master nodes with DNS performing load > balancing, you can also make use of the maintenance mode I have > planned which will use a separate IP address and update your DNS > records to remove the master which you put into maintenance mode and > put in the maintenance IP which is assigned to the slave node which is > being promoted as master. This allows calls to drain off the master > while new calls come up on the slave, and then you can gracefully shut > down the master when all calls are finished. Once you are done with > maintenance, you simply bring it back online and it comes up as a > slave or a master as needed (according to your configured master / > slave ratio). The IP which it was using as a master node becomes the > new designated maintenance IP. This is the only case where a 3rd party > software package is required, and it is an optional feature. > > > On Sat, Feb 9, 2013 at 6:49 PM, Gilad Abada > wrote: > > Hey Eliot > > > > Sounds exciting. Which DB will you be using Postgres? I will try to > > get my company to pitch in a few dollars. > > > > Sent from my mobile device. > > > > On Feb 9, 2013, at 12:11 PM, Eliot Gable > wrote: > > > >> For the past two years, I have been working on and off on a high > >> availability cluster module for FreeSWITCH. If you would like to see > >> HA built into FreeSWITCH directly, instead of dealing with Pacemaker + > >> Corosync or other 3rd party systems, please check out the module, > >> voice your support, provide feedback and suggestions, and consider > >> supporting it. > >> > >> Some advantages over a 3rd party solution like Pacemaker include: > >> > >> - Sub-second failure detection and response (meaning it can start > >> recovering calls less than a second after a failure occurs) > >> - Failure detection is built into FreeSWITCH so it can detect > >> conditions which 3rd party modules cannot detect > >> - One shared configuration file between all nodes (no per-node IP > >> configuration required) > >> - No fixed number of nodes > >> - Graceful migration of calls with a maintenance mode (requires DNS) > >> > >> These are just some of the benefits of building a FreeSWITCH-specific > >> HA system. For additional details about the project, please visit: > >> > >> http://wiki.freeswitch.org/wiki/Mod_ha_cluster > >> > >> > >> If you would like to see this project come to fruition, please > >> consider financially supporting it through Kickstarter: > >> > >> > http://www.kickstarter.com/projects/1175121367/high-availability-cluster-module-for-freeswitch > >> > >> > >> > >> Thanks in advance for anyone who contributes, either with just a > >> shout-out, with feedback and/or suggestions, and especially to those > >> who pledge some financial support. > >> > > > > > ---------- Forwarded message ---------- > From: Yuriy Ostapchuk > To: freeswitch-users at lists.freeswitch.org > Cc: > Date: Sun, 10 Feb 2013 00:10:39 -0500 > Subject: [Freeswitch-users] USSD support for mod_gsmopen > Hi, > I've submitted a patch that implements USSD messaging support. > (useful for checking and refilling account balance and other > provider-specific usage) > > Here is the JIRA ticket: > http://jira.freeswitch.org/browse/FS-5078 > > If this feature is considered useful, I would appreciate if it's > integrated in git. > > Thanks, > Yuriy > > > ---------- Forwarded message ---------- > From: Yehavi Bourvine > To: FreeSWITCH Users Help > Cc: > Date: Sun, 10 Feb 2013 11:21:25 +0200 > Subject: Re: [Freeswitch-users] VoIP gateway with PRI > Hi, > I just checked with our Audiocodes support, and you can do what you want. > > In order to route a call from one PRI to the other, you insert an entry in > two tables: > Tel->IP: the destination is localhost. > IP->Tel - the source is either any or localhost. > > Regards, __Yehavi: > > > 2013/2/7 Mimiko > >> Hello. >> >> I want to propose for purchasing a VoIP gateway to connect some PRI >> lines with FS. Basic operation has to be routing like this: >> >> One line PRI will be connected to phone provider. Some calls to certain >> numbers thru this connection must be routed to FS via IP. Other >> destination numbers must be routed to second PRI connection for legacy >> PBX for analog phones. So the VoIP gateway must have two PRI (T1/E1) >> connections and two LAN connection for failover. >> >> After a while this VoIP gateway will be as a tehnology converter from IP >> to PRI. So LAN connectors will be used for connection to FS in a >> failover manner. And two PRI connectors will be used to connect to >> legacy PBX which have two PRI inputs to handle 60 concurrent connection. >> >> I've spotted three models: >> 1) Patton SN4951/4E60V2GS/EUI 4T1/E1/PRI 60 VoIP Channels G.SHDSL for >> $5,000 >> >> 2) Audiocodes Mediant 1000 M1K-D2 2T1/E1 for $3,800 >> >> 3) Sangoma Vega 400 2 T1/E1 Digital Gateway for $4,437 >> >> Which of this is a better working solution with failover and extended >> call routing plans? >> >> Thansk. >> >> -- >> Mimiko desu. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> > > > > ---------- Forwarded message ---------- > From: Giovanni Maruzzelli > To: FreeSWITCH Users Help > Cc: > Date: Sun, 10 Feb 2013 12:36:28 +0100 > Subject: Re: [Freeswitch-users] USSD support for mod_gsmopen > > On Sun, Feb 10, 2013 at 6:10 AM, Yuriy Ostapchuk wrot > >> Hi, >> I've submitted a patch that implements USSD messaging support. >> (useful for checking and refilling account balance and other >> provider-specific usage) >> >> >> Here is the JIRA ticket: >> http://jira.freeswitch.org/browse/FS-5078 >> >> If this feature is considered useful, I would appreciate if it's >> integrated in git. >> > > Thanks Yuriy, I'll have a look into it asap next week. > > Thanks again, yes is considered useful and grateful received. > > -giovanni > > > -- > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > > > ---------- Forwarded message ---------- > From: Marcin Gozdalik > To: FreeSWITCH Users Help > Cc: > Date: Sun, 10 Feb 2013 12:54:12 +0100 > Subject: Re: [Freeswitch-users] High Availability Cluster Module for > FreeSWITCH > I wonder how do you handle network partition? Simply "knowing" which nodes > are up in a distributed system is hard enough and there is a reason why > solutions like pacemaker/corosync have years of development behind them. > > W dniu niedziela, 10 lutego 2013 u?ytkownik Eliot Gable napisa?: > >> The beauty of the module is that it will not require any 3rd party >> software to operate. I am designing it to run entirely in-memory using >> multi-cast events to communicate between cluster nodes. No external >> database will be required to enable the automatic fail over and call >> recovery. >> >> If you have your customers / clients using DNS to point traffic to >> your system and you run multiple master nodes with DNS performing load >> balancing, you can also make use of the maintenance mode I have >> planned which will use a separate IP address and update your DNS >> records to remove the master which you put into maintenance mode and >> put in the maintenance IP which is assigned to the slave node which is >> being promoted as master. This allows calls to drain off the master >> while new calls come up on the slave, and then you can gracefully shut >> down the master when all calls are finished. Once you are done with >> maintenance, you simply bring it back online and it comes up as a >> slave or a master as needed (according to your configured master / >> slave ratio). The IP which it was using as a master node becomes the >> new designated maintenance IP. This is the only case where a 3rd party >> software package is required, and it is an optional feature. >> >> >> On Sat, Feb 9, 2013 at 6:49 PM, Gilad Abada >> wrote: >> > Hey Eliot >> > >> > Sounds exciting. Which DB will you be using Postgres? I will try to >> > get my company to pitch in a few dollars. >> > >> > Sent from my mobile device. >> > >> > On Feb 9, 2013, at 12:11 PM, Eliot Gable >> wrote: >> > >> >> For the past two years, I have been working on and off on a high >> >> availability cluster module for FreeSWITCH. If you would like to see >> >> HA built into FreeSWITCH directly, instead of dealing with Pacemaker + >> >> Corosync or other 3rd party systems, please check out the module, >> >> voice your support, provide feedback and suggestions, and consider >> >> supporting it. >> >> >> >> Some advantages over a 3rd party solution like Pacemaker include: >> >> >> >> - Sub-second failure detection and response (meaning it can start >> >> recovering calls less than a second after a failure occurs) >> >> - Failure detection is built into FreeSWITCH so it can detect >> >> conditions which 3rd party modules cannot detect >> >> - One shared configuration file between all nodes (no per-node IP >> >> configuration required) >> >> - No fixed number of nodes >> >> - Graceful migration of calls with a maintenance mode (requires DNS) >> >> >> >> These are just some of the benefits of building a FreeSWITCH-specific >> >> HA system. For additional details about the project, please visit: >> >> >> >> http://wiki.freeswitch.org/wiki/Mod_ha_cluster >> >> >> >> >> >> If you would like to see this project come to fruition, please >> >> consider financially supporting it through Kickstarter: >> >> >> >> >> http://www.kickstarter.com/projects/1175121367/high-availability-cluster-module-for-freeswitch >> >> >> >> >> >> >> >> Thanks in advance for anyone who contributes, either with just a >> >> shout-out, with feedback and/or suggestions, and especially to those >> >> who pledge some financial support. >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> > > > -- > Marcin Gozdalik > > _______________________________________________ > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130210/e8c6aff3/attachment-0001.html From steveayre at gmail.com Sun Feb 10 20:08:28 2013 From: steveayre at gmail.com (Steven Ayre) Date: Sun, 10 Feb 2013 17:08:28 +0000 Subject: [Freeswitch-users] High Availability Cluster Module for FreeSWITCH In-Reply-To: References: <-2120316468757263517@unknownmsgid> Message-ID: That covers redundancy in case of a network card or cable failure, but isn't what partitioning is about. Multiple NICs cannot prevent partitioning. As an example, partitioning might happen when a network switch between two network segments fails so you have nodes A+B in segment 1 able to talk to each other but unable to talk to nodes C+D in segment 2, while C+D can talk to each other but not A+B. Pacemaker/corosync contain a lot of algorithms to fence off partitions without quorum and can resort to things like STONITH if required to force a node to shutdown rather than risk it causing disruption to the cluster (for example if it tries to take over traffic to a virtual IP you could end up in a case where you have two servers sending ARP responses for the same IP). -Steve On 10 February 2013 15:27, Eliot Gable wrote: > You use multiple NICs in the systems and send heartbeats out all of > them. There is no other way to do it. Two NICs are required, and 3 are > recommended. > > > On Sun, Feb 10, 2013 at 6:54 AM, Marcin Gozdalik wrote: > > I wonder how do you handle network partition? Simply "knowing" which > nodes > > are up in a distributed system is hard enough and there is a reason why > > solutions like pacemaker/corosync have years of development behind them. > > > > W dniu niedziela, 10 lutego 2013 u?ytkownik Eliot Gable napisa?: > >> > >> The beauty of the module is that it will not require any 3rd party > >> software to operate. I am designing it to run entirely in-memory using > >> multi-cast events to communicate between cluster nodes. No external > >> database will be required to enable the automatic fail over and call > >> recovery. > >> > >> If you have your customers / clients using DNS to point traffic to > >> your system and you run multiple master nodes with DNS performing load > >> balancing, you can also make use of the maintenance mode I have > >> planned which will use a separate IP address and update your DNS > >> records to remove the master which you put into maintenance mode and > >> put in the maintenance IP which is assigned to the slave node which is > >> being promoted as master. This allows calls to drain off the master > >> while new calls come up on the slave, and then you can gracefully shut > >> down the master when all calls are finished. Once you are done with > >> maintenance, you simply bring it back online and it comes up as a > >> slave or a master as needed (according to your configured master / > >> slave ratio). The IP which it was using as a master node becomes the > >> new designated maintenance IP. This is the only case where a 3rd party > >> software package is required, and it is an optional feature. > >> > >> > >> On Sat, Feb 9, 2013 at 6:49 PM, Gilad Abada > >> wrote: > >> > Hey Eliot > >> > > >> > Sounds exciting. Which DB will you be using Postgres? I will try to > >> > get my company to pitch in a few dollars. > >> > > >> > Sent from my mobile device. > >> > > >> > On Feb 9, 2013, at 12:11 PM, Eliot Gable > > >> > wrote: > >> > > >> >> For the past two years, I have been working on and off on a high > >> >> availability cluster module for FreeSWITCH. If you would like to see > >> >> HA built into FreeSWITCH directly, instead of dealing with Pacemaker > + > >> >> Corosync or other 3rd party systems, please check out the module, > >> >> voice your support, provide feedback and suggestions, and consider > >> >> supporting it. > >> >> > >> >> Some advantages over a 3rd party solution like Pacemaker include: > >> >> > >> >> - Sub-second failure detection and response (meaning it can start > >> >> recovering calls less than a second after a failure occurs) > >> >> - Failure detection is built into FreeSWITCH so it can detect > >> >> conditions which 3rd party modules cannot detect > >> >> - One shared configuration file between all nodes (no per-node IP > >> >> configuration required) > >> >> - No fixed number of nodes > >> >> - Graceful migration of calls with a maintenance mode (requires DNS) > >> >> > >> >> These are just some of the benefits of building a FreeSWITCH-specific > >> >> HA system. For additional details about the project, please visit: > >> >> > >> >> http://wiki.freeswitch.org/wiki/Mod_ha_cluster > >> >> > >> >> > >> >> If you would like to see this project come to fruition, please > >> >> consider financially supporting it through Kickstarter: > >> >> > >> >> > >> >> > http://www.kickstarter.com/projects/1175121367/high-availability-cluster-module-for-freeswitch > >> >> > >> >> > >> >> > >> >> Thanks in advance for anyone who contributes, either with just a > >> >> shout-out, with feedback and/or suggestions, and especially to those > >> >> who pledge some financial support. > >> >> > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> 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 > > > > > > > > -- > > Marcin Gozdalik > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > 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 > > > > > > -- > Eliot Gable > > "We do not inherit the Earth from our ancestors: we borrow it from our > children." ~David Brower > > "I decided the words were too conservative for me. We're not borrowing > from our children, we're stealing from them--and it's not even > considered to be a crime." ~David Brower > > "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to > live; not live to eat.) ~Marcus Tullius Cicero > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130210/186cb720/attachment.html From avi at avimarcus.net Sun Feb 10 20:17:11 2013 From: avi at avimarcus.net (Avi Marcus) Date: Sun, 10 Feb 2013 19:17:11 +0200 Subject: [Freeswitch-users] High Availability Cluster Module for FreeSWITCH In-Reply-To: References: <-2120316468757263517@unknownmsgid> Message-ID: Steven: Eliot specifically said 2 (or three!) entire networks: each with their own NIC, cables, and network switches. -Avi On Sun, Feb 10, 2013 at 7:08 PM, Steven Ayre wrote: > That covers redundancy in case of a network card or cable failure, but > isn't what partitioning is about. Multiple NICs cannot prevent partitioning. > > As an example, partitioning might happen when a network switch between two > network segments fails so you have nodes A+B in segment 1 able to talk to > each other but unable to talk to nodes C+D in segment 2, while C+D can talk > to each other but not A+B. > > Pacemaker/corosync contain a lot of algorithms to fence off partitions > without quorum and can resort to things like STONITH if required to force a > node to shutdown rather than risk it causing disruption to the cluster (for > example if it tries to take over traffic to a virtual IP you could end up > in a case where you have two servers sending ARP responses for the same IP). > > -Steve > > > > On 10 February 2013 15:27, Eliot Gable wrote: > >> You use multiple NICs in the systems and send heartbeats out all of >> them. There is no other way to do it. Two NICs are required, and 3 are >> recommended. >> >> >> On Sun, Feb 10, 2013 at 6:54 AM, Marcin Gozdalik >> wrote: >> > I wonder how do you handle network partition? Simply "knowing" which >> nodes >> > are up in a distributed system is hard enough and there is a reason why >> > solutions like pacemaker/corosync have years of development behind them. >> > >> > W dniu niedziela, 10 lutego 2013 u?ytkownik Eliot Gable napisa?: >> >> >> >> The beauty of the module is that it will not require any 3rd party >> >> software to operate. I am designing it to run entirely in-memory using >> >> multi-cast events to communicate between cluster nodes. No external >> >> database will be required to enable the automatic fail over and call >> >> recovery. >> >> >> >> If you have your customers / clients using DNS to point traffic to >> >> your system and you run multiple master nodes with DNS performing load >> >> balancing, you can also make use of the maintenance mode I have >> >> planned which will use a separate IP address and update your DNS >> >> records to remove the master which you put into maintenance mode and >> >> put in the maintenance IP which is assigned to the slave node which is >> >> being promoted as master. This allows calls to drain off the master >> >> while new calls come up on the slave, and then you can gracefully shut >> >> down the master when all calls are finished. Once you are done with >> >> maintenance, you simply bring it back online and it comes up as a >> >> slave or a master as needed (according to your configured master / >> >> slave ratio). The IP which it was using as a master node becomes the >> >> new designated maintenance IP. This is the only case where a 3rd party >> >> software package is required, and it is an optional feature. >> >> >> >> >> >> On Sat, Feb 9, 2013 at 6:49 PM, Gilad Abada >> >> wrote: >> >> > Hey Eliot >> >> > >> >> > Sounds exciting. Which DB will you be using Postgres? I will try to >> >> > get my company to pitch in a few dollars. >> >> > >> >> > Sent from my mobile device. >> >> > >> >> > On Feb 9, 2013, at 12:11 PM, Eliot Gable < >> egable+freeswitch at gmail.com> >> >> > wrote: >> >> > >> >> >> For the past two years, I have been working on and off on a high >> >> >> availability cluster module for FreeSWITCH. If you would like to see >> >> >> HA built into FreeSWITCH directly, instead of dealing with >> Pacemaker + >> >> >> Corosync or other 3rd party systems, please check out the module, >> >> >> voice your support, provide feedback and suggestions, and consider >> >> >> supporting it. >> >> >> >> >> >> Some advantages over a 3rd party solution like Pacemaker include: >> >> >> >> >> >> - Sub-second failure detection and response (meaning it can start >> >> >> recovering calls less than a second after a failure occurs) >> >> >> - Failure detection is built into FreeSWITCH so it can detect >> >> >> conditions which 3rd party modules cannot detect >> >> >> - One shared configuration file between all nodes (no per-node IP >> >> >> configuration required) >> >> >> - No fixed number of nodes >> >> >> - Graceful migration of calls with a maintenance mode (requires DNS) >> >> >> >> >> >> These are just some of the benefits of building a >> FreeSWITCH-specific >> >> >> HA system. For additional details about the project, please visit: >> >> >> >> >> >> http://wiki.freeswitch.org/wiki/Mod_ha_cluster >> >> >> >> >> >> >> >> >> If you would like to see this project come to fruition, please >> >> >> consider financially supporting it through Kickstarter: >> >> >> >> >> >> >> >> >> >> http://www.kickstarter.com/projects/1175121367/high-availability-cluster-module-for-freeswitch >> >> >> >> >> >> >> >> >> >> >> >> Thanks in advance for anyone who contributes, either with just a >> >> >> shout-out, with feedback and/or suggestions, and especially to those >> >> >> who pledge some financial support. >> >> >> >> >> >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://wiki.freeswitch.org >> >> http://www.cluecon.com >> >> >> >> 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 >> > >> > >> > >> > -- >> > Marcin Gozdalik >> > >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > 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 >> > >> >> >> >> -- >> Eliot Gable >> >> "We do not inherit the Earth from our ancestors: we borrow it from our >> children." ~David Brower >> >> "I decided the words were too conservative for me. We're not borrowing >> from our children, we're stealing from them--and it's not even >> considered to be a crime." ~David Brower >> >> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to >> live; not live to eat.) ~Marcus Tullius Cicero >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130210/e6819eba/attachment-0001.html From egable+freeswitch at gmail.com Sun Feb 10 21:14:23 2013 From: egable+freeswitch at gmail.com (Eliot Gable) Date: Sun, 10 Feb 2013 13:14:23 -0500 Subject: [Freeswitch-users] High Availability Cluster Module for FreeSWITCH In-Reply-To: References: <-2120316468757263517@unknownmsgid> Message-ID: On Sun, Feb 10, 2013 at 12:08 PM, Steven Ayre wrote: > That covers redundancy in case of a network card or cable failure, but isn't > what partitioning is about. Multiple NICs cannot prevent partitioning. > > As an example, partitioning might happen when a network switch between two > network segments fails so you have nodes A+B in segment 1 able to talk to > each other but unable to talk to nodes C+D in segment 2, while C+D can talk > to each other but not A+B. > > Pacemaker/corosync contain a lot of algorithms to fence off partitions > without quorum and can resort to things like STONITH if required to force a > node to shutdown rather than risk it causing disruption to the cluster (for > example if it tries to take over traffic to a virtual IP you could end up in > a case where you have two servers sending ARP responses for the same IP). > Steve, As Avi pointed out, I mentioned having multiple physical networks as a guard against a network split / partition. If one network is split such that A and B can talk to each other over it and C and D can talk to each other over it, you would indeed have an issue if you only had one network. However, with two or more networks, all four nodes will still be able to talk to each other over the other network(s). Now, granted, if you have a network split in all networks, then you are still screwed. Pacemaker and other solutions deal with this, as you mentioned, using something called "quorum" where you need a majority of nodes to be able to see each other, and they fence the remaining nodes. As I documented on my wiki page for the module, I do have plans to eventually support such functionality. However, that is a bit further down the road as it will take some time to develop STONITH interfaces to various hardware or even to reuse the STONITH modules from Pacemaker or another project. In any case, I feel it is more important to get the base functionality developed and debugged as utilizing multiple networks is a good way to prevent network splits from being an issue. That being said, there are other issues to contend with when discussing network splits. For example, if A and B can see the Internet but C and D cannot, but C is a Master and B is a slave, you still have an issue to address. In this case, mod_ha_cluster must be able to determine that C and D cannot see the Internet. They need to perform very fast pings to some IP address, or have some external host sending them data in some way that they can detect when traffic to/from the Internet has stopped. I can place a media bug on the audio streams to make this determination fairly accurately. I can also rely on a ping mechanism to make the determination. Once the determination is made, mod_ha_cluster then has to promote B to a master to take over C. So, there are certainly still other issues to address when a network split occurs, but split-brain is easily avoided by simply adding redundant networks. From sertys at gmail.com Sun Feb 10 21:43:12 2013 From: sertys at gmail.com (Daniel Ivanov) Date: Sun, 10 Feb 2013 19:43:12 +0100 Subject: [Freeswitch-users] USSD support for mod_gsmopen In-Reply-To: References: Message-ID: A great job that is. Ussd is becoming more and more widespread. I already have some ideas about it. On Feb 10, 2013 12:40 PM, "Giovanni Maruzzelli" wrote: > > On Sun, Feb 10, 2013 at 6:10 AM, Yuriy Ostapchuk wrot > >> Hi, >> I've submitted a patch that implements USSD messaging support. >> (useful for checking and refilling account balance and other >> provider-specific usage) >> >> >> Here is the JIRA ticket: >> http://jira.freeswitch.org/browse/FS-5078 >> >> If this feature is considered useful, I would appreciate if it's >> integrated in git. >> > > Thanks Yuriy, I'll have a look into it asap next week. > > Thanks again, yes is considered useful and grateful received. > > -giovanni > > > -- > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130210/153a4ae5/attachment.html From gozdal at gmail.com Sun Feb 10 23:11:22 2013 From: gozdal at gmail.com (Marcin Gozdalik) Date: Sun, 10 Feb 2013 21:11:22 +0100 Subject: [Freeswitch-users] High Availability Cluster Module for FreeSWITCH In-Reply-To: References: <-2120316468757263517@unknownmsgid> Message-ID: Don't get me wrong, I'd love to fund good HA module for FS, if not for any other reason that I could benefit from that. But having done a few installations of systems that were supposed to be "HA" and seen them fail when real problems came I know it ain't easy. Redundant networks are fine but following scenarios usually lead to both machines reply to ARPs for virtual IP and whole HA falls apart: 1) FS stops responding (e.g. due to heavy swapping or disk full), yet kernel manages to reply to ARPs 2) the HA module fails (like in crashes) but FS manages to work 3) some firewall rule is activated that stops multicast traffic (all or some) STONITH based on separate technology (like USB-USB connection connected to some KVM-over-IP with control over power) is indispensable in such scenarios. 2013/2/10 Eliot Gable : > On Sun, Feb 10, 2013 at 12:08 PM, Steven Ayre wrote: >> That covers redundancy in case of a network card or cable failure, but isn't >> what partitioning is about. Multiple NICs cannot prevent partitioning. >> >> As an example, partitioning might happen when a network switch between two >> network segments fails so you have nodes A+B in segment 1 able to talk to >> each other but unable to talk to nodes C+D in segment 2, while C+D can talk >> to each other but not A+B. >> >> Pacemaker/corosync contain a lot of algorithms to fence off partitions >> without quorum and can resort to things like STONITH if required to force a >> node to shutdown rather than risk it causing disruption to the cluster (for >> example if it tries to take over traffic to a virtual IP you could end up in >> a case where you have two servers sending ARP responses for the same IP). >> > > Steve, > > As Avi pointed out, I mentioned having multiple physical networks as a > guard against a network split / partition. If one network is split > such that A and B can talk to each other over it and C and D can talk > to each other over it, you would indeed have an issue if you only had > one network. However, with two or more networks, all four nodes will > still be able to talk to each other over the other network(s). > > Now, granted, if you have a network split in all networks, then you > are still screwed. Pacemaker and other solutions deal with this, as > you mentioned, using something called "quorum" where you need a > majority of nodes to be able to see each other, and they fence the > remaining nodes. As I documented on my wiki page for the module, I do > have plans to eventually support such functionality. However, that is > a bit further down the road as it will take some time to develop > STONITH interfaces to various hardware or even to reuse the STONITH > modules from Pacemaker or another project. In any case, I feel it is > more important to get the base functionality developed and debugged as > utilizing multiple networks is a good way to prevent network splits > from being an issue. > > That being said, there are other issues to contend with when > discussing network splits. For example, if A and B can see the > Internet but C and D cannot, but C is a Master and B is a slave, you > still have an issue to address. In this case, mod_ha_cluster must be > able to determine that C and D cannot see the Internet. They need to > perform very fast pings to some IP address, or have some external host > sending them data in some way that they can detect when traffic > to/from the Internet has stopped. I can place a media bug on the audio > streams to make this determination fairly accurately. I can also rely > on a ping mechanism to make the determination. Once the determination > is made, mod_ha_cluster then has to promote B to a master to take over > C. > > So, there are certainly still other issues to address when a network > split occurs, but split-brain is easily avoided by simply adding > redundant networks. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -- Marcin Gozdalik From steveayre at gmail.com Sun Feb 10 23:16:54 2013 From: steveayre at gmail.com (Steven Ayre) Date: Sun, 10 Feb 2013 20:16:54 +0000 Subject: [Freeswitch-users] High Availability Cluster Module for FreeSWITCH In-Reply-To: References: <-2120316468757263517@unknownmsgid> Message-ID: Fair enough. I still don't see the need to reinvent the wheel just to avoid depending on a 3rd party well developed and extensively tested piece of software. Corosync provides an API that can be used for passing messages between nodes in a cluster. -Steve On 10 February 2013 18:14, Eliot Gable wrote: > On Sun, Feb 10, 2013 at 12:08 PM, Steven Ayre wrote: > > That covers redundancy in case of a network card or cable failure, but > isn't > > what partitioning is about. Multiple NICs cannot prevent partitioning. > > > > As an example, partitioning might happen when a network switch between > two > > network segments fails so you have nodes A+B in segment 1 able to talk to > > each other but unable to talk to nodes C+D in segment 2, while C+D can > talk > > to each other but not A+B. > > > > Pacemaker/corosync contain a lot of algorithms to fence off partitions > > without quorum and can resort to things like STONITH if required to > force a > > node to shutdown rather than risk it causing disruption to the cluster > (for > > example if it tries to take over traffic to a virtual IP you could end > up in > > a case where you have two servers sending ARP responses for the same IP). > > > > Steve, > > As Avi pointed out, I mentioned having multiple physical networks as a > guard against a network split / partition. If one network is split > such that A and B can talk to each other over it and C and D can talk > to each other over it, you would indeed have an issue if you only had > one network. However, with two or more networks, all four nodes will > still be able to talk to each other over the other network(s). > > Now, granted, if you have a network split in all networks, then you > are still screwed. Pacemaker and other solutions deal with this, as > you mentioned, using something called "quorum" where you need a > majority of nodes to be able to see each other, and they fence the > remaining nodes. As I documented on my wiki page for the module, I do > have plans to eventually support such functionality. However, that is > a bit further down the road as it will take some time to develop > STONITH interfaces to various hardware or even to reuse the STONITH > modules from Pacemaker or another project. In any case, I feel it is > more important to get the base functionality developed and debugged as > utilizing multiple networks is a good way to prevent network splits > from being an issue. > > That being said, there are other issues to contend with when > discussing network splits. For example, if A and B can see the > Internet but C and D cannot, but C is a Master and B is a slave, you > still have an issue to address. In this case, mod_ha_cluster must be > able to determine that C and D cannot see the Internet. They need to > perform very fast pings to some IP address, or have some external host > sending them data in some way that they can detect when traffic > to/from the Internet has stopped. I can place a media bug on the audio > streams to make this determination fairly accurately. I can also rely > on a ping mechanism to make the determination. Once the determination > is made, mod_ha_cluster then has to promote B to a master to take over > C. > > So, there are certainly still other issues to address when a network > split occurs, but split-brain is easily avoided by simply adding > redundant networks. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130210/5a12ae47/attachment-0001.html From avi at avimarcus.net Sun Feb 10 23:40:21 2013 From: avi at avimarcus.net (Avi Marcus) Date: Sun, 10 Feb 2013 22:40:21 +0200 Subject: [Freeswitch-users] High Availability Cluster Module for FreeSWITCH In-Reply-To: References: <-2120316468757263517@unknownmsgid> Message-ID: Steve: Several reasons. One is the mod_ha_cluster is an N+1 Cluster, not a active/passive pair. That way if any single node fails, there's something to pick up the slack. Secondly, in order to recover live calls, you need a list of the calls. That currently requires some sort of odbc (or postgres) with replication. Again, that's abstracted as part of mod_ha_cluster. Third: The docs mention a similar of pooling for registration, that you can register to one server and you're regged on them all without needing a DB to sync everything. Fourth, according to the docs: single configuration for all FS instances, rather than manually ensuring each one has the same config. Fifth: Voicemail clustering? Or we'll have to wait for mod_voicemail's APIs to be rewritten for that, perhaps... There's certainly *something *special possible with mod_ha_cluster that can't be done with existing solutions cleanly, if at all... -Avi On Sun, Feb 10, 2013 at 10:16 PM, Steven Ayre wrote: > Fair enough. > > I still don't see the need to reinvent the wheel just to avoid depending > on a 3rd party well developed and extensively tested piece of software. > Corosync provides an API that can be used for passing messages between > nodes in a cluster. > > -Steve > > > > On 10 February 2013 18:14, Eliot Gable wrote: > >> On Sun, Feb 10, 2013 at 12:08 PM, Steven Ayre >> wrote: >> > That covers redundancy in case of a network card or cable failure, but >> isn't >> > what partitioning is about. Multiple NICs cannot prevent partitioning. >> > >> > As an example, partitioning might happen when a network switch between >> two >> > network segments fails so you have nodes A+B in segment 1 able to talk >> to >> > each other but unable to talk to nodes C+D in segment 2, while C+D can >> talk >> > to each other but not A+B. >> > >> > Pacemaker/corosync contain a lot of algorithms to fence off partitions >> > without quorum and can resort to things like STONITH if required to >> force a >> > node to shutdown rather than risk it causing disruption to the cluster >> (for >> > example if it tries to take over traffic to a virtual IP you could end >> up in >> > a case where you have two servers sending ARP responses for the same >> IP). >> > >> >> Steve, >> >> As Avi pointed out, I mentioned having multiple physical networks as a >> guard against a network split / partition. If one network is split >> such that A and B can talk to each other over it and C and D can talk >> to each other over it, you would indeed have an issue if you only had >> one network. However, with two or more networks, all four nodes will >> still be able to talk to each other over the other network(s). >> >> Now, granted, if you have a network split in all networks, then you >> are still screwed. Pacemaker and other solutions deal with this, as >> you mentioned, using something called "quorum" where you need a >> majority of nodes to be able to see each other, and they fence the >> remaining nodes. As I documented on my wiki page for the module, I do >> have plans to eventually support such functionality. However, that is >> a bit further down the road as it will take some time to develop >> STONITH interfaces to various hardware or even to reuse the STONITH >> modules from Pacemaker or another project. In any case, I feel it is >> more important to get the base functionality developed and debugged as >> utilizing multiple networks is a good way to prevent network splits >> from being an issue. >> >> That being said, there are other issues to contend with when >> discussing network splits. For example, if A and B can see the >> Internet but C and D cannot, but C is a Master and B is a slave, you >> still have an issue to address. In this case, mod_ha_cluster must be >> able to determine that C and D cannot see the Internet. They need to >> perform very fast pings to some IP address, or have some external host >> sending them data in some way that they can detect when traffic >> to/from the Internet has stopped. I can place a media bug on the audio >> streams to make this determination fairly accurately. I can also rely >> on a ping mechanism to make the determination. Once the determination >> is made, mod_ha_cluster then has to promote B to a master to take over >> C. >> >> So, there are certainly still other issues to address when a network >> split occurs, but split-brain is easily avoided by simply adding >> redundant networks. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130210/1405b24b/attachment.html From egable+freeswitch at gmail.com Mon Feb 11 00:00:21 2013 From: egable+freeswitch at gmail.com (Eliot Gable) Date: Sun, 10 Feb 2013 16:00:21 -0500 Subject: [Freeswitch-users] High Availability Cluster Module for FreeSWITCH In-Reply-To: References: <-2120316468757263517@unknownmsgid> Message-ID: On Sun, Feb 10, 2013 at 3:11 PM, Marcin Gozdalik wrote: > Don't get me wrong, I'd love to fund good HA module for FS, if not for > any other reason that I could benefit from that. > But having done a few installations of systems that were supposed to > be "HA" and seen them fail when real problems came I know it ain't > easy. > Redundant networks are fine but following scenarios usually lead to > both machines reply to ARPs for virtual IP and whole HA falls apart: > > 1) FS stops responding (e.g. due to heavy swapping or disk full), yet > kernel manages to reply to ARPs > 2) the HA module fails (like in crashes) but FS manages to work > 3) some firewall rule is activated that stops multicast traffic (all or > some) > > STONITH based on separate technology (like USB-USB connection > connected to some KVM-over-IP with control over power) is > indispensable in such scenarios. > I have also seen each of these cases when dealing with HA setups for FreeSWITCH. That is part of why I want to write one specifically for FreeSWITCH. General purpose HA systems cannot catch and properly deal with the sorts of things I see when using one to run FreeSWITCH in a high availability configuration. For #1, it is pretty easy to detect when FS stops responding for whatever reason. A watchdog thread inside FS can shut it down in a lot of those cases, as well as remove the IP from the system. Aside from that, future fencing will be useful to bounce the entire machine if the watchdog thread does not or cannot handle the situation. This is another advantage of doing the module inside FS. For #2, a lot of testing is required to make sure that if the HA system fails, it takes down FS with it, or to ensure that it cannot simply "not work" while FS continues to work. It's very hard for a module inside FS to fail in a way that FS keeps working yet the module doesn't. When using Pacemaker and Corosync, it is easy for the HA system to fail in a way that leaves FS running on a node yet the HA system thinks it is not there. This is one of the advantages of doing a module inside FS. For #3, I would like to eventually have the module scanning the firewall rules for changes and enforcing a specific, pre-determined "known-good" set of firewall rules. That is a ways off, but it is planned. Besides that, this is yet one more reason why having the module in FS is the best option. When running in FS, it is easy to determine if the traffic is being blocked by a firewall rule (you simply do not receive the traffic). Again, a module can more effectively catch and respond to this type of situation compared to a general purpose solution like Pacemaker and Corosync. I do not disagree about STONITH being indispensable in certain cases. However, most companies I have worked with who try to do HA right now using Pacemaker and Corosync completely ignore STONITH entirely. STONITH is another case where the biggest problem is simply getting people to follow best practices. And, interestingly enough, most companies actually get by just fine without deploying a STONITH solution. They simply accept that there is a risk of hitting a case where they need it and will not have it, and they just plan to have someone manually reset something if that case arises. This is why STONITH is not one of my first priorities in development of the module. It is on the list because it is important (and I feel it should be used in any carrier-grade deployment), but most companies are perfectly OK to do without it. This is especially true for companies who want HA on their home-grown PBX systems for their 10-person organization. They couldn't care less whether the reset happens manually or automatically in 1 out of 1,000 failures. As long as those other 999 failures are automatic, they will be happy. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130210/30610730/attachment.html From egable+freeswitch at gmail.com Mon Feb 11 00:21:00 2013 From: egable+freeswitch at gmail.com (Eliot Gable) Date: Sun, 10 Feb 2013 16:21:00 -0500 Subject: [Freeswitch-users] High Availability Cluster Module for FreeSWITCH In-Reply-To: References: <-2120316468757263517@unknownmsgid> Message-ID: On Sun, Feb 10, 2013 at 3:40 PM, Avi Marcus wrote: > Steve: > Several reasons. > One is the mod_ha_cluster is an N+1 Cluster, not a active/passive pair. > That way if any single node fails, there's something to pick up the slack. > > Secondly, in order to recover live calls, you need a list of the calls. > That currently requires some sort of odbc (or postgres) with replication. > Again, that's abstracted as part of mod_ha_cluster. > > Third: The docs mention a similar of pooling for registration, that you > can register to one server and you're regged on them all without needing a > DB to sync everything. > > Fourth, according to the docs: single configuration for all FS instances, > rather than manually ensuring each one has the same config. > > Fifth: Voicemail clustering? Or we'll have to wait for mod_voicemail's > APIs to be rewritten for that, perhaps... > > There's certainly *something *special possible with mod_ha_cluster that > can't be done with existing solutions cleanly, if at all... > > -Avi > Thanks, Avi, for the voice of support, and for actually reading my documentation. :) First, however, I need to correct you on one point. I have actually designed mod_ha_cluster as a N + x cluster, meaning N master nodes and x slave nodes. What this means is that you can get better availability than having 1 slave node, while also getting more cost effective scalability than having 1 slave node per master (which is all that is really possible with most other solutions). So, for example, you could run 16 master nodes and 3 slave nodes in the clusters. In addition, you do not need to deal with the cost of a database system to synchronize the calls and registration information between all the nodes. Keep in mind, also, that you would also then have to have that database system set up in a high availability configuration, and it would have to be a pretty big box to handle the load from so many servers. So, bottom line is that mod_ha_cluster can dramatically reduce the cost of deploying a high availability solution for FreeSWITCH, as well as ease the configuration. The reduced deployment complexity not only makes it more approachable for a much larger audience, it can also actually improve availability simply due to fewer components involved and fewer places failures or bugs to cause issues. Also, as you pointed out, this is more than simply a high availability module. It is also a clustering module. This means it shares registration information, call limiting information, call state, and all sorts of other essential information. Using a shared disk resource, you can have voicemail and menus shared between multiple nodes and your entire set of potentially dozens of nodes can act as one single cluster on a single DNS address. Because of the N + x design, you can use smaller nodes in your cluster and scale by adding more nodes vs adding more horsepower to individual nodes. And, as Avi pointed out, this means that if a single node fails, you have other nodes to pick up the slack. Using Pacemaker and Corosync with an active/passive configuration, 16 master nodes requires an additional 16 slave nodes of the same size. That is a lot of unused and expensive gear. Using mod_ha_cluster, you could take those 32 nodes and run 4 as slaves while the other 28 are active. That's a much more efficient and effective design. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130210/539761cc/attachment-0001.html From steveayre at gmail.com Mon Feb 11 00:32:07 2013 From: steveayre at gmail.com (Steven Ayre) Date: Sun, 10 Feb 2013 21:32:07 +0000 Subject: [Freeswitch-users] High Availability Cluster Module for FreeSWITCH In-Reply-To: References: <-2120316468757263517@unknownmsgid> Message-ID: > > One is the mod_ha_cluster is an N+1 Cluster, not a active/passive pair. > That way if any single node fails, there's something to pick up the slack. Corosync clusters are not limited to active/passive pairs. It's just a very common setup. For example you could have resource agents 1) to keep FS running on all nodes 2) for virtual IPs 3) for IP:port Sofia profiles. You can then define dependancies between them. That should let you keep FS running at all times and move an IP and the associated Sofia profiles to a new node that's already running FS when the original node fails. For maintance you can simply trigger that from the CRM. Secondly, in order to recover live calls, you need a list of the calls. > That currently requires some sort of odbc (or postgres) with replication. > Again, that's abstracted as part of mod_ha_cluster. > Third: The docs mention a similar of pooling for registration, that you > can register to one server and you're regged on them all without needing a > DB to sync everything. Which can also be done using Corosync's IPC messaging API. (Personally I prefer using MySQL Cluster via ODBC - which is HA, synchronous and offloads all load off of the FS nodes, but that's offtopic). Fourth, according to the docs: single configuration for all FS instances, > rather than manually ensuring each one has the same config. This could be achieved with the Corosync API. Fifth: Voicemail clustering? Or we'll have to wait for mod_voicemail's APIs > to be rewritten for that, perhaps... That's not going to happen without a storage API added to the FS core - you're always going to need some 3rd party such NFS, CIFS, DRBD etc. mod_voicemail is hardcoded to use the ODBC interface but that's only for the index not the recordings.nc Corosync would allow you to make FS depend on the NFS/whatever service running so if the storage backend has failed FS would move to a node where it is available - not necessarily possibly from within mod_ha_cluster itself. There's certainly *something *special possible with mod_ha_cluster that > can't be done with existing solutions cleanly, if at all... Don't misunderstand me, I think it's a great idea to have a module aimed at HA and sharing state across a cluster while being able to detect new failure conditions. I just think that in the specific area of node monitoring and messaging across the cluster it would be better to use a well tested and proven solution such as Corosync which is based on a large number of papers, algorithms, and generally decades of work. Every time I've seen/heard of an attempt to redo that from scratch it's been unreliable especially in unexpected failure conditions. Simply because it's a dependency on another program isn't a good enough reason not to use it - you're already depending on various other programs (Linux, sysvinit, monit, cron, syslog etc) anyway. By not using it you're just adding extra work, adding unnecessarily complexity and increasing the risk of bugs. Corosync would also have advantages because the tools to migrate services to another node for maintenance, detect and restart resources on failure etc already exist. -Steve On 10 February 2013 20:40, Avi Marcus wrote: > Steve: > Several reasons. > One is the mod_ha_cluster is an N+1 Cluster, not a active/passive pair. > That way if any single node fails, there's something to pick up the slack. > > Secondly, in order to recover live calls, you need a list of the calls. > That currently requires some sort of odbc (or postgres) with replication. > Again, that's abstracted as part of mod_ha_cluster. > > Third: The docs mention a similar of pooling for registration, that you > can register to one server and you're regged on them all without needing a > DB to sync everything. > > Fourth, according to the docs: single configuration for all FS instances, > rather than manually ensuring each one has the same config. > > Fifth: Voicemail clustering? Or we'll have to wait for mod_voicemail's > APIs to be rewritten for that, perhaps... > > There's certainly *something *special possible with mod_ha_cluster that > can't be done with existing solutions cleanly, if at all... > > -Avi > > On Sun, Feb 10, 2013 at 10:16 PM, Steven Ayre wrote: > >> Fair enough. >> >> I still don't see the need to reinvent the wheel just to avoid depending >> on a 3rd party well developed and extensively tested piece of software. >> Corosync provides an API that can be used for passing messages between >> nodes in a cluster. >> >> -Steve >> >> >> >> On 10 February 2013 18:14, Eliot Gable wrote: >> >>> On Sun, Feb 10, 2013 at 12:08 PM, Steven Ayre >>> wrote: >>> > That covers redundancy in case of a network card or cable failure, but >>> isn't >>> > what partitioning is about. Multiple NICs cannot prevent partitioning. >>> > >>> > As an example, partitioning might happen when a network switch between >>> two >>> > network segments fails so you have nodes A+B in segment 1 able to talk >>> to >>> > each other but unable to talk to nodes C+D in segment 2, while C+D can >>> talk >>> > to each other but not A+B. >>> > >>> > Pacemaker/corosync contain a lot of algorithms to fence off partitions >>> > without quorum and can resort to things like STONITH if required to >>> force a >>> > node to shutdown rather than risk it causing disruption to the cluster >>> (for >>> > example if it tries to take over traffic to a virtual IP you could end >>> up in >>> > a case where you have two servers sending ARP responses for the same >>> IP). >>> > >>> >>> Steve, >>> >>> As Avi pointed out, I mentioned having multiple physical networks as a >>> guard against a network split / partition. If one network is split >>> such that A and B can talk to each other over it and C and D can talk >>> to each other over it, you would indeed have an issue if you only had >>> one network. However, with two or more networks, all four nodes will >>> still be able to talk to each other over the other network(s). >>> >>> Now, granted, if you have a network split in all networks, then you >>> are still screwed. Pacemaker and other solutions deal with this, as >>> you mentioned, using something called "quorum" where you need a >>> majority of nodes to be able to see each other, and they fence the >>> remaining nodes. As I documented on my wiki page for the module, I do >>> have plans to eventually support such functionality. However, that is >>> a bit further down the road as it will take some time to develop >>> STONITH interfaces to various hardware or even to reuse the STONITH >>> modules from Pacemaker or another project. In any case, I feel it is >>> more important to get the base functionality developed and debugged as >>> utilizing multiple networks is a good way to prevent network splits >>> from being an issue. >>> >>> That being said, there are other issues to contend with when >>> discussing network splits. For example, if A and B can see the >>> Internet but C and D cannot, but C is a Master and B is a slave, you >>> still have an issue to address. In this case, mod_ha_cluster must be >>> able to determine that C and D cannot see the Internet. They need to >>> perform very fast pings to some IP address, or have some external host >>> sending them data in some way that they can detect when traffic >>> to/from the Internet has stopped. I can place a media bug on the audio >>> streams to make this determination fairly accurately. I can also rely >>> on a ping mechanism to make the determination. Once the determination >>> is made, mod_ha_cluster then has to promote B to a master to take over >>> C. >>> >>> So, there are certainly still other issues to address when a network >>> split occurs, but split-brain is easily avoided by simply adding >>> redundant networks. >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130210/71d7bb26/attachment.html From dvl36.ripe.nick at gmail.com Mon Feb 11 01:59:36 2013 From: dvl36.ripe.nick at gmail.com (dvl36) Date: Sun, 10 Feb 2013 14:59:36 -0800 (PST) Subject: [Freeswitch-users] High Availability Cluster Module for FreeSWITCH In-Reply-To: References: <-2120316468757263517@unknownmsgid> Message-ID: <1360537176787-7587194.post@n2.nabble.com> For case #1: How watchdog thread will remove ip address if freeswitch coredumped? I think this should be done by other process. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/High-Availability-Cluster-Module-for-FreeSWITCH-tp7587172p7587194.html Sent from the freeswitch-users mailing list archive at Nabble.com. From egable+freeswitch at gmail.com Mon Feb 11 04:07:14 2013 From: egable+freeswitch at gmail.com (Eliot Gable) Date: Sun, 10 Feb 2013 20:07:14 -0500 Subject: [Freeswitch-users] High Availability Cluster Module for FreeSWITCH In-Reply-To: References: <-2120316468757263517@unknownmsgid> Message-ID: On Sun, Feb 10, 2013 at 4:32 PM, Steven Ayre wrote: > One is the mod_ha_cluster is an N+1 Cluster, not a active/passive pair. >> That way if any single node fails, there's something to pick up the slack. > > > Corosync clusters are not limited to active/passive pairs. It's just a > very common setup. > > For example you could have resource agents 1) to keep FS running on all > nodes 2) for virtual IPs 3) for IP:port Sofia profiles. You can then define > dependancies between them. That should let you keep FS running at all times > and move an IP and the associated Sofia profiles to a new node that's > already running FS when the original node fails. For maintance you can > simply trigger that from the CRM. This is true, however it would require a very complicated resource agent to manage FreeSWITCH in a similar configuration to what mod_ha_cluster is designed to do. In addition, the functionality simply does not exist in FreeSWITCH right now to tell it to take over for a specific failed master and recover those specific calls. So, right now, using Pacemaker and Corosync, there is absolutely no way to run a N + x FreeSWITCH cluster. Also, the response time on a Pacemaker + Corosync cluster for failure detection and recovery is measured in seconds, which is not ideal for a real-time communications platform. Obviously, there is nothing at all preventing you from running Pacemaker and Corosync in addition to mod_ha_cluster. In fact, I was even considering providing some CLI arguments to allow FreeSWITCH (with mod_ha_cluster enabled) to be commanded from Pacemaker and act as its own resource agent. If you think that would be an interesting feature, I can look into what it would take to work that out. I previously wrote a resource agent for Broadvox when I worked for them, and I wrote another one after leaving them. Both were intended to manage FreeSWITCH as a master/slave pair, so I have some idea on how to do it, just not necessarily the specifics on making it do N + x the way I intend to have mod_ha_cluster operate. > Secondly, in order to recover live calls, you need a list of the calls. >> That currently requires some sort of odbc (or postgres) with replication. >> Again, that's abstracted as part of mod_ha_cluster. >> Third: The docs mention a similar of pooling for registration, that you >> can register to one server and you're regged on them all without needing a >> DB to sync everything. > > > Which can also be done using Corosync's IPC messaging API. > > (Personally I prefer using MySQL Cluster via ODBC - which is > HA, synchronous and offloads all load off of the FS nodes, but that's > offtopic). > > I have used MySQL cluster and Postgres with my own replication daemon. Neither are ideal solutions for a multitude of reasons. Also, using something like Corosync's IPC messaging is not ideal either. Besides that, the FreeSWITCH core has a sufficiently robust API for doing network messaging, and also its own event system which happens to be perfectly suited for exactly the kind of messaging I need to accomplish. Using the Corosync IPC messaging API would be like trying to shove a large round peg through a small square hole. > Fourth, according to the docs: single configuration for all FS instances, >> rather than manually ensuring each one has the same config. > > > This could be achieved with the Corosync API. > > Fifth: Voicemail clustering? Or we'll have to wait for mod_voicemail's >> APIs to be rewritten for that, perhaps... > > > That's not going to happen without a storage API added to the FS core - > you're always going to need some 3rd party such NFS, CIFS, DRBD etc. > mod_voicemail is hardcoded to use the ODBC interface but that's only for > the index not the recordings.nc > > Corosync would allow you to make FS depend on the NFS/whatever service > running so if the storage backend has failed FS would move to a node where > it is available - not necessarily possibly from within mod_ha_cluster > itself. > Well, I *could* broadcast an entire voicemail to all nodes, but that seems like a bit of overkill, and I don't see any real reason to code a storage system into FreeSWITCH. That really would be reinventing the wheel. I don't think the majority of users would find it all that difficult to set up a shared NFS space somewhere, or even use something like MooseFS. In fact, MooseFS is so easy to use, I went from reading about it for the first time to having a fully functional 6-node shared storage cluster in about 45 minutes. This is one place where a 3rd party solution is really the optimal approach. > There's certainly *something *special possible with mod_ha_cluster that >> can't be done with existing solutions cleanly, if at all... > > > Don't misunderstand me, I think it's a great idea to have a module aimed > at HA and sharing state across a cluster while being able to detect new > failure conditions. > > I just think that in the specific area of node monitoring and messaging > across the cluster it would be better to use a well tested and proven > solution such as Corosync which is based on a large number of papers, > algorithms, and generally decades of work. Every time I've seen/heard of an > attempt to redo that from scratch it's been unreliable especially in > unexpected failure conditions. Simply because it's a dependency on another > program isn't a good enough reason not to use it - you're already depending > on various other programs (Linux, sysvinit, monit, cron, syslog etc) > anyway. By not using it you're just adding extra work, adding unnecessarily > complexity and increasing the risk of bugs. Corosync would also have > advantages because the tools to migrate services to another node > for maintenance, detect and restart resources on failure etc already exist. > I think you greatly overestimate the complexity of the messaging and node monitoring required to make FS run as a multi-master, multi-slave cluster, detect a node failure, and have it automatically recover calls on a slave system. You seem to be thinking of the task as if it were a general purpose HA solution which needs to work for any random piece of software across every imaginable network configuration and software deployment scenario. The messaging I need to do is simple: 1) I need to send events from the local FS node to all other slave nodes in the cluster so they can synchronize call state, registration information, call limiting information, etc. 2) I need to send heartbeats from all nodes to all nodes out all configured NICs so they can keep track of who is in which state and which is the designated slave in case of failure. These two tasks accomplish everything needed to make this system work. The heartbeats let each node calculate the state of the cluster. There is no "single brain" in the cluster. There is a very specific and well-defined set of rules by which the state of the cluster is calculated by each node, and all nodes will always arrive at the same conclusion so long as you have multiple physical networks to ensure communication is never broken between sets of nodes. The decision making happens on the slave nodes only. The masters are already in their role and will stay that way unless something catastrophic occurs. They have ways to detect internal failures, and I will have a way set up for them to detect and deal with a segfault of FS, as well. If such a failure occurs, their single purpose task is to shut down as cleanly as possible and get themselves back into a stable state. The slaves also have only one goal: to recover the calls of the failed master and become the new master. One slave is always designated as the primary candidate, and all slaves know which it is at all times (once they exchange their first heartbeats). They also know which is the secondary, tertiary, etc. It is all pre-determined up-front and only changes if a failure, reconfiguration, or maintenance event occurs. When the primary sees a master go down, it will perform a sanity check on itself first (to make sure it didn't have an issue) and then take over for the first master it saw go down. Once it has started that process, it broadcasts that it is switching to master and which master it is taking over for. At this point, the secondary immediately becomes primary to all remaining slaves and it is immediately available to take over for any other master nodes which might also have failed simultaneously. The whole process is a very specific and very orchestrated process. The whole thing is barely comparable to the complexity of the failure detection and fail over process which Pacemaker has to deal with. I am not trying to recreate something like Pacemaker. It is a hugely complex system which has to deal with all sorts of random and generalized configurations of endless types of software. Pacemaker is like an interstellar spacecraft while I just need a rocket to put a satellite in orbit. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130210/58d0243d/attachment-0001.html From dvl36.ripe.nick at gmail.com Mon Feb 11 04:44:03 2013 From: dvl36.ripe.nick at gmail.com (Dmitry Lysenko) Date: Mon, 11 Feb 2013 03:44:03 +0200 Subject: [Freeswitch-users] Reg. Freeswitch Performace on Freescale Board In-Reply-To: <5114EDE0.6020400@cem-solutions.net> References: <5110FD13.3080406@cem-solutions.net> <51121A98.6070109@cem-solutions.net> <5114EDE0.6020400@cem-solutions.net> Message-ID: Hello, Varghese, did you try to increase ptime? 2013/2/8 Varghese > Hi all, > > Any update on this issue. Any other options we need to try with freeswitch. > > E1 PRI with freetdm freeswitch in freescale and cpu load is going to 100 > with 15 calls. > > Thanks And Regards > > Varghese Paul > > Hi Anthony, > > > > Thanks for the information. > > > > Kernel is Linux-2.6.32 patched by Freescale team for the hardware. > > > > Further we tested the freeswitch-1.2.6 on Freescale and following are > > observations. > > > > We set the ulimit options as per the wiki link and ulimit ?s 240 > > > > 1. With switches *-np* and *?lp* the result is same i.e. maximum of 90 > > calls and top command shows 100% CPU for freeswitch process > > > > 2. With RTP Bypass mode we could make 250 calls and CPU usage is 20% > > > > 3. With Proxy media mode, and default rtp mode the CPU is very high. > > Approximately 1 to 1.5% per call and scales up to 90 calls > > > > 4. Set the option enable-timer value = false in SIP profile, > > stun-enabled=false given all IP address in ext-rtp, > > rtp-timer-name=none/soft etc., still the results are same > > > > Q1) Can you explain ?asterisk also uses blocking reads in its rtp > > stack where we have timers. You can disable the timers as documented > > in the wiki? which parameter you are referring to and where we need to > > disable timers. > > > > Q2) Do we need to try any other options to increase the performance > > numbers? > > > > F.Y.I, We used Empirix Hammer SIP Call Generator for testing SIP calls. > > > > Thanks > > Varghese Paul > > > > Anthony Minessale wrote: > >> That depends on the version of linux, the kernel version and several > >> other factors. > >> > >> Were you actually moving media when testing, what tool are you > >> testing with? > >> > >> On 32 bit you should make sure you have the stack size at 240 (ulimit > >> -s 240 before starting) > >> Also you may need to add the -lp or -np startup flags to reduce the > >> realtime threads. > >> > >> asterisk also uses blocking reads in its rtp stack where we have > >> timers. You can disable the timers as documented in the wiki.. > >> > >> > >> > >> > >> On Tue, Feb 5, 2013 at 6:37 AM, Varghese > >> > > >> wrote: > >> > >> Hi all, > >> We are facing following observations/issues with freeswitch on > >> freescale > >> processor boards. > >> We ported freeswitch with freescale provided tool chain. > >> _*Freeswitch Configuration: version 1.2.6 (git version)*_ > >> 1. Only enabled mod_sofia, mod_dialplan, mod_command,mod_dptool and > >> mod_console. > >> 2. set the ulimit options and followed the steps in the following > >> link > >> > >> http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations > >> 3. SIP test tool run the load test for 500 users in freeswitch and > >> configured the dial plan to just bridge the channels > >> 4. Freeswitch media processing is in default mode: i.e media will go > >> through freeswitch > >> _*Freescale Board Configuration:*_ > >> CPU : 1GHZ PowerQUICC III, 32 bit > >> RAM: 1 GB > >> Linux- 2.6.32 > >> Load test is conducted for SIP to SIP calls with RTP. > >> _*Observations:*_ > >> Linux TOP command shows increasing CPU usage per call and > >> increases to > >> 100% with only 90 calls. After that system response is very slow > >> We could also found delay in media. > >> > >> _Surprisingly, with Asterisk 1.8 the same system configuration > >> works 250 > >> calls without any problems. > >> _ > >> _*Questions: > >> *_1. Any one ported Freeswitch on Freescale or any embedded > >> processors? > >> Any performance figures or references will be appreciated > >> 2. How many simultaneous calls can be possible with 1GHz > >> PowerQUICC III > >> processor ? > >> 3. Any more settings are required in freeswitch or linux for > >> scalability? > >> > >> Thanks And Regards > >> > >> Varghese Paul > >> > >> > >> > >> > _________________________________________________________________________ > >> > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> 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 > >> > >> > >> > >> > >> -- > >> Anthony Minessale II > >> > >> FreeSWITCH http://www.freeswitch.org/ > >> ClueCon http://www.cluecon.com/ > >> Twitter: http://twitter.com/FreeSWITCH_wire > >> > >> AIM: anthm > >> MSN:anthony_minessale at hotmail.com > >> > >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > >> > >> IRC: irc.freenode.net #freeswitch > >> > >> FreeSWITCH Developer Conference > >> sip:888 at conference.freeswitch.org > >> > >> googletalk:conf+888 at conference.freeswitch.org > >> > >> pstn:+19193869900 > >> ------------------------------------------------------------------------ > >> > >> > _________________________________________________________________________ > >> > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> 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 > >> > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130211/9b17f599/attachment.html From bdfoster at endigotech.com Mon Feb 11 06:19:24 2013 From: bdfoster at endigotech.com (Brian Foster) Date: Sun, 10 Feb 2013 22:19:24 -0500 Subject: [Freeswitch-users] High Availability Cluster Module for FreeSWITCH In-Reply-To: References: Message-ID: I applaud your efforts on getting a specialized mechanism in the form of a freeswitch module implemented to correctly perform high availability functions. I see a great potential here for the freeswitch project to attain a whole new level of performance and viability, and I really really hope others agree. I've been struggling with this very issue; how to do correct HA with freeswitch to the point where we aren't dropping calls when something bad happens. It's held me back in a few ways, and while ill never look into a non-open source solution, I've got way to much on my plate right now to play with it now. I'll see if my company can send some $$ your way. Ive got a few clients who would be interested in this as well. People, start donating! Sent from my iPhone On Feb 9, 2013, at 12:00 PM, Eliot Gable wrote: > For the past two years, I have been working on and off on a high > availability cluster module for FreeSWITCH. If you would like to see > HA built into FreeSWITCH directly, instead of dealing with Pacemaker + > Corosync or other 3rd party systems, please check out the module, > voice your support, provide feedback and suggestions, and consider > supporting it. > > Some advantages over a 3rd party solution like Pacemaker include: > > - Sub-second failure detection and response (meaning it can start > recovering calls less than a second after a failure occurs) > - Failure detection is built into FreeSWITCH so it can detect > conditions which 3rd party modules cannot detect > - One shared configuration file between all nodes (no per-node IP > configuration required) > - No fixed number of nodes > - Graceful migration of calls with a maintenance mode (requires DNS) > > These are just some of the benefits of building a FreeSWITCH-specific > HA system. For additional details about the project, please visit: > > http://wiki.freeswitch.org/wiki/Mod_ha_cluster > > > If you would like to see this project come to fruition, please > consider financially supporting it through Kickstarter: > > http://www.kickstarter.com/projects/1175121367/high-availability-cluster-module-for-freeswitch > > > > Thanks in advance for anyone who contributes, either with just a > shout-out, with feedback and/or suggestions, and especially to those > who pledge some financial support. > > -- > Eliot Gable > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 benoit.raymond at amiconcept.com Mon Feb 11 06:42:32 2013 From: benoit.raymond at amiconcept.com (Benoit Raymond) Date: Sun, 10 Feb 2013 22:42:32 -0500 Subject: [Freeswitch-users] How to handle refer calls when ATA has call foward turned on Message-ID: <050501ce0809$d30ebf50$792c3df0$@amiconcept.com> Hi gang, I want to keep the *72 feature on my Cisco SPA122 ATAs so I don't have to deal keeping the call forward numbers in FS. But in the standard dialplan, I hit the deflect application and since the call hasn't been answered yet, I assume I need to use the redirect or even the transfer application. Despite my attempts, I am struggling at finding the right way to deal with this refer call. Anybody has a dialplan recipe to handle the following scenarios? External Gatewasy -> FS (bridge app) -> ATA -> REFER -> FS (???) -> External Gateway Internal device -> FS (bridge app) -> ATA -> REFER -> FS(???) -> External Gateway My most wanted scenario is an inbound call coming from the External Gateway. Once FS gets the REFER from the ATA, I need to connect the a-leg to the proper destination but instead, call is dropped. I am a bit at lost here so any hint is appreciated. Benoit -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130210/b9b1f2e2/attachment-0001.html From vbvbrj at gmail.com Mon Feb 11 10:06:06 2013 From: vbvbrj at gmail.com (Mimiko) Date: Mon, 11 Feb 2013 09:06:06 +0200 Subject: [Freeswitch-users] VoIP gateway with PRI In-Reply-To: References: <51140221.4040107@gmail.com> Message-ID: <5118985E.6090008@gmail.com> On 10.02.2013 11:21, Yehavi Bourvine wrote: > Hi, > I just checked with our Audiocodes support, and you can do what you want. > In order to route a call from one PRI to the other, you insert an entry > in two tables: > Tel->IP: the destination is localhost. > IP->Tel - the source is either any or localhost. > Regards, __Yehavi: Ok. thank you. -- Mimiko desu. From mwhapples at aim.com Sun Feb 10 23:05:03 2013 From: mwhapples at aim.com (Michael Whapples) Date: Sun, 10 Feb 2013 20:05:03 +0000 Subject: [Freeswitch-users] Trouble autostarting freeswitch on boot Message-ID: <5117FD6F.3050309@aim.com> Hello, I am trying to work out how to get freeswitch to be automatically be started when I boot the system. First of all I tried using ArchLinux to run freeswitch as that is the distribution of Linux I am most familiar with. I compiled freeswitch using the AUR package (https://aur.archlinux.org/packages/freeswitch-git/). If I just run freeswitch from the command line it runs fine. As ArchLinux uses systemd for start up, I tried the systemd service file linked to in the AUR package and also tried modifying the one included from the debian directory (modifying paths as appropriate). Neither worked, the AUR package I think gave exit codes of 214 and the freeswitch debian one I cannot remember (although its probably the same as I am getting with debian). I decided as the Debian one was written for Debian, and may be there might be something Debian specific in that service file I would try on a Debian system. Still no luck, the service just fails to start, although running freeswitch from the command line works fine. On Debian I first got an error from the mkdir command (looked like it was to do with the directory not existing, probably caused by the WorkingDirectory line referring to the directory the mkdir command creates) so I decided to create /run/freeswitch manually and then try again to start the freeswitch service, but it still fails, this time at the launching of freeswitch which gets the SEGV signal. Also on debian I tried the sysvinit script from the debian folder of the freeswitch distribution, and while I get a message appearing saying backgrounding the process, freeswitch fails to continue to run. Ideally I want it so that I can control the freeswitch process, so using systemd seems a reasonable choice, but I will consider other options which meet that ability to control/monitor the process. A search on the internet does not seem to be yielding any answers to this problem. How can I get this running? I might consider another distribution, but it must be one which will run on the raspberrypi as that will be where it will be run in the end. I did try the ArchLinux stuff on both an X86_64 system as well as the raspberrypi, Debian, I mainly tried on the RaspberryPi using the raspbian distribution. Any help would be welcome as I have reached the point that I cannot think of anything else to try other than another system than FreeSwitch. Michael Whapples From gerald.weber at besharp.at Mon Feb 11 12:31:11 2013 From: gerald.weber at besharp.at (Gerald Weber) Date: Mon, 11 Feb 2013 09:31:11 +0000 Subject: [Freeswitch-users] High Availability Cluster Module for FreeSWITCH In-Reply-To: References: Message-ID: Just my 2 eurocent, i know you said without 3rd party solution. But wouldn't this be a perfect task for Erlangs distribution and fault tolerance capabilities ? -----Urspr?ngliche Nachricht----- Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Brian Foster Gesendet: Montag, 11. Februar 2013 04:19 An: FreeSWITCH Users Help Betreff: Re: [Freeswitch-users] High Availability Cluster Module for FreeSWITCH I applaud your efforts on getting a specialized mechanism in the form of a freeswitch module implemented to correctly perform high availability functions. I see a great potential here for the freeswitch project to attain a whole new level of performance and viability, and I really really hope others agree. I've been struggling with this very issue; how to do correct HA with freeswitch to the point where we aren't dropping calls when something bad happens. It's held me back in a few ways, and while ill never look into a non-open source solution, I've got way to much on my plate right now to play with it now. I'll see if my company can send some $$ your way. Ive got a few clients who would be interested in this as well. People, start donating! Sent from my iPhone On Feb 9, 2013, at 12:00 PM, Eliot Gable wrote: > For the past two years, I have been working on and off on a high > availability cluster module for FreeSWITCH. If you would like to see > HA built into FreeSWITCH directly, instead of dealing with Pacemaker + > Corosync or other 3rd party systems, please check out the module, > voice your support, provide feedback and suggestions, and consider > supporting it. > > Some advantages over a 3rd party solution like Pacemaker include: > > - Sub-second failure detection and response (meaning it can start > recovering calls less than a second after a failure occurs) > - Failure detection is built into FreeSWITCH so it can detect > conditions which 3rd party modules cannot detect > - One shared configuration file between all nodes (no per-node IP > configuration required) > - No fixed number of nodes > - Graceful migration of calls with a maintenance mode (requires DNS) > > These are just some of the benefits of building a FreeSWITCH-specific > HA system. For additional details about the project, please visit: > > http://wiki.freeswitch.org/wiki/Mod_ha_cluster > > > If you would like to see this project come to fruition, please > consider financially supporting it through Kickstarter: > > http://www.kickstarter.com/projects/1175121367/high-availability-clust > er-module-for-freeswitch > > > > Thanks in advance for anyone who contributes, either with just a > shout-out, with feedback and/or suggestions, and especially to those > who pledge some financial support. > > -- > Eliot Gable > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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-use > rs > http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 paul at cupis.co.uk Mon Feb 11 13:15:40 2013 From: paul at cupis.co.uk (Paul Cupis) Date: Mon, 11 Feb 2013 10:15:40 +0000 Subject: [Freeswitch-users] High Availability Cluster Module for FreeSWITCH In-Reply-To: References: <-2120316468757263517@unknownmsgid> Message-ID: <20130211101539.GA27841@eagle.cupis.co.uk> On Sun, Feb 10, 2013 at 04:00:21PM -0500, Eliot Gable wrote: > For #3, I would like to eventually have the module scanning the > firewall rules for changes and enforcing a specific, pre-determined > "known-good" set of firewall rules. In this scenario, don't forget to consider users who run freeswitch under a dedicated user account, not as root - some method for the "freeswitch" user to be able to run iptables/other will need to be implemented (whether that is sudo, or a seperate (root) process or otherwise). Regards, From shouldbeq931 at gmail.com Mon Feb 11 15:18:54 2013 From: shouldbeq931 at gmail.com (shouldbe q931) Date: Mon, 11 Feb 2013 12:18:54 +0000 Subject: [Freeswitch-users] VoIP gateway with PRI In-Reply-To: <51140221.4040107@gmail.com> References: <51140221.4040107@gmail.com> Message-ID: On Thu, Feb 7, 2013 at 7:36 PM, Mimiko wrote: > One line PRI will be connected to phone provider. Some calls to certain > numbers thru this connection must be routed to FS via IP. Other > destination numbers must be routed to second PRI connection for legacy > PBX for analog phones. So the VoIP gateway must have two PRI (T1/E1) > connections and two LAN connection for failover. > > After a while this VoIP gateway will be as a tehnology converter from IP > to PRI. So LAN connectors will be used for connection to FS in a > failover manner. And two PRI connectors will be used to connect to > legacy PBX which have two PRI inputs to handle 60 concurrent connection. > > I've spotted three models: > 1) Patton SN4951/4E60V2GS/EUI 4T1/E1/PRI 60 VoIP Channels G.SHDSL for > $5,000 > > 2) Audiocodes Mediant 1000 M1K-D2 2T1/E1 for $3,800 > > 3) Sangoma Vega 400 2 T1/E1 Digital Gateway for $4,437 > > Which of this is a better working solution with failover and extended > call routing plans? > > Thansk. > Some time ago (~2007) I used a Vega unit to connect to a Telco PRI, a PRI file transfer unit (predates broadband and FTP) and a PRI connection on a PaBX, a total of 3 PRI connections. The (very basic) dialplan sent 6 DDI numbers to the ISDN file transfer box, and the rest to the PaBX, it had the capability (with an extra licence) to also add IP destinations to the dialplan, but this was of no use to us. I found the UK support to be very good. Cheers -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130211/98a3028d/attachment.html From gozdal at gmail.com Mon Feb 11 15:36:14 2013 From: gozdal at gmail.com (Marcin Gozdalik) Date: Mon, 11 Feb 2013 13:36:14 +0100 Subject: [Freeswitch-users] High Availability Cluster Module for FreeSWITCH In-Reply-To: References: <-2120316468757263517@unknownmsgid> Message-ID: +1 I do not doubt mod_ha is necessary inside of FS and it may be better/simpler than writing Pacemaker resource agent, but writing yet-another-cluster-communication-engine is IMHO the wrong way to go and using Corosync for communication will give a lot of value from mature codebase. 2013/2/10 Steven Ayre : >> One is the mod_ha_cluster is an N+1 Cluster, not a active/passive pair. >> That way if any single node fails, there's something to pick up the slack. > > > Corosync clusters are not limited to active/passive pairs. It's just a very > common setup. > > For example you could have resource agents 1) to keep FS running on all > nodes 2) for virtual IPs 3) for IP:port Sofia profiles. You can then define > dependancies between them. That should let you keep FS running at all times > and move an IP and the associated Sofia profiles to a new node that's > already running FS when the original node fails. For maintance you can > simply trigger that from the CRM. > >> Secondly, in order to recover live calls, you need a list of the calls. >> That currently requires some sort of odbc (or postgres) with replication. >> Again, that's abstracted as part of mod_ha_cluster. >> Third: The docs mention a similar of pooling for registration, that you >> can register to one server and you're regged on them all without needing a >> DB to sync everything. > > > Which can also be done using Corosync's IPC messaging API. > > (Personally I prefer using MySQL Cluster via ODBC - which is HA, synchronous > and offloads all load off of the FS nodes, but that's offtopic). > >> Fourth, according to the docs: single configuration for all FS instances, >> rather than manually ensuring each one has the same config. > > > This could be achieved with the Corosync API. > >> Fifth: Voicemail clustering? Or we'll have to wait for mod_voicemail's >> APIs to be rewritten for that, perhaps... > > > That's not going to happen without a storage API added to the FS core - > you're always going to need some 3rd party such NFS, CIFS, DRBD etc. > mod_voicemail is hardcoded to use the ODBC interface but that's only for the > index not the recordings.nc > > Corosync would allow you to make FS depend on the NFS/whatever service > running so if the storage backend has failed FS would move to a node where > it is available - not necessarily possibly from within mod_ha_cluster > itself. > >> There's certainly something special possible with mod_ha_cluster that >> can't be done with existing solutions cleanly, if at all... > > > Don't misunderstand me, I think it's a great idea to have a module aimed at > HA and sharing state across a cluster while being able to detect new failure > conditions. > > I just think that in the specific area of node monitoring and messaging > across the cluster it would be better to use a well tested and proven > solution such as Corosync which is based on a large number of papers, > algorithms, and generally decades of work. Every time I've seen/heard of an > attempt to redo that from scratch it's been unreliable especially in > unexpected failure conditions. Simply because it's a dependency on another > program isn't a good enough reason not to use it - you're already depending > on various other programs (Linux, sysvinit, monit, cron, syslog etc) anyway. > By not using it you're just adding extra work, adding unnecessarily > complexity and increasing the risk of bugs. Corosync would also have > advantages because the tools to migrate services to another node for > maintenance, detect and restart resources on failure etc already exist. > > -Steve > > > > > > On 10 February 2013 20:40, Avi Marcus wrote: >> >> Steve: >> Several reasons. >> One is the mod_ha_cluster is an N+1 Cluster, not a active/passive pair. >> That way if any single node fails, there's something to pick up the slack. >> >> Secondly, in order to recover live calls, you need a list of the calls. >> That currently requires some sort of odbc (or postgres) with replication. >> Again, that's abstracted as part of mod_ha_cluster. >> >> Third: The docs mention a similar of pooling for registration, that you >> can register to one server and you're regged on them all without needing a >> DB to sync everything. >> >> Fourth, according to the docs: single configuration for all FS instances, >> rather than manually ensuring each one has the same config. >> >> Fifth: Voicemail clustering? Or we'll have to wait for mod_voicemail's >> APIs to be rewritten for that, perhaps... >> >> There's certainly something special possible with mod_ha_cluster that >> can't be done with existing solutions cleanly, if at all... >> >> -Avi >> >> On Sun, Feb 10, 2013 at 10:16 PM, Steven Ayre wrote: >>> >>> Fair enough. >>> >>> I still don't see the need to reinvent the wheel just to avoid depending >>> on a 3rd party well developed and extensively tested piece of software. >>> Corosync provides an API that can be used for passing messages between nodes >>> in a cluster. >>> >>> -Steve >>> >>> >>> >>> On 10 February 2013 18:14, Eliot Gable >>> wrote: >>>> >>>> On Sun, Feb 10, 2013 at 12:08 PM, Steven Ayre >>>> wrote: >>>> > That covers redundancy in case of a network card or cable failure, but >>>> > isn't >>>> > what partitioning is about. Multiple NICs cannot prevent partitioning. >>>> > >>>> > As an example, partitioning might happen when a network switch between >>>> > two >>>> > network segments fails so you have nodes A+B in segment 1 able to talk >>>> > to >>>> > each other but unable to talk to nodes C+D in segment 2, while C+D can >>>> > talk >>>> > to each other but not A+B. >>>> > >>>> > Pacemaker/corosync contain a lot of algorithms to fence off partitions >>>> > without quorum and can resort to things like STONITH if required to >>>> > force a >>>> > node to shutdown rather than risk it causing disruption to the cluster >>>> > (for >>>> > example if it tries to take over traffic to a virtual IP you could end >>>> > up in >>>> > a case where you have two servers sending ARP responses for the same >>>> > IP). >>>> > >>>> >>>> Steve, >>>> >>>> As Avi pointed out, I mentioned having multiple physical networks as a >>>> guard against a network split / partition. If one network is split >>>> such that A and B can talk to each other over it and C and D can talk >>>> to each other over it, you would indeed have an issue if you only had >>>> one network. However, with two or more networks, all four nodes will >>>> still be able to talk to each other over the other network(s). >>>> >>>> Now, granted, if you have a network split in all networks, then you >>>> are still screwed. Pacemaker and other solutions deal with this, as >>>> you mentioned, using something called "quorum" where you need a >>>> majority of nodes to be able to see each other, and they fence the >>>> remaining nodes. As I documented on my wiki page for the module, I do >>>> have plans to eventually support such functionality. However, that is >>>> a bit further down the road as it will take some time to develop >>>> STONITH interfaces to various hardware or even to reuse the STONITH >>>> modules from Pacemaker or another project. In any case, I feel it is >>>> more important to get the base functionality developed and debugged as >>>> utilizing multiple networks is a good way to prevent network splits >>>> from being an issue. >>>> >>>> That being said, there are other issues to contend with when >>>> discussing network splits. For example, if A and B can see the >>>> Internet but C and D cannot, but C is a Master and B is a slave, you >>>> still have an issue to address. In this case, mod_ha_cluster must be >>>> able to determine that C and D cannot see the Internet. They need to >>>> perform very fast pings to some IP address, or have some external host >>>> sending them data in some way that they can detect when traffic >>>> to/from the Internet has stopped. I can place a media bug on the audio >>>> streams to make this determination fairly accurately. I can also rely >>>> on a ping mechanism to make the determination. Once the determination >>>> is made, mod_ha_cluster then has to promote B to a master to take over >>>> C. >>>> >>>> So, there are certainly still other issues to address when a network >>>> split occurs, but split-brain is easily avoided by simply adding >>>> redundant networks. >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -- Marcin Gozdalik From dujinfang at gmail.com Mon Feb 11 17:03:00 2013 From: dujinfang at gmail.com (Seven Du) Date: Mon, 11 Feb 2013 22:03:00 +0800 Subject: [Freeswitch-users] High Availability Cluster Module for reeSWITCH In-Reply-To: References: Message-ID: I also had considered the erlang way: - born clustered, can run an erlang node besides each FS instance, or run the erlang cluster on some dedicated servers - can share reg, session information with mnesia DB, in memory or on disk - add some functions in FS to recover sessions based on mnesia I wonder how the 2600hz guys are doing HA right now. Don't get me wrong, I'd saw the HA N+M wiki a long time ago and very interested how will it work. And, besides the FS HA, how will 3rd party softwares work with FS, if, say, someone connected via ESL and FS1 crashes and the float IP bind to FS2, will the ESL client fail and re-connect or it can magically connect to FS2 without any addintional work? Thanks -- Seven Du http://www.freeswitch.org.cn http://about.me/dujinfang http://www.dujinfang.com Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Monday, February 11, 2013 at 5:31 PM, Gerald Weber wrote: > Just my 2 eurocent, i know you said without 3rd party solution. > But wouldn't this be a perfect task for Erlangs distribution and fault tolerance capabilities ? > > -----Urspr?ngliche Nachricht----- > Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Brian Foster > Gesendet: Montag, 11. Februar 2013 04:19 > An: FreeSWITCH Users Help > Betreff: Re: [Freeswitch-users] High Availability Cluster Module for FreeSWITCH > > I applaud your efforts on getting a specialized mechanism in the form of a freeswitch module implemented to correctly perform high availability functions. I see a great potential here for the freeswitch project to attain a whole new level of performance and viability, and I really really hope others agree. > > I've been struggling with this very issue; how to do correct HA with freeswitch to the point where we aren't dropping calls when something bad happens. It's held me back in a few ways, and while ill never look into a non-open source solution, I've got way to much on my plate right now to play with it now. > > I'll see if my company can send some $$ your way. Ive got a few clients who would be interested in this as well. People, start donating! > > Sent from my iPhone > > On Feb 9, 2013, at 12:00 PM, Eliot Gable wrote: > > > For the past two years, I have been working on and off on a high > > availability cluster module for FreeSWITCH. If you would like to see > > HA built into FreeSWITCH directly, instead of dealing with Pacemaker + > > Corosync or other 3rd party systems, please check out the module, > > voice your support, provide feedback and suggestions, and consider > > supporting it. > > > > Some advantages over a 3rd party solution like Pacemaker include: > > > > - Sub-second failure detection and response (meaning it can start > > recovering calls less than a second after a failure occurs) > > - Failure detection is built into FreeSWITCH so it can detect > > conditions which 3rd party modules cannot detect > > - One shared configuration file between all nodes (no per-node IP > > configuration required) > > - No fixed number of nodes > > - Graceful migration of calls with a maintenance mode (requires DNS) > > > > These are just some of the benefits of building a FreeSWITCH-specific > > HA system. For additional details about the project, please visit: > > > > http://wiki.freeswitch.org/wiki/Mod_ha_cluster > > > > > > If you would like to see this project come to fruition, please > > consider financially supporting it through Kickstarter: > > > > http://www.kickstarter.com/projects/1175121367/high-availability-clust > > er-module-for-freeswitch > > > > > > > > Thanks in advance for anyone who contributes, either with just a > > shout-out, with feedback and/or suggestions, and especially to those > > who pledge some financial support. > > > > -- > > Eliot Gable > > > > ______________________________________________________________________ > > ___ Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > > rs > > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto: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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto: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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130211/c7391940/attachment.html From egable+freeswitch at gmail.com Mon Feb 11 17:06:59 2013 From: egable+freeswitch at gmail.com (Eliot Gable) Date: Mon, 11 Feb 2013 09:06:59 -0500 Subject: [Freeswitch-users] High Availability Cluster Module for FreeSWITCH In-Reply-To: References: <-2120316468757263517@unknownmsgid> Message-ID: On Mon, Feb 11, 2013 at 7:36 AM, Marcin Gozdalik wrote: > +1 > > I do not doubt mod_ha is necessary inside of FS and it may be > better/simpler than writing Pacemaker resource agent, but writing > yet-another-cluster-communication-engine is IMHO the wrong way to go > and using Corosync for communication will give a lot of value from > mature codebase. > > I understand what you are saying, but what I am trying to get across is that I am not writing yet-another-cluster-communication-engine. All I am really doing is combining a multicast messaging API written by Tony and the event API in FS to broadcast existing state information between multiple FS nodes, as well as adding a tiny amount of logic on top of that to coordinate call fail over and recovery. That's probably a little over-simplified, but it gets the point across. The network communication code is already in FS and well tested. The event system is already in FS and well tested. I have already written the code to the point that it parses the configuration files and starts sending heartbeats out all of the interfaces configured. I have also already written a lot of the code that deals with the state transitions. All I am talking about doing is implementing a tiny little finite state machine. It's a pretty trivial programming task. In fact, I think it was covered in my first year at Carnegie Mellon University. Of course, I had already figured out how to write such things in high school, I just did not know what it was called at that point. My point is, that this is not yet-another-cluster-communication-engine. It is a very specific and small finite state machine designed solely with the goal in mind of making FS have just enough information to coordinate call fail over internally. If I recall correctly, a lot of people also said writing yet-another-VoIP-server was a waste of time, but now we have FreeSWITCH, and it was obviously worth the effort. And I am not even trying to do something as complex as that. If you think this is yet-another-cluster-communication-engine, you are missing the point. It is not. It never will be. Look at Sonus, Genband, Broadsoft, Veraz, etc. All the big-name carrier-grade telecom providers have a built-in solution for automatic call fail over. The only way FreeSWITCH will ever compete with such solutions is if it also has that feature. Pacemaker and Corosync are overkill just to get FS to handle single node failures and provide call recovery. It took me a full 3 months of working with them every day to really understand how to deploy them properly in conjunction with FreeSWITCH and Postgres to provide a carrier-grade hot-standby solution which was robust enough to handle 99% of the failures I could throw at it. Granted, this was back when the configuration still needed to be written by hand in XML and prior the existence of any resource agent for FreeSWITCH. But, even with those changes, deploying Pacemaker and Corosync is not a simple task. If that is the requirement for FS to have HA, it will never truly stand a chance against commercial offerings. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130211/99dfdcf7/attachment.html From ravik0568 at gmail.com Sun Feb 10 04:04:16 2013 From: ravik0568 at gmail.com (Ravi K) Date: Sat, 9 Feb 2013 17:04:16 -0800 Subject: [Freeswitch-users] Passing the original caller-id after a call is transferred Message-ID: I am using Freeswitch with Cisco phones(mode 303) and have the following setup: User 1001 is calling user 1002, user 1002 does a blind/supervised transfer to 1003, but user 1003 see the caller-id of 1002. I am trying to get the caller-id of 1001 on 1003. I have gone through the earlier posts related to this topic and tried the following suggestions: 1. I configured this before bridging the call, but did not work. 2. and set >I might consider another distribution, but it must be one which will run >on the raspberrypi as that will be where it will be run in the end. I >did try the ArchLinux stuff on both an X86_64 system as well as the >raspberrypi, Debian, I mainly tried on the RaspberryPi using the >raspbian distribution. in /etc/rc.local on raspbian su pi -c "ulimit -s 240; /usr/local/freeswitch/bin/freeswitch -rp -nc" -- David Matthews mail at dmatthews.org From mike at jerris.com Mon Feb 11 18:08:30 2013 From: mike at jerris.com (Michael Jerris) Date: Mon, 11 Feb 2013 10:08:30 -0500 Subject: [Freeswitch-users] Running FS with "daemon" group. In-Reply-To: References: Message-ID: <79B039C3-503C-4B40-8236-00814D1CA42F@jerris.com> You can run freeswitch with whatever user/group you like as long as it has the necessary permissions. The more isolated typically the better, depending on your specific needs. On Feb 9, 2013, at 11:29 AM, Mimiko Mimiko wrote: > Hello. > > On this page: http://wiki.freeswitch.org/wiki/Freeswitch_init I see that FS is started under "daemon" group. Is it safe to start it that way, and is it safe if other services will be started under the same group? > > About "daemon" group debian says that is better that each service is started in its own uid/guid. But that way services cannot share same directory for storing and reading common files. Especially that FS creates files with 750 permission. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130211/15e6c07e/attachment.html From egable+freeswitch at gmail.com Mon Feb 11 18:09:59 2013 From: egable+freeswitch at gmail.com (Eliot Gable) Date: Mon, 11 Feb 2013 10:09:59 -0500 Subject: [Freeswitch-users] High Availability Cluster Module for reeSWITCH In-Reply-To: References: Message-ID: On Mon, Feb 11, 2013 at 9:03 AM, Seven Du wrote: > > And, besides the FS HA, how will 3rd party softwares work with FS, if, > say, someone connected via ESL and FS1 crashes and the float IP bind to > FS2, will the ESL client fail and re-connect or it can magically connect to > FS2 without any addintional work? > > The ESL client would have to reconnect. We could make a little ESL helper daemon you could run on some host which would do connection management for you, but it's probably a better idea to just handle it in your own code. It's not that difficult to notice a failed connection and reconnect. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130211/0e2dc8da/attachment.html From gaz.foreman at gmail.com Mon Feb 11 18:30:36 2013 From: gaz.foreman at gmail.com (Gary Foreman) Date: Mon, 11 Feb 2013 15:30:36 +0000 Subject: [Freeswitch-users] Terminating calls prior to answer using api Message-ID: Hi, I'm having an issue gracefully terminating calls prior to them being answered. When I issue the uuid_kill command prior to the call being answered my sip client gives an error "unable to connect" with an error tone. Is it possible to cancel the client dial without causing an error? Maybe using a sip notify message or something? I can't seem to find a list of valid sip notify messages anywhere. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130211/139779bc/attachment.html From andrew at cassidywebservices.co.uk Mon Feb 11 18:43:22 2013 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Mon, 11 Feb 2013 15:43:22 +0000 Subject: [Freeswitch-users] Terminating calls prior to answer using api In-Reply-To: References: Message-ID: probably need to use the cause parameter to uuid_kill This doesn't seem to be documented well, though. On 11 February 2013 15:30, Gary Foreman wrote: > Hi, > > I'm having an issue gracefully terminating calls prior to them being > answered. > > When I issue the uuid_kill command prior to the call being answered my sip > client gives an error "unable to connect" with an error tone. > > Is it possible to cancel the client dial without causing an error? Maybe > using a sip notify message or something? > > I can't seem to find a list of valid sip notify messages anywhere. > > Thanks > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130211/04fdf4e0/attachment.html From a.venugopan at mundio.com Mon Feb 11 18:51:56 2013 From: a.venugopan at mundio.com (Archana Venugopan) Date: Mon, 11 Feb 2013 15:51:56 +0000 Subject: [Freeswitch-users] Throttle Error Message-ID: <592A9CF93E12394E8472A6CC66E66BF2358566@Mail-Kilo.squay.com> Hi, I have been getting this error in freeswitch for group calls. Can anyone please let me know what is the issue? I tried increasing sessions_per_second in switch.conf.xml but even after that I have been facing this issue. Please help. Thanks 2013-02-11 15:35:12.237145 [CRIT] switch_core_session.c:1813 Throttle Error! 33 2013-02-11 15:35:12.237145 [CRIT] mod_sofia.c:4278 Error Creating Session 2013-02-11 15:35:12.317103 [CRIT] switch_time.c:869 Over Session Rate of 10! Regards, Archana -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130211/bf14c31e/attachment-0001.html From krice at freeswitch.org Mon Feb 11 19:10:17 2013 From: krice at freeswitch.org (Ken Rice) Date: Mon, 11 Feb 2013 10:10:17 -0600 Subject: [Freeswitch-users] Throttle Error In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF2358566@Mail-Kilo.squay.com> Message-ID: Did you restart after increasing sessions_per_second? Check ?status? from the cli... You can change it on the fly inside freeswitch with the fsctl sps command On 2/11/13 9:51 AM, "Archana Venugopan" wrote: > Hi, > > I have been getting this error in freeswitch for group calls. Can anyone > please let me know what is the issue? > I tried increasing sessions_per_second in switch.conf.xml but even after that > I have been facing this issue. Please help. Thanks > > 2013-02-11 15:35:12.237145 [CRIT] switch_core_session.c:1813 Throttle Error! > 33 > 2013-02-11 15:35:12.237145 [CRIT] mod_sofia.c:4278 Error Creating Session > 2013-02-11 15:35:12.317103 [CRIT] switch_time.c:869 Over Session Rate of 10! > > Regards, > Archana > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130211/6905e113/attachment.html From a.venugopan at mundio.com Mon Feb 11 19:37:07 2013 From: a.venugopan at mundio.com (Archana Venugopan) Date: Mon, 11 Feb 2013 16:37:07 +0000 Subject: [Freeswitch-users] Throttle Error In-Reply-To: References: <592A9CF93E12394E8472A6CC66E66BF2358566@Mail-Kilo.squay.com> Message-ID: <592A9CF93E12394E8472A6CC66E66BF2358588@Mail-Kilo.squay.com> Ya i did re-loaded the xml. And this is what I see in status UP 0 years, 6 days, 9 hours, 3 minutes, 17 seconds, 803 milliseconds, 420 microseconds FreeSWITCH is ready 62552 session(s) since startup 16 session(s) 0/10 50 session(s) max min idle cpu 0.00/91.00 Regards, Archana From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice Sent: 11 February 2013 16:10 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Throttle Error Did you restart after increasing sessions_per_second? Check "status" from the cli... You can change it on the fly inside freeswitch with the fsctl sps command On 2/11/13 9:51 AM, "Archana Venugopan" wrote: Hi, I have been getting this error in freeswitch for group calls. Can anyone please let me know what is the issue? I tried increasing sessions_per_second in switch.conf.xml but even after that I have been facing this issue. Please help. Thanks 2013-02-11 15:35:12.237145 [CRIT] switch_core_session.c:1813 Throttle Error! 33 2013-02-11 15:35:12.237145 [CRIT] mod_sofia.c:4278 Error Creating Session 2013-02-11 15:35:12.317103 [CRIT] switch_time.c:869 Over Session Rate of 10! Regards, Archana ________________________________ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130211/e4bcf47d/attachment.html From a.venugopan at mundio.com Mon Feb 11 19:53:31 2013 From: a.venugopan at mundio.com (Archana Venugopan) Date: Mon, 11 Feb 2013 16:53:31 +0000 Subject: [Freeswitch-users] Throttle Error In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF2358588@Mail-Kilo.squay.com> References: <592A9CF93E12394E8472A6CC66E66BF2358566@Mail-Kilo.squay.com> <592A9CF93E12394E8472A6CC66E66BF2358588@Mail-Kilo.squay.com> Message-ID: <592A9CF93E12394E8472A6CC66E66BF235859D@Mail-Kilo.squay.com> Hi, When I gave fsctl sps it shows still 10. Can you please tell me apart from reload_xml do we have to anyother thing after changing switch.conf.xml? Regards, Archana From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Archana Venugopan Sent: 11 February 2013 16:37 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Throttle Error Ya i did re-loaded the xml. And this is what I see in status UP 0 years, 6 days, 9 hours, 3 minutes, 17 seconds, 803 milliseconds, 420 microseconds FreeSWITCH is ready 62552 session(s) since startup 16 session(s) 0/10 50 session(s) max min idle cpu 0.00/91.00 Regards, Archana From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice Sent: 11 February 2013 16:10 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Throttle Error Did you restart after increasing sessions_per_second? Check "status" from the cli... You can change it on the fly inside freeswitch with the fsctl sps command On 2/11/13 9:51 AM, "Archana Venugopan" wrote: Hi, I have been getting this error in freeswitch for group calls. Can anyone please let me know what is the issue? I tried increasing sessions_per_second in switch.conf.xml but even after that I have been facing this issue. Please help. Thanks 2013-02-11 15:35:12.237145 [CRIT] switch_core_session.c:1813 Throttle Error! 33 2013-02-11 15:35:12.237145 [CRIT] mod_sofia.c:4278 Error Creating Session 2013-02-11 15:35:12.317103 [CRIT] switch_time.c:869 Over Session Rate of 10! Regards, Archana ________________________________ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130211/8dff18df/attachment-0001.html From gaz.foreman at gmail.com Mon Feb 11 19:40:06 2013 From: gaz.foreman at gmail.com (Gary Foreman) Date: Mon, 11 Feb 2013 16:40:06 +0000 Subject: [Freeswitch-users] Terminating calls prior to answer using api Message-ID: Thanks for your help. I might be doing something wrong but I had tried using a cause, the command I ran was... bgapi uuid_kill 0292147c-7469-11e2-817e-bd1d1e5a02e8 normal_clearing This seemed to work as the hangup event showed "NORMAL_CLEARING" as the hangup cause but the sip client still reported the error. On Mon, Feb 11, 2013 at 3:52 PM, < freeswitch-users-request at lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users at lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request at lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner at lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. Re: Trouble autostarting freeswitch on boot (David Matthews) > 2. Re: Running FS with "daemon" group. (Michael Jerris) > 3. Re: High Availability Cluster Module for reeSWITCH (Eliot Gable) > 4. Terminating calls prior to answer using api (Gary Foreman) > 5. Re: Terminating calls prior to answer using api (Andrew Cassidy) > 6. Throttle Error (Archana Venugopan) > > > ---------- Forwarded message ---------- > From: David Matthews > To: FreeSWITCH Users Help > Cc: > Date: Mon, 11 Feb 2013 13:00:33 +0000 > Subject: Re: [Freeswitch-users] Trouble autostarting freeswitch on boot > >> I might consider another distribution, but it must be one which will run >> on the raspberrypi as that will be where it will be run in the end. I >> did try the ArchLinux stuff on both an X86_64 system as well as the >> raspberrypi, Debian, I mainly tried on the RaspberryPi using the >> raspbian distribution. >> > > in /etc/rc.local on raspbian > > su pi -c "ulimit -s 240; /usr/local/freeswitch/bin/**freeswitch -rp -nc" > > -- > David Matthews > mail at dmatthews.org > > > > > ---------- Forwarded message ---------- > From: Michael Jerris > To: FreeSWITCH Users Help > Cc: > Date: Mon, 11 Feb 2013 10:08:30 -0500 > Subject: Re: [Freeswitch-users] Running FS with "daemon" group. > You can run freeswitch with whatever user/group you like as long as it has > the necessary permissions. The more isolated typically the better, > depending on your specific needs. > > > On Feb 9, 2013, at 11:29 AM, Mimiko Mimiko wrote: > > Hello. > > On this page: http://wiki.freeswitch.org/wiki/Freeswitch_init I see that > FS is started under "daemon" group. Is it safe to start it that way, and is > it safe if other services will be started under the same group? > > About "daemon" group debian says that is better that each service is > started in its own uid/guid. But that way services cannot share same > directory for storing and reading common files. Especially that FS creates > files with 750 permission. > > > > ---------- Forwarded message ---------- > From: Eliot Gable > To: FreeSWITCH Users Help > Cc: > Date: Mon, 11 Feb 2013 10:09:59 -0500 > Subject: Re: [Freeswitch-users] High Availability Cluster Module for > reeSWITCH > On Mon, Feb 11, 2013 at 9:03 AM, Seven Du wrote: > >> >> And, besides the FS HA, how will 3rd party softwares work with FS, if, >> say, someone connected via ESL and FS1 crashes and the float IP bind to >> FS2, will the ESL client fail and re-connect or it can magically connect to >> FS2 without any addintional work? >> >> > The ESL client would have to reconnect. We could make a little ESL helper > daemon you could run on some host which would do connection management for > you, but it's probably a better idea to just handle it in your own code. > It's not that difficult to notice a failed connection and reconnect. > > > > ---------- Forwarded message ---------- > From: Gary Foreman > To: freeswitch-users at lists.freeswitch.org > Cc: > Date: Mon, 11 Feb 2013 15:30:36 +0000 > Subject: [Freeswitch-users] Terminating calls prior to answer using api > Hi, > > I'm having an issue gracefully terminating calls prior to them being > answered. > > When I issue the uuid_kill command prior to the call being answered my sip > client gives an error "unable to connect" with an error tone. > > Is it possible to cancel the client dial without causing an error? Maybe > using a sip notify message or something? > > I can't seem to find a list of valid sip notify messages anywhere. > > Thanks > > > ---------- Forwarded message ---------- > From: Andrew Cassidy > To: FreeSWITCH Users Help > Cc: > Date: Mon, 11 Feb 2013 15:43:22 +0000 > Subject: Re: [Freeswitch-users] Terminating calls prior to answer using api > probably need to use the cause parameter to uuid_kill > > This doesn't seem to be documented well, though. > > On 11 February 2013 15:30, Gary Foreman wrote: > >> Hi, >> >> I'm having an issue gracefully terminating calls prior to them being >> answered. >> >> When I issue the uuid_kill command prior to the call being answered my >> sip client gives an error "unable to connect" with an error tone. >> >> Is it possible to cancel the client dial without causing an error? Maybe >> using a sip notify message or something? >> >> I can't seem to find a list of valid sip notify messages anywhere. >> >> Thanks >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > > -- > *Andrew Cassidy BSc (Hons) MBCS SSCA* > Managing Director > > > *T *03300 100 960 *F > *03300 100 961 > *E *andrew at cassidywebservices.co.uk > *W *www.cassidywebservices.co.uk > > > ---------- Forwarded message ---------- > From: Archana Venugopan > To: FreeSWITCH Users Help > Cc: > Date: Mon, 11 Feb 2013 15:51:56 +0000 > Subject: [Freeswitch-users] Throttle Error > > Hi,**** > > ** ** > > I have been getting this error in freeswitch for group calls. Can anyone > please let me know what is the issue?**** > > I tried increasing sessions_per_second in switch.conf.xml but even after > that I have been facing this issue. Please help. Thanks**** > > ** ** > > 2013-02-11 15:35:12.237145 [CRIT] switch_core_session.c:1813 Throttle > Error! 33**** > > 2013-02-11 15:35:12.237145 [CRIT] mod_sofia.c:4278 Error Creating Session* > *** > > 2013-02-11 15:35:12.317103 [CRIT] switch_time.c:869 Over Session Rate of > 10!**** > > ** ** > > Regards,**** > > Archana**** > > ** ** > > _______________________________________________ > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130211/a05a57e1/attachment.html From martyn at magiccow.co.uk Mon Feb 11 19:56:38 2013 From: martyn at magiccow.co.uk (Martyn Davies) Date: Mon, 11 Feb 2013 16:56:38 +0000 Subject: [Freeswitch-users] Hold/unhold events for a conference participant Message-ID: I'm trying to process CHANNEL_CALLSTATE events (in an ESL socket application) to extract the HELD/ACTIVE messages for conference participants. One problem for me is that the name of the conference and the member-id for that call are not available to me in the callstate messages, and therefore it is difficult to associated calls (going HELD and ACTIVE) with the conference. Is there any way to get the callstate messages to include custom variables? This would fix the problem for me. Regards, Martyn -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130211/9e501916/attachment.html From a.venugopan at mundio.com Mon Feb 11 20:00:25 2013 From: a.venugopan at mundio.com (Archana Venugopan) Date: Mon, 11 Feb 2013 17:00:25 +0000 Subject: [Freeswitch-users] Throttle Error In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF235859D@Mail-Kilo.squay.com> References: <592A9CF93E12394E8472A6CC66E66BF2358566@Mail-Kilo.squay.com> <592A9CF93E12394E8472A6CC66E66BF2358588@Mail-Kilo.squay.com> <592A9CF93E12394E8472A6CC66E66BF235859D@Mail-Kilo.squay.com> Message-ID: <592A9CF93E12394E8472A6CC66E66BF23585AE@Mail-Kilo.squay.com> Thanks I got it:) Regards, Archana From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Archana Venugopan Sent: 11 February 2013 16:54 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Throttle Error Hi, When I gave fsctl sps it shows still 10. Can you please tell me apart from reload_xml do we have to anyother thing after changing switch.conf.xml? Regards, Archana From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Archana Venugopan Sent: 11 February 2013 16:37 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Throttle Error Ya i did re-loaded the xml. And this is what I see in status UP 0 years, 6 days, 9 hours, 3 minutes, 17 seconds, 803 milliseconds, 420 microseconds FreeSWITCH is ready 62552 session(s) since startup 16 session(s) 0/10 50 session(s) max min idle cpu 0.00/91.00 Regards, Archana From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice Sent: 11 February 2013 16:10 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Throttle Error Did you restart after increasing sessions_per_second? Check "status" from the cli... You can change it on the fly inside freeswitch with the fsctl sps command On 2/11/13 9:51 AM, "Archana Venugopan" wrote: Hi, I have been getting this error in freeswitch for group calls. Can anyone please let me know what is the issue? I tried increasing sessions_per_second in switch.conf.xml but even after that I have been facing this issue. Please help. Thanks 2013-02-11 15:35:12.237145 [CRIT] switch_core_session.c:1813 Throttle Error! 33 2013-02-11 15:35:12.237145 [CRIT] mod_sofia.c:4278 Error Creating Session 2013-02-11 15:35:12.317103 [CRIT] switch_time.c:869 Over Session Rate of 10! Regards, Archana ________________________________ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130211/1a3fba82/attachment-0001.html From avi at avimarcus.net Mon Feb 11 20:09:33 2013 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 11 Feb 2013 19:09:33 +0200 Subject: [Freeswitch-users] Throttle Error In-Reply-To: References: <592A9CF93E12394E8472A6CC66E66BF2358566@Mail-Kilo.squay.com> <592A9CF93E12394E8472A6CC66E66BF2358588@Mail-Kilo.squay.com> <592A9CF93E12394E8472A6CC66E66BF235859D@Mail-Kilo.squay.com> Message-ID: As Ken's original message said, try "fsctl sps 40" or the like. He said that otherwise only a restart, not reload xml would work. -Avi (This message was painstakingly thumbed out on my mobile, so apologies for brevity and errors.) On Feb 11, 2013 6:57 PM, "Archana Venugopan" wrote: Hi,**** ** ** When I gave fsctl sps it shows still 10. Can you please tell me apart from reload_xml do we have to anyother thing after changing switch.conf.xml?**** ** ** Regards,**** Archana**** ** ** *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Archana Venugopan *Sent:* 11 February 2013 16:37 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Throttle Error Ya i did re-loaded the xml. And this is what I see in status UP 0 years, 6 days, 9 hours, 3... _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130211/c4c385da/attachment.html From steveayre at gmail.com Mon Feb 11 20:40:57 2013 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 11 Feb 2013 17:40:57 +0000 Subject: [Freeswitch-users] Terminating calls prior to answer using api In-Reply-To: References: Message-ID: You either answer a call with 200, redirect with 3xx, or give an error 4xx 5xx or 6xx... basically without answering your only option is to return an error. The specific error code can show a varying number of levels of 'error' though. Some might be a perfectly valid result (eg 486 User Busy) while other's imply a more serious error (eg 500 Server Internal Error). But there's not really one for a successful unanswered call. How your phone behaves on the various errors is really up to the SIP client though. A generic unable to connect message doesn't sound very useful, I'd expect different error tones depending on the response code. -Steve On 11 February 2013 15:30, Gary Foreman wrote: > Hi, > > I'm having an issue gracefully terminating calls prior to them being > answered. > > When I issue the uuid_kill command prior to the call being answered my sip > client gives an error "unable to connect" with an error tone. > > Is it possible to cancel the client dial without causing an error? Maybe > using a sip notify message or something? > > I can't seem to find a list of valid sip notify messages anywhere. > > Thanks > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130211/db9c1e67/attachment.html From steveayre at gmail.com Mon Feb 11 20:42:51 2013 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 11 Feb 2013 17:42:51 +0000 Subject: [Freeswitch-users] Terminating calls prior to answer using api In-Reply-To: References: Message-ID: The NORMAL_CLEARING ISDN clearing cause (16) has no mapping to any SIP code for example, it's a code you normally would only see a) when the caller hangs up at any time (regardless of answered or not) or b) when call has been answered *then* callee hangs up. -Steve On 11 February 2013 17:40, Steven Ayre wrote: > You either answer a call with 200, redirect with 3xx, or give an error 4xx > 5xx or 6xx... basically without answering your only option is to return an > error. > > The specific error code can show a varying number of levels of 'error' > though. Some might be a perfectly valid result (eg 486 User Busy) while > other's imply a more serious error (eg 500 Server Internal Error). But > there's not really one for a successful unanswered call. > > How your phone behaves on the various errors is really up to the SIP > client though. A generic unable to connect message doesn't sound very > useful, I'd expect different error tones depending on the response code. > > -Steve > > > > On 11 February 2013 15:30, Gary Foreman wrote: > >> Hi, >> >> I'm having an issue gracefully terminating calls prior to them being >> answered. >> >> When I issue the uuid_kill command prior to the call being answered my >> sip client gives an error "unable to connect" with an error tone. >> >> Is it possible to cancel the client dial without causing an error? Maybe >> using a sip notify message or something? >> >> I can't seem to find a list of valid sip notify messages anywhere. >> >> Thanks >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130211/e7bef017/attachment.html From egable+freeswitch at gmail.com Mon Feb 11 20:44:55 2013 From: egable+freeswitch at gmail.com (Eliot Gable) Date: Mon, 11 Feb 2013 12:44:55 -0500 Subject: [Freeswitch-users] High Availability Cluster Module for FreeSWITCH - Major News Update Message-ID: After discussing this project in further detail with Tony, I have decided to cancel the Kickstarter project. Funding will now go through FreeSWITCH Solutions. If you would like to contribute towards making mod_ha_cluster a reality, please send contributions to FreeSWITCH Solutions. Thanks! -- Eliot Gable -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130211/81f26242/attachment.html From luis.daniel.lucio at gmail.com Mon Feb 11 21:06:16 2013 From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz) Date: Mon, 11 Feb 2013 13:06:16 -0500 Subject: [Freeswitch-users] High Availability Cluster Module for FreeSWITCH In-Reply-To: References: <-2120316468757263517@unknownmsgid> Message-ID: Is the latest snapshot ofyour module at git://git.freeswitch.org/freeswitch-contrib.git ? 2013/2/11 Eliot Gable : > On Mon, Feb 11, 2013 at 7:36 AM, Marcin Gozdalik wrote: >> >> +1 >> >> I do not doubt mod_ha is necessary inside of FS and it may be >> better/simpler than writing Pacemaker resource agent, but writing >> yet-another-cluster-communication-engine is IMHO the wrong way to go >> and using Corosync for communication will give a lot of value from >> mature codebase. >> > > I understand what you are saying, but what I am trying to get across is that > I am not writing yet-another-cluster-communication-engine. All I am really > doing is combining a multicast messaging API written by Tony and the event > API in FS to broadcast existing state information between multiple FS nodes, > as well as adding a tiny amount of logic on top of that to coordinate call > fail over and recovery. That's probably a little over-simplified, but it > gets the point across. The network communication code is already in FS and > well tested. The event system is already in FS and well tested. I have > already written the code to the point that it parses the configuration files > and starts sending heartbeats out all of the interfaces configured. I have > also already written a lot of the code that deals with the state > transitions. All I am talking about doing is implementing a tiny little > finite state machine. It's a pretty trivial programming task. In fact, I > think it was covered in my first year at Carnegie Mellon University. Of > course, I had already figured out how to write such things in high school, I > just did not know what it was called at that point. My point is, that this > is not yet-another-cluster-communication-engine. It is a very specific and > small finite state machine designed solely with the goal in mind of making > FS have just enough information to coordinate call fail over internally. If > I recall correctly, a lot of people also said writing > yet-another-VoIP-server was a waste of time, but now we have FreeSWITCH, and > it was obviously worth the effort. And I am not even trying to do something > as complex as that. If you think this is > yet-another-cluster-communication-engine, you are missing the point. It is > not. It never will be. > > Look at Sonus, Genband, Broadsoft, Veraz, etc. All the big-name > carrier-grade telecom providers have a built-in solution for automatic call > fail over. The only way FreeSWITCH will ever compete with such solutions is > if it also has that feature. Pacemaker and Corosync are overkill just to get > FS to handle single node failures and provide call recovery. It took me a > full 3 months of working with them every day to really understand how to > deploy them properly in conjunction with FreeSWITCH and Postgres to provide > a carrier-grade hot-standby solution which was robust enough to handle 99% > of the failures I could throw at it. Granted, this was back when the > configuration still needed to be written by hand in XML and prior the > existence of any resource agent for FreeSWITCH. But, even with those > changes, deploying Pacemaker and Corosync is not a simple task. If that is > the requirement for FS to have HA, it will never truly stand a chance > against commercial offerings. > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 gaz.foreman at gmail.com Mon Feb 11 21:37:17 2013 From: gaz.foreman at gmail.com (Gary Foreman) Date: Mon, 11 Feb 2013 18:37:17 +0000 Subject: [Freeswitch-users] Terminating calls prior to answer using api Message-ID: I was concerned that might be the case, I have managed to get different errors by send alternative hangup codes "user_busy" for example is recognised but as you say all are still errors. I'm planning on using Polycom sip phones, does anyone know if it is possible to end an unanswered call using a notify message or something? Like the talk sip notify message used to answer. I'm writing a CTI application so terminating the call via the api prior to answer without errors is really important. Thanks again On Mon, Feb 11, 2013 at 6:07 PM, < freeswitch-users-request at lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users at lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request at lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner at lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. Re: Throttle Error (Avi Marcus) > 2. Re: Terminating calls prior to answer using api (Steven Ayre) > 3. Re: Terminating calls prior to answer using api (Steven Ayre) > 4. High Availability Cluster Module for FreeSWITCH - Major News > Update (Eliot Gable) > 5. Re: High Availability Cluster Module for FreeSWITCH > (Luis Daniel Lucio Quiroz) > > > ---------- Forwarded message ---------- > From: Avi Marcus > To: FreeSWITCH Users Help > Cc: > Date: Mon, 11 Feb 2013 19:09:33 +0200 > Subject: Re: [Freeswitch-users] Throttle Error > > As Ken's original message said, try "fsctl sps 40" or the like. > He said that otherwise only a restart, not reload xml would work. > > -Avi > (This message was painstakingly thumbed out on my mobile, so apologies for > brevity and errors.) > > On Feb 11, 2013 6:57 PM, "Archana Venugopan" > wrote: > > Hi,**** > > ** ** > > When I gave fsctl sps it shows still 10. Can you please tell me apart from > reload_xml do we have to anyother thing after changing switch.conf.xml?*** > * > > ** ** > > Regards,**** > > Archana**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Archana > Venugopan > *Sent:* 11 February 2013 16:37 > > > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Throttle Error > > > > > > Ya i did re-loaded the xml. And this is what I see in status > > > > UP 0 years, 6 days, 9 hours, 3... > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > ---------- Forwarded message ---------- > From: Steven Ayre > To: FreeSWITCH Users Help > Cc: > Date: Mon, 11 Feb 2013 17:40:57 +0000 > Subject: Re: [Freeswitch-users] Terminating calls prior to answer using api > You either answer a call with 200, redirect with 3xx, or give an error 4xx > 5xx or 6xx... basically without answering your only option is to return an > error. > > The specific error code can show a varying number of levels of 'error' > though. Some might be a perfectly valid result (eg 486 User Busy) while > other's imply a more serious error (eg 500 Server Internal Error). But > there's not really one for a successful unanswered call. > > How your phone behaves on the various errors is really up to the SIP > client though. A generic unable to connect message doesn't sound very > useful, I'd expect different error tones depending on the response code. > > -Steve > > > > On 11 February 2013 15:30, Gary Foreman wrote: > >> Hi, >> >> I'm having an issue gracefully terminating calls prior to them being >> answered. >> >> When I issue the uuid_kill command prior to the call being answered my >> sip client gives an error "unable to connect" with an error tone. >> >> Is it possible to cancel the client dial without causing an error? Maybe >> using a sip notify message or something? >> >> I can't seem to find a list of valid sip notify messages anywhere. >> >> Thanks >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > > ---------- Forwarded message ---------- > From: Steven Ayre > To: FreeSWITCH Users Help > Cc: > Date: Mon, 11 Feb 2013 17:42:51 +0000 > Subject: Re: [Freeswitch-users] Terminating calls prior to answer using api > The NORMAL_CLEARING ISDN clearing cause (16) has no mapping to any SIP > code for example, it's a code you normally would only see a) when the > caller hangs up at any time (regardless of answered or not) or b) when call > has been answered *then* callee hangs up. > > -Steve > > > > > On 11 February 2013 17:40, Steven Ayre wrote: > >> You either answer a call with 200, redirect with 3xx, or give an error >> 4xx 5xx or 6xx... basically without answering your only option is to return >> an error. >> >> The specific error code can show a varying number of levels of 'error' >> though. Some might be a perfectly valid result (eg 486 User Busy) while >> other's imply a more serious error (eg 500 Server Internal Error). But >> there's not really one for a successful unanswered call. >> >> How your phone behaves on the various errors is really up to the SIP >> client though. A generic unable to connect message doesn't sound very >> useful, I'd expect different error tones depending on the response code. >> >> -Steve >> >> >> >> On 11 February 2013 15:30, Gary Foreman wrote: >> >>> Hi, >>> >>> I'm having an issue gracefully terminating calls prior to them being >>> answered. >>> >>> When I issue the uuid_kill command prior to the call being answered my >>> sip client gives an error "unable to connect" with an error tone. >>> >>> Is it possible to cancel the client dial without causing an error? Maybe >>> using a sip notify message or something? >>> >>> I can't seem to find a list of valid sip notify messages anywhere. >>> >>> Thanks >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> > > > ---------- Forwarded message ---------- > From: Eliot Gable > To: FreeSWITCH Users Help > Cc: > Date: Mon, 11 Feb 2013 12:44:55 -0500 > Subject: [Freeswitch-users] High Availability Cluster Module for > FreeSWITCH - Major News Update > After discussing this project in further detail with Tony, I have decided > to cancel the Kickstarter project. Funding will now go through FreeSWITCH > Solutions. If you would like to contribute towards making mod_ha_cluster a > reality, please send contributions to FreeSWITCH Solutions. > > Thanks! > > -- > Eliot Gable > > > > ---------- Forwarded message ---------- > From: Luis Daniel Lucio Quiroz > To: FreeSWITCH Users Help > Cc: > Date: Mon, 11 Feb 2013 13:06:16 -0500 > Subject: Re: [Freeswitch-users] High Availability Cluster Module for > FreeSWITCH > Is the latest snapshot ofyour module at > git://git.freeswitch.org/freeswitch-contrib.git ? > > 2013/2/11 Eliot Gable : > > On Mon, Feb 11, 2013 at 7:36 AM, Marcin Gozdalik > wrote: > >> > >> +1 > >> > >> I do not doubt mod_ha is necessary inside of FS and it may be > >> better/simpler than writing Pacemaker resource agent, but writing > >> yet-another-cluster-communication-engine is IMHO the wrong way to go > >> and using Corosync for communication will give a lot of value from > >> mature codebase. > >> > > > > I understand what you are saying, but what I am trying to get across is > that > > I am not writing yet-another-cluster-communication-engine. All I am > really > > doing is combining a multicast messaging API written by Tony and the > event > > API in FS to broadcast existing state information between multiple FS > nodes, > > as well as adding a tiny amount of logic on top of that to coordinate > call > > fail over and recovery. That's probably a little over-simplified, but it > > gets the point across. The network communication code is already in FS > and > > well tested. The event system is already in FS and well tested. I have > > already written the code to the point that it parses the configuration > files > > and starts sending heartbeats out all of the interfaces configured. I > have > > also already written a lot of the code that deals with the state > > transitions. All I am talking about doing is implementing a tiny little > > finite state machine. It's a pretty trivial programming task. In fact, I > > think it was covered in my first year at Carnegie Mellon University. Of > > course, I had already figured out how to write such things in high > school, I > > just did not know what it was called at that point. My point is, that > this > > is not yet-another-cluster-communication-engine. It is a very specific > and > > small finite state machine designed solely with the goal in mind of > making > > FS have just enough information to coordinate call fail over internally. > If > > I recall correctly, a lot of people also said writing > > yet-another-VoIP-server was a waste of time, but now we have FreeSWITCH, > and > > it was obviously worth the effort. And I am not even trying to do > something > > as complex as that. If you think this is > > yet-another-cluster-communication-engine, you are missing the point. It > is > > not. It never will be. > > > > Look at Sonus, Genband, Broadsoft, Veraz, etc. All the big-name > > carrier-grade telecom providers have a built-in solution for automatic > call > > fail over. The only way FreeSWITCH will ever compete with such solutions > is > > if it also has that feature. Pacemaker and Corosync are overkill just to > get > > FS to handle single node failures and provide call recovery. It took me a > > full 3 months of working with them every day to really understand how to > > deploy them properly in conjunction with FreeSWITCH and Postgres to > provide > > a carrier-grade hot-standby solution which was robust enough to handle > 99% > > of the failures I could throw at it. Granted, this was back when the > > configuration still needed to be written by hand in XML and prior the > > existence of any resource agent for FreeSWITCH. But, even with those > > changes, deploying Pacemaker and Corosync is not a simple task. If that > is > > the requirement for FS to have HA, it will never truly stand a chance > > against commercial offerings. > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > 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 > > > > > > _______________________________________________ > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130211/732f8116/attachment-0001.html From red.rain.seven at gmail.com Mon Feb 11 21:37:22 2013 From: red.rain.seven at gmail.com (Henry Huang) Date: Mon, 11 Feb 2013 10:37:22 -0800 Subject: [Freeswitch-users] High Availability Cluster Module for FreeSWITCH In-Reply-To: References: <-2120316468757263517@unknownmsgid> Message-ID: I would choose to go with Elliot's idea of having the HA being build within the FS itself under the condition that it's a easy adaptation comparing to having to use and struggling with Pacemaker and Corosync. They can literally take months if someone is starting from scratch. And it would be nice to have this kind of capability to go against those brand name solutions. On Mon, Feb 11, 2013 at 6:06 AM, Eliot Gable wrote: > On Mon, Feb 11, 2013 at 7:36 AM, Marcin Gozdalik wrote: > >> +1 >> >> I do not doubt mod_ha is necessary inside of FS and it may be >> better/simpler than writing Pacemaker resource agent, but writing >> yet-another-cluster-communication-engine is IMHO the wrong way to go >> and using Corosync for communication will give a lot of value from >> mature codebase. >> >> > I understand what you are saying, but what I am trying to get across is > that I am not writing yet-another-cluster-communication-engine. All I am > really doing is combining a multicast messaging API written by Tony and the > event API in FS to broadcast existing state information between multiple FS > nodes, as well as adding a tiny amount of logic on top of that to > coordinate call fail over and recovery. That's probably a little > over-simplified, but it gets the point across. The network communication > code is already in FS and well tested. The event system is already in FS > and well tested. I have already written the code to the point that it > parses the configuration files and starts sending heartbeats out all of the > interfaces configured. I have also already written a lot of the code that > deals with the state transitions. All I am talking about doing is > implementing a tiny little finite state machine. It's a pretty trivial > programming task. In fact, I think it was covered in my first year at > Carnegie Mellon University. Of course, I had already figured out how to > write such things in high school, I just did not know what it was called at > that point. My point is, that this is not > yet-another-cluster-communication-engine. It is a very specific and small > finite state machine designed solely with the goal in mind of making FS > have just enough information to coordinate call fail over internally. If I > recall correctly, a lot of people also said writing yet-another-VoIP-server > was a waste of time, but now we have FreeSWITCH, and it was obviously worth > the effort. And I am not even trying to do something as complex as that. If > you think this is yet-another-cluster-communication-engine, you are missing > the point. It is not. It never will be. > > Look at Sonus, Genband, Broadsoft, Veraz, etc. All the big-name > carrier-grade telecom providers have a built-in solution for automatic call > fail over. The only way FreeSWITCH will ever compete with such solutions is > if it also has that feature. Pacemaker and Corosync are overkill just to > get FS to handle single node failures and provide call recovery. It took me > a full 3 months of working with them every day to really understand how to > deploy them properly in conjunction with FreeSWITCH and Postgres to provide > a carrier-grade hot-standby solution which was robust enough to handle 99% > of the failures I could throw at it. Granted, this was back when the > configuration still needed to be written by hand in XML and prior the > existence of any resource agent for FreeSWITCH. But, even with those > changes, deploying Pacemaker and Corosync is not a simple task. If that is > the requirement for FS to have HA, it will never truly stand a chance > against commercial offerings. > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130211/8c0fcdd1/attachment.html From avi at avimarcus.net Mon Feb 11 22:13:56 2013 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 11 Feb 2013 21:13:56 +0200 Subject: [Freeswitch-users] Terminating calls prior to answer using api In-Reply-To: References: Message-ID: Or maybe transfer to an extension that answers then hangs up immediately if your phone doesn't like anything else. -Avi (This message was painstakingly thumbed out on my mobile, so apologies for brevity and errors.) On Feb 11, 2013 6:57 PM, "Archana Venugopan" wrote: Hi,**** ** ** When I gave fsctl sps it shows still 10. Can you please tell me apart from reload_xml do we have to anyother thing after changing switch.conf.xml?**** ** ** Regards,**** Archana**** ** ** *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Archana Venugopan *Sent:* 11 February 2013 16:37 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Throttle Error Ya i did re-loaded the xml. And this is what I see in status UP 0 years, 6 days, 9 hours, 3... >> >> >> _________________________________________________________________________ >> Professional ... > > > ---------- Forwarded message ---------- > From: Steven Ayre > To: FreeSW... > Date: Mon, 11 Feb 2013 17:40:57 +0000 > Subject: Re: [Freeswitch-users] Terminating calls prior to... > You either answer a call with 200, redirect with 3xx, or give an error 4xx 5xx or 6xx... basically... > ---------- Forwarded message ---------- > From: Steven Ayre > To: FreeSWITCH... > Date: Mon, 11 Feb 2013 17:42:51 +0000 > Subject: Re: [Freeswitch-users] Terminating calls prior to... > The NORMAL_CLEARING ISDN clearing cause (16) has no mapping to any SIP code for example, it's a co... > ---------- Forwarded message ---------- > From: Eliot Gable > To: Fr... Date: Mon, 11 Feb 2013 12:44:55 -0500 Subject: [Freeswitch-users] High Availability Cluster Module for FreeSWITCH - Major News Update After discussing this project in further detail with Tony, I have decided to cancel the Kickstarter project. Funding will now go through FreeSWITCH Solutions. If you would like to contribute towards making mod_ha_cluster a reality, please send contributions to FreeSWITCH Solutions. Thanks! -- Eliot Gable > > > ---------- Forwarded message ---------- > From: Luis Daniel Lucio Quiroz : > On Mon, Feb 11, 2013 at 7:36 AM, Marcin Gozdalik wrote: >> >> +1 >> >> I do not doubt mod_ha is necessary inside of FS and it may be >> better/simpler than writing Pacemaker resource agent, but writing >> yet-another-cluster-communication-engine is IMHO the wrong way to go >> and using Corosync for communication will give a lot of value from >> mature codebase. >> > > I understand what you are saying, but what I am trying to get across is that > I am not writing yet-another-cluster-communication-engine. All I am really > doing is combining a multicast messaging API written by Tony and the event > API in FS to broadcast existing state information between multiple FS nodes, > as well as adding a tiny amount of logic on top of that to coordinate call > fail over and recovery. That's probably a little over-simplified, but it > gets the point across. The network communication code is already in FS and > well tested. The event system is already in FS and well tested. I have > already written the code to the point that it parses the configuration files > and starts sending heartbeats out all of the interfaces configured. I have > also already written a lot of the code that deals with the state > transitions. All I am talking about doing is implementing a tiny little > finite state machine. It's a pretty trivial programming task. In fact, I > think it was covered in my first year at Carnegie Mellon University. Of > course, I had already figured out how to write such things in high school, I > just did not know what it was called at that point. My point is, that this > is not yet-another-cluster-communication-engine. It is a very specific and > small finite state machine designed solely with the goal in mind of making > FS have just enough information to coordinate call fail over internally. If > I recall correctly, a lot of people also said writing > yet-another-VoIP-server was a waste of time, but now we have FreeSWITCH, and > it was obviously worth the effort. And I am not even trying to do something > as complex as that. If you think this is > yet-another-cluster-communication-engine, you are missing the point. It is > not. It never will be. > > Look at Sonus, Genband, Broadsoft, Veraz, etc. All the big-name > carrier-grade telecom providers have a built-in solution for automatic call > fail over. The only way FreeSWITCH will ever compete with such solutions is > if it also has that feature. Pacemaker and Corosync are overkill just to get > FS to handle single node failures and provide call recovery. It took me a > full 3 months of working with them every day to really understand how to > deploy them properly in conjunction with FreeSWITCH and Postgres to provide > a carrier-grade hot-standby solution which was robust enough to handle 99% > of the failures I could throw at it. Granted, this was back when the > configuration still needed to be written by hand in XML and prior the > existence of any resource agent for FreeSWITCH. But, even with those > changes, deploying Pacemaker and Corosync is not a simple task. If that is > the requirement for FS to have HA, it will never truly stand a chance > against commercial offerings. > > > > > > > > _________________________________________________________________________ > > Profe... _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.o... _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130211/26e74ae8/attachment-0001.html From steveayre at gmail.com Mon Feb 11 23:07:44 2013 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 11 Feb 2013 20:07:44 +0000 Subject: [Freeswitch-users] Terminating calls prior to answer using api In-Reply-To: References: Message-ID: NOTIFY messages are a SIP feature outside of an INVITE call flow. Ending the call using 200+BYE, 4xx, 5xx or 6xx are your only options I'm afraid. I'm writing a CTI application so terminating the call via the api prior to > answer without errors is really important. Any call that's unanswered is a failed call, but that's not necessarily an error (eg user busy). I understand your concern that the caller might not know if their call succeeded or not... Is there any possibility perhaps to answer the call, send it to a playback (eg 'your request was successful') and then hangup? -Steve On 11 February 2013 18:37, Gary Foreman wrote: > I was concerned that might be the case, I have managed to get different > errors by send alternative hangup codes "user_busy" for example is > recognised but as you say all are still errors. > > I'm planning on using Polycom sip phones, does anyone know if it is > possible to end an unanswered call using a notify message or something? > Like the talk sip notify message used to answer. > > I'm writing a CTI application so terminating the call via the api prior to > answer without errors is really important. > > Thanks again > > > On Mon, Feb 11, 2013 at 6:07 PM, < > freeswitch-users-request at lists.freeswitch.org> wrote: > >> Send FreeSWITCH-users mailing list submissions to >> freeswitch-users at lists.freeswitch.org >> >> To subscribe or unsubscribe via the World Wide Web, visit >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> or, via email, send a message with subject or body 'help' to >> freeswitch-users-request at lists.freeswitch.org >> >> You can reach the person managing the list at >> freeswitch-users-owner at lists.freeswitch.org >> >> When replying, please edit your Subject line so it is more specific >> than "Re: Contents of FreeSWITCH-users digest..." >> >> Today's Topics: >> >> 1. Re: Throttle Error (Avi Marcus) >> 2. Re: Terminating calls prior to answer using api (Steven Ayre) >> 3. Re: Terminating calls prior to answer using api (Steven Ayre) >> 4. High Availability Cluster Module for FreeSWITCH - Major News >> Update (Eliot Gable) >> 5. Re: High Availability Cluster Module for FreeSWITCH >> (Luis Daniel Lucio Quiroz) >> >> >> >> ---------- Forwarded message ---------- >> From: Avi Marcus >> To: FreeSWITCH Users Help >> Cc: >> Date: Mon, 11 Feb 2013 19:09:33 +0200 >> Subject: Re: [Freeswitch-users] Throttle Error >> >> As Ken's original message said, try "fsctl sps 40" or the like. >> He said that otherwise only a restart, not reload xml would work. >> >> -Avi >> (This message was painstakingly thumbed out on my mobile, so apologies >> for brevity and errors.) >> >> On Feb 11, 2013 6:57 PM, "Archana Venugopan" >> wrote: >> >> Hi,**** >> >> ** ** >> >> When I gave fsctl sps it shows still 10. Can you please tell me apart >> from reload_xml do we have to anyother thing after changing switch.conf.xml? >> **** >> >> ** ** >> >> Regards,**** >> >> Archana**** >> >> ** ** >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Archana >> Venugopan >> *Sent:* 11 February 2013 16:37 >> >> >> To: FreeSWITCH Users Help >> Subject: Re: [Freeswitch-users] Throttle Error >> >> >> >> >> >> Ya i did re-loaded the xml. And this is what I see in status >> >> >> >> UP 0 years, 6 days, 9 hours, 3... >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> >> >> ---------- Forwarded message ---------- >> From: Steven Ayre >> To: FreeSWITCH Users Help >> Cc: >> Date: Mon, 11 Feb 2013 17:40:57 +0000 >> Subject: Re: [Freeswitch-users] Terminating calls prior to answer using >> api >> You either answer a call with 200, redirect with 3xx, or give an error >> 4xx 5xx or 6xx... basically without answering your only option is to return >> an error. >> >> The specific error code can show a varying number of levels of 'error' >> though. Some might be a perfectly valid result (eg 486 User Busy) while >> other's imply a more serious error (eg 500 Server Internal Error). But >> there's not really one for a successful unanswered call. >> >> How your phone behaves on the various errors is really up to the SIP >> client though. A generic unable to connect message doesn't sound very >> useful, I'd expect different error tones depending on the response code. >> >> -Steve >> >> >> >> On 11 February 2013 15:30, Gary Foreman wrote: >> >>> Hi, >>> >>> I'm having an issue gracefully terminating calls prior to them being >>> answered. >>> >>> When I issue the uuid_kill command prior to the call being answered my >>> sip client gives an error "unable to connect" with an error tone. >>> >>> Is it possible to cancel the client dial without causing an error? Maybe >>> using a sip notify message or something? >>> >>> I can't seem to find a list of valid sip notify messages anywhere. >>> >>> Thanks >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> >> ---------- Forwarded message ---------- >> From: Steven Ayre >> To: FreeSWITCH Users Help >> Cc: >> Date: Mon, 11 Feb 2013 17:42:51 +0000 >> Subject: Re: [Freeswitch-users] Terminating calls prior to answer using >> api >> The NORMAL_CLEARING ISDN clearing cause (16) has no mapping to any SIP >> code for example, it's a code you normally would only see a) when the >> caller hangs up at any time (regardless of answered or not) or b) when call >> has been answered *then* callee hangs up. >> >> -Steve >> >> >> >> >> On 11 February 2013 17:40, Steven Ayre wrote: >> >>> You either answer a call with 200, redirect with 3xx, or give an error >>> 4xx 5xx or 6xx... basically without answering your only option is to return >>> an error. >>> >>> The specific error code can show a varying number of levels of 'error' >>> though. Some might be a perfectly valid result (eg 486 User Busy) while >>> other's imply a more serious error (eg 500 Server Internal Error). But >>> there's not really one for a successful unanswered call. >>> >>> How your phone behaves on the various errors is really up to the SIP >>> client though. A generic unable to connect message doesn't sound very >>> useful, I'd expect different error tones depending on the response code. >>> >>> -Steve >>> >>> >>> >>> On 11 February 2013 15:30, Gary Foreman wrote: >>> >>>> Hi, >>>> >>>> I'm having an issue gracefully terminating calls prior to them being >>>> answered. >>>> >>>> When I issue the uuid_kill command prior to the call being answered my >>>> sip client gives an error "unable to connect" with an error tone. >>>> >>>> Is it possible to cancel the client dial without causing an error? >>>> Maybe using a sip notify message or something? >>>> >>>> I can't seem to find a list of valid sip notify messages anywhere. >>>> >>>> Thanks >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>>> >>>> >>> >> >> >> ---------- Forwarded message ---------- >> From: Eliot Gable >> To: FreeSWITCH Users Help >> Cc: >> Date: Mon, 11 Feb 2013 12:44:55 -0500 >> Subject: [Freeswitch-users] High Availability Cluster Module for >> FreeSWITCH - Major News Update >> After discussing this project in further detail with Tony, I have decided >> to cancel the Kickstarter project. Funding will now go through FreeSWITCH >> Solutions. If you would like to contribute towards making mod_ha_cluster a >> reality, please send contributions to FreeSWITCH Solutions. >> >> Thanks! >> >> -- >> Eliot Gable >> >> >> >> ---------- Forwarded message ---------- >> From: Luis Daniel Lucio Quiroz >> To: FreeSWITCH Users Help >> Cc: >> Date: Mon, 11 Feb 2013 13:06:16 -0500 >> Subject: Re: [Freeswitch-users] High Availability Cluster Module for >> FreeSWITCH >> Is the latest snapshot ofyour module at >> git://git.freeswitch.org/freeswitch-contrib.git ? >> >> 2013/2/11 Eliot Gable : >> > On Mon, Feb 11, 2013 at 7:36 AM, Marcin Gozdalik >> wrote: >> >> >> >> +1 >> >> >> >> I do not doubt mod_ha is necessary inside of FS and it may be >> >> better/simpler than writing Pacemaker resource agent, but writing >> >> yet-another-cluster-communication-engine is IMHO the wrong way to go >> >> and using Corosync for communication will give a lot of value from >> >> mature codebase. >> >> >> > >> > I understand what you are saying, but what I am trying to get across is >> that >> > I am not writing yet-another-cluster-communication-engine. All I am >> really >> > doing is combining a multicast messaging API written by Tony and the >> event >> > API in FS to broadcast existing state information between multiple FS >> nodes, >> > as well as adding a tiny amount of logic on top of that to coordinate >> call >> > fail over and recovery. That's probably a little over-simplified, but it >> > gets the point across. The network communication code is already in FS >> and >> > well tested. The event system is already in FS and well tested. I have >> > already written the code to the point that it parses the configuration >> files >> > and starts sending heartbeats out all of the interfaces configured. I >> have >> > also already written a lot of the code that deals with the state >> > transitions. All I am talking about doing is implementing a tiny little >> > finite state machine. It's a pretty trivial programming task. In fact, I >> > think it was covered in my first year at Carnegie Mellon University. Of >> > course, I had already figured out how to write such things in high >> school, I >> > just did not know what it was called at that point. My point is, that >> this >> > is not yet-another-cluster-communication-engine. It is a very specific >> and >> > small finite state machine designed solely with the goal in mind of >> making >> > FS have just enough information to coordinate call fail over >> internally. If >> > I recall correctly, a lot of people also said writing >> > yet-another-VoIP-server was a waste of time, but now we have >> FreeSWITCH, and >> > it was obviously worth the effort. And I am not even trying to do >> something >> > as complex as that. If you think this is >> > yet-another-cluster-communication-engine, you are missing the point. It >> is >> > not. It never will be. >> > >> > Look at Sonus, Genband, Broadsoft, Veraz, etc. All the big-name >> > carrier-grade telecom providers have a built-in solution for automatic >> call >> > fail over. The only way FreeSWITCH will ever compete with such >> solutions is >> > if it also has that feature. Pacemaker and Corosync are overkill just >> to get >> > FS to handle single node failures and provide call recovery. It took me >> a >> > full 3 months of working with them every day to really understand how to >> > deploy them properly in conjunction with FreeSWITCH and Postgres to >> provide >> > a carrier-grade hot-standby solution which was robust enough to handle >> 99% >> > of the failures I could throw at it. Granted, this was back when the >> > configuration still needed to be written by hand in XML and prior the >> > existence of any resource agent for FreeSWITCH. But, even with those >> > changes, deploying Pacemaker and Corosync is not a simple task. If that >> is >> > the requirement for FS to have HA, it will never truly stand a chance >> > against commercial offerings. >> >> > >> > >> > >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > 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 >> > >> >> >> >> _______________________________________________ >> >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130211/e105af89/attachment-0001.html From msc at freeswitch.org Mon Feb 11 23:57:34 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 11 Feb 2013 12:57:34 -0800 Subject: [Freeswitch-users] Can local_stream MOH have a time limit/max duration? In-Reply-To: References: <219BCB63-3A5A-419F-B374-BDD45FDE3FAB@mgtech.com> Message-ID: For #2 use transfer_ringback channel variable and set it to MOH and it should work the same way, i.e. instead of ringing the caller will hear music. -MC On Fri, Feb 8, 2013 at 8:55 AM, Mario G wrote: > I looked into chime-ins, but it only chimes in at fixed intervals, and it > was not flexible enough for true customization. I could not find anything > else on the wiki that would play moh (or any other file) for a specific > time. See my previous post for a details of how it works and what I wanted > to do. Thanks. > Mario G > > On Feb 8, 2013, at 5:48 AM, Vik Killa wrote: > > > I don't quite understand what you are trying to do here. > > mod_local_stream plays audio in a loop constantly with chime-ins > > (announcements) set to play periodically. If you need something to > > play from the beginning for a certain amount of time, you need to use > > something other than mod_local_stream. > > > > > >> On Wed, Feb 6, 2013 at 9:04 AM, Mario G wrote: > >>> > >>> Could not find anything in the wiki and this may not be possible > but... I > >>> have early media with instructions, is has voice, ring, voice, music. > I want > >>> to change the ring to music but I could not find a way to limit the > time the > >>> music plays. See the second example, I would like to limit the first > moh to > >>> a maximum time. If not possible, is this a "bounty" enhancement? > Thanks for > >>> any help. > >>> Mario G > >>> > >>> > >>> Currently: > >>> >>> > data="ringback=file_string://${lua_ring1}!tone_stream://${us-ring};loops=3!file_string://${lua_ring2}!local_stream://moh"/> > >>> > >>> Would Like (add something to limit the first moh): > >>> >>> > data="ringback=file_string://${lua_ring1}!local_stream://moh!file_string://${lua_ring2}!local_stream://moh"/> > >>> > >>> > >>> ______________________________________________________________________ > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130211/c2a120d6/attachment.html From msc at freeswitch.org Tue Feb 12 00:15:24 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 11 Feb 2013 13:15:24 -0800 Subject: [Freeswitch-users] DTMF in IVR menu In-Reply-To: References: Message-ID: Yes, there is a bit of a delay between the the time the digit is entered and the time that it is officially detected and the playback of the IVR greeting is interrupted. I've not noticed that being a problem, other than people maybe wondering if their digits were not properly entered so they hit it again. Is that what is happening? -MC On Fri, Feb 8, 2013 at 4:03 PM, JP wrote: > Hi, > I am observing that when I use IVR menu, the main menu greeting does not > get stopped as soon as I enter the dtmf digit. It plays for a couple of > seconds longer and then stops and then the dtmf is processed. Is there any > tuning parameters that I can tweak to make it stop sooner. > > Thanks, > JP > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130211/37e2e4c5/attachment.html From grcamauer at gmail.com Tue Feb 12 00:21:29 2013 From: grcamauer at gmail.com (Guillermo Ruiz Camauer) Date: Mon, 11 Feb 2013 18:21:29 -0300 Subject: [Freeswitch-users] DTMF in IVR menu In-Reply-To: References: Message-ID: <2516165109928082112@unknownmsgid> I have ported an automated polling (surveys) system from Dialogic to FreeSwitch and noticed the same thing. People are used to interrupting the prompts as soon as they hear the option they like. Is there any way around this delay? Guillermo Sent from my iPhone On 11/02/2013, at 18:17, Michael Collins wrote: Yes, there is a bit of a delay between the the time the digit is entered and the time that it is officially detected and the playback of the IVR greeting is interrupted. I've not noticed that being a problem, other than people maybe wondering if their digits were not properly entered so they hit it again. Is that what is happening? -MC On Fri, Feb 8, 2013 at 4:03 PM, JP wrote: > Hi, > I am observing that when I use IVR menu, the main menu greeting does not > get stopped as soon as I enter the dtmf digit. It plays for a couple of > seconds longer and then stops and then the dtmf is processed. Is there any > tuning parameters that I can tweak to make it stop sooner. > > Thanks, > JP > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130211/69d20bc9/attachment.html From krice at freeswitch.org Tue Feb 12 00:26:36 2013 From: krice at freeswitch.org (Ken Rice) Date: Mon, 11 Feb 2013 15:26:36 -0600 Subject: [Freeswitch-users] Ken's Monday Notes: Jira, and , ZRTP updated in FreeSWITCH Message-ID: Hey Guys, Doing a little Clean up on Jira, I?ve been going thru stale bugs and trying to clean some of them out, (note I am looking at bug tickets not other types). Thanks to you who have already responded, for those who have not responded, if you have an open ticket, and one of the FS developers has asked for more information, testing etc, please follow up on the ticket. If you don?t follow up in a reasonable amount of time, we will poke you for a follow up. At that point, if you don?t follow up we will close the bug due to inactivity as we cant confirm the current status. This will free up the developers to move on to active issues. If you are not the original reporter and its a ticket you are monitoring and have valid information please chime in. We want to squash the bugs, but we need your help to make sure they are dead! Also, Big thanks to PRZ and mr Travis Cross on some updated ZRTP support. TC posted to the an open ticket today the following ?I convinced Phil to relicense the elliptic curve (ECC) code. I've already pushed this code to the libzrtp upstream repository, and I have a series of FS commits ready to push that will add this code to FS's libzrtp along with numerous other fixes and improvements that have occurred upstream. After this commit series is pushed, however, everyone who has ZRTP enabled in their build is going to need to re-bootstrap because of changes to the way libzrtp is built. Therefore Ken and I are coordinating on when to push this. I've tested with CSipSimple against a copy of FS running this patch and EC25 was successfully negotiated.? These patches have now been pushed, if you arent using ZRTP you wont notice a change, but if you are using ZRTP, you?ll want to pull, and rebuild starting with a new bootstrap and configure... Let us know how this is working! So far it is good! Have a great week! Don?t forget to join us on Wed for the regular weekly call, and join us on Friday for the Free For All. Mr Collins will be sending out his Weekly Update sooner or later with more information on these. K -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130211/08580fdb/attachment-0001.html From anthony.minessale at gmail.com Tue Feb 12 00:30:25 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 11 Feb 2013 15:30:25 -0600 Subject: [Freeswitch-users] High Availability Cluster Module for FreeSWITCH In-Reply-To: References: <-2120316468757263517@unknownmsgid> Message-ID: I believe some more planning and thought needs to go into this before proceeding with code. I recommend a continued discussion. My point of view is we should specifically review the way FS does the message exchange since its the cornerstone of many other possible cluster scenarios besides just fault tolerance. Some are starting to build this externally and I think the real answer is a more clean abstraction between the framework for many FS communicating with each other and the business logic. So some of what people like plivo and 2600hz do externally would be better served as part of FS at the comms level and then still separate the logic. That way everyone can benefit from the low level code. On Mon, Feb 11, 2013 at 12:37 PM, Henry Huang wrote: > I would choose to go with Elliot's idea of having the HA being build > within the FS itself under the condition that it's a easy adaptation > comparing to having to use and struggling with Pacemaker and Corosync. They > can literally take months if someone is starting from scratch. > > And it would be nice to have this kind of capability to go against those > brand name solutions. > > > On Mon, Feb 11, 2013 at 6:06 AM, Eliot Gable wrote: > >> On Mon, Feb 11, 2013 at 7:36 AM, Marcin Gozdalik wrote: >> >>> +1 >>> >>> I do not doubt mod_ha is necessary inside of FS and it may be >>> better/simpler than writing Pacemaker resource agent, but writing >>> yet-another-cluster-communication-engine is IMHO the wrong way to go >>> and using Corosync for communication will give a lot of value from >>> mature codebase. >>> >>> >> I understand what you are saying, but what I am trying to get across is >> that I am not writing yet-another-cluster-communication-engine. All I am >> really doing is combining a multicast messaging API written by Tony and the >> event API in FS to broadcast existing state information between multiple FS >> nodes, as well as adding a tiny amount of logic on top of that to >> coordinate call fail over and recovery. That's probably a little >> over-simplified, but it gets the point across. The network communication >> code is already in FS and well tested. The event system is already in FS >> and well tested. I have already written the code to the point that it >> parses the configuration files and starts sending heartbeats out all of the >> interfaces configured. I have also already written a lot of the code that >> deals with the state transitions. All I am talking about doing is >> implementing a tiny little finite state machine. It's a pretty trivial >> programming task. In fact, I think it was covered in my first year at >> Carnegie Mellon University. Of course, I had already figured out how to >> write such things in high school, I just did not know what it was called at >> that point. My point is, that this is not >> yet-another-cluster-communication-engine. It is a very specific and small >> finite state machine designed solely with the goal in mind of making FS >> have just enough information to coordinate call fail over internally. If I >> recall correctly, a lot of people also said writing yet-another-VoIP-server >> was a waste of time, but now we have FreeSWITCH, and it was obviously worth >> the effort. And I am not even trying to do something as complex as that. If >> you think this is yet-another-cluster-communication-engine, you are missing >> the point. It is not. It never will be. >> >> Look at Sonus, Genband, Broadsoft, Veraz, etc. All the big-name >> carrier-grade telecom providers have a built-in solution for automatic call >> fail over. The only way FreeSWITCH will ever compete with such solutions is >> if it also has that feature. Pacemaker and Corosync are overkill just to >> get FS to handle single node failures and provide call recovery. It took me >> a full 3 months of working with them every day to really understand how to >> deploy them properly in conjunction with FreeSWITCH and Postgres to provide >> a carrier-grade hot-standby solution which was robust enough to handle 99% >> of the failures I could throw at it. Granted, this was back when the >> configuration still needed to be written by hand in XML and prior the >> existence of any resource agent for FreeSWITCH. But, even with those >> changes, deploying Pacemaker and Corosync is not a simple task. If that is >> the requirement for FS to have HA, it will never truly stand a chance >> against commercial offerings. >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130211/b5810288/attachment.html From jaykris at gmail.com Tue Feb 12 00:35:01 2013 From: jaykris at gmail.com (JP) Date: Mon, 11 Feb 2013 13:35:01 -0800 Subject: [Freeswitch-users] DTMF in IVR menu In-Reply-To: References: Message-ID: Yes, that is what is happening. Is there anything that can be done? -JP On Mon, Feb 11, 2013 at 1:15 PM, Michael Collins wrote: > Yes, there is a bit of a delay between the the time the digit is entered > and the time that it is officially detected and the playback of the IVR > greeting is interrupted. I've not noticed that being a problem, other than > people maybe wondering if their digits were not properly entered so they > hit it again. Is that what is happening? > > -MC > > On Fri, Feb 8, 2013 at 4:03 PM, JP wrote: > >> Hi, >> I am observing that when I use IVR menu, the main menu greeting does not >> get stopped as soon as I enter the dtmf digit. It plays for a couple of >> seconds longer and then stops and then the dtmf is processed. Is there any >> tuning parameters that I can tweak to make it stop sooner. >> >> Thanks, >> JP >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130211/ee94b998/attachment.html From anthony.minessale at gmail.com Tue Feb 12 00:37:51 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 11 Feb 2013 15:37:51 -0600 Subject: [Freeswitch-users] Terminating calls prior to answer using api In-Reply-To: References: Message-ID: Try SWITCH_CAUSE_PICKED_OFF or SWITCH_CAUSE_LOSE_RACE that will inject headers to the phone telling it that the call was completed elsewhere which makes most of them skip counting it as an error. On Mon, Feb 11, 2013 at 2:07 PM, Steven Ayre wrote: > NOTIFY messages are a SIP feature outside of an INVITE call flow. Ending > the call using 200+BYE, 4xx, 5xx or 6xx are your only options I'm afraid. > > I'm writing a CTI application so terminating the call via the api prior to >> answer without errors is really important. > > > Any call that's unanswered is a failed call, but that's not necessarily an > error (eg user busy). > > I understand your concern that the caller might not know if their call > succeeded or not... Is there any possibility perhaps to answer the call, > send it to a playback (eg 'your request was successful') and then hangup? > > -Steve > > > > > On 11 February 2013 18:37, Gary Foreman wrote: > >> I was concerned that might be the case, I have managed to get different >> errors by send alternative hangup codes "user_busy" for example is >> recognised but as you say all are still errors. >> >> I'm planning on using Polycom sip phones, does anyone know if it is >> possible to end an unanswered call using a notify message or something? >> Like the talk sip notify message used to answer. >> >> I'm writing a CTI application so terminating the call via the api prior >> to answer without errors is really important. >> >> Thanks again >> >> >> On Mon, Feb 11, 2013 at 6:07 PM, < >> freeswitch-users-request at lists.freeswitch.org> wrote: >> >>> Send FreeSWITCH-users mailing list submissions to >>> freeswitch-users at lists.freeswitch.org >>> >>> To subscribe or unsubscribe via the World Wide Web, visit >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> or, via email, send a message with subject or body 'help' to >>> freeswitch-users-request at lists.freeswitch.org >>> >>> You can reach the person managing the list at >>> freeswitch-users-owner at lists.freeswitch.org >>> >>> When replying, please edit your Subject line so it is more specific >>> than "Re: Contents of FreeSWITCH-users digest..." >>> >>> Today's Topics: >>> >>> 1. Re: Throttle Error (Avi Marcus) >>> 2. Re: Terminating calls prior to answer using api (Steven Ayre) >>> 3. Re: Terminating calls prior to answer using api (Steven Ayre) >>> 4. High Availability Cluster Module for FreeSWITCH - Major News >>> Update (Eliot Gable) >>> 5. Re: High Availability Cluster Module for FreeSWITCH >>> (Luis Daniel Lucio Quiroz) >>> >>> >>> >>> ---------- Forwarded message ---------- >>> From: Avi Marcus >>> To: FreeSWITCH Users Help >>> Cc: >>> Date: Mon, 11 Feb 2013 19:09:33 +0200 >>> Subject: Re: [Freeswitch-users] Throttle Error >>> >>> As Ken's original message said, try "fsctl sps 40" or the like. >>> He said that otherwise only a restart, not reload xml would work. >>> >>> -Avi >>> (This message was painstakingly thumbed out on my mobile, so apologies >>> for brevity and errors.) >>> >>> On Feb 11, 2013 6:57 PM, "Archana Venugopan" >>> wrote: >>> >>> Hi,**** >>> >>> ** ** >>> >>> When I gave fsctl sps it shows still 10. Can you please tell me apart >>> from reload_xml do we have to anyother thing after changing switch.conf.xml? >>> **** >>> >>> ** ** >>> >>> Regards,**** >>> >>> Archana**** >>> >>> ** ** >>> >>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Archana >>> Venugopan >>> *Sent:* 11 February 2013 16:37 >>> >>> >>> To: FreeSWITCH Users Help >>> Subject: Re: [Freeswitch-users] Throttle Error >>> >>> >>> >>> >>> >>> Ya i did re-loaded the xml. And this is what I see in status >>> >>> >>> >>> UP 0 years, 6 days, 9 hours, 3... >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >>> >>> ---------- Forwarded message ---------- >>> From: Steven Ayre >>> To: FreeSWITCH Users Help >>> Cc: >>> Date: Mon, 11 Feb 2013 17:40:57 +0000 >>> Subject: Re: [Freeswitch-users] Terminating calls prior to answer using >>> api >>> You either answer a call with 200, redirect with 3xx, or give an error >>> 4xx 5xx or 6xx... basically without answering your only option is to return >>> an error. >>> >>> The specific error code can show a varying number of levels of 'error' >>> though. Some might be a perfectly valid result (eg 486 User Busy) while >>> other's imply a more serious error (eg 500 Server Internal Error). But >>> there's not really one for a successful unanswered call. >>> >>> How your phone behaves on the various errors is really up to the SIP >>> client though. A generic unable to connect message doesn't sound very >>> useful, I'd expect different error tones depending on the response code. >>> >>> -Steve >>> >>> >>> >>> On 11 February 2013 15:30, Gary Foreman wrote: >>> >>>> Hi, >>>> >>>> I'm having an issue gracefully terminating calls prior to them being >>>> answered. >>>> >>>> When I issue the uuid_kill command prior to the call being answered my >>>> sip client gives an error "unable to connect" with an error tone. >>>> >>>> Is it possible to cancel the client dial without causing an error? >>>> Maybe using a sip notify message or something? >>>> >>>> I can't seem to find a list of valid sip notify messages anywhere. >>>> >>>> Thanks >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>>> >>>> >>> >>> >>> ---------- Forwarded message ---------- >>> From: Steven Ayre >>> To: FreeSWITCH Users Help >>> Cc: >>> Date: Mon, 11 Feb 2013 17:42:51 +0000 >>> Subject: Re: [Freeswitch-users] Terminating calls prior to answer using >>> api >>> The NORMAL_CLEARING ISDN clearing cause (16) has no mapping to any SIP >>> code for example, it's a code you normally would only see a) when the >>> caller hangs up at any time (regardless of answered or not) or b) when call >>> has been answered *then* callee hangs up. >>> >>> -Steve >>> >>> >>> >>> >>> On 11 February 2013 17:40, Steven Ayre wrote: >>> >>>> You either answer a call with 200, redirect with 3xx, or give an error >>>> 4xx 5xx or 6xx... basically without answering your only option is to return >>>> an error. >>>> >>>> The specific error code can show a varying number of levels of 'error' >>>> though. Some might be a perfectly valid result (eg 486 User Busy) while >>>> other's imply a more serious error (eg 500 Server Internal Error). But >>>> there's not really one for a successful unanswered call. >>>> >>>> How your phone behaves on the various errors is really up to the SIP >>>> client though. A generic unable to connect message doesn't sound very >>>> useful, I'd expect different error tones depending on the response code. >>>> >>>> -Steve >>>> >>>> >>>> >>>> On 11 February 2013 15:30, Gary Foreman wrote: >>>> >>>>> Hi, >>>>> >>>>> I'm having an issue gracefully terminating calls prior to them being >>>>> answered. >>>>> >>>>> When I issue the uuid_kill command prior to the call being answered my >>>>> sip client gives an error "unable to connect" with an error tone. >>>>> >>>>> Is it possible to cancel the client dial without causing an error? >>>>> Maybe using a sip notify message or something? >>>>> >>>>> I can't seem to find a list of valid sip notify messages anywhere. >>>>> >>>>> Thanks >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> 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 >>>>> >>>>> >>>> >>> >>> >>> ---------- Forwarded message ---------- >>> From: Eliot Gable >>> To: FreeSWITCH Users Help >>> Cc: >>> Date: Mon, 11 Feb 2013 12:44:55 -0500 >>> Subject: [Freeswitch-users] High Availability Cluster Module for >>> FreeSWITCH - Major News Update >>> After discussing this project in further detail with Tony, I have >>> decided to cancel the Kickstarter project. Funding will now go through >>> FreeSWITCH Solutions. If you would like to contribute towards making >>> mod_ha_cluster a reality, please send contributions to FreeSWITCH Solutions. >>> >>> Thanks! >>> >>> -- >>> Eliot Gable >>> >>> >>> >>> ---------- Forwarded message ---------- >>> From: Luis Daniel Lucio Quiroz >>> To: FreeSWITCH Users Help >>> Cc: >>> Date: Mon, 11 Feb 2013 13:06:16 -0500 >>> Subject: Re: [Freeswitch-users] High Availability Cluster Module for >>> FreeSWITCH >>> Is the latest snapshot ofyour module at >>> git://git.freeswitch.org/freeswitch-contrib.git ? >>> >>> 2013/2/11 Eliot Gable : >>> > On Mon, Feb 11, 2013 at 7:36 AM, Marcin Gozdalik >>> wrote: >>> >> >>> >> +1 >>> >> >>> >> I do not doubt mod_ha is necessary inside of FS and it may be >>> >> better/simpler than writing Pacemaker resource agent, but writing >>> >> yet-another-cluster-communication-engine is IMHO the wrong way to go >>> >> and using Corosync for communication will give a lot of value from >>> >> mature codebase. >>> >> >>> > >>> > I understand what you are saying, but what I am trying to get across >>> is that >>> > I am not writing yet-another-cluster-communication-engine. All I am >>> really >>> > doing is combining a multicast messaging API written by Tony and the >>> event >>> > API in FS to broadcast existing state information between multiple FS >>> nodes, >>> > as well as adding a tiny amount of logic on top of that to coordinate >>> call >>> > fail over and recovery. That's probably a little over-simplified, but >>> it >>> > gets the point across. The network communication code is already in FS >>> and >>> > well tested. The event system is already in FS and well tested. I have >>> > already written the code to the point that it parses the configuration >>> files >>> > and starts sending heartbeats out all of the interfaces configured. I >>> have >>> > also already written a lot of the code that deals with the state >>> > transitions. All I am talking about doing is implementing a tiny little >>> > finite state machine. It's a pretty trivial programming task. In fact, >>> I >>> > think it was covered in my first year at Carnegie Mellon University. Of >>> > course, I had already figured out how to write such things in high >>> school, I >>> > just did not know what it was called at that point. My point is, that >>> this >>> > is not yet-another-cluster-communication-engine. It is a very specific >>> and >>> > small finite state machine designed solely with the goal in mind of >>> making >>> > FS have just enough information to coordinate call fail over >>> internally. If >>> > I recall correctly, a lot of people also said writing >>> > yet-another-VoIP-server was a waste of time, but now we have >>> FreeSWITCH, and >>> > it was obviously worth the effort. And I am not even trying to do >>> something >>> > as complex as that. If you think this is >>> > yet-another-cluster-communication-engine, you are missing the point. >>> It is >>> > not. It never will be. >>> > >>> > Look at Sonus, Genband, Broadsoft, Veraz, etc. All the big-name >>> > carrier-grade telecom providers have a built-in solution for automatic >>> call >>> > fail over. The only way FreeSWITCH will ever compete with such >>> solutions is >>> > if it also has that feature. Pacemaker and Corosync are overkill just >>> to get >>> > FS to handle single node failures and provide call recovery. It took >>> me a >>> > full 3 months of working with them every day to really understand how >>> to >>> > deploy them properly in conjunction with FreeSWITCH and Postgres to >>> provide >>> > a carrier-grade hot-standby solution which was robust enough to handle >>> 99% >>> > of the failures I could throw at it. Granted, this was back when the >>> > configuration still needed to be written by hand in XML and prior the >>> > existence of any resource agent for FreeSWITCH. But, even with those >>> > changes, deploying Pacemaker and Corosync is not a simple task. If >>> that is >>> > the requirement for FS to have HA, it will never truly stand a chance >>> > against commercial offerings. >>> >>> > >>> > >>> > >>> > >>> _________________________________________________________________________ >>> > Professional FreeSWITCH Consulting Services: >>> > consulting at freeswitch.org >>> > http://www.freeswitchsolutions.com >>> > >>> > >>> > >>> > >>> > Official FreeSWITCH Sites >>> > http://www.freeswitch.org >>> > http://wiki.freeswitch.org >>> > http://www.cluecon.com >>> > >>> > 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 >>> > >>> >>> >>> >>> _______________________________________________ >>> >>> 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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130211/06f1eb0c/attachment-0001.html From anthony.minessale at gmail.com Tue Feb 12 00:38:40 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 11 Feb 2013 15:38:40 -0600 Subject: [Freeswitch-users] Terminating calls prior to answer using api In-Reply-To: References: Message-ID: Oh, and remember to take the "SWITCH_CAUSE" out when using it with uuid_kill uuid_kill picked_off On Mon, Feb 11, 2013 at 3:37 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > Try SWITCH_CAUSE_PICKED_OFF or SWITCH_CAUSE_LOSE_RACE that will inject > headers to the phone telling it that the call was completed elsewhere which > makes most of them skip counting it as an error. > > > > > On Mon, Feb 11, 2013 at 2:07 PM, Steven Ayre wrote: > >> NOTIFY messages are a SIP feature outside of an INVITE call flow. Ending >> the call using 200+BYE, 4xx, 5xx or 6xx are your only options I'm afraid. >> >> I'm writing a CTI application so terminating the call via the api prior >>> to answer without errors is really important. >> >> >> Any call that's unanswered is a failed call, but that's not necessarily >> an error (eg user busy). >> >> I understand your concern that the caller might not know if their call >> succeeded or not... Is there any possibility perhaps to answer the call, >> send it to a playback (eg 'your request was successful') and then hangup? >> >> -Steve >> >> >> >> >> On 11 February 2013 18:37, Gary Foreman wrote: >> >>> I was concerned that might be the case, I have managed to get different >>> errors by send alternative hangup codes "user_busy" for example is >>> recognised but as you say all are still errors. >>> >>> I'm planning on using Polycom sip phones, does anyone know if it is >>> possible to end an unanswered call using a notify message or something? >>> Like the talk sip notify message used to answer. >>> >>> I'm writing a CTI application so terminating the call via the api prior >>> to answer without errors is really important. >>> >>> Thanks again >>> >>> >>> On Mon, Feb 11, 2013 at 6:07 PM, < >>> freeswitch-users-request at lists.freeswitch.org> wrote: >>> >>>> Send FreeSWITCH-users mailing list submissions to >>>> freeswitch-users at lists.freeswitch.org >>>> >>>> To subscribe or unsubscribe via the World Wide Web, visit >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> or, via email, send a message with subject or body 'help' to >>>> freeswitch-users-request at lists.freeswitch.org >>>> >>>> You can reach the person managing the list at >>>> freeswitch-users-owner at lists.freeswitch.org >>>> >>>> When replying, please edit your Subject line so it is more specific >>>> than "Re: Contents of FreeSWITCH-users digest..." >>>> >>>> Today's Topics: >>>> >>>> 1. Re: Throttle Error (Avi Marcus) >>>> 2. Re: Terminating calls prior to answer using api (Steven Ayre) >>>> 3. Re: Terminating calls prior to answer using api (Steven Ayre) >>>> 4. High Availability Cluster Module for FreeSWITCH - Major News >>>> Update (Eliot Gable) >>>> 5. Re: High Availability Cluster Module for FreeSWITCH >>>> (Luis Daniel Lucio Quiroz) >>>> >>>> >>>> >>>> ---------- Forwarded message ---------- >>>> From: Avi Marcus >>>> To: FreeSWITCH Users Help >>>> Cc: >>>> Date: Mon, 11 Feb 2013 19:09:33 +0200 >>>> Subject: Re: [Freeswitch-users] Throttle Error >>>> >>>> As Ken's original message said, try "fsctl sps 40" or the like. >>>> He said that otherwise only a restart, not reload xml would work. >>>> >>>> -Avi >>>> (This message was painstakingly thumbed out on my mobile, so apologies >>>> for brevity and errors.) >>>> >>>> On Feb 11, 2013 6:57 PM, "Archana Venugopan" >>>> wrote: >>>> >>>> Hi,**** >>>> >>>> ** ** >>>> >>>> When I gave fsctl sps it shows still 10. Can you please tell me apart >>>> from reload_xml do we have to anyother thing after changing switch.conf.xml? >>>> **** >>>> >>>> ** ** >>>> >>>> Regards,**** >>>> >>>> Archana**** >>>> >>>> ** ** >>>> >>>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >>>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Archana >>>> Venugopan >>>> *Sent:* 11 February 2013 16:37 >>>> >>>> >>>> To: FreeSWITCH Users Help >>>> Subject: Re: [Freeswitch-users] Throttle Error >>>> >>>> >>>> >>>> >>>> >>>> Ya i did re-loaded the xml. And this is what I see in status >>>> >>>> >>>> >>>> UP 0 years, 6 days, 9 hours, 3... >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>>> >>>> >>>> >>>> ---------- Forwarded message ---------- >>>> From: Steven Ayre >>>> To: FreeSWITCH Users Help >>>> Cc: >>>> Date: Mon, 11 Feb 2013 17:40:57 +0000 >>>> Subject: Re: [Freeswitch-users] Terminating calls prior to answer using >>>> api >>>> You either answer a call with 200, redirect with 3xx, or give an error >>>> 4xx 5xx or 6xx... basically without answering your only option is to return >>>> an error. >>>> >>>> The specific error code can show a varying number of levels of 'error' >>>> though. Some might be a perfectly valid result (eg 486 User Busy) while >>>> other's imply a more serious error (eg 500 Server Internal Error). But >>>> there's not really one for a successful unanswered call. >>>> >>>> How your phone behaves on the various errors is really up to the SIP >>>> client though. A generic unable to connect message doesn't sound very >>>> useful, I'd expect different error tones depending on the response code. >>>> >>>> -Steve >>>> >>>> >>>> >>>> On 11 February 2013 15:30, Gary Foreman wrote: >>>> >>>>> Hi, >>>>> >>>>> I'm having an issue gracefully terminating calls prior to them being >>>>> answered. >>>>> >>>>> When I issue the uuid_kill command prior to the call being answered my >>>>> sip client gives an error "unable to connect" with an error tone. >>>>> >>>>> Is it possible to cancel the client dial without causing an error? >>>>> Maybe using a sip notify message or something? >>>>> >>>>> I can't seem to find a list of valid sip notify messages anywhere. >>>>> >>>>> Thanks >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> 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 >>>>> >>>>> >>>> >>>> >>>> ---------- Forwarded message ---------- >>>> From: Steven Ayre >>>> To: FreeSWITCH Users Help >>>> Cc: >>>> Date: Mon, 11 Feb 2013 17:42:51 +0000 >>>> Subject: Re: [Freeswitch-users] Terminating calls prior to answer using >>>> api >>>> The NORMAL_CLEARING ISDN clearing cause (16) has no mapping to any SIP >>>> code for example, it's a code you normally would only see a) when the >>>> caller hangs up at any time (regardless of answered or not) or b) when call >>>> has been answered *then* callee hangs up. >>>> >>>> -Steve >>>> >>>> >>>> >>>> >>>> On 11 February 2013 17:40, Steven Ayre wrote: >>>> >>>>> You either answer a call with 200, redirect with 3xx, or give an error >>>>> 4xx 5xx or 6xx... basically without answering your only option is to return >>>>> an error. >>>>> >>>>> The specific error code can show a varying number of levels of 'error' >>>>> though. Some might be a perfectly valid result (eg 486 User Busy) while >>>>> other's imply a more serious error (eg 500 Server Internal Error). But >>>>> there's not really one for a successful unanswered call. >>>>> >>>>> How your phone behaves on the various errors is really up to the SIP >>>>> client though. A generic unable to connect message doesn't sound very >>>>> useful, I'd expect different error tones depending on the response code. >>>>> >>>>> -Steve >>>>> >>>>> >>>>> >>>>> On 11 February 2013 15:30, Gary Foreman wrote: >>>>> >>>>>> Hi, >>>>>> >>>>>> I'm having an issue gracefully terminating calls prior to them being >>>>>> answered. >>>>>> >>>>>> When I issue the uuid_kill command prior to the call being answered >>>>>> my sip client gives an error "unable to connect" with an error tone. >>>>>> >>>>>> Is it possible to cancel the client dial without causing an error? >>>>>> Maybe using a sip notify message or something? >>>>>> >>>>>> I can't seem to find a list of valid sip notify messages anywhere. >>>>>> >>>>>> Thanks >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> 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 >>>>>> >>>>>> >>>>> >>>> >>>> >>>> ---------- Forwarded message ---------- >>>> From: Eliot Gable >>>> To: FreeSWITCH Users Help >>>> Cc: >>>> Date: Mon, 11 Feb 2013 12:44:55 -0500 >>>> Subject: [Freeswitch-users] High Availability Cluster Module for >>>> FreeSWITCH - Major News Update >>>> After discussing this project in further detail with Tony, I have >>>> decided to cancel the Kickstarter project. Funding will now go through >>>> FreeSWITCH Solutions. If you would like to contribute towards making >>>> mod_ha_cluster a reality, please send contributions to FreeSWITCH Solutions. >>>> >>>> Thanks! >>>> >>>> -- >>>> Eliot Gable >>>> >>>> >>>> >>>> ---------- Forwarded message ---------- >>>> From: Luis Daniel Lucio Quiroz >>>> To: FreeSWITCH Users Help >>>> Cc: >>>> Date: Mon, 11 Feb 2013 13:06:16 -0500 >>>> Subject: Re: [Freeswitch-users] High Availability Cluster Module for >>>> FreeSWITCH >>>> Is the latest snapshot ofyour module at >>>> git://git.freeswitch.org/freeswitch-contrib.git ? >>>> >>>> 2013/2/11 Eliot Gable : >>>> > On Mon, Feb 11, 2013 at 7:36 AM, Marcin Gozdalik >>>> wrote: >>>> >> >>>> >> +1 >>>> >> >>>> >> I do not doubt mod_ha is necessary inside of FS and it may be >>>> >> better/simpler than writing Pacemaker resource agent, but writing >>>> >> yet-another-cluster-communication-engine is IMHO the wrong way to go >>>> >> and using Corosync for communication will give a lot of value from >>>> >> mature codebase. >>>> >> >>>> > >>>> > I understand what you are saying, but what I am trying to get across >>>> is that >>>> > I am not writing yet-another-cluster-communication-engine. All I am >>>> really >>>> > doing is combining a multicast messaging API written by Tony and the >>>> event >>>> > API in FS to broadcast existing state information between multiple FS >>>> nodes, >>>> > as well as adding a tiny amount of logic on top of that to coordinate >>>> call >>>> > fail over and recovery. That's probably a little over-simplified, but >>>> it >>>> > gets the point across. The network communication code is already in >>>> FS and >>>> > well tested. The event system is already in FS and well tested. I have >>>> > already written the code to the point that it parses the >>>> configuration files >>>> > and starts sending heartbeats out all of the interfaces configured. I >>>> have >>>> > also already written a lot of the code that deals with the state >>>> > transitions. All I am talking about doing is implementing a tiny >>>> little >>>> > finite state machine. It's a pretty trivial programming task. In >>>> fact, I >>>> > think it was covered in my first year at Carnegie Mellon University. >>>> Of >>>> > course, I had already figured out how to write such things in high >>>> school, I >>>> > just did not know what it was called at that point. My point is, that >>>> this >>>> > is not yet-another-cluster-communication-engine. It is a very >>>> specific and >>>> > small finite state machine designed solely with the goal in mind of >>>> making >>>> > FS have just enough information to coordinate call fail over >>>> internally. If >>>> > I recall correctly, a lot of people also said writing >>>> > yet-another-VoIP-server was a waste of time, but now we have >>>> FreeSWITCH, and >>>> > it was obviously worth the effort. And I am not even trying to do >>>> something >>>> > as complex as that. If you think this is >>>> > yet-another-cluster-communication-engine, you are missing the point. >>>> It is >>>> > not. It never will be. >>>> > >>>> > Look at Sonus, Genband, Broadsoft, Veraz, etc. All the big-name >>>> > carrier-grade telecom providers have a built-in solution for >>>> automatic call >>>> > fail over. The only way FreeSWITCH will ever compete with such >>>> solutions is >>>> > if it also has that feature. Pacemaker and Corosync are overkill just >>>> to get >>>> > FS to handle single node failures and provide call recovery. It took >>>> me a >>>> > full 3 months of working with them every day to really understand how >>>> to >>>> > deploy them properly in conjunction with FreeSWITCH and Postgres to >>>> provide >>>> > a carrier-grade hot-standby solution which was robust enough to >>>> handle 99% >>>> > of the failures I could throw at it. Granted, this was back when the >>>> > configuration still needed to be written by hand in XML and prior the >>>> > existence of any resource agent for FreeSWITCH. But, even with those >>>> > changes, deploying Pacemaker and Corosync is not a simple task. If >>>> that is >>>> > the requirement for FS to have HA, it will never truly stand a chance >>>> > against commercial offerings. >>>> >>>> > >>>> > >>>> > >>>> > >>>> _________________________________________________________________________ >>>> > Professional FreeSWITCH Consulting Services: >>>> > consulting at freeswitch.org >>>> > http://www.freeswitchsolutions.com >>>> > >>>> > >>>> > >>>> > >>>> > Official FreeSWITCH Sites >>>> > http://www.freeswitch.org >>>> > http://wiki.freeswitch.org >>>> > http://www.cluecon.com >>>> > >>>> > 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 >>>> > >>>> >>>> >>>> >>>> _______________________________________________ >>>> >>>> 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 >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130211/3035c200/attachment-0001.html From POlsson at enghouse.com Tue Feb 12 00:46:03 2013 From: POlsson at enghouse.com (Peter Olsson) Date: Mon, 11 Feb 2013 21:46:03 +0000 Subject: [Freeswitch-users] DTMF in IVR menu In-Reply-To: References: , Message-ID: <1FFF97C269757C458224B7C895F35F1521EC89@cantor.std.visionutv.se> Do a pcap and try to find out where the delay for a couple of seconds occur. I've seen delays of maximum 200ms, so a couple of seconds seems like a very long time. /Peter ________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r JP [jaykris at gmail.com] Skickat: den 11 februari 2013 22:35 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] DTMF in IVR menu Yes, that is what is happening. Is there anything that can be done? -JP On Mon, Feb 11, 2013 at 1:15 PM, Michael Collins > wrote: Yes, there is a bit of a delay between the the time the digit is entered and the time that it is officially detected and the playback of the IVR greeting is interrupted. I've not noticed that being a problem, other than people maybe wondering if their digits were not properly entered so they hit it again. Is that what is happening? -MC On Fri, Feb 8, 2013 at 4:03 PM, JP > wrote: Hi, I am observing that when I use IVR menu, the main menu greeting does not get stopped as soon as I enter the dtmf digit. It plays for a couple of seconds longer and then stops and then the dtmf is processed. Is there any tuning parameters that I can tweak to make it stop sooner. Thanks, JP _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 !DSPAM:5119606c32761732238334! From steveayre at gmail.com Tue Feb 12 00:49:56 2013 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 11 Feb 2013 21:49:56 +0000 Subject: [Freeswitch-users] Terminating calls prior to answer using api In-Reply-To: References: Message-ID: Interesting :) On 11 February 2013 21:38, Anthony Minessale wrote: > Oh, and remember to take the "SWITCH_CAUSE" out when using it with > uuid_kill > > uuid_kill picked_off > > > > On Mon, Feb 11, 2013 at 3:37 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> Try SWITCH_CAUSE_PICKED_OFF or SWITCH_CAUSE_LOSE_RACE that will inject >> headers to the phone telling it that the call was completed elsewhere which >> makes most of them skip counting it as an error. >> >> >> >> >> On Mon, Feb 11, 2013 at 2:07 PM, Steven Ayre wrote: >> >>> NOTIFY messages are a SIP feature outside of an INVITE call flow. Ending >>> the call using 200+BYE, 4xx, 5xx or 6xx are your only options I'm afraid. >>> >>> I'm writing a CTI application so terminating the call via the api prior >>>> to answer without errors is really important. >>> >>> >>> Any call that's unanswered is a failed call, but that's not necessarily >>> an error (eg user busy). >>> >>> I understand your concern that the caller might not know if their call >>> succeeded or not... Is there any possibility perhaps to answer the call, >>> send it to a playback (eg 'your request was successful') and then hangup? >>> >>> -Steve >>> >>> >>> >>> >>> On 11 February 2013 18:37, Gary Foreman wrote: >>> >>>> I was concerned that might be the case, I have managed to get different >>>> errors by send alternative hangup codes "user_busy" for example is >>>> recognised but as you say all are still errors. >>>> >>>> I'm planning on using Polycom sip phones, does anyone know if it is >>>> possible to end an unanswered call using a notify message or something? >>>> Like the talk sip notify message used to answer. >>>> >>>> I'm writing a CTI application so terminating the call via the api prior >>>> to answer without errors is really important. >>>> >>>> Thanks again >>>> >>>> >>>> On Mon, Feb 11, 2013 at 6:07 PM, < >>>> freeswitch-users-request at lists.freeswitch.org> wrote: >>>> >>>>> Send FreeSWITCH-users mailing list submissions to >>>>> freeswitch-users at lists.freeswitch.org >>>>> >>>>> To subscribe or unsubscribe via the World Wide Web, visit >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> or, via email, send a message with subject or body 'help' to >>>>> freeswitch-users-request at lists.freeswitch.org >>>>> >>>>> You can reach the person managing the list at >>>>> freeswitch-users-owner at lists.freeswitch.org >>>>> >>>>> When replying, please edit your Subject line so it is more specific >>>>> than "Re: Contents of FreeSWITCH-users digest..." >>>>> >>>>> Today's Topics: >>>>> >>>>> 1. Re: Throttle Error (Avi Marcus) >>>>> 2. Re: Terminating calls prior to answer using api (Steven Ayre) >>>>> 3. Re: Terminating calls prior to answer using api (Steven Ayre) >>>>> 4. High Availability Cluster Module for FreeSWITCH - Major News >>>>> Update (Eliot Gable) >>>>> 5. Re: High Availability Cluster Module for FreeSWITCH >>>>> (Luis Daniel Lucio Quiroz) >>>>> >>>>> >>>>> >>>>> ---------- Forwarded message ---------- >>>>> From: Avi Marcus >>>>> To: FreeSWITCH Users Help >>>>> Cc: >>>>> Date: Mon, 11 Feb 2013 19:09:33 +0200 >>>>> Subject: Re: [Freeswitch-users] Throttle Error >>>>> >>>>> As Ken's original message said, try "fsctl sps 40" or the like. >>>>> He said that otherwise only a restart, not reload xml would work. >>>>> >>>>> -Avi >>>>> (This message was painstakingly thumbed out on my mobile, so apologies >>>>> for brevity and errors.) >>>>> >>>>> On Feb 11, 2013 6:57 PM, "Archana Venugopan" >>>>> wrote: >>>>> >>>>> Hi,**** >>>>> >>>>> ** ** >>>>> >>>>> When I gave fsctl sps it shows still 10. Can you please tell me apart >>>>> from reload_xml do we have to anyother thing after changing switch.conf.xml? >>>>> **** >>>>> >>>>> ** ** >>>>> >>>>> Regards,**** >>>>> >>>>> Archana**** >>>>> >>>>> ** ** >>>>> >>>>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >>>>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Archana >>>>> Venugopan >>>>> *Sent:* 11 February 2013 16:37 >>>>> >>>>> >>>>> To: FreeSWITCH Users Help >>>>> Subject: Re: [Freeswitch-users] Throttle Error >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> Ya i did re-loaded the xml. And this is what I see in status >>>>> >>>>> >>>>> >>>>> UP 0 years, 6 days, 9 hours, 3... >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> 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 >>>>> >>>>> >>>>> >>>>> ---------- Forwarded message ---------- >>>>> From: Steven Ayre >>>>> To: FreeSWITCH Users Help >>>>> Cc: >>>>> Date: Mon, 11 Feb 2013 17:40:57 +0000 >>>>> Subject: Re: [Freeswitch-users] Terminating calls prior to answer >>>>> using api >>>>> You either answer a call with 200, redirect with 3xx, or give an error >>>>> 4xx 5xx or 6xx... basically without answering your only option is to return >>>>> an error. >>>>> >>>>> The specific error code can show a varying number of levels of 'error' >>>>> though. Some might be a perfectly valid result (eg 486 User Busy) while >>>>> other's imply a more serious error (eg 500 Server Internal Error). But >>>>> there's not really one for a successful unanswered call. >>>>> >>>>> How your phone behaves on the various errors is really up to the SIP >>>>> client though. A generic unable to connect message doesn't sound very >>>>> useful, I'd expect different error tones depending on the response code. >>>>> >>>>> -Steve >>>>> >>>>> >>>>> >>>>> On 11 February 2013 15:30, Gary Foreman wrote: >>>>> >>>>>> Hi, >>>>>> >>>>>> I'm having an issue gracefully terminating calls prior to them being >>>>>> answered. >>>>>> >>>>>> When I issue the uuid_kill command prior to the call being answered >>>>>> my sip client gives an error "unable to connect" with an error tone. >>>>>> >>>>>> Is it possible to cancel the client dial without causing an error? >>>>>> Maybe using a sip notify message or something? >>>>>> >>>>>> I can't seem to find a list of valid sip notify messages anywhere. >>>>>> >>>>>> Thanks >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> 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 >>>>>> >>>>>> >>>>> >>>>> >>>>> ---------- Forwarded message ---------- >>>>> From: Steven Ayre >>>>> To: FreeSWITCH Users Help >>>>> Cc: >>>>> Date: Mon, 11 Feb 2013 17:42:51 +0000 >>>>> Subject: Re: [Freeswitch-users] Terminating calls prior to answer >>>>> using api >>>>> The NORMAL_CLEARING ISDN clearing cause (16) has no mapping to any SIP >>>>> code for example, it's a code you normally would only see a) when the >>>>> caller hangs up at any time (regardless of answered or not) or b) when call >>>>> has been answered *then* callee hangs up. >>>>> >>>>> -Steve >>>>> >>>>> >>>>> >>>>> >>>>> On 11 February 2013 17:40, Steven Ayre wrote: >>>>> >>>>>> You either answer a call with 200, redirect with 3xx, or give an >>>>>> error 4xx 5xx or 6xx... basically without answering your only option is to >>>>>> return an error. >>>>>> >>>>>> The specific error code can show a varying number of levels of >>>>>> 'error' though. Some might be a perfectly valid result (eg 486 User Busy) >>>>>> while other's imply a more serious error (eg 500 Server Internal Error). >>>>>> But there's not really one for a successful unanswered call. >>>>>> >>>>>> How your phone behaves on the various errors is really up to the SIP >>>>>> client though. A generic unable to connect message doesn't sound very >>>>>> useful, I'd expect different error tones depending on the response code. >>>>>> >>>>>> -Steve >>>>>> >>>>>> >>>>>> >>>>>> On 11 February 2013 15:30, Gary Foreman wrote: >>>>>> >>>>>>> Hi, >>>>>>> >>>>>>> I'm having an issue gracefully terminating calls prior to them being >>>>>>> answered. >>>>>>> >>>>>>> When I issue the uuid_kill command prior to the call being answered >>>>>>> my sip client gives an error "unable to connect" with an error tone. >>>>>>> >>>>>>> Is it possible to cancel the client dial without causing an error? >>>>>>> Maybe using a sip notify message or something? >>>>>>> >>>>>>> I can't seem to find a list of valid sip notify messages anywhere. >>>>>>> >>>>>>> Thanks >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://wiki.freeswitch.org >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> 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 >>>>>>> >>>>>>> >>>>>> >>>>> >>>>> >>>>> ---------- Forwarded message ---------- >>>>> From: Eliot Gable >>>>> To: FreeSWITCH Users Help >>>>> Cc: >>>>> Date: Mon, 11 Feb 2013 12:44:55 -0500 >>>>> Subject: [Freeswitch-users] High Availability Cluster Module for >>>>> FreeSWITCH - Major News Update >>>>> After discussing this project in further detail with Tony, I have >>>>> decided to cancel the Kickstarter project. Funding will now go through >>>>> FreeSWITCH Solutions. If you would like to contribute towards making >>>>> mod_ha_cluster a reality, please send contributions to FreeSWITCH Solutions. >>>>> >>>>> Thanks! >>>>> >>>>> -- >>>>> Eliot Gable >>>>> >>>>> >>>>> >>>>> ---------- Forwarded message ---------- >>>>> From: Luis Daniel Lucio Quiroz >>>>> To: FreeSWITCH Users Help >>>>> Cc: >>>>> Date: Mon, 11 Feb 2013 13:06:16 -0500 >>>>> Subject: Re: [Freeswitch-users] High Availability Cluster Module for >>>>> FreeSWITCH >>>>> Is the latest snapshot ofyour module at >>>>> git://git.freeswitch.org/freeswitch-contrib.git ? >>>>> >>>>> 2013/2/11 Eliot Gable : >>>>> > On Mon, Feb 11, 2013 at 7:36 AM, Marcin Gozdalik >>>>> wrote: >>>>> >> >>>>> >> +1 >>>>> >> >>>>> >> I do not doubt mod_ha is necessary inside of FS and it may be >>>>> >> better/simpler than writing Pacemaker resource agent, but writing >>>>> >> yet-another-cluster-communication-engine is IMHO the wrong way to go >>>>> >> and using Corosync for communication will give a lot of value from >>>>> >> mature codebase. >>>>> >> >>>>> > >>>>> > I understand what you are saying, but what I am trying to get across >>>>> is that >>>>> > I am not writing yet-another-cluster-communication-engine. All I am >>>>> really >>>>> > doing is combining a multicast messaging API written by Tony and the >>>>> event >>>>> > API in FS to broadcast existing state information between multiple >>>>> FS nodes, >>>>> > as well as adding a tiny amount of logic on top of that to >>>>> coordinate call >>>>> > fail over and recovery. That's probably a little over-simplified, >>>>> but it >>>>> > gets the point across. The network communication code is already in >>>>> FS and >>>>> > well tested. The event system is already in FS and well tested. I >>>>> have >>>>> > already written the code to the point that it parses the >>>>> configuration files >>>>> > and starts sending heartbeats out all of the interfaces configured. >>>>> I have >>>>> > also already written a lot of the code that deals with the state >>>>> > transitions. All I am talking about doing is implementing a tiny >>>>> little >>>>> > finite state machine. It's a pretty trivial programming task. In >>>>> fact, I >>>>> > think it was covered in my first year at Carnegie Mellon University. >>>>> Of >>>>> > course, I had already figured out how to write such things in high >>>>> school, I >>>>> > just did not know what it was called at that point. My point is, >>>>> that this >>>>> > is not yet-another-cluster-communication-engine. It is a very >>>>> specific and >>>>> > small finite state machine designed solely with the goal in mind of >>>>> making >>>>> > FS have just enough information to coordinate call fail over >>>>> internally. If >>>>> > I recall correctly, a lot of people also said writing >>>>> > yet-another-VoIP-server was a waste of time, but now we have >>>>> FreeSWITCH, and >>>>> > it was obviously worth the effort. And I am not even trying to do >>>>> something >>>>> > as complex as that. If you think this is >>>>> > yet-another-cluster-communication-engine, you are missing the point. >>>>> It is >>>>> > not. It never will be. >>>>> > >>>>> > Look at Sonus, Genband, Broadsoft, Veraz, etc. All the big-name >>>>> > carrier-grade telecom providers have a built-in solution for >>>>> automatic call >>>>> > fail over. The only way FreeSWITCH will ever compete with such >>>>> solutions is >>>>> > if it also has that feature. Pacemaker and Corosync are overkill >>>>> just to get >>>>> > FS to handle single node failures and provide call recovery. It took >>>>> me a >>>>> > full 3 months of working with them every day to really understand >>>>> how to >>>>> > deploy them properly in conjunction with FreeSWITCH and Postgres to >>>>> provide >>>>> > a carrier-grade hot-standby solution which was robust enough to >>>>> handle 99% >>>>> > of the failures I could throw at it. Granted, this was back when the >>>>> > configuration still needed to be written by hand in XML and prior the >>>>> > existence of any resource agent for FreeSWITCH. But, even with those >>>>> > changes, deploying Pacemaker and Corosync is not a simple task. If >>>>> that is >>>>> > the requirement for FS to have HA, it will never truly stand a chance >>>>> > against commercial offerings. >>>>> >>>>> > >>>>> > >>>>> > >>>>> > >>>>> _________________________________________________________________________ >>>>> > Professional FreeSWITCH Consulting Services: >>>>> > consulting at freeswitch.org >>>>> > http://www.freeswitchsolutions.com >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > Official FreeSWITCH Sites >>>>> > http://www.freeswitch.org >>>>> > http://wiki.freeswitch.org >>>>> > http://www.cluecon.com >>>>> > >>>>> > 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 >>>>> > >>>>> >>>>> >>>>> >>>>> _______________________________________________ >>>>> >>>>> 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 >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130211/0646c698/attachment-0001.html From steveayre at gmail.com Tue Feb 12 00:52:16 2013 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 11 Feb 2013 21:52:16 +0000 Subject: [Freeswitch-users] DTMF in IVR menu In-Reply-To: <2516165109928082112@unknownmsgid> References: <2516165109928082112@unknownmsgid> Message-ID: Out of interest, do you know the RTT from phone to server, and the size of the jitterbuffers? Just wondering whether the lag is a result of the two combined. DTMF digits can also last a noticeable length of time (often ~400ms) and I believe get detected when they 'end'. -Steve On 11 February 2013 21:21, Guillermo Ruiz Camauer wrote: > I have ported an automated polling (surveys) system from Dialogic to > FreeSwitch and noticed the same thing. People are used to interrupting the > prompts as soon as they hear the option they like. Is there any way around > this delay? > > Guillermo > > Sent from my iPhone > > On 11/02/2013, at 18:17, Michael Collins wrote: > > Yes, there is a bit of a delay between the the time the digit is entered > and the time that it is officially detected and the playback of the IVR > greeting is interrupted. I've not noticed that being a problem, other than > people maybe wondering if their digits were not properly entered so they > hit it again. Is that what is happening? > > -MC > > On Fri, Feb 8, 2013 at 4:03 PM, JP wrote: > >> Hi, >> I am observing that when I use IVR menu, the main menu greeting does not >> get stopped as soon as I enter the dtmf digit. It plays for a couple of >> seconds longer and then stops and then the dtmf is processed. Is there any >> tuning parameters that I can tweak to make it stop sooner. >> >> Thanks, >> JP >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130211/7d46ccc3/attachment.html From jaykris at gmail.com Tue Feb 12 01:18:30 2013 From: jaykris at gmail.com (JP) Date: Mon, 11 Feb 2013 14:18:30 -0800 Subject: [Freeswitch-users] DTMF in IVR menu In-Reply-To: References: <2516165109928082112@unknownmsgid> Message-ID: I am using Counterpath's eyebeam and their documentation says jitterbuffer is dynamic and cannot be configured. I don't have this problem with play-and-get-digits, so I don't think the RTT is a problem. I was wondering would tweaking the following parameters help in any way? min_dtmf_duration max_dtmf_duration default_dtmf_duration Thanks, JP On Mon, Feb 11, 2013 at 1:52 PM, Steven Ayre wrote: > Out of interest, do you know the RTT from phone to server, and the size of > the jitterbuffers? > > Just wondering whether the lag is a result of the two combined. DTMF > digits can also last a noticeable length of time (often ~400ms) and I > believe get detected when they 'end'. > > -Steve > > > > > On 11 February 2013 21:21, Guillermo Ruiz Camauer wrote: > >> I have ported an automated polling (surveys) system from Dialogic to >> FreeSwitch and noticed the same thing. People are used to interrupting the >> prompts as soon as they hear the option they like. Is there any way around >> this delay? >> >> Guillermo >> >> Sent from my iPhone >> >> On 11/02/2013, at 18:17, Michael Collins wrote: >> >> Yes, there is a bit of a delay between the the time the digit is entered >> and the time that it is officially detected and the playback of the IVR >> greeting is interrupted. I've not noticed that being a problem, other than >> people maybe wondering if their digits were not properly entered so they >> hit it again. Is that what is happening? >> >> -MC >> >> On Fri, Feb 8, 2013 at 4:03 PM, JP wrote: >> >>> Hi, >>> I am observing that when I use IVR menu, the main menu greeting does not >>> get stopped as soon as I enter the dtmf digit. It plays for a couple of >>> seconds longer and then stops and then the dtmf is processed. Is there any >>> tuning parameters that I can tweak to make it stop sooner. >>> >>> Thanks, >>> JP >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130211/04cf90b7/attachment.html From msc at freeswitch.org Tue Feb 12 01:52:20 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 11 Feb 2013 14:52:20 -0800 Subject: [Freeswitch-users] Conference moderator via the ESL API? In-Reply-To: References: Message-ID: You can set the moderator flag on a member prior to his/her entry into the conference, but not after they've entered. Instead of using conference dial you could do something like this: originate user/1000 &conference(confname++flags{moderator}) or originate sofia/gateway/18005551212 &conference(confname++flags{moderator}) Those are how you'd type them at fs_cli. You can also use event socket/ESL and the api (or bgapi) event socket command to send the originate. -MC On Sat, Feb 9, 2013 at 1:01 AM, Martyn Davies wrote: > Is there a way to switch the moderator flag on/off for members of an > active conference, and can this be done from the API? > > Also, can the moderator flag be switched on at "conference dial..."? The > only way I can see to control mod status seems to be to define two profiles > in the config. > > Regards, > Martyn > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130211/9c6f0dce/attachment-0001.html From msc at freeswitch.org Tue Feb 12 01:53:21 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 11 Feb 2013 14:53:21 -0800 Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 80, Issue 70 In-Reply-To: References: Message-ID: See my other post about using originate, where you can specify all sorts of flags when sending the dialed leg into the conference. -MC On Sun, Feb 10, 2013 at 8:27 AM, Martyn Davies wrote: > Hi Jun, > > There is a member-flag called endconf which does what you want. However, > I'm not sure how to specify that as part of "conference dial"... > > Regards, > Martyn > > On 10 February 2013 11:54, wrote: > >> Send FreeSWITCH-users mailing list submissions to >> freeswitch-users at lists.freeswitch.org >> >> To subscribe or unsubscribe via the World Wide Web, visit >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> or, via email, send a message with subject or body 'help' to >> freeswitch-users-request at lists.freeswitch.org >> >> You can reach the person managing the list at >> freeswitch-users-owner at lists.freeswitch.org >> >> When replying, please edit your Subject line so it is more specific >> than "Re: Contents of FreeSWITCH-users digest..." >> >> Today's Topics: >> >> 1. how to end conference after initiator hangs up (dialout >> style) (Jun Sun) >> 2. Re: High Availability Cluster Module for FreeSWITCH (Gilad Abada) >> 3. Re: High Availability Cluster Module for FreeSWITCH (Eliot Gable) >> 4. USSD support for mod_gsmopen (Yuriy Ostapchuk) >> 5. Re: VoIP gateway with PRI (Yehavi Bourvine) >> 6. Re: USSD support for mod_gsmopen (Giovanni Maruzzelli) >> 7. Re: High Availability Cluster Module for FreeSWITCH >> (Marcin Gozdalik) >> >> >> ---------- Forwarded message ---------- >> From: Jun Sun >> To: freeswitch-users at lists.freeswitch.org >> Cc: >> Date: Sat, 09 Feb 2013 14:52:15 -0800 >> Subject: [Freeswitch-users] how to end conference after initiator hangs >> up (dialout style) >> >> I have set up a conferencing call node using dialout style. That is, I >> use the following commands to bridge everyone into a conference: >> >> conference $myRoom dial sofia/internal/$myNumber at x.x.**x.x:5060 >> $myNumber conference_test"; >> conference $myRoom dial sofia/internal/$theirNumber1@**x.x.x.x:5060 >> $myNumber conference_test"; >> conference $myRoom dial sofia/internal/$theirNumber2@**x.x.x.x:5060 >> $myNumber conference_test"; >> ... >> >> Right now things are working fine except for one thing. I like to end the >> conference (i.e., all legs are disconnected) when the initiator ($myNumber) >> hangs up. But it is not doing that (understandably so!). >> >> How can I achieve that? >> >> I searched around and it seems most conf call controls only apply to >> dial-in style conference calls. >> >> Thanks for any pointers! >> >> Jun >> >> >> >> >> ---------- Forwarded message ---------- >> From: Gilad Abada >> To: FreeSWITCH Users Help >> Cc: >> Date: Sat, 9 Feb 2013 18:49:56 -0500 >> Subject: Re: [Freeswitch-users] High Availability Cluster Module for >> FreeSWITCH >> Hey Eliot >> >> Sounds exciting. Which DB will you be using Postgres? I will try to >> get my company to pitch in a few dollars. >> >> Sent from my mobile device. >> >> On Feb 9, 2013, at 12:11 PM, Eliot Gable >> wrote: >> >> > For the past two years, I have been working on and off on a high >> > availability cluster module for FreeSWITCH. If you would like to see >> > HA built into FreeSWITCH directly, instead of dealing with Pacemaker + >> > Corosync or other 3rd party systems, please check out the module, >> > voice your support, provide feedback and suggestions, and consider >> > supporting it. >> > >> > Some advantages over a 3rd party solution like Pacemaker include: >> > >> > - Sub-second failure detection and response (meaning it can start >> > recovering calls less than a second after a failure occurs) >> > - Failure detection is built into FreeSWITCH so it can detect >> > conditions which 3rd party modules cannot detect >> > - One shared configuration file between all nodes (no per-node IP >> > configuration required) >> > - No fixed number of nodes >> > - Graceful migration of calls with a maintenance mode (requires DNS) >> > >> > These are just some of the benefits of building a FreeSWITCH-specific >> > HA system. For additional details about the project, please visit: >> > >> > http://wiki.freeswitch.org/wiki/Mod_ha_cluster >> > >> > >> > If you would like to see this project come to fruition, please >> > consider financially supporting it through Kickstarter: >> > >> > >> http://www.kickstarter.com/projects/1175121367/high-availability-cluster-module-for-freeswitch >> > >> > >> > >> > Thanks in advance for anyone who contributes, either with just a >> > shout-out, with feedback and/or suggestions, and especially to those >> > who pledge some financial support. >> > >> > -- >> > Eliot Gable >> > >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > 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 >> >> >> >> >> ---------- Forwarded message ---------- >> From: Eliot Gable >> To: FreeSWITCH Users Help >> Cc: >> Date: Sat, 9 Feb 2013 23:00:59 -0500 >> Subject: Re: [Freeswitch-users] High Availability Cluster Module for >> FreeSWITCH >> The beauty of the module is that it will not require any 3rd party >> software to operate. I am designing it to run entirely in-memory using >> multi-cast events to communicate between cluster nodes. No external >> database will be required to enable the automatic fail over and call >> recovery. >> >> If you have your customers / clients using DNS to point traffic to >> your system and you run multiple master nodes with DNS performing load >> balancing, you can also make use of the maintenance mode I have >> planned which will use a separate IP address and update your DNS >> records to remove the master which you put into maintenance mode and >> put in the maintenance IP which is assigned to the slave node which is >> being promoted as master. This allows calls to drain off the master >> while new calls come up on the slave, and then you can gracefully shut >> down the master when all calls are finished. Once you are done with >> maintenance, you simply bring it back online and it comes up as a >> slave or a master as needed (according to your configured master / >> slave ratio). The IP which it was using as a master node becomes the >> new designated maintenance IP. This is the only case where a 3rd party >> software package is required, and it is an optional feature. >> >> >> On Sat, Feb 9, 2013 at 6:49 PM, Gilad Abada >> wrote: >> > Hey Eliot >> > >> > Sounds exciting. Which DB will you be using Postgres? I will try to >> > get my company to pitch in a few dollars. >> > >> > Sent from my mobile device. >> > >> > On Feb 9, 2013, at 12:11 PM, Eliot Gable >> wrote: >> > >> >> For the past two years, I have been working on and off on a high >> >> availability cluster module for FreeSWITCH. If you would like to see >> >> HA built into FreeSWITCH directly, instead of dealing with Pacemaker + >> >> Corosync or other 3rd party systems, please check out the module, >> >> voice your support, provide feedback and suggestions, and consider >> >> supporting it. >> >> >> >> Some advantages over a 3rd party solution like Pacemaker include: >> >> >> >> - Sub-second failure detection and response (meaning it can start >> >> recovering calls less than a second after a failure occurs) >> >> - Failure detection is built into FreeSWITCH so it can detect >> >> conditions which 3rd party modules cannot detect >> >> - One shared configuration file between all nodes (no per-node IP >> >> configuration required) >> >> - No fixed number of nodes >> >> - Graceful migration of calls with a maintenance mode (requires DNS) >> >> >> >> These are just some of the benefits of building a FreeSWITCH-specific >> >> HA system. For additional details about the project, please visit: >> >> >> >> http://wiki.freeswitch.org/wiki/Mod_ha_cluster >> >> >> >> >> >> If you would like to see this project come to fruition, please >> >> consider financially supporting it through Kickstarter: >> >> >> >> >> http://www.kickstarter.com/projects/1175121367/high-availability-cluster-module-for-freeswitch >> >> >> >> >> >> >> >> Thanks in advance for anyone who contributes, either with just a >> >> shout-out, with feedback and/or suggestions, and especially to those >> >> who pledge some financial support. >> >> >> >> >> >> >> ---------- Forwarded message ---------- >> From: Yuriy Ostapchuk >> To: freeswitch-users at lists.freeswitch.org >> Cc: >> Date: Sun, 10 Feb 2013 00:10:39 -0500 >> Subject: [Freeswitch-users] USSD support for mod_gsmopen >> Hi, >> I've submitted a patch that implements USSD messaging support. >> (useful for checking and refilling account balance and other >> provider-specific usage) >> >> Here is the JIRA ticket: >> http://jira.freeswitch.org/browse/FS-5078 >> >> If this feature is considered useful, I would appreciate if it's >> integrated in git. >> >> Thanks, >> Yuriy >> >> >> ---------- Forwarded message ---------- >> From: Yehavi Bourvine >> To: FreeSWITCH Users Help >> Cc: >> Date: Sun, 10 Feb 2013 11:21:25 +0200 >> Subject: Re: [Freeswitch-users] VoIP gateway with PRI >> Hi, >> I just checked with our Audiocodes support, and you can do what you >> want. >> >> In order to route a call from one PRI to the other, you insert an entry >> in two tables: >> Tel->IP: the destination is localhost. >> IP->Tel - the source is either any or localhost. >> >> Regards, __Yehavi: >> >> >> 2013/2/7 Mimiko >> >>> Hello. >>> >>> I want to propose for purchasing a VoIP gateway to connect some PRI >>> lines with FS. Basic operation has to be routing like this: >>> >>> One line PRI will be connected to phone provider. Some calls to certain >>> numbers thru this connection must be routed to FS via IP. Other >>> destination numbers must be routed to second PRI connection for legacy >>> PBX for analog phones. So the VoIP gateway must have two PRI (T1/E1) >>> connections and two LAN connection for failover. >>> >>> After a while this VoIP gateway will be as a tehnology converter from IP >>> to PRI. So LAN connectors will be used for connection to FS in a >>> failover manner. And two PRI connectors will be used to connect to >>> legacy PBX which have two PRI inputs to handle 60 concurrent connection. >>> >>> I've spotted three models: >>> 1) Patton SN4951/4E60V2GS/EUI 4T1/E1/PRI 60 VoIP Channels G.SHDSL for >>> $5,000 >>> >>> 2) Audiocodes Mediant 1000 M1K-D2 2T1/E1 for $3,800 >>> >>> 3) Sangoma Vega 400 2 T1/E1 Digital Gateway for $4,437 >>> >>> Which of this is a better working solution with failover and extended >>> call routing plans? >>> >>> Thansk. >>> >>> -- >>> Mimiko desu. >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >> >> >> >> ---------- Forwarded message ---------- >> From: Giovanni Maruzzelli >> To: FreeSWITCH Users Help >> Cc: >> Date: Sun, 10 Feb 2013 12:36:28 +0100 >> Subject: Re: [Freeswitch-users] USSD support for mod_gsmopen >> >> On Sun, Feb 10, 2013 at 6:10 AM, Yuriy Ostapchuk wrot >> >>> Hi, >>> I've submitted a patch that implements USSD messaging support. >>> (useful for checking and refilling account balance and other >>> provider-specific usage) >>> >>> >>> Here is the JIRA ticket: >>> http://jira.freeswitch.org/browse/FS-5078 >>> >>> If this feature is considered useful, I would appreciate if it's >>> integrated in git. >>> >> >> Thanks Yuriy, I'll have a look into it asap next week. >> >> Thanks again, yes is considered useful and grateful received. >> >> -giovanni >> >> >> -- >> Sincerely, >> >> Giovanni Maruzzelli >> Cell : +39-347-2665618 >> >> >> ---------- Forwarded message ---------- >> From: Marcin Gozdalik >> To: FreeSWITCH Users Help >> Cc: >> Date: Sun, 10 Feb 2013 12:54:12 +0100 >> Subject: Re: [Freeswitch-users] High Availability Cluster Module for >> FreeSWITCH >> I wonder how do you handle network partition? Simply "knowing" which >> nodes are up in a distributed system is hard enough and there is a >> reason why solutions like pacemaker/corosync have years of development >> behind them. >> >> W dniu niedziela, 10 lutego 2013 u?ytkownik Eliot Gable napisa?: >> >>> The beauty of the module is that it will not require any 3rd party >>> software to operate. I am designing it to run entirely in-memory using >>> multi-cast events to communicate between cluster nodes. No external >>> database will be required to enable the automatic fail over and call >>> recovery. >>> >>> If you have your customers / clients using DNS to point traffic to >>> your system and you run multiple master nodes with DNS performing load >>> balancing, you can also make use of the maintenance mode I have >>> planned which will use a separate IP address and update your DNS >>> records to remove the master which you put into maintenance mode and >>> put in the maintenance IP which is assigned to the slave node which is >>> being promoted as master. This allows calls to drain off the master >>> while new calls come up on the slave, and then you can gracefully shut >>> down the master when all calls are finished. Once you are done with >>> maintenance, you simply bring it back online and it comes up as a >>> slave or a master as needed (according to your configured master / >>> slave ratio). The IP which it was using as a master node becomes the >>> new designated maintenance IP. This is the only case where a 3rd party >>> software package is required, and it is an optional feature. >>> >>> >>> On Sat, Feb 9, 2013 at 6:49 PM, Gilad Abada >>> wrote: >>> > Hey Eliot >>> > >>> > Sounds exciting. Which DB will you be using Postgres? I will try to >>> > get my company to pitch in a few dollars. >>> > >>> > Sent from my mobile device. >>> > >>> > On Feb 9, 2013, at 12:11 PM, Eliot Gable >>> wrote: >>> > >>> >> For the past two years, I have been working on and off on a high >>> >> availability cluster module for FreeSWITCH. If you would like to see >>> >> HA built into FreeSWITCH directly, instead of dealing with Pacemaker + >>> >> Corosync or other 3rd party systems, please check out the module, >>> >> voice your support, provide feedback and suggestions, and consider >>> >> supporting it. >>> >> >>> >> Some advantages over a 3rd party solution like Pacemaker include: >>> >> >>> >> - Sub-second failure detection and response (meaning it can start >>> >> recovering calls less than a second after a failure occurs) >>> >> - Failure detection is built into FreeSWITCH so it can detect >>> >> conditions which 3rd party modules cannot detect >>> >> - One shared configuration file between all nodes (no per-node IP >>> >> configuration required) >>> >> - No fixed number of nodes >>> >> - Graceful migration of calls with a maintenance mode (requires DNS) >>> >> >>> >> These are just some of the benefits of building a FreeSWITCH-specific >>> >> HA system. For additional details about the project, please visit: >>> >> >>> >> http://wiki.freeswitch.org/wiki/Mod_ha_cluster >>> >> >>> >> >>> >> If you would like to see this project come to fruition, please >>> >> consider financially supporting it through Kickstarter: >>> >> >>> >> >>> http://www.kickstarter.com/projects/1175121367/high-availability-cluster-module-for-freeswitch >>> >> >>> >> >>> >> >>> >> Thanks in advance for anyone who contributes, either with just a >>> >> shout-out, with feedback and/or suggestions, and especially to those >>> >> who pledge some financial support. >>> >> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >> >> >> -- >> Marcin Gozdalik >> >> _______________________________________________ >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130211/bcec5a15/attachment-0001.html From gaz.foreman at gmail.com Tue Feb 12 02:03:21 2013 From: gaz.foreman at gmail.com (Gary Foreman) Date: Mon, 11 Feb 2013 23:03:21 +0000 Subject: [Freeswitch-users] Terminating calls prior to answer using api Message-ID: No joy, even using the "picked_off" cause I'm still getting a tone back indicating the call failed to connect. I have managed to hack a solution together using an idea from one of the previous suggestions. I'm sending a uuid_answer followed by a uuid_kill event on a timer half a second later. Obviously this isn't ideal as it gives me channel_answer events etc and timed events are rarely a good thing. Is the SIP notify route out of the question? I know next to nothing about SIP (but I'm enjoying the learning curve!) I'm using a Polycom IP335 it that helps. On Mon, Feb 11, 2013 at 9:39 PM, < freeswitch-users-request at lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users at lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request at lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner at lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. Re: Terminating calls prior to answer using api > (Anthony Minessale) > > > ---------- Forwarded message ---------- > From: Anthony Minessale > To: FreeSWITCH Users Help > Cc: > Date: Mon, 11 Feb 2013 15:38:40 -0600 > Subject: Re: [Freeswitch-users] Terminating calls prior to answer using api > Oh, and remember to take the "SWITCH_CAUSE" out when using it with > uuid_kill > > uuid_kill picked_off > > > > On Mon, Feb 11, 2013 at 3:37 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> Try SWITCH_CAUSE_PICKED_OFF or SWITCH_CAUSE_LOSE_RACE that will inject >> headers to the phone telling it that the call was completed elsewhere which >> makes most of them skip counting it as an error. >> >> >> >> >> On Mon, Feb 11, 2013 at 2:07 PM, Steven Ayre wrote: >> >>> NOTIFY messages are a SIP feature outside of an INVITE call flow. Ending >>> the call using 200+BYE, 4xx, 5xx or 6xx are your only options I'm afraid. >>> >>> I'm writing a CTI application so terminating the call via the api prior >>>> to answer without errors is really important. >>> >>> >>> Any call that's unanswered is a failed call, but that's not necessarily >>> an error (eg user busy). >>> >>> I understand your concern that the caller might not know if their call >>> succeeded or not... Is there any possibility perhaps to answer the call, >>> send it to a playback (eg 'your request was successful') and then hangup? >>> >>> -Steve >>> >>> >>> >>> >>> On 11 February 2013 18:37, Gary Foreman wrote: >>> >>>> I was concerned that might be the case, I have managed to get different >>>> errors by send alternative hangup codes "user_busy" for example is >>>> recognised but as you say all are still errors. >>>> >>>> I'm planning on using Polycom sip phones, does anyone know if it is >>>> possible to end an unanswered call using a notify message or something? >>>> Like the talk sip notify message used to answer. >>>> >>>> I'm writing a CTI application so terminating the call via the api prior >>>> to answer without errors is really important. >>>> >>>> Thanks again >>>> >>>> >>>> On Mon, Feb 11, 2013 at 6:07 PM, < >>>> freeswitch-users-request at lists.freeswitch.org> wrote: >>>> >>>>> Send FreeSWITCH-users mailing list submissions to >>>>> freeswitch-users at lists.freeswitch.org >>>>> >>>>> To subscribe or unsubscribe via the World Wide Web, visit >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> or, via email, send a message with subject or body 'help' to >>>>> freeswitch-users-request at lists.freeswitch.org >>>>> >>>>> You can reach the person managing the list at >>>>> freeswitch-users-owner at lists.freeswitch.org >>>>> >>>>> When replying, please edit your Subject line so it is more specific >>>>> than "Re: Contents of FreeSWITCH-users digest..." >>>>> >>>>> Today's Topics: >>>>> >>>>> 1. Re: Throttle Error (Avi Marcus) >>>>> 2. Re: Terminating calls prior to answer using api (Steven Ayre) >>>>> 3. Re: Terminating calls prior to answer using api (Steven Ayre) >>>>> 4. High Availability Cluster Module for FreeSWITCH - Major News >>>>> Update (Eliot Gable) >>>>> 5. Re: High Availability Cluster Module for FreeSWITCH >>>>> (Luis Daniel Lucio Quiroz) >>>>> >>>>> >>>>> >>>>> ---------- Forwarded message ---------- >>>>> From: Avi Marcus >>>>> To: FreeSWITCH Users Help >>>>> Cc: >>>>> Date: Mon, 11 Feb 2013 19:09:33 +0200 >>>>> Subject: Re: [Freeswitch-users] Throttle Error >>>>> >>>>> As Ken's original message said, try "fsctl sps 40" or the like. >>>>> He said that otherwise only a restart, not reload xml would work. >>>>> >>>>> -Avi >>>>> (This message was painstakingly thumbed out on my mobile, so apologies >>>>> for brevity and errors.) >>>>> >>>>> On Feb 11, 2013 6:57 PM, "Archana Venugopan" >>>>> wrote: >>>>> >>>>> Hi,**** >>>>> >>>>> ** ** >>>>> >>>>> When I gave fsctl sps it shows still 10. Can you please tell me apart >>>>> from reload_xml do we have to anyother thing after changing switch.conf.xml? >>>>> **** >>>>> >>>>> ** ** >>>>> >>>>> Regards,**** >>>>> >>>>> Archana**** >>>>> >>>>> ** ** >>>>> >>>>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >>>>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Archana >>>>> Venugopan >>>>> *Sent:* 11 February 2013 16:37 >>>>> >>>>> >>>>> To: FreeSWITCH Users Help >>>>> Subject: Re: [Freeswitch-users] Throttle Error >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> Ya i did re-loaded the xml. And this is what I see in status >>>>> >>>>> >>>>> >>>>> UP 0 years, 6 days, 9 hours, 3... >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> 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 >>>>> >>>>> >>>>> >>>>> ---------- Forwarded message ---------- >>>>> From: Steven Ayre >>>>> To: FreeSWITCH Users Help >>>>> Cc: >>>>> Date: Mon, 11 Feb 2013 17:40:57 +0000 >>>>> Subject: Re: [Freeswitch-users] Terminating calls prior to answer >>>>> using api >>>>> You either answer a call with 200, redirect with 3xx, or give an error >>>>> 4xx 5xx or 6xx... basically without answering your only option is to return >>>>> an error. >>>>> >>>>> The specific error code can show a varying number of levels of 'error' >>>>> though. Some might be a perfectly valid result (eg 486 User Busy) while >>>>> other's imply a more serious error (eg 500 Server Internal Error). But >>>>> there's not really one for a successful unanswered call. >>>>> >>>>> How your phone behaves on the various errors is really up to the SIP >>>>> client though. A generic unable to connect message doesn't sound very >>>>> useful, I'd expect different error tones depending on the response code. >>>>> >>>>> -Steve >>>>> >>>>> >>>>> >>>>> On 11 February 2013 15:30, Gary Foreman wrote: >>>>> >>>>>> Hi, >>>>>> >>>>>> I'm having an issue gracefully terminating calls prior to them being >>>>>> answered. >>>>>> >>>>>> When I issue the uuid_kill command prior to the call being answered >>>>>> my sip client gives an error "unable to connect" with an error tone. >>>>>> >>>>>> Is it possible to cancel the client dial without causing an error? >>>>>> Maybe using a sip notify message or something? >>>>>> >>>>>> I can't seem to find a list of valid sip notify messages anywhere. >>>>>> >>>>>> Thanks >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> 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 >>>>>> >>>>>> >>>>> >>>>> >>>>> ---------- Forwarded message ---------- >>>>> From: Steven Ayre >>>>> To: FreeSWITCH Users Help >>>>> Cc: >>>>> Date: Mon, 11 Feb 2013 17:42:51 +0000 >>>>> Subject: Re: [Freeswitch-users] Terminating calls prior to answer >>>>> using api >>>>> The NORMAL_CLEARING ISDN clearing cause (16) has no mapping to any SIP >>>>> code for example, it's a code you normally would only see a) when the >>>>> caller hangs up at any time (regardless of answered or not) or b) when call >>>>> has been answered *then* callee hangs up. >>>>> >>>>> -Steve >>>>> >>>>> >>>>> >>>>> >>>>> On 11 February 2013 17:40, Steven Ayre wrote: >>>>> >>>>>> You either answer a call with 200, redirect with 3xx, or give an >>>>>> error 4xx 5xx or 6xx... basically without answering your only option is to >>>>>> return an error. >>>>>> >>>>>> The specific error code can show a varying number of levels of >>>>>> 'error' though. Some might be a perfectly valid result (eg 486 User Busy) >>>>>> while other's imply a more serious error (eg 500 Server Internal Error). >>>>>> But there's not really one for a successful unanswered call. >>>>>> >>>>>> How your phone behaves on the various errors is really up to the SIP >>>>>> client though. A generic unable to connect message doesn't sound very >>>>>> useful, I'd expect different error tones depending on the response code. >>>>>> >>>>>> -Steve >>>>>> >>>>>> >>>>>> >>>>>> On 11 February 2013 15:30, Gary Foreman wrote: >>>>>> >>>>>>> Hi, >>>>>>> >>>>>>> I'm having an issue gracefully terminating calls prior to them being >>>>>>> answered. >>>>>>> >>>>>>> When I issue the uuid_kill command prior to the call being answered >>>>>>> my sip client gives an error "unable to connect" with an error tone. >>>>>>> >>>>>>> Is it possible to cancel the client dial without causing an error? >>>>>>> Maybe using a sip notify message or something? >>>>>>> >>>>>>> I can't seem to find a list of valid sip notify messages anywhere. >>>>>>> >>>>>>> Thanks >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://wiki.freeswitch.org >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> 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 >>>>>>> >>>>>>> >>>>>> >>>>> >>>>> >>>>> ---------- Forwarded message ---------- >>>>> From: Eliot Gable >>>>> To: FreeSWITCH Users Help >>>>> Cc: >>>>> Date: Mon, 11 Feb 2013 12:44:55 -0500 >>>>> Subject: [Freeswitch-users] High Availability Cluster Module for >>>>> FreeSWITCH - Major News Update >>>>> After discussing this project in further detail with Tony, I have >>>>> decided to cancel the Kickstarter project. Funding will now go through >>>>> FreeSWITCH Solutions. If you would like to contribute towards making >>>>> mod_ha_cluster a reality, please send contributions to FreeSWITCH Solutions. >>>>> >>>>> Thanks! >>>>> >>>>> -- >>>>> Eliot Gable >>>>> >>>>> >>>>> >>>>> ---------- Forwarded message ---------- >>>>> From: Luis Daniel Lucio Quiroz >>>>> To: FreeSWITCH Users Help >>>>> Cc: >>>>> Date: Mon, 11 Feb 2013 13:06:16 -0500 >>>>> Subject: Re: [Freeswitch-users] High Availability Cluster Module for >>>>> FreeSWITCH >>>>> Is the latest snapshot ofyour module at >>>>> git://git.freeswitch.org/freeswitch-contrib.git ? >>>>> >>>>> 2013/2/11 Eliot Gable : >>>>> > On Mon, Feb 11, 2013 at 7:36 AM, Marcin Gozdalik >>>>> wrote: >>>>> >> >>>>> >> +1 >>>>> >> >>>>> >> I do not doubt mod_ha is necessary inside of FS and it may be >>>>> >> better/simpler than writing Pacemaker resource agent, but writing >>>>> >> yet-another-cluster-communication-engine is IMHO the wrong way to go >>>>> >> and using Corosync for communication will give a lot of value from >>>>> >> mature codebase. >>>>> >> >>>>> > >>>>> > I understand what you are saying, but what I am trying to get across >>>>> is that >>>>> > I am not writing yet-another-cluster-communication-engine. All I am >>>>> really >>>>> > doing is combining a multicast messaging API written by Tony and the >>>>> event >>>>> > API in FS to broadcast existing state information between multiple >>>>> FS nodes, >>>>> > as well as adding a tiny amount of logic on top of that to >>>>> coordinate call >>>>> > fail over and recovery. That's probably a little over-simplified, >>>>> but it >>>>> > gets the point across. The network communication code is already in >>>>> FS and >>>>> > well tested. The event system is already in FS and well tested. I >>>>> have >>>>> > already written the code to the point that it parses the >>>>> configuration files >>>>> > and starts sending heartbeats out all of the interfaces configured. >>>>> I have >>>>> > also already written a lot of the code that deals with the state >>>>> > transitions. All I am talking about doing is implementing a tiny >>>>> little >>>>> > finite state machine. It's a pretty trivial programming task. In >>>>> fact, I >>>>> > think it was covered in my first year at Carnegie Mellon University. >>>>> Of >>>>> > course, I had already figured out how to write such things in high >>>>> school, I >>>>> > just did not know what it was called at that point. My point is, >>>>> that this >>>>> > is not yet-another-cluster-communication-engine. It is a very >>>>> specific and >>>>> > small finite state machine designed solely with the goal in mind of >>>>> making >>>>> > FS have just enough information to coordinate call fail over >>>>> internally. If >>>>> > I recall correctly, a lot of people also said writing >>>>> > yet-another-VoIP-server was a waste of time, but now we have >>>>> FreeSWITCH, and >>>>> > it was obviously worth the effort. And I am not even trying to do >>>>> something >>>>> > as complex as that. If you think this is >>>>> > yet-another-cluster-communication-engine, you are missing the point. >>>>> It is >>>>> > not. It never will be. >>>>> > >>>>> > Look at Sonus, Genband, Broadsoft, Veraz, etc. All the big-name >>>>> > carrier-grade telecom providers have a built-in solution for >>>>> automatic call >>>>> > fail over. The only way FreeSWITCH will ever compete with such >>>>> solutions is >>>>> > if it also has that feature. Pacemaker and Corosync are overkill >>>>> just to get >>>>> > FS to handle single node failures and provide call recovery. It took >>>>> me a >>>>> > full 3 months of working with them every day to really understand >>>>> how to >>>>> > deploy them properly in conjunction with FreeSWITCH and Postgres to >>>>> provide >>>>> > a carrier-grade hot-standby solution which was robust enough to >>>>> handle 99% >>>>> > of the failures I could throw at it. Granted, this was back when the >>>>> > configuration still needed to be written by hand in XML and prior the >>>>> > existence of any resource agent for FreeSWITCH. But, even with those >>>>> > changes, deploying Pacemaker and Corosync is not a simple task. If >>>>> that is >>>>> > the requirement for FS to have HA, it will never truly stand a chance >>>>> > against commercial offerings. >>>>> >>>>> > >>>>> > >>>>> > >>>>> > >>>>> _________________________________________________________________________ >>>>> > Professional FreeSWITCH Consulting Services: >>>>> > consulting at freeswitch.org >>>>> > http://www.freeswitchsolutions.com >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > Official FreeSWITCH Sites >>>>> > http://www.freeswitch.org >>>>> > http://wiki.freeswitch.org >>>>> > http://www.cluecon.com >>>>> > >>>>> > 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 >>>>> > >>>>> >>>>> >>>>> >>>>> _______________________________________________ >>>>> >>>>> 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 >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _______________________________________________ > FreeSWITCH-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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130211/e3c9417f/attachment-0001.html From avi at avimarcus.net Tue Feb 12 02:11:30 2013 From: avi at avimarcus.net (Avi Marcus) Date: Tue, 12 Feb 2013 01:11:30 +0200 Subject: [Freeswitch-users] Terminating calls prior to answer using api In-Reply-To: References: Message-ID: As I said, another option is a uuid_transfer to a dialplan extension that does the answer then hangup. That way you only need one command. If you don't want to see a CDR at all (but you usually get one even if the call doesn't connect) you can set process_cdr =false. Why are you concerned about channel_answer events? Don't listen / ignore it. -Avi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130212/8c5d792b/attachment.html From schoch+freeswitch.org at xwin32.com Tue Feb 12 02:46:00 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Mon, 11 Feb 2013 15:46:00 -0800 Subject: [Freeswitch-users] FAX problems In-Reply-To: References: <20130119191943.ca09ab6d@mail.tritonwest.net> <5102AF93.2000201@coppice.org> Message-ID: It turns out I was too optimistic. It works *sometimes*. It makes a successful FAX transmission only about 20% of the time. Usually it fails. A typical failure is in http://pastebin.freeswitch.org/20570 This is HylaFax running on the Freeswitch machine, using the SpanDSP modems. I'm calling a FAX number that's an analog PSTN line. Can anybody tell me what I'm doing wrong? -- Steve On Wed, Feb 6, 2013 at 5:58 PM, Steven Schoch < schoch+freeswitch.org at xwin32.com> wrote: > On Fri, Feb 1, 2013 at 4:53 AM, shouldbe q931 wrote: > >> As a replacement for the faxback plugin. >> >> how about using hylafax+ with t38modem ? http://hylafax.sourceforge.net/ >> >> or the commercial "fork" from iFax ? >> http://www.ifax.com/products/hylafax/fax-over-ip.html >> > > I am happy to report that I have (partial) success! I used HylaFSP > (commercial HylaFAX Windows client) to send a FAX to HylaFAX running on the > FS machine, connecting to /dev/FS0. > > The FAX wasn't received 100%, because it didn't go through the T.38 > gateway, and the Caller-ID didn't get set, so that's what I'd like to fix > next. > > I already have this rule in the outgoing dialplan: > > > > > > > I set the $fax_call variable (and the $outbound_caller_id_number variable) > in the directory entry that's used by the ATA that the analog FAX machine > uses. > > What I want to do now is set the same variables on channels opened by the > /dev/FS0 modem. Where should I put these variables? > > -- > Steve > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130211/db828cb8/attachment.html From msc at freeswitch.org Tue Feb 12 02:56:51 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 11 Feb 2013 15:56:51 -0800 Subject: [Freeswitch-users] Terminating calls prior to answer using api In-Reply-To: References: Message-ID: What version of the Polycom firmware? Also, I think you have to tell the phone to support the "call answered elsewhere" feature. I don't know it off the top of my head so google around for that. We do this all the time on the CudaTel so I know it is possible. (I just am not an expert in the nitty gritty details...) -MC On Mon, Feb 11, 2013 at 3:03 PM, Gary Foreman wrote: > No joy, even using the "picked_off" cause I'm still getting a tone back > indicating the call failed to connect. > > I have managed to hack a solution together using an idea from one of the > previous suggestions. I'm sending a uuid_answer followed by a uuid_kill > event on a timer half a second later. > > Obviously this isn't ideal as it gives me channel_answer events etc and > timed events are rarely a good thing. > > Is the SIP notify route out of the question? I know next to nothing about > SIP (but I'm enjoying the learning curve!) > > I'm using a Polycom IP335 it that helps. > > > On Mon, Feb 11, 2013 at 9:39 PM, < > freeswitch-users-request at lists.freeswitch.org> wrote: > >> Send FreeSWITCH-users mailing list submissions to >> freeswitch-users at lists.freeswitch.org >> >> To subscribe or unsubscribe via the World Wide Web, visit >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> or, via email, send a message with subject or body 'help' to >> freeswitch-users-request at lists.freeswitch.org >> >> You can reach the person managing the list at >> freeswitch-users-owner at lists.freeswitch.org >> >> When replying, please edit your Subject line so it is more specific >> than "Re: Contents of FreeSWITCH-users digest..." >> >> Today's Topics: >> >> 1. Re: Terminating calls prior to answer using api >> (Anthony Minessale) >> >> >> >> ---------- Forwarded message ---------- >> From: Anthony Minessale >> To: FreeSWITCH Users Help >> Cc: >> Date: Mon, 11 Feb 2013 15:38:40 -0600 >> Subject: Re: [Freeswitch-users] Terminating calls prior to answer using >> api >> Oh, and remember to take the "SWITCH_CAUSE" out when using it with >> uuid_kill >> >> uuid_kill picked_off >> >> >> >> On Mon, Feb 11, 2013 at 3:37 PM, Anthony Minessale < >> anthony.minessale at gmail.com> wrote: >> >>> Try SWITCH_CAUSE_PICKED_OFF or SWITCH_CAUSE_LOSE_RACE that will inject >>> headers to the phone telling it that the call was completed elsewhere which >>> makes most of them skip counting it as an error. >>> >>> >>> >>> >>> On Mon, Feb 11, 2013 at 2:07 PM, Steven Ayre wrote: >>> >>>> NOTIFY messages are a SIP feature outside of an INVITE call flow. >>>> Ending the call using 200+BYE, 4xx, 5xx or 6xx are your only options I'm >>>> afraid. >>>> >>>> I'm writing a CTI application so terminating the call via the api prior >>>>> to answer without errors is really important. >>>> >>>> >>>> Any call that's unanswered is a failed call, but that's not necessarily >>>> an error (eg user busy). >>>> >>>> I understand your concern that the caller might not know if their call >>>> succeeded or not... Is there any possibility perhaps to answer the call, >>>> send it to a playback (eg 'your request was successful') and then hangup? >>>> >>>> -Steve >>>> >>>> >>>> >>>> >>>> On 11 February 2013 18:37, Gary Foreman wrote: >>>> >>>>> I was concerned that might be the case, I have managed to get >>>>> different errors by send alternative hangup codes "user_busy" for example >>>>> is recognised but as you say all are still errors. >>>>> >>>>> I'm planning on using Polycom sip phones, does anyone know if it is >>>>> possible to end an unanswered call using a notify message or something? >>>>> Like the talk sip notify message used to answer. >>>>> >>>>> I'm writing a CTI application so terminating the call via the api >>>>> prior to answer without errors is really important. >>>>> >>>>> Thanks again >>>>> >>>>> >>>>> On Mon, Feb 11, 2013 at 6:07 PM, < >>>>> freeswitch-users-request at lists.freeswitch.org> wrote: >>>>> >>>>>> Send FreeSWITCH-users mailing list submissions to >>>>>> freeswitch-users at lists.freeswitch.org >>>>>> >>>>>> To subscribe or unsubscribe via the World Wide Web, visit >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> or, via email, send a message with subject or body 'help' to >>>>>> freeswitch-users-request at lists.freeswitch.org >>>>>> >>>>>> You can reach the person managing the list at >>>>>> freeswitch-users-owner at lists.freeswitch.org >>>>>> >>>>>> When replying, please edit your Subject line so it is more specific >>>>>> than "Re: Contents of FreeSWITCH-users digest..." >>>>>> >>>>>> Today's Topics: >>>>>> >>>>>> 1. Re: Throttle Error (Avi Marcus) >>>>>> 2. Re: Terminating calls prior to answer using api (Steven Ayre) >>>>>> 3. Re: Terminating calls prior to answer using api (Steven Ayre) >>>>>> 4. High Availability Cluster Module for FreeSWITCH - Major News >>>>>> Update (Eliot Gable) >>>>>> 5. Re: High Availability Cluster Module for FreeSWITCH >>>>>> (Luis Daniel Lucio Quiroz) >>>>>> >>>>>> >>>>>> >>>>>> ---------- Forwarded message ---------- >>>>>> From: Avi Marcus >>>>>> To: FreeSWITCH Users Help >>>>>> Cc: >>>>>> Date: Mon, 11 Feb 2013 19:09:33 +0200 >>>>>> Subject: Re: [Freeswitch-users] Throttle Error >>>>>> >>>>>> As Ken's original message said, try "fsctl sps 40" or the like. >>>>>> He said that otherwise only a restart, not reload xml would work. >>>>>> >>>>>> -Avi >>>>>> (This message was painstakingly thumbed out on my mobile, so >>>>>> apologies for brevity and errors.) >>>>>> >>>>>> On Feb 11, 2013 6:57 PM, "Archana Venugopan" >>>>>> wrote: >>>>>> >>>>>> Hi,**** >>>>>> >>>>>> ** ** >>>>>> >>>>>> When I gave fsctl sps it shows still 10. Can you please tell me apart >>>>>> from reload_xml do we have to anyother thing after changing switch.conf.xml? >>>>>> **** >>>>>> >>>>>> ** ** >>>>>> >>>>>> Regards,**** >>>>>> >>>>>> Archana**** >>>>>> >>>>>> ** ** >>>>>> >>>>>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >>>>>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Archana >>>>>> Venugopan >>>>>> *Sent:* 11 February 2013 16:37 >>>>>> >>>>>> >>>>>> To: FreeSWITCH Users Help >>>>>> Subject: Re: [Freeswitch-users] Throttle Error >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Ya i did re-loaded the xml. And this is what I see in status >>>>>> >>>>>> >>>>>> >>>>>> UP 0 years, 6 days, 9 hours, 3... >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> 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 >>>>>> >>>>>> >>>>>> >>>>>> ---------- Forwarded message ---------- >>>>>> From: Steven Ayre >>>>>> To: FreeSWITCH Users Help >>>>>> Cc: >>>>>> Date: Mon, 11 Feb 2013 17:40:57 +0000 >>>>>> Subject: Re: [Freeswitch-users] Terminating calls prior to answer >>>>>> using api >>>>>> You either answer a call with 200, redirect with 3xx, or give an >>>>>> error 4xx 5xx or 6xx... basically without answering your only option is to >>>>>> return an error. >>>>>> >>>>>> The specific error code can show a varying number of levels of >>>>>> 'error' though. Some might be a perfectly valid result (eg 486 User Busy) >>>>>> while other's imply a more serious error (eg 500 Server Internal Error). >>>>>> But there's not really one for a successful unanswered call. >>>>>> >>>>>> How your phone behaves on the various errors is really up to the SIP >>>>>> client though. A generic unable to connect message doesn't sound very >>>>>> useful, I'd expect different error tones depending on the response code. >>>>>> >>>>>> -Steve >>>>>> >>>>>> >>>>>> >>>>>> On 11 February 2013 15:30, Gary Foreman wrote: >>>>>> >>>>>>> Hi, >>>>>>> >>>>>>> I'm having an issue gracefully terminating calls prior to them being >>>>>>> answered. >>>>>>> >>>>>>> When I issue the uuid_kill command prior to the call being answered >>>>>>> my sip client gives an error "unable to connect" with an error tone. >>>>>>> >>>>>>> Is it possible to cancel the client dial without causing an error? >>>>>>> Maybe using a sip notify message or something? >>>>>>> >>>>>>> I can't seem to find a list of valid sip notify messages anywhere. >>>>>>> >>>>>>> Thanks >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://wiki.freeswitch.org >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> 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 >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> ---------- Forwarded message ---------- >>>>>> From: Steven Ayre >>>>>> To: FreeSWITCH Users Help >>>>>> Cc: >>>>>> Date: Mon, 11 Feb 2013 17:42:51 +0000 >>>>>> Subject: Re: [Freeswitch-users] Terminating calls prior to answer >>>>>> using api >>>>>> The NORMAL_CLEARING ISDN clearing cause (16) has no mapping to any >>>>>> SIP code for example, it's a code you normally would only see a) when the >>>>>> caller hangs up at any time (regardless of answered or not) or b) when call >>>>>> has been answered *then* callee hangs up. >>>>>> >>>>>> -Steve >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> On 11 February 2013 17:40, Steven Ayre wrote: >>>>>> >>>>>>> You either answer a call with 200, redirect with 3xx, or give an >>>>>>> error 4xx 5xx or 6xx... basically without answering your only option is to >>>>>>> return an error. >>>>>>> >>>>>>> The specific error code can show a varying number of levels of >>>>>>> 'error' though. Some might be a perfectly valid result (eg 486 User Busy) >>>>>>> while other's imply a more serious error (eg 500 Server Internal Error). >>>>>>> But there's not really one for a successful unanswered call. >>>>>>> >>>>>>> How your phone behaves on the various errors is really up to the SIP >>>>>>> client though. A generic unable to connect message doesn't sound very >>>>>>> useful, I'd expect different error tones depending on the response code. >>>>>>> >>>>>>> -Steve >>>>>>> >>>>>>> >>>>>>> >>>>>>> On 11 February 2013 15:30, Gary Foreman wrote: >>>>>>> >>>>>>>> Hi, >>>>>>>> >>>>>>>> I'm having an issue gracefully terminating calls prior to them >>>>>>>> being answered. >>>>>>>> >>>>>>>> When I issue the uuid_kill command prior to the call being answered >>>>>>>> my sip client gives an error "unable to connect" with an error tone. >>>>>>>> >>>>>>>> Is it possible to cancel the client dial without causing an error? >>>>>>>> Maybe using a sip notify message or something? >>>>>>>> >>>>>>>> I can't seem to find a list of valid sip notify messages anywhere. >>>>>>>> >>>>>>>> Thanks >>>>>>>> >>>>>>>> >>>>>>>> _________________________________________________________________________ >>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>> consulting at freeswitch.org >>>>>>>> http://www.freeswitchsolutions.com >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> Official FreeSWITCH Sites >>>>>>>> http://www.freeswitch.org >>>>>>>> http://wiki.freeswitch.org >>>>>>>> http://www.cluecon.com >>>>>>>> >>>>>>>> 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 >>>>>>>> >>>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> ---------- Forwarded message ---------- >>>>>> From: Eliot Gable >>>>>> To: FreeSWITCH Users Help >>>>>> Cc: >>>>>> Date: Mon, 11 Feb 2013 12:44:55 -0500 >>>>>> Subject: [Freeswitch-users] High Availability Cluster Module for >>>>>> FreeSWITCH - Major News Update >>>>>> After discussing this project in further detail with Tony, I have >>>>>> decided to cancel the Kickstarter project. Funding will now go through >>>>>> FreeSWITCH Solutions. If you would like to contribute towards making >>>>>> mod_ha_cluster a reality, please send contributions to FreeSWITCH Solutions. >>>>>> >>>>>> Thanks! >>>>>> >>>>>> -- >>>>>> Eliot Gable >>>>>> >>>>>> >>>>>> >>>>>> ---------- Forwarded message ---------- >>>>>> From: Luis Daniel Lucio Quiroz >>>>>> To: FreeSWITCH Users Help >>>>>> Cc: >>>>>> Date: Mon, 11 Feb 2013 13:06:16 -0500 >>>>>> Subject: Re: [Freeswitch-users] High Availability Cluster Module for >>>>>> FreeSWITCH >>>>>> Is the latest snapshot ofyour module at >>>>>> git://git.freeswitch.org/freeswitch-contrib.git ? >>>>>> >>>>>> 2013/2/11 Eliot Gable : >>>>>> > On Mon, Feb 11, 2013 at 7:36 AM, Marcin Gozdalik >>>>>> wrote: >>>>>> >> >>>>>> >> +1 >>>>>> >> >>>>>> >> I do not doubt mod_ha is necessary inside of FS and it may be >>>>>> >> better/simpler than writing Pacemaker resource agent, but writing >>>>>> >> yet-another-cluster-communication-engine is IMHO the wrong way to >>>>>> go >>>>>> >> and using Corosync for communication will give a lot of value from >>>>>> >> mature codebase. >>>>>> >> >>>>>> > >>>>>> > I understand what you are saying, but what I am trying to get >>>>>> across is that >>>>>> > I am not writing yet-another-cluster-communication-engine. All I am >>>>>> really >>>>>> > doing is combining a multicast messaging API written by Tony and >>>>>> the event >>>>>> > API in FS to broadcast existing state information between multiple >>>>>> FS nodes, >>>>>> > as well as adding a tiny amount of logic on top of that to >>>>>> coordinate call >>>>>> > fail over and recovery. That's probably a little over-simplified, >>>>>> but it >>>>>> > gets the point across. The network communication code is already in >>>>>> FS and >>>>>> > well tested. The event system is already in FS and well tested. I >>>>>> have >>>>>> > already written the code to the point that it parses the >>>>>> configuration files >>>>>> > and starts sending heartbeats out all of the interfaces configured. >>>>>> I have >>>>>> > also already written a lot of the code that deals with the state >>>>>> > transitions. All I am talking about doing is implementing a tiny >>>>>> little >>>>>> > finite state machine. It's a pretty trivial programming task. In >>>>>> fact, I >>>>>> > think it was covered in my first year at Carnegie Mellon >>>>>> University. Of >>>>>> > course, I had already figured out how to write such things in high >>>>>> school, I >>>>>> > just did not know what it was called at that point. My point is, >>>>>> that this >>>>>> > is not yet-another-cluster-communication-engine. It is a very >>>>>> specific and >>>>>> > small finite state machine designed solely with the goal in mind of >>>>>> making >>>>>> > FS have just enough information to coordinate call fail over >>>>>> internally. If >>>>>> > I recall correctly, a lot of people also said writing >>>>>> > yet-another-VoIP-server was a waste of time, but now we have >>>>>> FreeSWITCH, and >>>>>> > it was obviously worth the effort. And I am not even trying to do >>>>>> something >>>>>> > as complex as that. If you think this is >>>>>> > yet-another-cluster-communication-engine, you are missing the >>>>>> point. It is >>>>>> > not. It never will be. >>>>>> > >>>>>> > Look at Sonus, Genband, Broadsoft, Veraz, etc. All the big-name >>>>>> > carrier-grade telecom providers have a built-in solution for >>>>>> automatic call >>>>>> > fail over. The only way FreeSWITCH will ever compete with such >>>>>> solutions is >>>>>> > if it also has that feature. Pacemaker and Corosync are overkill >>>>>> just to get >>>>>> > FS to handle single node failures and provide call recovery. It >>>>>> took me a >>>>>> > full 3 months of working with them every day to really understand >>>>>> how to >>>>>> > deploy them properly in conjunction with FreeSWITCH and Postgres to >>>>>> provide >>>>>> > a carrier-grade hot-standby solution which was robust enough to >>>>>> handle 99% >>>>>> > of the failures I could throw at it. Granted, this was back when the >>>>>> > configuration still needed to be written by hand in XML and prior >>>>>> the >>>>>> > existence of any resource agent for FreeSWITCH. But, even with those >>>>>> > changes, deploying Pacemaker and Corosync is not a simple task. If >>>>>> that is >>>>>> > the requirement for FS to have HA, it will never truly stand a >>>>>> chance >>>>>> > against commercial offerings. >>>>>> >>>>>> > >>>>>> > >>>>>> > >>>>>> > >>>>>> _________________________________________________________________________ >>>>>> > Professional FreeSWITCH Consulting Services: >>>>>> > consulting at freeswitch.org >>>>>> > http://www.freeswitchsolutions.com >>>>>> > >>>>>> > >>>>>> > >>>>>> > >>>>>> > Official FreeSWITCH Sites >>>>>> > http://www.freeswitch.org >>>>>> > http://wiki.freeswitch.org >>>>>> > http://www.cluecon.com >>>>>> > >>>>>> > 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 >>>>>> > >>>>>> >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> >>>>>> 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 >>>>>> >>>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> 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 >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>>> >>>> >>> >>> >>> -- >>> Anthony Minessale II >>> >>> FreeSWITCH http://www.freeswitch.org/ >>> ClueCon http://www.cluecon.com/ >>> Twitter: http://twitter.com/FreeSWITCH_wire >>> >>> AIM: anthm >>> MSN:anthony_minessale at hotmail.com >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>> IRC: irc.freenode.net #freeswitch >>> >>> FreeSWITCH Developer Conference >>> sip:888 at conference.freeswitch.org >>> googletalk:conf+888 at conference.freeswitch.org >>> pstn:+19193869900 >>> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> >> _______________________________________________ >> FreeSWITCH-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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130211/562b81bc/attachment-0001.html From msc at freeswitch.org Tue Feb 12 03:00:32 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 11 Feb 2013 16:00:32 -0800 Subject: [Freeswitch-users] FAX problems In-Reply-To: References: <20130119191943.ca09ab6d@mail.tritonwest.net> <5102AF93.2000201@coppice.org> Message-ID: On Mon, Feb 11, 2013 at 3:46 PM, Steven Schoch < schoch+freeswitch.org at xwin32.com> wrote: > It turns out I was too optimistic. It works *sometimes*. It makes a > successful FAX transmission only about 20% of the time. Usually it fails. > A typical failure is in http://pastebin.freeswitch.org/20570 > > This is HylaFax running on the Freeswitch machine, using the SpanDSP > modems. I'm calling a FAX number that's an analog PSTN line. Can anybody > tell me what I'm doing wrong? > Wanting to use faxes? ;) Sorry, as a fax hater I couldn't resist. Question: do you have the capture of a successful call? Can you get pcaps/siptrace of failed vs. successful and open them in Wireshark and do a comparison? I didn't see anything in the FS console log that suggested what the error might be, although if you compare that to a working fax then maybe that will yield a clue as to where to look next. Same w/ the SIP trace. -MC > -- > Steve > > On Wed, Feb 6, 2013 at 5:58 PM, Steven Schoch < > schoch+freeswitch.org at xwin32.com> wrote: > >> On Fri, Feb 1, 2013 at 4:53 AM, shouldbe q931 wrote: >> >>> As a replacement for the faxback plugin. >>> >>> how about using hylafax+ with t38modem ? http://hylafax.sourceforge.net/ >>> >>> or the commercial "fork" from iFax ? >>> http://www.ifax.com/products/hylafax/fax-over-ip.html >>> >> >> I am happy to report that I have (partial) success! I used HylaFSP >> (commercial HylaFAX Windows client) to send a FAX to HylaFAX running on the >> FS machine, connecting to /dev/FS0. >> >> The FAX wasn't received 100%, because it didn't go through the T.38 >> gateway, and the Caller-ID didn't get set, so that's what I'd like to fix >> next. >> >> I already have this rule in the outgoing dialplan: >> >> >> >> >> >> >> I set the $fax_call variable (and the $outbound_caller_id_number >> variable) in the directory entry that's used by the ATA that the analog FAX >> machine uses. >> >> What I want to do now is set the same variables on channels opened by the >> /dev/FS0 modem. Where should I put these variables? >> >> -- >> Steve >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130211/773a51b7/attachment.html From spencer at 5ninesolutions.com Tue Feb 12 04:00:18 2013 From: spencer at 5ninesolutions.com (Spencer Thomason) Date: Mon, 11 Feb 2013 17:00:18 -0800 Subject: [Freeswitch-users] Sofia Profile Parameter rtp-notimer-during-bridge Message-ID: <6396884E-F7A3-41A9-B620-7930248FF1D3@5ninesolutions.com> Hello, Looking through the source of sofia.c I noticed this parameter, rtp-notimer-during-bridge. Could someone elaborate on its usage and when one might want to enable this? Thanks, Spencer From itsusama at gmail.com Tue Feb 12 04:00:11 2013 From: itsusama at gmail.com (Usama Zaidi) Date: Tue, 12 Feb 2013 06:00:11 +0500 Subject: [Freeswitch-users] Hold/unhold events for a conference participant (Martyn Davies) Message-ID: Hi, I'm stuck on something similar myself, I've tried filtering based on conference UUID by doing something like '$con->sendRecv("filter Unique-ID $confUUID");' but that doesn't give me custom events like started/stopped talking events for that conference, it might through pause/unpause because it does through dtmf events for that conference. The other thing you could try would be to have an inbound ESL script for every user and do something like ''$con->sendRecv("filter Unique-ID $CallerUUID");' and that will give you all events for that user, the only drawback is that you'll have to run the script using a local channel and there will be n scripts for n participants. Regards. -Usama On Mon, Feb 11, 2013 at 10:00 PM, < freeswitch-users-request at lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users at lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request at lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner at lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. Re: Terminating calls prior to answer using api (Gary Foreman) > 2. Hold/unhold events for a conference participant (Martyn Davies) > 3. Re: Throttle Error (Archana Venugopan) > > > ---------- Forwarded message ---------- > From: Gary Foreman > To: freeswitch-users at lists.freeswitch.org > Cc: > Date: Mon, 11 Feb 2013 16:40:06 +0000 > Subject: Re: [Freeswitch-users] Terminating calls prior to answer using api > Thanks for your help. I might be doing something wrong but I had tried > using a cause, the command I ran was... > > bgapi uuid_kill 0292147c-7469-11e2-817e-bd1d1e5a02e8 normal_clearing > > This seemed to work as the hangup event showed "NORMAL_CLEARING" as the > hangup cause but the sip client still reported the error. > > On Mon, Feb 11, 2013 at 3:52 PM, < > freeswitch-users-request at lists.freeswitch.org> wrote: > >> Send FreeSWITCH-users mailing list submissions to >> freeswitch-users at lists.freeswitch.org >> >> To subscribe or unsubscribe via the World Wide Web, visit >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> or, via email, send a message with subject or body 'help' to >> freeswitch-users-request at lists.freeswitch.org >> >> You can reach the person managing the list at >> freeswitch-users-owner at lists.freeswitch.org >> >> When replying, please edit your Subject line so it is more specific >> than "Re: Contents of FreeSWITCH-users digest..." >> >> Today's Topics: >> >> 1. Re: Trouble autostarting freeswitch on boot (David Matthews) >> 2. Re: Running FS with "daemon" group. (Michael Jerris) >> 3. Re: High Availability Cluster Module for reeSWITCH (Eliot Gable) >> 4. Terminating calls prior to answer using api (Gary Foreman) >> 5. Re: Terminating calls prior to answer using api (Andrew Cassidy) >> 6. Throttle Error (Archana Venugopan) >> >> >> ---------- Forwarded message ---------- >> From: David Matthews >> To: FreeSWITCH Users Help >> Cc: >> Date: Mon, 11 Feb 2013 13:00:33 +0000 >> Subject: Re: [Freeswitch-users] Trouble autostarting freeswitch on boot >> >>> I might consider another distribution, but it must be one which will run >>> on the raspberrypi as that will be where it will be run in the end. I >>> did try the ArchLinux stuff on both an X86_64 system as well as the >>> raspberrypi, Debian, I mainly tried on the RaspberryPi using the >>> raspbian distribution. >>> >> >> in /etc/rc.local on raspbian >> >> su pi -c "ulimit -s 240; /usr/local/freeswitch/bin/**freeswitch -rp -nc" >> >> -- >> David Matthews >> mail at dmatthews.org >> >> >> >> >> ---------- Forwarded message ---------- >> From: Michael Jerris >> To: FreeSWITCH Users Help >> Cc: >> Date: Mon, 11 Feb 2013 10:08:30 -0500 >> Subject: Re: [Freeswitch-users] Running FS with "daemon" group. >> You can run freeswitch with whatever user/group you like as long as it >> has the necessary permissions. The more isolated typically the better, >> depending on your specific needs. >> >> >> On Feb 9, 2013, at 11:29 AM, Mimiko Mimiko wrote: >> >> Hello. >> >> On this page: http://wiki.freeswitch.org/wiki/Freeswitch_init I see that >> FS is started under "daemon" group. Is it safe to start it that way, and is >> it safe if other services will be started under the same group? >> >> About "daemon" group debian says that is better that each service is >> started in its own uid/guid. But that way services cannot share same >> directory for storing and reading common files. Especially that FS creates >> files with 750 permission. >> >> >> >> ---------- Forwarded message ---------- >> From: Eliot Gable >> To: FreeSWITCH Users Help >> Cc: >> Date: Mon, 11 Feb 2013 10:09:59 -0500 >> Subject: Re: [Freeswitch-users] High Availability Cluster Module for >> reeSWITCH >> On Mon, Feb 11, 2013 at 9:03 AM, Seven Du wrote: >> >>> >>> And, besides the FS HA, how will 3rd party softwares work with FS, if, >>> say, someone connected via ESL and FS1 crashes and the float IP bind to >>> FS2, will the ESL client fail and re-connect or it can magically connect to >>> FS2 without any addintional work? >>> >>> >> The ESL client would have to reconnect. We could make a little ESL helper >> daemon you could run on some host which would do connection management for >> you, but it's probably a better idea to just handle it in your own code. >> It's not that difficult to notice a failed connection and reconnect. >> >> >> >> ---------- Forwarded message ---------- >> From: Gary Foreman >> To: freeswitch-users at lists.freeswitch.org >> Cc: >> Date: Mon, 11 Feb 2013 15:30:36 +0000 >> Subject: [Freeswitch-users] Terminating calls prior to answer using api >> Hi, >> >> I'm having an issue gracefully terminating calls prior to them being >> answered. >> >> When I issue the uuid_kill command prior to the call being answered my >> sip client gives an error "unable to connect" with an error tone. >> >> Is it possible to cancel the client dial without causing an error? Maybe >> using a sip notify message or something? >> >> I can't seem to find a list of valid sip notify messages anywhere. >> >> Thanks >> >> >> ---------- Forwarded message ---------- >> From: Andrew Cassidy >> To: FreeSWITCH Users Help >> Cc: >> Date: Mon, 11 Feb 2013 15:43:22 +0000 >> Subject: Re: [Freeswitch-users] Terminating calls prior to answer using >> api >> probably need to use the cause parameter to uuid_kill >> >> This doesn't seem to be documented well, though. >> >> On 11 February 2013 15:30, Gary Foreman wrote: >> >>> Hi, >>> >>> I'm having an issue gracefully terminating calls prior to them being >>> answered. >>> >>> When I issue the uuid_kill command prior to the call being answered my >>> sip client gives an error "unable to connect" with an error tone. >>> >>> Is it possible to cancel the client dial without causing an error? Maybe >>> using a sip notify message or something? >>> >>> I can't seem to find a list of valid sip notify messages anywhere. >>> >>> Thanks >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> >> -- >> *Andrew Cassidy BSc (Hons) MBCS SSCA* >> Managing Director >> >> >> *T *03300 100 960 *F >> *03300 100 961 >> *E *andrew at cassidywebservices.co.uk >> *W *www.cassidywebservices.co.uk >> >> >> ---------- Forwarded message ---------- >> From: Archana Venugopan >> To: FreeSWITCH Users Help >> Cc: >> Date: Mon, 11 Feb 2013 15:51:56 +0000 >> Subject: [Freeswitch-users] Throttle Error >> >> Hi,**** >> >> ** ** >> >> I have been getting this error in freeswitch for group calls. Can anyone >> please let me know what is the issue?**** >> >> I tried increasing sessions_per_second in switch.conf.xml but even after >> that I have been facing this issue. Please help. Thanks**** >> >> ** ** >> >> 2013-02-11 15:35:12.237145 [CRIT] switch_core_session.c:1813 Throttle >> Error! 33**** >> >> 2013-02-11 15:35:12.237145 [CRIT] mod_sofia.c:4278 Error Creating Session >> **** >> >> 2013-02-11 15:35:12.317103 [CRIT] switch_time.c:869 Over Session Rate of >> 10!**** >> >> ** ** >> >> Regards,**** >> >> Archana**** >> >> ** ** >> >> _______________________________________________ >> 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 >> >> > > > ---------- Forwarded message ---------- > From: Martyn Davies > To: FreeSWITCH Users Help > Cc: > Date: Mon, 11 Feb 2013 16:56:38 +0000 > Subject: [Freeswitch-users] Hold/unhold events for a conference participant > I'm trying to process CHANNEL_CALLSTATE events (in an ESL socket > application) to extract the HELD/ACTIVE messages for conference > participants. > > One problem for me is that the name of the conference and the member-id > for that call are not available to me in the callstate messages, and > therefore it is difficult to associated calls (going HELD and ACTIVE) with > the conference. Is there any way to get the callstate messages to include > custom variables? This would fix the problem for me. > > Regards, > Martyn > > > > > ---------- Forwarded message ---------- > From: Archana Venugopan > To: FreeSWITCH Users Help > Cc: > Date: Mon, 11 Feb 2013 17:00:25 +0000 > Subject: Re: [Freeswitch-users] Throttle Error > > Thanks I got itJ**** > > ** ** > > Regards,**** > > Archana**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Archana > Venugopan > *Sent:* 11 February 2013 16:54 > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Throttle Error**** > > ** ** > > Hi,**** > > ** ** > > When I gave fsctl sps it shows still 10. Can you please tell me apart from > reload_xml do we have to anyother thing after changing switch.conf.xml?*** > * > > ** ** > > Regards,**** > > Archana**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Archana > Venugopan > *Sent:* 11 February 2013 16:37 > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Throttle Error**** > > ** ** > > Ya i did re-loaded the xml. And this is what I see in status**** > > ** ** > > UP 0 years, 6 days, 9 hours, 3 minutes, 17 seconds, 803 milliseconds, 420 > microseconds**** > > FreeSWITCH is ready**** > > 62552 session(s) since startup**** > > 16 session(s) 0/10**** > > 50 session(s) max**** > > min idle cpu 0.00/91.00**** > > ** ** > > Regards,**** > > Archana**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Ken Rice > *Sent:* 11 February 2013 16:10 > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Throttle Error**** > > ** ** > > Did you restart after increasing sessions_per_second? Check ?status? from > the cli... > > You can change it on the fly inside freeswitch with the fsctl sps command > > > On 2/11/13 9:51 AM, "Archana Venugopan" wrote:*** > * > > Hi, > > I have been getting this error in freeswitch for group calls. Can anyone > please let me know what is the issue? > I tried increasing sessions_per_second in switch.conf.xml but even after > that I have been facing this issue. Please help. Thanks > > 2013-02-11 15:35:12.237145 [CRIT] switch_core_session.c:1813 Throttle > Error! 33 > 2013-02-11 15:35:12.237145 [CRIT] mod_sofia.c:4278 Error Creating Session > 2013-02-11 15:35:12.317103 [CRIT] switch_time.c:869 Over Session Rate of > 10! > > Regards, > Archana > **** > ------------------------------ > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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**** > > > -- > Ken > *http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > *irc.freenode.net #freeswitch**** > > _______________________________________________ > 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 > > -- I'd love to change the world, but they wont gimme the source code to it -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130212/cab89d49/attachment-0001.html From dujinfang at gmail.com Tue Feb 12 04:02:24 2013 From: dujinfang at gmail.com (Seven Du) Date: Tue, 12 Feb 2013 09:02:24 +0800 Subject: [Freeswitch-users] Hold/unhold events for a conference participant In-Reply-To: References: Message-ID: There's CHANNEL_HOLD and CHANNEL_UNHOLD messages And take a look at http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_verbose_events Is there a member-id var bind on channel? I only see it on conference::maintenance messages, not sure though. On Tuesday, February 12, 2013 at 12:56 AM, Martyn Davies wrote: > I'm trying to process CHANNEL_CALLSTATE events (in an ESL socket application) to extract the HELD/ACTIVE messages for conference participants. > > One problem for me is that the name of the conference and the member-id for that call are not available to me in the callstate messages, and therefore it is difficult to associated calls (going HELD and ACTIVE) with the conference. Is there any way to get the callstate messages to include custom variables? This would fix the problem for me. > > Regards, > Martyn > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto: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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130212/c446f78f/attachment.html From msc at freeswitch.org Tue Feb 12 04:05:31 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 11 Feb 2013 17:05:31 -0800 Subject: [Freeswitch-users] FreeSWITCH Weekly News and Notes Message-ID: Hello all! News and notes are back after a brief hiatus last week. We (the FreeSWITCH team) were in Milwaukee last week and we appreciated being fed by the community. Thank you! We love this community. It seems the big news last week came courtesy of our friend Kristian Kielhofner . As reported in this blog post, Kristian ran into a rather unusual set of circumstances that resulted in Packets of Death for some Intel NICs. In addition to being Slashdotted, Kristian's research ended up being featured in a Wired.com story. Don't forget to read Kristian's update post that includes information about Intel's response to the whole situation. This week things will start to return to normal with our weekly conference call . We are going to spend a few weeks talking about various FreeSWITCH GUIs, starting with the CudaTel <>. For the past four years or so we've been building the CudaTel Communication Server and we'd like to show off some of the cool things it can do. We have two other news items. The first one has to do with mod_ha_cluster, which our very own Eliot Gable is building. Financial support for this open-source module will be done through FreeSWITCH Solutions. An interesting discussion can be found in this email thread. The other item comes from Ken Rice who has some updates on Jira and ZRTP. Check out his mailing list postfor more information. Have a great week and we'll talk to you on Wednesday. -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130211/de6befc0/attachment.html From itsusama at gmail.com Tue Feb 12 04:11:20 2013 From: itsusama at gmail.com (Usama Zaidi) Date: Tue, 12 Feb 2013 06:11:20 +0500 Subject: [Freeswitch-users] Mod_Conference ESL: Filtering Events Message-ID: Hi, I want to filter user started/stopped talking events for all users in a particular conference using inbound ESL, if I do something like $con->sendRecv("filter Unique-ID $ConfUUID"); I don't get those events so I'm forced to do something like $con->sendRecv("filter Unique-ID $participantUUID"); which basically means that I need n scripts and local channels (as conference participants to spawn that inbound ESL script) for n conference participants, and I'd really love to avoid that, I don't want to do something like $con->events("plain","CUSTOM conference::maintenance"); because, if there's 100 conferences going on I'll have 100 scripts running for each conference and they'd be getting events for all 100 conferences, so basically FS is sending 100*100 * (num events) to all the scripts. Thanks. -Usama -- I'd love to change the world, but they wont gimme the source code to it -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130212/997c91b4/attachment.html From mario_fs at mgtech.com Tue Feb 12 04:22:02 2013 From: mario_fs at mgtech.com (Mario G) Date: Mon, 11 Feb 2013 17:22:02 -0800 Subject: [Freeswitch-users] unexplained RAM usage increase In-Reply-To: References: <2E3139C8-9CCD-4042-8FCD-F791BD4482B3@mgtech.com> <06CDE1C3-916A-45B6-927E-9F59449305A1@mgtech.com> Message-ID: Jeff and Anthony, an update: I have the commit pegged and valgrind log at http://jira.freeswitch.org/browse/FS-5095. Reading Jeff's jira I don't think they are related but you never know. Sorry to take so long but git bisects resulted in many versions that would not build (I opened about 5 jiras last 2 months) and I had to hand patch to test. Family emergencies took a big toll. I put about 30 hours into this since November and tested day/night the last 6 days, have many pages of notes and logs. I hope the jira has enough to solve it. BTW, I used the valgrind line below. Mario G On Jan 29, 2013, at 6:15 PM, Anthony Minessale wrote: > If its still there on head, try valgrind. > > valgrind --tool=memcheck --log-file=vg.log --leak-check=full --leak-resolution=high --show-reachable=yes /usr/local/freeswitch/bin/freeswitch -vg > > > On Tue, Jan 29, 2013 at 8:09 PM, Mario G wrote: > I have updated and if by some miracle it's fixed I will report back. Storage increases with no calls so I assume it's registrations (internal/external?). Please understand this is probably this worse week of the year for me to work on FS (or anything else). I promise to try and narrow down the commit that started it. The memory issue started at a bad time when I could not work on FS (emergencies) and by the time I updated other errors crept in that prevented use and I had to open JIRAs for them, so testing memory was out. Thanks for letting me know about todays fix. > Mario G > > > On Jan 29, 2013, at 5:41 PM, Anthony Minessale wrote: > >> There is a commit today that found some sql leaks but I would not label them as massive. Its worth it to check and maybe run valgrind. From our perspective; Any time someone mentions nonchalantly, "Oh yeah I've had this glaring problem for months", breaks our heart =p >> I was just about to push the tag to unlock the 1.2.6 you so desperately want but now I am halting it to confirm this report. >> >> TEST TEST TEST HEAD... Today's head is tomorrow's stable.... >> >> >> >> On Tue, Jan 29, 2013 at 7:23 PM, Ken Rice wrote: >> Mario, >> >> Is it fixed in the master branch? If it is fixed there it will be rolled down to 1.2 branch before long... >> >> >> On 1/29/13 6:40 PM, "Mario G" wrote: >> >> Anthony, I tried bisect and the problem is that there are other issues (RTP error still in 1.2 stable, waiting for 1.2.6), and about 4-5 other issues in head that prevented it from working for a good amount of time. I started jiras for them and they are fixed, but doing bisect brings them back so it's had been really hard to pinpoint the memory issue. Believe me, I have put a LOT of time into trying to narrow it down before opening a JIRA and will continue to do so. Right now it's hard since there were personal emergencies the last 2 months so other pressing things had to take priority. Still, I am working on FreeSwitch keeping up-to-date with head in case other issues pop up I can open a JIRA on them. >> Mario G >> >> On Jan 29, 2013, at 3:14 PM, Anthony Minessale wrote: >> >> Things like this are sad. We depend on testing and reporting for our releases. If you wait months to bring up a problem. It will spoil the whole release. >> >> If you you feel some leak has appeared suddenly, why can't you do git bisect and find it? >> >> >> >> On Tue, Jan 29, 2013 at 12:58 PM, Mario G wrote: >> Probably not be related, but you never know: on OSX since Nov/Dec there has been a memory leak on 1.2 and head that occurs for in/out/and registrations. I had to triple memory and recycle FreeSwitch every 2-3 days since then. Will open a Jira when I can obtain detailed info and possibly run valgrind. I am also also waiting to update the main FS computer from 10.6.8 to 10.8.3 to see its effect. Hopefully in Feb. >> Mario G >> >> On Jan 29, 2013, at 9:04 AM, Anthony Minessale wrote: >> >> Can you systematically increase the current call count and see where you do see something? >> On a system with limited ram you can also consider stripping all the .so files in the mods and lib dir but you will need to put non-stripped ones in for any debugging. >> >> I don't really see a correlation on how not using a timer could trigger an sustained memory increase so that's why I'd like you to step up the number and see if you can find a number of calls that tops out because usually there always is a magic number where it will hover and go up and down a meg at a time. >> >> SIP calls are required to keep state data around for at least 30 seconds after a call ends and there are a number of pools in the code that inflate once and do not return the memory. Its usually possible to identify the high watermark on a particular box. >> >> For instance the machine we host the conference call on launches using 25 megs and hovers at about 350 megs once it has accumulated all the pool memory it needs over time. >> >> >> >> >> >> On Tue, Jan 29, 2013 at 8:33 AM, Jeff Pyle wrote: >> Version: >> FreeSWITCH version: 1.3.4-n20130122T122521Z-1~squeeze+1 (-n20130122T122521Z-1~squeeze+1) >> >> The calls are bridged, from one sofia profile to another. >> >> Unfortunately two concurrent calls doesn't seem to trigger the same behavior. >> >> >> - Jeff >> >> >> On Mon, Jan 28, 2013 at 11:21 PM, Ken Rice wrote: >> Ok the package timestamps/versions there don?t do us a lot of good, we need to know the version line from the FreeSWITCH CLI.. >> >> >> On 1/28/13 9:59 PM, "Jeff Pyle" > wrote: >> >> I just updated from repo.profhost.eu > . The most recent timestamp on the packages was 2013-01-28 03:41:21 GMT. Same behavior. At 5 minutes it was using 12.5% RAM. At 40 minutes, 60.4%. After disconnecting the calls the usage returned to 8.5%. >> >> >> >> I started toggling config items to see if I could impact this. I found one that seems to have an effect: rtp-timer-name in the sofia profile config. By changing it from 'soft' to 'none', the CPU utilization with 30 calls dropped from ~70% to ~46%, and the RAM usage is rock solid at 5.8%. >> >> That's great, but does it make any sense? >> >> Does an rtp-timer-name of 'none' pose any risks? >> >> >> >> - Jeff >> >> >> >> On Mon, Jan 28, 2013 at 8:43 PM, Jeff Pyle > wrote: >> It's on Voyage Linux, a cousin of Debian. I believe it uses glibc. >> >> >> - Jeff >> >> >> On Mon, Jan 28, 2013 at 6:45 PM, Kristian Kielhofner > wrote: >> Out of curiosity does your distro use uclibc, eglibc, or glibc? >> >> On Mon, Jan 28, 2013 at 5:41 PM, Jeff Pyle > wrote: >> > Hello, >> > >> > I'm running HEAD version from Jan 22 on an Alix board with an AMD Geode LX >> > processor (i386). I can sustain 30 concurrent calls averaging around 70% >> > CPU utilization by the freeswitch process, measured by top. Bypass media >> > and proxy media are disabled. PCMU is forced on both endpoints (no >> > transcoding). >> > >> > The problem is the RAM usage over time. The board has 256M. Idle, >> > freeswitch occupies around 4% after a fresh restart. A minute or so after >> > 30 calls are nailed up the RAM usage is about 7.2%. After 5 minutes, 13.6%. >> > After 60 minutes, near 65%. Disconnecting the calls returns the RAM usage >> > to 6-8%. >> > >> > I've not tried to troubleshoot an issue like this before. Is valgrind the >> > next step, or would something else make more sense? >> > >> > >> > Regards, >> > Jeff >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> >> >> >> -- >> Ken >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> irc.freenode.net #freeswitch >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130211/40efc2f4/attachment-0001.html From krice at freeswitch.org Tue Feb 12 06:04:49 2013 From: krice at freeswitch.org (Ken Rice) Date: Mon, 11 Feb 2013 21:04:49 -0600 Subject: [Freeswitch-users] Attention FreeSWITCH Users Message-ID: Hey Guys, Who wants to help with Jira? Whats this ?Help with Jira? mean? We are in need of a bug marshal to help out... If you think you can help there, and you can dedicate a few hours a week to review bugs, see if you can duplicate them, and poke bug reporters to update their tickets if they havent responded to a developer with additional questions this might be something for you... Contact me offlist or visit us on the IRC channel, and talk to me there about it (I?m SwK on IRC) K -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130211/b36c377e/attachment.html From anthony.minessale at gmail.com Tue Feb 12 07:16:01 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 11 Feb 2013 22:16:01 -0600 Subject: [Freeswitch-users] unexplained RAM usage increase In-Reply-To: References: <2E3139C8-9CCD-4042-8FCD-F791BD4482B3@mgtech.com> <06CDE1C3-916A-45B6-927E-9F59449305A1@mgtech.com> Message-ID: Still need some more .. see Jira for questions.... If all else fails maybe we can visit the box on ssh to examine the situation. On Mon, Feb 11, 2013 at 7:22 PM, Mario G wrote: > Jeff and Anthony, an update: I have the commit pegged and valgrind log at > http://jira.freeswitch.org/browse/FS-5095. Reading Jeff's jira I don't > think they are related but you never know. Sorry to take so long but git > bisects resulted in many versions that would not build (I opened about 5 > jiras last 2 months) and I had to hand patch to test. Family emergencies > took a big toll. I put about 30 hours into this since November and tested > day/night the last 6 days, have many pages of notes and logs. I hope the > jira has enough to solve it. BTW, I used the valgrind line below. > Mario G > > On Jan 29, 2013, at 6:15 PM, Anthony Minessale wrote: > > If its still there on head, try valgrind. > > valgrind --tool=memcheck --log-file=vg.log --leak-check=full > --leak-resolution=high --show-reachable=yes > /usr/local/freeswitch/bin/freeswitch -vg > > > On Tue, Jan 29, 2013 at 8:09 PM, Mario G wrote: > >> I have updated and if by some miracle it's fixed I will report back. >> Storage increases with no calls so I assume it's registrations >> (internal/external?). Please understand this is probably this worse week of >> the year for me to work on FS (or anything else). I promise to try and >> narrow down the commit that started it. The memory issue started at a bad >> time when I could not work on FS (emergencies) and by the time I updated >> other errors crept in that prevented use and I had to open JIRAs for them, >> so testing memory was out. Thanks for letting me know about todays fix. >> Mario G >> >> >> On Jan 29, 2013, at 5:41 PM, Anthony Minessale wrote: >> >> There is a commit today that found some sql leaks but I would not label >> them as massive. Its worth it to check and maybe run valgrind. From our >> perspective; Any time someone mentions nonchalantly, "Oh yeah I've had this >> glaring problem for months", breaks our heart =p >> I was just about to push the tag to unlock the 1.2.6 you so desperately >> want but now I am halting it to confirm this report. >> >> TEST TEST TEST HEAD... Today's head is tomorrow's stable.... >> >> >> >> On Tue, Jan 29, 2013 at 7:23 PM, Ken Rice wrote: >> >>> Mario, >>> >>> Is it fixed in the master branch? If it is fixed there it will be rolled >>> down to 1.2 branch before long... >>> >>> >>> On 1/29/13 6:40 PM, "Mario G" wrote: >>> >>> Anthony, I tried bisect and the problem is that there are other issues >>> (RTP error still in 1.2 stable, waiting for 1.2.6), and about 4-5 other >>> issues in head that prevented it from working for a good amount of time. I >>> started jiras for them and they are fixed, but doing bisect brings them >>> back so it's had been really hard to pinpoint the memory issue. Believe me, >>> I have put a LOT of time into trying to narrow it down before opening a >>> JIRA and will continue to do so. Right now it's hard since there were >>> personal emergencies the last 2 months so other pressing things had to take >>> priority. Still, I am working on FreeSwitch keeping up-to-date with head in >>> case other issues pop up I can open a JIRA on them. >>> Mario G >>> >>> On Jan 29, 2013, at 3:14 PM, Anthony Minessale wrote: >>> >>> Things like this are sad. We depend on testing and reporting for our >>> releases. If you wait months to bring up a problem. It will spoil the >>> whole release. >>> >>> If you you feel some leak has appeared suddenly, why can't you do git >>> bisect and find it? >>> >>> >>> >>> On Tue, Jan 29, 2013 at 12:58 PM, Mario G wrote: >>> >>> Probably not be related, but you never know: on OSX since Nov/Dec there >>> has been a memory leak on 1.2 and head that occurs for in/out/and >>> registrations. I had to triple memory and recycle FreeSwitch every 2-3 days >>> since then. Will open a Jira when I can obtain detailed info and possibly >>> run valgrind. I am also also waiting to update the main FS computer from >>> 10.6.8 to 10.8.3 to see its effect. Hopefully in Feb. >>> Mario G >>> >>> On Jan 29, 2013, at 9:04 AM, Anthony Minessale wrote: >>> >>> Can you systematically increase the current call count and see where you >>> do see something? >>> On a system with limited ram you can also consider stripping all the .so >>> files in the mods and lib dir but you will need to put non-stripped ones in >>> for any debugging. >>> >>> I don't really see a correlation on how not using a timer could trigger >>> an sustained memory increase so that's why I'd like you to step up the >>> number and see if you can find a number of calls that tops out because >>> usually there always is a magic number where it will hover and go up and >>> down a meg at a time. >>> >>> SIP calls are required to keep state data around for at least 30 seconds >>> after a call ends and there are a number of pools in the code that inflate >>> once and do not return the memory. Its usually possible to identify the >>> high watermark on a particular box. >>> >>> For instance the machine we host the conference call on launches using >>> 25 megs and hovers at about 350 megs once it has accumulated all the pool >>> memory it needs over time. >>> >>> >>> >>> >>> >>> On Tue, Jan 29, 2013 at 8:33 AM, Jeff Pyle >>> wrote: >>> >>> Version: >>> FreeSWITCH version: 1.3.4-n20130122T122521Z-1~squeeze+1 >>> (-n20130122T122521Z-1~squeeze+1) >>> >>> The calls are bridged, from one sofia profile to another. >>> >>> Unfortunately two concurrent calls doesn't seem to trigger the same >>> behavior. >>> >>> >>> - Jeff >>> >>> >>> On Mon, Jan 28, 2013 at 11:21 PM, Ken Rice wrote: >>> >>> Ok the package timestamps/versions there don?t do us a lot of good, we >>> need to know the version line from the FreeSWITCH CLI.. >>> >>> >>> On 1/28/13 9:59 PM, "Jeff Pyle" >> http://jpyle at fidelityvoice.com/> > wrote: >>> >>> I just updated from repo.profhost.eu < >>> http://repo.profhost.eu > . The most recent >>> timestamp on the packages was 2013-01-28 03:41:21 GMT. Same behavior. At >>> 5 minutes it was using 12.5% RAM. At 40 minutes, 60.4%. After >>> disconnecting the calls the usage returned to 8.5%. >>> >>> >>> >>> I started toggling config items to see if I could impact this. I found >>> one that seems to have an effect: rtp-timer-name in the sofia profile >>> config. By changing it from 'soft' to 'none', the CPU utilization with 30 >>> calls dropped from ~70% to ~46%, and the RAM usage is rock solid at 5.8%. >>> >>> That's great, but does it make any sense? >>> >>> Does an rtp-timer-name of 'none' pose any risks? >>> >>> >>> >>> - Jeff >>> >>> >>> >>> On Mon, Jan 28, 2013 at 8:43 PM, Jeff Pyle >> http://jpyle at fidelityvoice.com/> > wrote: >>> >>> It's on Voyage Linux, a cousin of Debian. I believe it uses glibc. >>> >>> >>> - Jeff >>> >>> >>> On Mon, Jan 28, 2013 at 6:45 PM, Kristian Kielhofner >> http://kris at kriskinc.com/> > wrote: >>> Out of curiosity does your distro use uclibc, eglibc, or glibc? >>> >>> On Mon, Jan 28, 2013 at 5:41 PM, Jeff Pyle >> http://jpyle at fidelityvoice.com/> > wrote: >>> > Hello, >>> > >>> > I'm running HEAD version from Jan 22 on an Alix board with an AMD >>> Geode LX >>> > processor (i386). I can sustain 30 concurrent calls averaging around >>> 70% >>> > CPU utilization by the freeswitch process, measured by top. Bypass >>> media >>> > and proxy media are disabled. PCMU is forced on both endpoints (no >>> > transcoding). >>> > >>> > The problem is the RAM usage over time. The board has 256M. Idle, >>> > freeswitch occupies around 4% after a fresh restart. A minute or so >>> after >>> > 30 calls are nailed up the RAM usage is about 7.2%. After 5 minutes, >>> 13.6%. >>> > After 60 minutes, near 65%. Disconnecting the calls returns the RAM >>> usage >>> > to 6-8%. >>> > >>> > I've not tried to troubleshoot an issue like this before. Is valgrind >>> the >>> > next step, or would something else make more sense? >>> > >>> > >>> > Regards, >>> > Jeff >>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >>> >>> >>> -- >>> Ken >>> *http://www.FreeSWITCH.org >>> http://www.ClueCon.com >>> http://www.OSTAG.org >>> *irc.freenode.net #freeswitch >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130211/3bce52e9/attachment-0001.html From jaybinks at gmail.com Tue Feb 12 10:01:29 2013 From: jaybinks at gmail.com (jay binks) Date: Tue, 12 Feb 2013 17:01:29 +1000 Subject: [Freeswitch-users] "Throttle Error!" and related fun Message-ID: So.... this could be related to my FS version, or who knows what else.... ( FreeSWITCH Version 1.2.0-rc2+git~20120611T214605Z~c85c8d7bbd+unclean~20120611T231109Z ) but this is what im seeing.. so one of my customers has an auto dialer... and my FS box is set with an SPS of 50. ( I know autodialer... 50SPS ... haha... ) so im looking into this traffic, and something throws me... I have B-Legs for calls that are hitting the limit. Infact I see Invites ( on the BLeg ) out to my route servers, and SIP 300's back. then FS returns the "SIP/2.0 502 Bad Gateway" on the A Leg. ( with Reason: Q.850;cause=27;text="DESTINATION_OUT_OF_ORDER" ) So a few questions... a) I was expecting the throttles, to reject on the A-Leg only and never see a B-Leg for these calls. b) surely there is a better message than SIP 502 Q850 27 ( like 603 *CALL_REJECTED 21 ) ???* how hard would it be to make this user configurable ?? ( any pointers about where, and ill look at patching this if others agree ) c) because "status" in fs_cli always shows the current CPS ... its largely useless by its self.. does anyone have a good way to get max sps in last 1, 5, 15 minutes ? how hard would this be to add ?? ( any pointers about where, and ill look at patching this if others agree ) I guess Im just hoping someone can confirm this for me ... it just dosnt sound right to have a throttled call even hit the dialplan. it could just be that I havnt had enough caffeine today, but its got me convinced enough to send an email to the list.. * * -- Sincerely Jay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130212/e75ea731/attachment.html From jaybinks at gmail.com Tue Feb 12 10:20:05 2013 From: jaybinks at gmail.com (jay binks) Date: Tue, 12 Feb 2013 17:20:05 +1000 Subject: [Freeswitch-users] "Throttle Error!" and related fun In-Reply-To: References: Message-ID: hmmm so after a little more investigation the 502 Bad Gateway may be unrelated ... and a red herring. Ive found a few 603 Declined - Reason: Q.850;cause=21;text="CALL_REJECTED" which may be what "Throttle Error" sends... however the number of log lines dosnt match the count of 603's in my Pcap... however my other questions are still valid :) On 12 February 2013 17:01, jay binks wrote: > So.... this could be related to my FS version, or who knows what else.... > ( FreeSWITCH Version > 1.2.0-rc2+git~20120611T214605Z~c85c8d7bbd+unclean~20120611T231109Z ) > but this is what im seeing.. > > so one of my customers has an auto dialer... and my FS box is set with an > SPS of 50. > ( I know autodialer... 50SPS ... haha... ) > > so im looking into this traffic, and something throws me... > I have B-Legs for calls that are hitting the limit. > Infact I see Invites ( on the BLeg ) out to my route servers, and SIP 300's > back. > then FS returns the "SIP/2.0 502 Bad Gateway" on the A Leg. > ( with Reason: Q.850;cause=27;text="DESTINATION_OUT_OF_ORDER" ) > > So a few questions... > > a) I was expecting the throttles, to reject on the A-Leg only and never see > a B-Leg for these calls. > > b) surely there is a better message than SIP 502 Q850 27 ( like 603 > CALL_REJECTED 21 ) ??? > how hard would it be to make this user configurable ?? > ( any pointers about where, and ill look at patching this if others agree ) > > c) because "status" in fs_cli always shows the current CPS ... its largely > useless by its self.. > does anyone have a good way to get max sps in last 1, 5, 15 minutes ? how > hard would this be to add ?? > ( any pointers about where, and ill look at patching this if others agree ) > > I guess Im just hoping someone can confirm this for me ... > it just dosnt sound right to have a throttled call even hit the dialplan. > > it could just be that I havnt had enough caffeine today, but its got me > convinced enough to send an email to the list.. > > > > > > > > > > > > > > > > > -- > Sincerely > > Jay -- Sincerely Jay From itsusama at gmail.com Tue Feb 12 11:50:17 2013 From: itsusama at gmail.com (Usama Zaidi) Date: Tue, 12 Feb 2013 13:50:17 +0500 Subject: [Freeswitch-users] Mod_Conference ESL: Filtering Events Message-ID: Hey All, So I found the solution to my problem and I'm updating the wiki page for event filters so it might help someone else too. Here's what I did: $con->events("plain","all"); $con->events("plain","CUSTOM conference::maintenance"); $con->sendRecv("filter Unique-ID $uuidMember1"); $con->sendRecv("filter Unique-ID $uuidMember2"); $con->sendRecv("filter Unique-ID $uuidMember3"); . . . Do this for all conference members in your script, say in a loop or something, and then you'll get all custom events, pause/unpause, started/stopped speaking, all in a single script. On Tue, Feb 12, 2013 at 6:22 AM, < freeswitch-users-request at lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users at lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request at lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner at lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. Re: Hold/unhold events for a conference participant (Seven Du) > 2. FreeSWITCH Weekly News and Notes (Michael Collins) > 3. Mod_Conference ESL: Filtering Events (Usama Zaidi) > 4. Re: unexplained RAM usage increase (Mario G) > > > ---------- Forwarded message ---------- > From: Seven Du > To: FreeSWITCH Users Help > Cc: > Date: Tue, 12 Feb 2013 09:02:24 +0800 > Subject: Re: [Freeswitch-users] Hold/unhold events for a conference > participant > There's CHANNEL_HOLD and CHANNEL_UNHOLD messages > > And take a look at > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_verbose_events > > Is there a member-id var bind on channel? I only see it on > conference::maintenance messages, not sure though. > > On Tuesday, February 12, 2013 at 12:56 AM, Martyn Davies wrote: > > I'm trying to process CHANNEL_CALLSTATE events (in an ESL socket > application) to extract the HELD/ACTIVE messages for conference > participants. > > One problem for me is that the name of the conference and the member-id > for that call are not available to me in the callstate messages, and > therefore it is difficult to associated calls (going HELD and ACTIVE) with > the conference. Is there any way to get the callstate messages to include > custom variables? This would fix the problem for me. > > Regards, > Martyn > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > > ---------- Forwarded message ---------- > From: Michael Collins > To: freeswitch-users at lists.freeswitch.org, > freeswitch-dev at lists.freeswitch.org, > freeswitch-cluecon at lists.freeswitch.org > Cc: > Date: Mon, 11 Feb 2013 17:05:31 -0800 > Subject: [Freeswitch-users] FreeSWITCH Weekly News and Notes > Hello all! > > News and notes are back after a brief hiatus last week. We (the FreeSWITCH > team) were in Milwaukee last week and we appreciated being fed by the > community. Thank you! We love this community. > > It seems the big news last week came > courtesy of our friend Kristian Kielhofner . As > reported in this blog post, > Kristian ran into a rather unusual set of circumstances that resulted in > Packets of Death for some Intel NICs. In addition to being Slashdotted, > Kristian's research ended up being featured in a Wired.com story. > Don't forget to read Kristian's update post that > includes information about Intel's response to the whole situation. > > This week things will start to return to normal with our weekly > conference call . > We are going to spend a few weeks talking about various FreeSWITCH GUIs, > starting with the CudaTel <>. For the past four > years or so we've been building the CudaTel Communication Server and we'd > like to show off some of the cool things it can do. > > We have two other news items. The first one has to do with mod_ha_cluster, > which our very own Eliot Gable is building. Financial support for this > open-source module will be done through FreeSWITCH Solutions. > An interesting discussion can be found in this email thread. > The other item comes from Ken Rice who has some updates on Jira and ZRTP. > Check out his mailing list postfor more information. > > Have a great week and we'll talk to you on Wednesday. > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > > ---------- Forwarded message ---------- > From: Usama Zaidi > To: freeswitch-users at lists.freeswitch.org > Cc: > Date: Tue, 12 Feb 2013 06:11:20 +0500 > Subject: [Freeswitch-users] Mod_Conference ESL: Filtering Events > Hi, > > I want to filter user started/stopped talking events for all users in a > particular conference using inbound ESL, if I do something like > $con->sendRecv("filter Unique-ID $ConfUUID"); I don't get those events so > I'm forced to do something like $con->sendRecv("filter Unique-ID > $participantUUID"); which basically means that I need n scripts and local > channels (as conference participants to spawn that inbound ESL script) for > n conference participants, and I'd really love to avoid that, I don't want > to do something like $con->events("plain","CUSTOM > conference::maintenance"); because, if there's 100 conferences going on > I'll have 100 scripts running for each conference and they'd be getting > events for all 100 conferences, so basically FS is sending 100*100 * (num > events) to all the scripts. > > Thanks. > > -Usama > > -- > I'd love to change the world, but they wont gimme the source code to it > > > ---------- Forwarded message ---------- > From: Mario G > To: FreeSWITCH Users Help > Cc: > Date: Mon, 11 Feb 2013 17:22:02 -0800 > Subject: Re: [Freeswitch-users] unexplained RAM usage increase > Jeff and Anthony, an update: I have the commit pegged and valgrind log at > http://jira.freeswitch.org/browse/FS-5095. Reading Jeff's jira I don't > think they are related but you never know. Sorry to take so long but git > bisects resulted in many versions that would not build (I opened about 5 > jiras last 2 months) and I had to hand patch to test. Family emergencies > took a big toll. I put about 30 hours into this since November and tested > day/night the last 6 days, have many pages of notes and logs. I hope the > jira has enough to solve it. BTW, I used the valgrind line below. > Mario G > > On Jan 29, 2013, at 6:15 PM, Anthony Minessale wrote: > > If its still there on head, try valgrind. > > valgrind --tool=memcheck --log-file=vg.log --leak-check=full > --leak-resolution=high --show-reachable=yes > /usr/local/freeswitch/bin/freeswitch -vg > > > On Tue, Jan 29, 2013 at 8:09 PM, Mario G wrote: > >> I have updated and if by some miracle it's fixed I will report back. >> Storage increases with no calls so I assume it's registrations >> (internal/external?). Please understand this is probably this worse week of >> the year for me to work on FS (or anything else). I promise to try and >> narrow down the commit that started it. The memory issue started at a bad >> time when I could not work on FS (emergencies) and by the time I updated >> other errors crept in that prevented use and I had to open JIRAs for them, >> so testing memory was out. Thanks for letting me know about todays fix. >> Mario G >> >> >> On Jan 29, 2013, at 5:41 PM, Anthony Minessale wrote: >> >> There is a commit today that found some sql leaks but I would not label >> them as massive. Its worth it to check and maybe run valgrind. From our >> perspective; Any time someone mentions nonchalantly, "Oh yeah I've had this >> glaring problem for months", breaks our heart =p >> I was just about to push the tag to unlock the 1.2.6 you so desperately >> want but now I am halting it to confirm this report. >> >> TEST TEST TEST HEAD... Today's head is tomorrow's stable.... >> >> >> >> On Tue, Jan 29, 2013 at 7:23 PM, Ken Rice wrote: >> >>> Mario, >>> >>> Is it fixed in the master branch? If it is fixed there it will be rolled >>> down to 1.2 branch before long... >>> >>> >>> On 1/29/13 6:40 PM, "Mario G" wrote: >>> >>> Anthony, I tried bisect and the problem is that there are other issues >>> (RTP error still in 1.2 stable, waiting for 1.2.6), and about 4-5 other >>> issues in head that prevented it from working for a good amount of time. I >>> started jiras for them and they are fixed, but doing bisect brings them >>> back so it's had been really hard to pinpoint the memory issue. Believe me, >>> I have put a LOT of time into trying to narrow it down before opening a >>> JIRA and will continue to do so. Right now it's hard since there were >>> personal emergencies the last 2 months so other pressing things had to take >>> priority. Still, I am working on FreeSwitch keeping up-to-date with head in >>> case other issues pop up I can open a JIRA on them. >>> Mario G >>> >>> On Jan 29, 2013, at 3:14 PM, Anthony Minessale wrote: >>> >>> Things like this are sad. We depend on testing and reporting for our >>> releases. If you wait months to bring up a problem. It will spoil the >>> whole release. >>> >>> If you you feel some leak has appeared suddenly, why can't you do git >>> bisect and find it? >>> >>> >>> >>> On Tue, Jan 29, 2013 at 12:58 PM, Mario G wrote: >>> >>> Probably not be related, but you never know: on OSX since Nov/Dec there >>> has been a memory leak on 1.2 and head that occurs for in/out/and >>> registrations. I had to triple memory and recycle FreeSwitch every 2-3 days >>> since then. Will open a Jira when I can obtain detailed info and possibly >>> run valgrind. I am also also waiting to update the main FS computer from >>> 10.6.8 to 10.8.3 to see its effect. Hopefully in Feb. >>> Mario G >>> >>> On Jan 29, 2013, at 9:04 AM, Anthony Minessale wrote: >>> >>> Can you systematically increase the current call count and see where you >>> do see something? >>> On a system with limited ram you can also consider stripping all the .so >>> files in the mods and lib dir but you will need to put non-stripped ones in >>> for any debugging. >>> >>> I don't really see a correlation on how not using a timer could trigger >>> an sustained memory increase so that's why I'd like you to step up the >>> number and see if you can find a number of calls that tops out because >>> usually there always is a magic number where it will hover and go up and >>> down a meg at a time. >>> >>> SIP calls are required to keep state data around for at least 30 seconds >>> after a call ends and there are a number of pools in the code that inflate >>> once and do not return the memory. Its usually possible to identify the >>> high watermark on a particular box. >>> >>> For instance the machine we host the conference call on launches using >>> 25 megs and hovers at about 350 megs once it has accumulated all the pool >>> memory it needs over time. >>> >>> >>> >>> >>> >>> On Tue, Jan 29, 2013 at 8:33 AM, Jeff Pyle >>> wrote: >>> >>> Version: >>> FreeSWITCH version: 1.3.4-n20130122T122521Z-1~squeeze+1 >>> (-n20130122T122521Z-1~squeeze+1) >>> >>> The calls are bridged, from one sofia profile to another. >>> >>> Unfortunately two concurrent calls doesn't seem to trigger the same >>> behavior. >>> >>> >>> - Jeff >>> >>> >>> On Mon, Jan 28, 2013 at 11:21 PM, Ken Rice wrote: >>> >>> Ok the package timestamps/versions there don?t do us a lot of good, we >>> need to know the version line from the FreeSWITCH CLI.. >>> >>> >>> On 1/28/13 9:59 PM, "Jeff Pyle" >> http://jpyle at fidelityvoice.com/> > wrote: >>> >>> I just updated from repo.profhost.eu < >>> http://repo.profhost.eu > . The most recent >>> timestamp on the packages was 2013-01-28 03:41:21 GMT. Same behavior. At >>> 5 minutes it was using 12.5% RAM. At 40 minutes, 60.4%. After >>> disconnecting the calls the usage returned to 8.5%. >>> >>> >>> >>> I started toggling config items to see if I could impact this. I found >>> one that seems to have an effect: rtp-timer-name in the sofia profile >>> config. By changing it from 'soft' to 'none', the CPU utilization with 30 >>> calls dropped from ~70% to ~46%, and the RAM usage is rock solid at 5.8%. >>> >>> That's great, but does it make any sense? >>> >>> Does an rtp-timer-name of 'none' pose any risks? >>> >>> >>> >>> - Jeff >>> >>> >>> >>> On Mon, Jan 28, 2013 at 8:43 PM, Jeff Pyle >> http://jpyle at fidelityvoice.com/> > wrote: >>> >>> It's on Voyage Linux, a cousin of Debian. I believe it uses glibc. >>> >>> >>> - Jeff >>> >>> >>> On Mon, Jan 28, 2013 at 6:45 PM, Kristian Kielhofner >> http://kris at kriskinc.com/> > wrote: >>> Out of curiosity does your distro use uclibc, eglibc, or glibc? >>> >>> On Mon, Jan 28, 2013 at 5:41 PM, Jeff Pyle >> http://jpyle at fidelityvoice.com/> > wrote: >>> > Hello, >>> > >>> > I'm running HEAD version from Jan 22 on an Alix board with an AMD >>> Geode LX >>> > processor (i386). I can sustain 30 concurrent calls averaging around >>> 70% >>> > CPU utilization by the freeswitch process, measured by top. Bypass >>> media >>> > and proxy media are disabled. PCMU is forced on both endpoints (no >>> > transcoding). >>> > >>> > The problem is the RAM usage over time. The board has 256M. Idle, >>> > freeswitch occupies around 4% after a fresh restart. A minute or so >>> after >>> > 30 calls are nailed up the RAM usage is about 7.2%. After 5 minutes, >>> 13.6%. >>> > After 60 minutes, near 65%. Disconnecting the calls returns the RAM >>> usage >>> > to 6-8%. >>> > >>> > I've not tried to troubleshoot an issue like this before. Is valgrind >>> the >>> > next step, or would something else make more sense? >>> > >>> > >>> > Regards, >>> > Jeff >>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >>> >>> >>> -- >>> Ken >>> *http://www.FreeSWITCH.org >>> http://www.ClueCon.com >>> http://www.OSTAG.org >>> *irc.freenode.net #freeswitch >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > _______________________________________________ > 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 > > -- I'd love to change the world, but they wont gimme the source code to it -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130212/6eebaf6a/attachment-0001.html From gmaruzz at gmail.com Tue Feb 12 14:25:12 2013 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Tue, 12 Feb 2013 12:25:12 +0100 Subject: [Freeswitch-users] ESL and DEBUG messages In-Reply-To: References: Message-ID: On Sat, Feb 9, 2013 at 6:15 PM, Anton Vojlenko wrote: > Hello! > Giovanni, can you recommend me how to check skype client availability? I > need this because skypopen module sometimes lost connection with skype > client. > > use PING skype api command. It supposed to answer PONG. In debug mode: "skypopen skype103 PING" > > 2013/2/5 Anton Vojlenko > >> Unfortunately* *$con->log(7) has incorrect syntax: >> # perl /home/checker.pl >> Use of inherited AUTOLOAD for non-method ESL::log() is deprecated at >> /home/checker.pl line 5. >> Can't locate auto/ESL/log.al in @INC (@INC contains: >> /usr/local/lib/perl5 /usr/local/share/perl5 /usr/lib/perl5/vendor_perl >> /usr/share/perl5/vendor_perl /usrlib/perl5 /usr/share/perl5 .) at /home/ >> checker.pl line 5 >> >> Who can help me? >> >> >> 2013/2/4 Steven Ayre >> >>> I believe so. They're probably received as the LOG event, which is >>> unfortunately undocumented >>> http://wiki.freeswitch.org/wiki/Event_list#LOG >>> >>> -Steve >>> >>> >>> >>> On 4 February 2013 14:58, Anton Vojlenko wrote: >>> >>>> Thank you, Steve. >>>> >>>> >>>> #!/usr/bin/perl >>>> use strict; >>>> use warnings; >>>> require ESL; >>>> #ESL::eslSetLogLevel(7); >>>> >>>> my $host = "127.0.0.1"; >>>> my $port = "8021"; >>>> my $pass = "ClueCon"; >>>> my $con = new ESL::ESLconnection($host, $port, $pass); >>>> >>>> *$con->log(7);* >>>> >>>> my $status = $con->api("skypopen sk1 GET USERSTATUS")->getBody(); >>>> print "$status"; >>>> $con->disconnect(); >>>> >>>> Am i right? >>>> >>>> >>>> 2013/2/4 Giovanni Maruzzelli >>>> >>>>> Thanks Steve! >>>>> >>>>> >>>>> On Mon, Feb 4, 2013 at 3:32 PM, Steven Ayre wrote: >>>>> >>>>>> 'fsctl loglevel 9' controls the core logging, all logs are routed >>>>>> through here and dropped if they're below the logging level set there. On >>>>>> top of that modules that handle log messages separately filter log levels. >>>>>> >>>>>> 'console loglevel 9' controls mod_console which is only for when >>>>>> freeswitch is started in the foreground. >>>>>> >>>>>> You don't want to use either of the above. >>>>>> >>>>>> Mod_event_socket controls logging via ESL connections. You need to >>>>>> issue the 'log' ESL command to set the log level on a per-connection basis >>>>>> (this is what /log does in fs_cli). >>>>>> >>>>>> See http://wiki.freeswitch.org/wiki/Mod_event_socket#log >>>>>> >>>>>> Note log is an ESL protocol command, not an api/app. >>>>>> >>>>>> -Steve >>>>>> >>>>>> >>>>>> >>>>>> On 4 February 2013 13:50, Anton Vojlenko wrote: >>>>>> >>>>>>> ****I >>>>>>> can't solve this problem alone. >>>>>>> In fs_cli i can see DEBUG messages: >>>>>>> freeswitch at internal> skypopen sk1 GET USERSTATUS >>>>>>> Using interface: globals.SKYPOPEN_INTERFACES[1].name=|||sk1||| >>>>>>> >>>>>>> 2013-02-04 15:31:44.565590 [DEBUG] skypopen_protocol.c:1732 [|] >>>>>>> [DEBUG_SKYPE 1732 ][sk1 ][IDLE,IDLE] SENDING: |||GET >>>>>>> USERSTATUS|||| >>>>>>> 2013-02-04 15:31:44.565590 [DEBUG] skypopen_protocol.c:207 [|] >>>>>>> [DEBUG_SKYPE 207 ][sk1 ][IDLE,IDLE] READING: |||USERSTATUS >>>>>>> ONLINE||| >>>>>>> >>>>>>> But when i execute perl script i can't see them: >>>>>>> #perl /home/checker.pl >>>>>>> Using interface: globals.SKYPOPEN_INTERFACES[1].name=|||sk1||| >>>>>>> >>>>>>> #cat /home/checker.pl >>>>>>> >>>>>>> #!/usr/bin/perl >>>>>>> use strict; >>>>>>> use warnings; >>>>>>> require ESL; >>>>>>> #ESL::eslSetLogLevel(7); >>>>>>> >>>>>>> my $host = "127.0.0.1"; >>>>>>> my $port = "8021"; >>>>>>> my $pass = "ClueCon"; >>>>>>> >>>>>>> my $con = new ESL::ESLconnection($host, $port, $pass); >>>>>>> >>>>>>> $con->api("console loglevel 9"); >>>>>>> $con->api("fsctl loglevel 9"); >>>>>>> >>>>>>> my $status = $con->api("skypopen sk1 GET USERSTATUS")->getBody(); >>>>>>> print "$status"; >>>>>>> $con->disconnect() >>>>>>> >>>>>>> >>>>>>> >>>>>>> 2013/1/23 Anton Vojlenko >>>>>>> >>>>>>>> Giovanni, I can see Skype API answers only when i using fs_cli. >>>>>>>> With perl script I can't see them. >>>>>>>> >>>>>>>> >>>>>>>> 2013/1/22 Giovanni Maruzzelli >>>>>>>> >>>>>>>>> On Tue, Jan 22, 2013 at 12:51 PM, Anton Vojlenko < >>>>>>>>> stargray at bigmir.net> wrote: >>>>>>>>> > Hello, >>>>>>>>> > >>>>>>>>> > I want to check user status from the Skype client instance, but >>>>>>>>> I can't see >>>>>>>>> > Skype API answers from it. How to fix this? >>>>>>>>> >>>>>>>>> Have you tried to give: >>>>>>>>> >>>>>>>>> "fsctl loglevel 9" in addition to "console loglevel 9" ? >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> > Script example: >>>>>>>>> > >>>>>>>>> > #!/usr/bin/perl >>>>>>>>> > use strict; >>>>>>>>> > use warnings; >>>>>>>>> > require ESL; >>>>>>>>> > >>>>>>>>> > my $host = "localhost"; >>>>>>>>> > my $port = "8021"; >>>>>>>>> > my $pass = "somepass"; >>>>>>>>> > my $con = new ESL::ESLconnection($host, $port, $pass); >>>>>>>>> > >>>>>>>>> > $con->api("console loglevel 9"); >>>>>>>>> > my $status = $con->api("skypopen sk1 GET USERSTATUS")->getBody(); >>>>>>>>> > print $status; >>>>>>>>> > $con->disconnect(); >>>>>>>>> > >>>>>>>>> >>>>>>>>> -- >>>>>>>>> Sincerely, >>>>>>>>> >>>>>>>>> Giovanni Maruzzelli >>>>>>>>> Cell : +39-347-2665618 >>>>>>>>> >>>>>>>>> >>>>>>>>> _________________________________________________________________________ >>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>> consulting at freeswitch.org >>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> Official FreeSWITCH Sites >>>>>>>>> http://www.freeswitch.org >>>>>>>>> http://wiki.freeswitch.org >>>>>>>>> http://www.cluecon.com >>>>>>>>> >>>>>>>>> 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 >>>>>>>>> >>>>>>>> >>>>>>>> >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://wiki.freeswitch.org >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> 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 >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> 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 >>>>>> >>>>>> >>>>> >>>>> >>>>> -- >>>>> Sincerely, >>>>> >>>>> Giovanni Maruzzelli >>>>> Cell : +39-347-2665618 >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> 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 >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130212/9f021728/attachment-0001.html From avi at avimarcus.net Tue Feb 12 16:21:28 2013 From: avi at avimarcus.net (Avi Marcus) Date: Tue, 12 Feb 2013 15:21:28 +0200 Subject: [Freeswitch-users] Passing the original caller-id after a call is transferred In-Reply-To: References: Message-ID: An attended / supervised transfer should always show 1002 -- he's initiating a call to 1003, e.g. the secretary calls him and asks if he should transfer the call. Are you sure you'd really want to show the caller ID of 1001 in this case? If you are doing a blind transfer, however, then that's right, you would want the CID of 1001 to show. Are you sure it's not? Can you provide an FS log (and/or a PCAP) of both attended and unattended transfer to show this issue? You can paste to http://pastebin.freeswitch.org -Avi Marcus BestFone On Sun, Feb 10, 2013 at 3:04 AM, Ravi K wrote: > > I am using Freeswitch with Cisco phones(mode 303) and have the following > setup: > > User 1001 is calling user 1002, user 1002 does a blind/supervised transfer > to 1003, but user 1003 see the caller-id of 1002. I am trying to get the > caller-id of 1001 on 1003. > > I have gone through the earlier posts related to this topic and tried the > following suggestions: > > 1. > > I configured this before bridging the call, but did not work. > > 2. > > and set data="effective_caller_id_number=${caller_id_number}" > but this did not work either > > 3. I see a message from Anthony: "use set_profile_var on the inbound leg > to set caller_id_name". What variable do I assign to caller_id_name. > > I am new to FS and most surely asking a dumb question. So I greatly > appreciate if any helpful souls can point me in the right direction. I will > update the wiki for sure. > > Thanks > Ravi > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130212/3f5cabf8/attachment.html From egable+freeswitch at gmail.com Tue Feb 12 16:42:36 2013 From: egable+freeswitch at gmail.com (Eliot Gable) Date: Tue, 12 Feb 2013 08:42:36 -0500 Subject: [Freeswitch-users] High Availability Cluster Module for FreeSWITCH In-Reply-To: References: <-2120316468757263517@unknownmsgid> Message-ID: On Mon, Feb 11, 2013 at 1:06 PM, Luis Daniel Lucio Quiroz < luis.daniel.lucio at gmail.com> wrote: > Is the latest snapshot ofyour module at > git://git.freeswitch.org/freeswitch-contrib.git ? > > For those who are interested, yes, the latest snapshot is available there. What is there right now is rather old code. I have newer stuff amounting to about 2k lines of changes which has not yet been checked in. I am in the middle of a rather large rearrangement of some things. When I finish with that rearrangement and have it compiling again, I will check in the latest code. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130212/86e6a4ab/attachment.html From egable+freeswitch at gmail.com Tue Feb 12 16:49:13 2013 From: egable+freeswitch at gmail.com (Eliot Gable) Date: Tue, 12 Feb 2013 08:49:13 -0500 Subject: [Freeswitch-users] FreeSWITCH Message Bus / Shared Key Value Store Message-ID: Tony and Mike and I had a discussion last night about FreeSWITCH with regards to implementing some form of core message bus or shared key-value store. We discussed a few different options, but did not really settle on anything. If you are writing modules or using FreeSWITCH in a multi-node setting, please share what features / functionality you would like to see implemented in this regard, how you would use it, and why you want to see the specific mechanism of your choice rather than some alternative. Also, please consider and mention whether "cluster awareness" is something that factors into your use case. By this, I mean having each FS node have some idea about the state / status of each other node in terms of taking calls vs acting as a standby or slave node, etc. -- Eliot Gable -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130212/b67d06ef/attachment.html From a.venugopan at mundio.com Tue Feb 12 17:15:47 2013 From: a.venugopan at mundio.com (Archana Venugopan) Date: Tue, 12 Feb 2013 14:15:47 +0000 Subject: [Freeswitch-users] Calling db in lua script Message-ID: <592A9CF93E12394E8472A6CC66E66BF2358C31@Mail-Kilo.squay.com> Hi, I tried executing a simple DB query in lua script in freeswitch. dbh:execute{"Select * from dir_users"} But I got this error 2013-02-12 14:09:56.247165 [ERR] mod_lua.cpp:198 /usr/local/freeswitch/scripts/directory.lua:184: attempt to call method 'execute' (a nil value) stack traceback: /usr/local/freeswitch/scripts/directory.lua:184: in main chunk 2013-02-12 14:09:56.247165 [ERR] mod_lua.cpp:264 LUA script parse/execute error! Can anyone please point what is the mistake I have done and why execute method is not being recognised? Thanks. Regards, Archana -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130212/e2bd1461/attachment.html From gozdal at gmail.com Tue Feb 12 17:21:23 2013 From: gozdal at gmail.com (Marcin Gozdalik) Date: Tue, 12 Feb 2013 15:21:23 +0100 Subject: [Freeswitch-users] High Availability Cluster Module for FreeSWITCH Message-ID: 2013/2/11 Eliot Gable : > On Mon, Feb 11, 2013 at 7:36 AM, Marcin Gozdalik wrote: >> >> +1 >> >> I do not doubt mod_ha is necessary inside of FS and it may be >> better/simpler than writing Pacemaker resource agent, but writing >> yet-another-cluster-communication-engine is IMHO the wrong way to go >> and using Corosync for communication will give a lot of value from >> mature codebase. >> > > I understand what you are saying, but what I am trying to get across is that > I am not writing yet-another-cluster-communication-engine. All I am really > doing is combining a multicast messaging API written by Tony and the event > API in FS to broadcast existing state information between multiple FS nodes, > as well as adding a tiny amount of logic on top of that to coordinate call > fail over and recovery. That's probably a little over-simplified, but it > gets the point across. The network communication code is already in FS and > well tested. The event system is already in FS and well tested. I also think I understand what you are saying. It means we have trouble putting thought into writing ;) >From what I understand what you are trying to achieve is that every node in FS "cluster" knows what are the nodes and whether they are down or up. What I am saying is that this simple task is fundamentally hard. Sending and receiving multicast is easy, but keeping distributed state consistent between nodes in cluster is hard (like in really hard, harder than writing VoIP softswitch all over again), especially in case of Byzantine failures (i.e. nodes lying that they are down when they are up or other way round). I am no big expert in the area but seen at least 2 cases (MMM - http://www.xaprb.com/blog/2011/05/04/whats-wrong-with-mmm/ and Chubby in Google - http://www.read.seas.harvard.edu/~kohler/class/08w-dsi/chandra07paxos.pdf) where people were trying to write (MMM) or use (Chubby) some kind of distributed code and failed. That's why whenever I see anything related to distributed state I say that it's way beyond my understanding and best is to use something that works. I have > already written the code to the point that it parses the configuration files > and starts sending heartbeats out all of the interfaces configured. I have > also already written a lot of the code that deals with the state > transitions. All I am talking about doing is implementing a tiny little > finite state machine. It's a pretty trivial programming task. In fact, I > think it was covered in my first year at Carnegie Mellon University. Of > course, I had already figured out how to write such things in high school, I > just did not know what it was called at that point. My point is, that this > is not yet-another-cluster-communication-engine. It is a very specific and > small finite state machine designed solely with the goal in mind of making > FS have just enough information to coordinate call fail over internally. If > I recall correctly, a lot of people also said writing > yet-another-VoIP-server was a waste of time, but now we have FreeSWITCH, and > it was obviously worth the effort. And I am not even trying to do something > as complex as that. If you think this is > yet-another-cluster-communication-engine, you are missing the point. It is > not. It never will be. See above - if it will never be and you are trying to achieve distributed, consistent state between nodes, IMHO you are going to get it wrong. Frankly I lack the knowledge and time to check if Corosync API is perfect for this task. As Anthony suggested elsewhere maybe it is possible to abstract the communication/keeping distributed state part so that it would be easy to provide Corosync or other (OpenReplica? Zookeeper?) implementations. > Look at Sonus, Genband, Broadsoft, Veraz, etc. All the big-name > carrier-grade telecom providers have a built-in solution for automatic call > fail over. The only way FreeSWITCH will ever compete with such solutions is > if it also has that feature. Pacemaker and Corosync are overkill just to get > FS to handle single node failures and provide call recovery. It took me a > full 3 months of working with them every day to really understand how to > deploy them properly in conjunction with FreeSWITCH and Postgres to provide > a carrier-grade hot-standby solution which was robust enough to handle 99% > of the failures I could throw at it. Granted, this was back when the > configuration still needed to be written by hand in XML and prior the > existence of any resource agent for FreeSWITCH. But, even with those > changes, deploying Pacemaker and Corosync is not a simple task. If that is > the requirement for FS to have HA, it will never truly stand a chance > against commercial offerings. I believe that Clusters from Scratch http://clusterlabs.org/doc/en-US/Pacemaker/1.1/html/Clusters_from_Scratch/) allowed me to setup a working Pacemaker/Corosync installation on Debian in less than a day. It is a fair point that FS to compete with big names (at least in marketing buzz-feature checklist) has to have HA. Trouble is HA is always hard. Making it simple and work is the ultimate goal but I'd rather shoot at "work" first and later for "simple". Comparing FS to commercial offerings is hard as the way FS is deployed is usually different than commercial competitors. If you buy Broadsoft you don't get a Debian package or sources to compile - you get a bunch of highly-paid consultants that install and configure everything for you. If HA in FS would work that way it would maybe be even better for building business around FS ;) Jokes aside - if somebody wants to configure FS in HA evironment I don't think it is much of an obstacle for him to configure Corosync as already he has to have some kind of DB failover as well, redundant switches, power, etc. If some commercial vendor is able to manufacture an out-of-the box solutions based on FS with a "HA" checkbox somewhere in Web configuration - well, congratulations to him and he I hope he would be deservedly reaping profits from his product. -- Marcin Gozdalik From fdelawarde at wirelessmundi.com Tue Feb 12 17:30:44 2013 From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde) Date: Tue, 12 Feb 2013 15:30:44 +0100 Subject: [Freeswitch-users] Calling db in lua script In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF2358C31@Mail-Kilo.squay.com> References: <592A9CF93E12394E8472A6CC66E66BF2358C31@Mail-Kilo.squay.com> Message-ID: <1360679444.22409.748.camel@luna.madrid.commsmundi.com> Maybe because dbh has no 'execute' method? Try with: dbh:query(sql) Regards, Fran?ois. On Tue, 2013-02-12 at 14:15 +0000, Archana Venugopan wrote: > Hi, > > > > I tried executing a simple DB query in lua script in freeswitch. > > > > dbh:execute{"Select * from dir_users"} > > > > But I got this error > > 2013-02-12 14:09:56.247165 [ERR] > mod_lua.cpp:198 /usr/local/freeswitch/scripts/directory.lua:184: > attempt to call method 'execute' (a nil value) > > stack traceback: > > /usr/local/freeswitch/scripts/directory.lua:184: in main chunk > > 2013-02-12 14:09:56.247165 [ERR] mod_lua.cpp:264 LUA script > parse/execute error! > > > > Can anyone please point what is the mistake I have done and why > execute method is not being recognised? Thanks. > > > > Regards, > > Archana > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 a.venugopan at mundio.com Tue Feb 12 17:35:30 2013 From: a.venugopan at mundio.com (Archana Venugopan) Date: Tue, 12 Feb 2013 14:35:30 +0000 Subject: [Freeswitch-users] Calling db in lua script In-Reply-To: <1360679444.22409.748.camel@luna.madrid.commsmundi.com> References: <592A9CF93E12394E8472A6CC66E66BF2358C31@Mail-Kilo.squay.com> <1360679444.22409.748.camel@luna.madrid.commsmundi.com> Message-ID: <592A9CF93E12394E8472A6CC66E66BF2358C7C@Mail-Kilo.squay.com> Dbh:query works. But I want execute a Stored procedure. In that case what can I use? Please suggest. Many thanks. Regards, Archana -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Fran?ois Delawarde Sent: 12 February 2013 14:31 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Calling db in lua script Maybe because dbh has no 'execute' method? Try with: dbh:query(sql) Regards, Fran?ois. On Tue, 2013-02-12 at 14:15 +0000, Archana Venugopan wrote: > Hi, > > > > I tried executing a simple DB query in lua script in freeswitch. > > > > dbh:execute{"Select * from dir_users"} > > > > But I got this error > > 2013-02-12 14:09:56.247165 [ERR] > mod_lua.cpp:198 /usr/local/freeswitch/scripts/directory.lua:184: > attempt to call method 'execute' (a nil value) > > stack traceback: > > /usr/local/freeswitch/scripts/directory.lua:184: in main chunk > > 2013-02-12 14:09:56.247165 [ERR] mod_lua.cpp:264 LUA script > parse/execute error! > > > > Can anyone please point what is the mistake I have done and why > execute method is not being recognised? Thanks. > > > > Regards, > > Archana > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 fdelawarde at wirelessmundi.com Tue Feb 12 18:45:58 2013 From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde) Date: Tue, 12 Feb 2013 16:45:58 +0100 Subject: [Freeswitch-users] Calling db in lua script In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF2358C7C@Mail-Kilo.squay.com> References: <592A9CF93E12394E8472A6CC66E66BF2358C31@Mail-Kilo.squay.com> <1360679444.22409.748.camel@luna.madrid.commsmundi.com> <592A9CF93E12394E8472A6CC66E66BF2358C7C@Mail-Kilo.squay.com> Message-ID: <1360683958.22409.752.camel@luna.madrid.commsmundi.com> Try that: sql=" {call procedure_name()}" dbh:query(sql, mycallback) (don't forget to define your callback function, mycallback in this case) Regards, Fran?ois. On Tue, 2013-02-12 at 14:35 +0000, Archana Venugopan wrote: > Dbh:query works. But I want execute a Stored procedure. In that case what can I use? Please suggest. > Many thanks. > > Regards, > Archana > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Fran?ois Delawarde > Sent: 12 February 2013 14:31 > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Calling db in lua script > > Maybe because dbh has no 'execute' method? > > Try with: > dbh:query(sql) > > Regards, > Fran?ois. > > On Tue, 2013-02-12 at 14:15 +0000, Archana Venugopan wrote: > > Hi, > > > > > > > > I tried executing a simple DB query in lua script in freeswitch. > > > > > > > > dbh:execute{"Select * from dir_users"} > > > > > > > > But I got this error > > > > 2013-02-12 14:09:56.247165 [ERR] > > mod_lua.cpp:198 /usr/local/freeswitch/scripts/directory.lua:184: > > attempt to call method 'execute' (a nil value) > > > > stack traceback: > > > > /usr/local/freeswitch/scripts/directory.lua:184: in main chunk > > > > 2013-02-12 14:09:56.247165 [ERR] mod_lua.cpp:264 LUA script > > parse/execute error! > > > > > > > > Can anyone please point what is the mistake I have done and why > > execute method is not being recognised? Thanks. > > > > > > > > Regards, > > > > Archana > > > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 martyn at magiccow.co.uk Tue Feb 12 18:47:20 2013 From: martyn at magiccow.co.uk (Martyn Davies) Date: Tue, 12 Feb 2013 15:47:20 +0000 Subject: [Freeswitch-users] Hold/unhold events for a conference participant In-Reply-To: References: Message-ID: Thanks for your suggestion. CHANNEL_HOLD/UNHOLD don't appear to produce any messages. On 12 February 2013 01:02, Seven Du wrote: > There's CHANNEL_HOLD and CHANNEL_UNHOLD messages > > And take a look at > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_verbose_events > > Is there a member-id var bind on channel? I only see it on > conference::maintenance messages, not sure though. > > On Tuesday, February 12, 2013 at 12:56 AM, Martyn Davies wrote: > > I'm trying to process CHANNEL_CALLSTATE events (in an ESL socket > application) to extract the HELD/ACTIVE messages for conference > participants. > > One problem for me is that the name of the conference and the member-id > for that call are not available to me in the callstate messages, and > therefore it is difficult to associated calls (going HELD and ACTIVE) with > the conference. Is there any way to get the callstate messages to include > custom variables? This would fix the problem for me. > > Regards, > Martyn > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130212/7f4e92a2/attachment.html From fdelawarde at wirelessmundi.com Tue Feb 12 19:04:56 2013 From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde) Date: Tue, 12 Feb 2013 17:04:56 +0100 Subject: [Freeswitch-users] best freeswitch timer Message-ID: <1360685096.22409.768.camel@luna.madrid.commsmundi.com> Hello, Could someone knowing the different FS timers clarify the difference between those: - default core softtimer - core heavy timer (does this option still exists?) - posix timer - timerfd Can any of those provide a significant performance improvement, or is it just not worth switching from the default? In any case, which one would be the recommended/best one under Linux with a relatively recent kernel (>3.0)? Thanks, Fran?ois. From krice at freeswitch.org Tue Feb 12 19:21:44 2013 From: krice at freeswitch.org (Ken Rice) Date: Tue, 12 Feb 2013 10:21:44 -0600 Subject: [Freeswitch-users] best freeswitch timer In-Reply-To: <1360685096.22409.768.camel@luna.madrid.commsmundi.com> Message-ID: The default one is the best one on linux with kernels starting around 2.6.32... Anything after that and you are good to go... The other timers are left around for other platforms K On 2/12/13 10:04 AM, "Fran?ois Delawarde" wrote: > Hello, Could someone knowing the different FS timers clarify the > difference between those: - default core softtimer - core heavy timer (does > this option still exists?) - posix timer - timerfd Can any of those provide a > significant performance improvement, or is it just not worth switching from > the default? In any case, which one would be the recommended/best one under > Linux with a relatively recent kernel > (>3.0)? Thanks, Fran?ois. ________________________________________________ > _________________________ Professional FreeSWITCH Consulting > Services: consulting at freeswitch.org http://www.freeswitchsolutions.com FreeSW > ITCH-powered IP PBX: The CudaTel Communication > Server Official FreeSWITCH > Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon. > com FreeSWITCH-users mailing > list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman > /listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/opt > ions/freeswitch-users http://www.freeswitch.org -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch From fdelawarde at wirelessmundi.com Tue Feb 12 20:27:01 2013 From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde) Date: Tue, 12 Feb 2013 18:27:01 +0100 Subject: [Freeswitch-users] best freeswitch timer In-Reply-To: References: Message-ID: <1360690021.22409.801.camel@luna.madrid.commsmundi.com> Thanks, that's what I wanted to know. I did a few tests a while back using timerfd on micro-servers, but didn't notice any difference at all (with my scenarios). Out of curiosity, why is timerfd even in the source? Is there a specific use-case where it could perform better (embedded system, ...)? Fran?ois. On Tue, 2013-02-12 at 10:21 -0600, Ken Rice wrote: > The default one is the best one on linux with kernels starting around > 2.6.32... Anything after that and you are good to go... > > The other timers are left around for other platforms > > K > > > > On 2/12/13 10:04 AM, "Fran?ois Delawarde" > wrote: > > > Hello, > > Could someone knowing the different FS timers clarify the > > difference > between those: > > - default core softtimer > - core heavy timer (does > > this option still exists?) > - posix timer > - timerfd > > Can any of those provide a > > significant performance improvement, or is it > just not worth switching from > > the default? > > In any case, which one would be the recommended/best one under > > Linux > with a relatively recent kernel > > (>3.0)? > > Thanks, > Fran?ois. > > > > ________________________________________________ > > _________________________ > Professional FreeSWITCH Consulting > > Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > FreeSW > > ITCH-powered IP PBX: The CudaTel Communication > > Server > > > Official FreeSWITCH > > Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon. > > com > > FreeSWITCH-users mailing > > list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman > > /listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/opt > > ions/freeswitch-users > http://www.freeswitch.org > > From c.barthes4 at free.fr Tue Feb 12 08:34:20 2013 From: c.barthes4 at free.fr (cbarthes35) Date: Mon, 11 Feb 2013 21:34:20 -0800 (PST) Subject: [Freeswitch-users] skinny (SCCP) dialing is impossible... Message-ID: <1360647260499-7587257.post@n2.nabble.com> Good morning everybody, First, i'm a current asterisk user very impatient to use freeswitch !! I've got a server ready for it, and cisco sccp phones also ready !! My problem is the following: I've installed the last 1.2 stable FS from Git: I've installed & registered a cisco 7940 SIP phone and it is OK. i've managed to register a cisco 7941g SCCP phone (not SIP),* but the dialing is impossible.* Here is the registration: However the dialing is the following: and the phone displays only ??? on the screen, and does not ring at all. Could anybody help me ? Thanks in advance. Chris. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/skinny-SCCP-dialing-is-impossible-tp7587257.html Sent from the freeswitch-users mailing list archive at Nabble.com. From hynek.cihlar at gmail.com Tue Feb 12 14:03:58 2013 From: hynek.cihlar at gmail.com (Hynek Cihlar) Date: Tue, 12 Feb 2013 12:03:58 +0100 Subject: [Freeswitch-users] CCS_RINGING too early Message-ID: When originating a call the respective call channel's call state is set to RINGING right after progress 100 is received. Here's the captured flow: |Time | | | | | | |2.848 | INVITE SDP (g711A g711U GSM telephone-eventRTP...e-101 CN) |SIP From: "" (5060) | |2.848 | 407 Proxy Authentication Required |SIP Status | |(5080) <------------------ (5060) | |2.849 | ACK | |SIP Request | |(5080) ------------------> (5060) | |2.849 | INVITE SDP (g711A g711U GSM telephone-eventRTP...e-101 CN) |SIP From: "" (5060) | |2.850 | 100 Trying| |SIP Status | |(5080) <------------------ (5060) | |13.444 | 180 Ringing |SIP Status | |(5080) <------------------ (5060) | |13.445 | 183 Session Progress SDP (g711A g711U GSM tele...ne-eventRTPType-101) |SIP Status | |(5080) <------------------ (5060) | |13.445 | RTP (g711A) |RTP Num packets:230 Duration:4.574s SSRC:0x1E777E26 | |(26056) <------------------ (19312) | |13.601 | RTP (g711A) |RTP Num packets:220 Duration:4.419s SSRC:0xA530C519 | |(26056) ------------------> (19312) | After the 100 Trying is received the switch executes switch_channel_perform_set_running_state (switch_channel.c) and the channel call state is set to RINGING and ESL event CHANNEL_CALLSTATE:RINGING is generated. I would expect the channel call state to be set to RINGING only after 180 Ringing is received from the far endpoint. Could anybody give me a hint what could be wrong or what steps to take to figure out? I am already out of ideas. Thanks! Hynek -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130212/692db3bf/attachment.html From krice at freeswitch.org Tue Feb 12 20:38:23 2013 From: krice at freeswitch.org (Ken Rice) Date: Tue, 12 Feb 2013 11:38:23 -0600 Subject: [Freeswitch-users] skinny (SCCP) dialing is impossible... In-Reply-To: <1360647260499-7587257.post@n2.nabble.com> Message-ID: I don't know if that module was ever actually completed to the point where it works... Someone else might chime in here tho On 2/11/13 11:34 PM, "cbarthes35" wrote: > Good morning everybody, > > > First, i'm a current asterisk user very impatient to use freeswitch !! > I've got a server ready for it, and cisco sccp phones also ready !! > > My problem is the following: > > I've installed the last 1.2 stable FS from Git: > > > I've installed & registered a cisco 7940 SIP phone and it is OK. > > i've managed to register a cisco 7941g SCCP phone (not SIP),* but the > dialing is impossible.* > > Here is the registration: > > > However the dialing is the following: > > > > > and the phone displays only ??? on the screen, and does not ring at all. > > Could anybody help me ? > > Thanks in advance. > > Chris. > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/skinny-SCCP-dialing-is-impossibl > e-tp7587257.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch From consultoriaticesar at gmail.com Tue Feb 12 20:42:28 2013 From: consultoriaticesar at gmail.com (Joao Cesar S. Pereira) Date: Tue, 12 Feb 2013 15:42:28 -0200 Subject: [Freeswitch-users] (no subject) In-Reply-To: <1360647260499-7587257.post@n2.nabble.com> References: <1360647260499-7587257.post@n2.nabble.com> Message-ID: <511A7F04.9010704@gmail.com> From msc at freeswitch.org Tue Feb 12 20:59:06 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 12 Feb 2013 09:59:06 -0800 Subject: [Freeswitch-users] High Availability Cluster Module for FreeSWITCH In-Reply-To: References: Message-ID: > building business around FS ;) Jokes aside - if somebody wants to > configure FS in HA evironment I don't think it is much of an obstacle > for him to configure Corosync as already he has to have some kind of > DB failover as well, redundant switches, power, etc. > I would like to see that statement put to the test. I would like to see someone attempt to put Corosync (or whatever existing solution) to use as the HA engine in a FS cluster. I suspect the amount of work involved is way more than has been suggested. If it really were that easy ("easy" being relative compared to what people perceive Eliot is doing) then I suspect it would have been done already. In any case, Tony makes good points about furthering the discussion. If we can improve FreeSWITCH in any way to make this process more feasible (regardless of which solution is used) then that is a win for everyone. We will *briefly* discuss this subject on tomorrow's conference call so that interested parties can decide when and how they'd like to further this discussion. -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130212/f2f121ef/attachment.html From msc at freeswitch.org Tue Feb 12 21:02:29 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 12 Feb 2013 10:02:29 -0800 Subject: [Freeswitch-users] Mod_Conference ESL: Filtering Events In-Reply-To: References: Message-ID: Thanks for sharing this tip! -MC On Tue, Feb 12, 2013 at 12:50 AM, Usama Zaidi wrote: > Hey All, > > So I found the solution to my problem and I'm updating the wiki page for > event filters so it might help someone else too. > > Here's what I did: > > $con->events("plain","all"); > $con->events("plain","CUSTOM conference::maintenance"); > $con->sendRecv("filter Unique-ID $uuidMember1"); > $con->sendRecv("filter Unique-ID $uuidMember2"); > $con->sendRecv("filter Unique-ID $uuidMember3"); > . > . > . > Do this for all conference members in your script, say in a loop or > something, and then you'll get all custom events, pause/unpause, > started/stopped speaking, all in a single script. > > On Tue, Feb 12, 2013 at 6:22 AM, < > freeswitch-users-request at lists.freeswitch.org> wrote: > >> Send FreeSWITCH-users mailing list submissions to >> freeswitch-users at lists.freeswitch.org >> >> To subscribe or unsubscribe via the World Wide Web, visit >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> or, via email, send a message with subject or body 'help' to >> freeswitch-users-request at lists.freeswitch.org >> >> You can reach the person managing the list at >> freeswitch-users-owner at lists.freeswitch.org >> >> When replying, please edit your Subject line so it is more specific >> than "Re: Contents of FreeSWITCH-users digest..." >> >> Today's Topics: >> >> 1. Re: Hold/unhold events for a conference participant (Seven Du) >> 2. FreeSWITCH Weekly News and Notes (Michael Collins) >> 3. Mod_Conference ESL: Filtering Events (Usama Zaidi) >> 4. Re: unexplained RAM usage increase (Mario G) >> >> >> ---------- Forwarded message ---------- >> From: Seven Du >> To: FreeSWITCH Users Help >> Cc: >> Date: Tue, 12 Feb 2013 09:02:24 +0800 >> Subject: Re: [Freeswitch-users] Hold/unhold events for a conference >> participant >> There's CHANNEL_HOLD and CHANNEL_UNHOLD messages >> >> And take a look at >> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_verbose_events >> >> Is there a member-id var bind on channel? I only see it on >> conference::maintenance messages, not sure though. >> >> On Tuesday, February 12, 2013 at 12:56 AM, Martyn Davies wrote: >> >> I'm trying to process CHANNEL_CALLSTATE events (in an ESL socket >> application) to extract the HELD/ACTIVE messages for conference >> participants. >> >> One problem for me is that the name of the conference and the member-id >> for that call are not available to me in the callstate messages, and >> therefore it is difficult to associated calls (going HELD and ACTIVE) with >> the conference. Is there any way to get the callstate messages to include >> custom variables? This would fix the problem for me. >> >> Regards, >> Martyn >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> >> >> >> ---------- Forwarded message ---------- >> From: Michael Collins >> To: freeswitch-users at lists.freeswitch.org, >> freeswitch-dev at lists.freeswitch.org, >> freeswitch-cluecon at lists.freeswitch.org >> Cc: >> Date: Mon, 11 Feb 2013 17:05:31 -0800 >> Subject: [Freeswitch-users] FreeSWITCH Weekly News and Notes >> Hello all! >> >> News and notes are back after a brief hiatus last week. We (the >> FreeSWITCH team) were in Milwaukee last week and we appreciated being fed >> by the community. Thank you! We love this community. >> >> It seems the big news last week came >> courtesy of our friend Kristian Kielhofner . As >> reported in this blog post, >> Kristian ran into a rather unusual set of circumstances that resulted in >> Packets of Death for some Intel NICs. In addition to being Slashdotted, >> Kristian's research ended up being featured in a Wired.com story. >> Don't forget to read Kristian's update post that >> includes information about Intel's response to the whole situation. >> >> This week things will start to return to normal with our weekly >> conference call . >> We are going to spend a few weeks talking about various FreeSWITCH GUIs, >> starting with the CudaTel <>. For the past four >> years or so we've been building the CudaTel Communication Server and we'd >> like to show off some of the cool things it can do. >> >> We have two other news items. The first one has to do with mod_ha_cluster, >> which our very own Eliot Gable is building. Financial support for this >> open-source module will be done through FreeSWITCH Solutions. >> An interesting discussion can be found in this email thread. >> The other item comes from Ken Rice who has some updates on Jira and ZRTP. >> Check out his mailing list postfor more information. >> >> Have a great week and we'll talk to you on Wednesday. >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> >> >> ---------- Forwarded message ---------- >> From: Usama Zaidi >> To: freeswitch-users at lists.freeswitch.org >> Cc: >> Date: Tue, 12 Feb 2013 06:11:20 +0500 >> Subject: [Freeswitch-users] Mod_Conference ESL: Filtering Events >> Hi, >> >> I want to filter user started/stopped talking events for all users in a >> particular conference using inbound ESL, if I do something like >> $con->sendRecv("filter Unique-ID $ConfUUID"); I don't get those events so >> I'm forced to do something like $con->sendRecv("filter Unique-ID >> $participantUUID"); which basically means that I need n scripts and local >> channels (as conference participants to spawn that inbound ESL script) for >> n conference participants, and I'd really love to avoid that, I don't want >> to do something like $con->events("plain","CUSTOM >> conference::maintenance"); because, if there's 100 conferences going on >> I'll have 100 scripts running for each conference and they'd be getting >> events for all 100 conferences, so basically FS is sending 100*100 * (num >> events) to all the scripts. >> >> Thanks. >> >> -Usama >> >> -- >> I'd love to change the world, but they wont gimme the source code to it >> >> >> ---------- Forwarded message ---------- >> From: Mario G >> To: FreeSWITCH Users Help >> Cc: >> Date: Mon, 11 Feb 2013 17:22:02 -0800 >> Subject: Re: [Freeswitch-users] unexplained RAM usage increase >> Jeff and Anthony, an update: I have the commit pegged and valgrind log at >> http://jira.freeswitch.org/browse/FS-5095. Reading Jeff's jira I don't >> think they are related but you never know. Sorry to take so long but git >> bisects resulted in many versions that would not build (I opened about 5 >> jiras last 2 months) and I had to hand patch to test. Family emergencies >> took a big toll. I put about 30 hours into this since November and tested >> day/night the last 6 days, have many pages of notes and logs. I hope the >> jira has enough to solve it. BTW, I used the valgrind line below. >> Mario G >> >> On Jan 29, 2013, at 6:15 PM, Anthony Minessale wrote: >> >> If its still there on head, try valgrind. >> >> valgrind --tool=memcheck --log-file=vg.log --leak-check=full >> --leak-resolution=high --show-reachable=yes >> /usr/local/freeswitch/bin/freeswitch -vg >> >> >> On Tue, Jan 29, 2013 at 8:09 PM, Mario G wrote: >> >>> I have updated and if by some miracle it's fixed I will report back. >>> Storage increases with no calls so I assume it's registrations >>> (internal/external?). Please understand this is probably this worse week of >>> the year for me to work on FS (or anything else). I promise to try and >>> narrow down the commit that started it. The memory issue started at a bad >>> time when I could not work on FS (emergencies) and by the time I updated >>> other errors crept in that prevented use and I had to open JIRAs for them, >>> so testing memory was out. Thanks for letting me know about todays fix. >>> Mario G >>> >>> >>> On Jan 29, 2013, at 5:41 PM, Anthony Minessale wrote: >>> >>> There is a commit today that found some sql leaks but I would not label >>> them as massive. Its worth it to check and maybe run valgrind. From our >>> perspective; Any time someone mentions nonchalantly, "Oh yeah I've had this >>> glaring problem for months", breaks our heart =p >>> I was just about to push the tag to unlock the 1.2.6 you so desperately >>> want but now I am halting it to confirm this report. >>> >>> TEST TEST TEST HEAD... Today's head is tomorrow's stable.... >>> >>> >>> >>> On Tue, Jan 29, 2013 at 7:23 PM, Ken Rice wrote: >>> >>>> Mario, >>>> >>>> Is it fixed in the master branch? If it is fixed there it will be >>>> rolled down to 1.2 branch before long... >>>> >>>> >>>> On 1/29/13 6:40 PM, "Mario G" wrote: >>>> >>>> Anthony, I tried bisect and the problem is that there are other issues >>>> (RTP error still in 1.2 stable, waiting for 1.2.6), and about 4-5 other >>>> issues in head that prevented it from working for a good amount of time. I >>>> started jiras for them and they are fixed, but doing bisect brings them >>>> back so it's had been really hard to pinpoint the memory issue. Believe me, >>>> I have put a LOT of time into trying to narrow it down before opening a >>>> JIRA and will continue to do so. Right now it's hard since there were >>>> personal emergencies the last 2 months so other pressing things had to take >>>> priority. Still, I am working on FreeSwitch keeping up-to-date with head in >>>> case other issues pop up I can open a JIRA on them. >>>> Mario G >>>> >>>> On Jan 29, 2013, at 3:14 PM, Anthony Minessale wrote: >>>> >>>> Things like this are sad. We depend on testing and reporting for our >>>> releases. If you wait months to bring up a problem. It will spoil the >>>> whole release. >>>> >>>> If you you feel some leak has appeared suddenly, why can't you do git >>>> bisect and find it? >>>> >>>> >>>> >>>> On Tue, Jan 29, 2013 at 12:58 PM, Mario G wrote: >>>> >>>> Probably not be related, but you never know: on OSX since Nov/Dec there >>>> has been a memory leak on 1.2 and head that occurs for in/out/and >>>> registrations. I had to triple memory and recycle FreeSwitch every 2-3 days >>>> since then. Will open a Jira when I can obtain detailed info and possibly >>>> run valgrind. I am also also waiting to update the main FS computer from >>>> 10.6.8 to 10.8.3 to see its effect. Hopefully in Feb. >>>> Mario G >>>> >>>> On Jan 29, 2013, at 9:04 AM, Anthony Minessale wrote: >>>> >>>> Can you systematically increase the current call count and see where >>>> you do see something? >>>> On a system with limited ram you can also consider stripping all the >>>> .so files in the mods and lib dir but you will need to put non-stripped >>>> ones in for any debugging. >>>> >>>> I don't really see a correlation on how not using a timer could trigger >>>> an sustained memory increase so that's why I'd like you to step up the >>>> number and see if you can find a number of calls that tops out because >>>> usually there always is a magic number where it will hover and go up and >>>> down a meg at a time. >>>> >>>> SIP calls are required to keep state data around for at least 30 >>>> seconds after a call ends and there are a number of pools in the code that >>>> inflate once and do not return the memory. Its usually possible to >>>> identify the high watermark on a particular box. >>>> >>>> For instance the machine we host the conference call on launches using >>>> 25 megs and hovers at about 350 megs once it has accumulated all the pool >>>> memory it needs over time. >>>> >>>> >>>> >>>> >>>> >>>> On Tue, Jan 29, 2013 at 8:33 AM, Jeff Pyle >>>> wrote: >>>> >>>> Version: >>>> FreeSWITCH version: 1.3.4-n20130122T122521Z-1~squeeze+1 >>>> (-n20130122T122521Z-1~squeeze+1) >>>> >>>> The calls are bridged, from one sofia profile to another. >>>> >>>> Unfortunately two concurrent calls doesn't seem to trigger the same >>>> behavior. >>>> >>>> >>>> - Jeff >>>> >>>> >>>> On Mon, Jan 28, 2013 at 11:21 PM, Ken Rice >>>> wrote: >>>> >>>> Ok the package timestamps/versions there don?t do us a lot of good, we >>>> need to know the version line from the FreeSWITCH CLI.. >>>> >>>> >>>> On 1/28/13 9:59 PM, "Jeff Pyle" >>> http://jpyle at fidelityvoice.com/> > wrote: >>>> >>>> I just updated from repo.profhost.eu < >>>> http://repo.profhost.eu > . The most >>>> recent timestamp on the packages was 2013-01-28 03:41:21 GMT. Same >>>> behavior. At 5 minutes it was using 12.5% RAM. At 40 minutes, 60.4%. >>>> After disconnecting the calls the usage returned to 8.5%. >>>> >>>> >>>> >>>> I started toggling config items to see if I could impact this. I found >>>> one that seems to have an effect: rtp-timer-name in the sofia profile >>>> config. By changing it from 'soft' to 'none', the CPU utilization with 30 >>>> calls dropped from ~70% to ~46%, and the RAM usage is rock solid at 5.8%. >>>> >>>> That's great, but does it make any sense? >>>> >>>> Does an rtp-timer-name of 'none' pose any risks? >>>> >>>> >>>> >>>> - Jeff >>>> >>>> >>>> >>>> On Mon, Jan 28, 2013 at 8:43 PM, Jeff Pyle >>> http://jpyle at fidelityvoice.com/> > wrote: >>>> >>>> It's on Voyage Linux, a cousin of Debian. I believe it uses glibc. >>>> >>>> >>>> - Jeff >>>> >>>> >>>> On Mon, Jan 28, 2013 at 6:45 PM, Kristian Kielhofner >>> http://kris at kriskinc.com/> > wrote: >>>> Out of curiosity does your distro use uclibc, eglibc, or glibc? >>>> >>>> On Mon, Jan 28, 2013 at 5:41 PM, Jeff Pyle >>> http://jpyle at fidelityvoice.com/> > wrote: >>>> > Hello, >>>> > >>>> > I'm running HEAD version from Jan 22 on an Alix board with an AMD >>>> Geode LX >>>> > processor (i386). I can sustain 30 concurrent calls averaging around >>>> 70% >>>> > CPU utilization by the freeswitch process, measured by top. Bypass >>>> media >>>> > and proxy media are disabled. PCMU is forced on both endpoints (no >>>> > transcoding). >>>> > >>>> > The problem is the RAM usage over time. The board has 256M. Idle, >>>> > freeswitch occupies around 4% after a fresh restart. A minute or so >>>> after >>>> > 30 calls are nailed up the RAM usage is about 7.2%. After 5 minutes, >>>> 13.6%. >>>> > After 60 minutes, near 65%. Disconnecting the calls returns the RAM >>>> usage >>>> > to 6-8%. >>>> > >>>> > I've not tried to troubleshoot an issue like this before. Is >>>> valgrind the >>>> > next step, or would something else make more sense? >>>> > >>>> > >>>> > Regards, >>>> > Jeff >>>> >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>>> >>>> >>>> >>>> >>>> -- >>>> Ken >>>> *http://www.FreeSWITCH.org >>>> http://www.ClueCon.com >>>> http://www.OSTAG.org >>>> *irc.freenode.net #freeswitch >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>>> >>>> >>> >>> >>> -- >>> Anthony Minessale II >>> >>> FreeSWITCH http://www.freeswitch.org/ >>> ClueCon http://www.cluecon.com/ >>> Twitter: http://twitter.com/FreeSWITCH_wire >>> >>> AIM: anthm >>> MSN:anthony_minessale at hotmail.com >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>> IRC: irc.freenode.net #freeswitch >>> >>> FreeSWITCH Developer Conference >>> sip:888 at conference.freeswitch.org >>> googletalk:conf+888 at conference.freeswitch.org >>> pstn:+19193869900 >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> >> >> _______________________________________________ >> 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 >> >> > > > -- > I'd love to change the world, but they wont gimme the source code to it > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130212/5a754cad/attachment-0001.html From david.villasmil.work at gmail.com Tue Feb 12 21:11:22 2013 From: david.villasmil.work at gmail.com (David Villasmil) Date: Tue, 12 Feb 2013 19:11:22 +0100 Subject: [Freeswitch-users] Nibblebill database question Message-ID: Hello guys, I have created 2 dsns, one for freeswitch core "freeswitch" and one for nibblebill ("whatever"). Is it possible to use different dsns in core and nibblebill? or does nibblebill uses the same dsn as the core?? thanks David -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130212/0cb53ad9/attachment.html From egable+freeswitch at gmail.com Tue Feb 12 21:19:48 2013 From: egable+freeswitch at gmail.com (Eliot Gable) Date: Tue, 12 Feb 2013 13:19:48 -0500 Subject: [Freeswitch-users] High Availability Cluster Module for FreeSWITCH In-Reply-To: References: Message-ID: On Tue, Feb 12, 2013 at 9:21 AM, Marcin Gozdalik wrote: > 2013/2/11 Eliot Gable : > > On Mon, Feb 11, 2013 at 7:36 AM, Marcin Gozdalik > wrote: > >> > >> +1 > >> > >> I do not doubt mod_ha is necessary inside of FS and it may be > >> better/simpler than writing Pacemaker resource agent, but writing > >> yet-another-cluster-communication-engine is IMHO the wrong way to go > >> and using Corosync for communication will give a lot of value from > >> mature codebase. > >> > > > > I understand what you are saying, but what I am trying to get across is > that > > I am not writing yet-another-cluster-communication-engine. All I am > really > > doing is combining a multicast messaging API written by Tony and the > event > > API in FS to broadcast existing state information between multiple FS > nodes, > > as well as adding a tiny amount of logic on top of that to coordinate > call > > fail over and recovery. That's probably a little over-simplified, but it > > gets the point across. The network communication code is already in FS > and > > well tested. The event system is already in FS and well tested. > > I also think I understand what you are saying. It means we have > trouble putting thought into writing ;) > >From what I understand what you are trying to achieve is that every > node in FS "cluster" knows what are the nodes and whether they are > down or up. > What I am saying is that this simple task is fundamentally hard. > Sending and receiving multicast is easy, but keeping distributed state > consistent between nodes in cluster is hard (like in really hard, > harder than writing VoIP softswitch all over again), especially in > case of Byzantine failures (i.e. nodes lying that they are down when > they are up or other way round). I am no big expert in the area but > seen at least 2 cases (MMM - > http://www.xaprb.com/blog/2011/05/04/whats-wrong-with-mmm/ and Chubby > in Google - > http://www.read.seas.harvard.edu/~kohler/class/08w-dsi/chandra07paxos.pdf) > where people were trying to write (MMM) or use (Chubby) some kind of > distributed code and failed. > That's why whenever I see anything related to distributed state I say > that it's way beyond my understanding and best is to use something > that works. > You were fortunate to have that resource available, as well as (I assume) an already made resource agent available for managing FreeSWITCH. I had to learn it from this: http://clusterlabs.org/doc/en-US/Pacemaker/1.0/pdf/Pacemaker_Explained/Pacemaker_Explained.pdf I also had to craft a resource agent to manage FreeSWITCH (none existed at the time). Then I found out Pacemaker was buggy (it has gotten much better since I started using it) and wouldn't properly honor colocation constraints or grouping correctly in certain failure conditions, so I had to make the resource agent handle managing all the IP addresses for FreeSWITCH (each instance had 12 Sofia profiles with each one running on a different IP). I spent months testing hundreds of different possible failure conditions and fixing dozens if not hundreds of bugs in the configuration and in how the resource agent managed everything and reported on the health of FreeSWITCH. Everything from someone accidentally removing a needed IP from the system to a failed hard drive to a Sofia profile failing to load to firewall rules accidentally blocking needed ports, etc. If you spent only one day setting up such a system, I am certain you failed to account for dozens if not hundreds of possible failure conditions. At the end of those 3 months of hell, I had a single pair of nodes which I could rely on to "do the right thing" under practically any failure condition. However, even then, I still had several dozen ways I could simulate FreeSWITCH failing which the system simply could not detect efficiently. I made attempts at testing some of them, but the load induced on the system to test them frequently enough to matter made the system fall outside the specifications I needed for the project to be profitable and workable. I have years of experience building and deploying FreeSWITCH clusters with Pacemaker and Corosync and hunting down and gracefully handling practically every conceivable way such a system could fail. I understand you think it's hard to do; and that is not without reason. I've lived it; I've done it. I know what's involved in the process. I simply want to take my experience with it and write it down in code in the form of mod_ha_cluster so that other people don't have to waste their time relearning all the things I already know with regard to making FreeSWITCH run in an HA setup. In the absence of Pacemaker and Corosync, my goal is to provide mod_ha_cluster enough awareness that the vast majority of failure cases are handled gracefully and FS can take care of itself for bringing a slave online to take over for a failed node. However, there is no reason I cannot also write it to let Pacemaker and Corosync give it direction as to which slave to turn into a master. So, if it makes you more comfortable, think of it as a glorified resource agent which always happens to know about the "deep state" of the nodes and can test for things that traditional resource agents can never do effectively. Then, when you do a "shallow" poll of the state, you can get back the "deep state" instead, but at the cost of doing a "shallow" test. And, on top of all that, it will handle synchronizing various data between the nodes so you don't need to rely on an external HA database. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130212/51d47513/attachment.html From cmrienzo at gmail.com Tue Feb 12 21:25:43 2013 From: cmrienzo at gmail.com (cmrienzo at gmail.com) Date: Tue, 12 Feb 2013 13:25:43 -0500 Subject: [Freeswitch-users] best freeswitch timer In-Reply-To: <1360690021.22409.801.camel@luna.madrid.commsmundi.com> References: <1360690021.22409.801.camel@luna.madrid.commsmundi.com> Message-ID: <84500D9D-B26F-401B-8B99-F25222B616F3@gmail.com> timerfd syscalls are automatically used by the default timer if your kernel supports it so mod_timerfd isn't necessary anymore. mod_posix_timer might give you better performance on old kernels that don't support timerfd. Chris On Feb 12, 2013, at 12:27 PM, Fran?ois Delawarde wrote: > Thanks, that's what I wanted to know. I did a few tests a while back > using timerfd on micro-servers, but didn't notice any difference at all > (with my scenarios). > > Out of curiosity, why is timerfd even in the source? Is there a specific > use-case where it could perform better (embedded system, ...)? > > Fran?ois. > > > > On Tue, 2013-02-12 at 10:21 -0600, Ken Rice wrote: >> The default one is the best one on linux with kernels starting around >> 2.6.32... Anything after that and you are good to go... >> >> The other timers are left around for other platforms >> >> K >> >> >> >> On 2/12/13 10:04 AM, "Fran?ois Delawarde" >> wrote: >> >>> Hello, >> >> Could someone knowing the different FS timers clarify the >>> difference >> between those: >> >> - default core softtimer >> - core heavy timer (does >>> this option still exists?) >> - posix timer >> - timerfd >> >> Can any of those provide a >>> significant performance improvement, or is it >> just not worth switching from >>> the default? >> >> In any case, which one would be the recommended/best one under >>> Linux >> with a relatively recent kernel >>> (>3.0)? >> >> Thanks, >> Fran?ois. >> >> >> >> ________________________________________________ >>> _________________________ >> Professional FreeSWITCH Consulting >>> Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSW >>> ITCH-powered IP PBX: The CudaTel Communication >>> Server >> >> >> Official FreeSWITCH >>> Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon. >>> com >> >> FreeSWITCH-users mailing >>> list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman >>> /listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/opt >>> ions/freeswitch-users >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 vipkilla at gmail.com Tue Feb 12 21:32:29 2013 From: vipkilla at gmail.com (Vik Killa) Date: Tue, 12 Feb 2013 13:32:29 -0500 Subject: [Freeswitch-users] High Availability Cluster Module for FreeSWITCH In-Reply-To: References: Message-ID: > > I would like to see that statement put to the test. I would like to see > someone attempt to put Corosync (or whatever existing solution) to use as > the HA engine in a FS cluster. I suspect the amount of work involved is way > more than has been suggested. If it really were that easy ("easy" being > relative compared to what people perceive Eliot is doing) then I suspect it > would have been done already. Hey MC, http://wiki.freeswitch.org/wiki/Enterprise_deployment_with_Corosync ^ I wrote that months ago. Pretty easy :) From msc at freeswitch.org Tue Feb 12 21:41:10 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 12 Feb 2013 10:41:10 -0800 Subject: [Freeswitch-users] High Availability Cluster Module for FreeSWITCH In-Reply-To: References: Message-ID: On Tue, Feb 12, 2013 at 10:19 AM, Eliot Gable wrote: > On Tue, Feb 12, 2013 at 9:21 AM, Marcin Gozdalik wrote: > >> 2013/2/11 Eliot Gable : >> > On Mon, Feb 11, 2013 at 7:36 AM, Marcin Gozdalik >> wrote: >> >> >> >> +1 >> >> >> >> I do not doubt mod_ha is necessary inside of FS and it may be >> >> better/simpler than writing Pacemaker resource agent, but writing >> >> yet-another-cluster-communication-engine is IMHO the wrong way to go >> >> and using Corosync for communication will give a lot of value from >> >> mature codebase. >> >> >> > >> > I understand what you are saying, but what I am trying to get across is >> that >> > I am not writing yet-another-cluster-communication-engine. All I am >> really >> > doing is combining a multicast messaging API written by Tony and the >> event >> > API in FS to broadcast existing state information between multiple FS >> nodes, >> > as well as adding a tiny amount of logic on top of that to coordinate >> call >> > fail over and recovery. That's probably a little over-simplified, but it >> > gets the point across. The network communication code is already in FS >> and >> > well tested. The event system is already in FS and well tested. >> >> I also think I understand what you are saying. It means we have >> trouble putting thought into writing ;) >> >From what I understand what you are trying to achieve is that every >> node in FS "cluster" knows what are the nodes and whether they are >> down or up. >> What I am saying is that this simple task is fundamentally hard. >> Sending and receiving multicast is easy, but keeping distributed state >> consistent between nodes in cluster is hard (like in really hard, >> harder than writing VoIP softswitch all over again), especially in >> case of Byzantine failures (i.e. nodes lying that they are down when >> they are up or other way round). I am no big expert in the area but >> seen at least 2 cases (MMM - >> http://www.xaprb.com/blog/2011/05/04/whats-wrong-with-mmm/ and Chubby >> in Google - >> http://www.read.seas.harvard.edu/~kohler/class/08w-dsi/chandra07paxos.pdf >> ) >> where people were trying to write (MMM) or use (Chubby) some kind of >> distributed code and failed. >> That's why whenever I see anything related to distributed state I say >> that it's way beyond my understanding and best is to use something >> that works. >> > > > You were fortunate to have that resource available, as well as (I assume) > an already made resource agent available for managing FreeSWITCH. I had to > learn it from this: > > > http://clusterlabs.org/doc/en-US/Pacemaker/1.0/pdf/Pacemaker_Explained/Pacemaker_Explained.pdf > > I also had to craft a resource agent to manage FreeSWITCH (none existed at > the time). Then I found out Pacemaker was buggy (it has gotten much better > since I started using it) and wouldn't properly honor colocation > constraints or grouping correctly in certain failure conditions, so I had > to make the resource agent handle managing all the IP addresses for > FreeSWITCH (each instance had 12 Sofia profiles with each one running on a > different IP). I spent months testing hundreds of different possible > failure conditions and fixing dozens if not hundreds of bugs in the > configuration and in how the resource agent managed everything and reported > on the health of FreeSWITCH. Everything from someone accidentally removing > a needed IP from the system to a failed hard drive to a Sofia profile > failing to load to firewall rules accidentally blocking needed ports, etc. > If you spent only one day setting up such a system, I am certain you failed > to account for dozens if not hundreds of possible failure conditions. At > the end of those 3 months of hell, I had a single pair of nodes which I > could rely on to "do the right thing" under practically any failure > condition. However, even then, I still had several dozen ways I could > simulate FreeSWITCH failing which the system simply could not > detect efficiently. I made attempts at testing some of them, but the load > induced on the system to test them frequently enough to matter made the > system fall outside the specifications I needed for the project to be > profitable and workable. > > I have years of experience building and deploying FreeSWITCH clusters with > Pacemaker and Corosync and hunting down and gracefully handling practically > every conceivable way such a system could fail. I understand you think it's > hard to do; and that is not without reason. I've lived it; I've done it. I > know what's involved in the process. I simply want to take my experience > with it and write it down in code in the form of mod_ha_cluster so that > other people don't have to waste their time relearning all the things I > already know with regard to making FreeSWITCH run in an HA setup. In the > absence of Pacemaker and Corosync, my goal is to provide mod_ha_cluster > enough awareness that the vast majority of failure cases are handled > gracefully and FS can take care of itself for bringing a slave online to > take over for a failed node. However, there is no reason I cannot also > write it to let Pacemaker and Corosync give it direction as to which slave > to turn into a master. So, if it makes you more comfortable, think of it as > a glorified resource agent which always happens to know about the "deep > state" of the nodes and can test for things that traditional resource > agents can never do effectively. Then, when you do a "shallow" poll of the > state, you can get back the "deep state" instead, but at the cost of doing > a "shallow" test. And, on top of all that, it will handle synchronizing > various data between the nodes so you don't need to rely on an external HA > database. > > +100! In my other post to this thread I postulated that if it were that easy then someone else would have done it already or that it should be "easily" doable. I was not aware of your excruciatingly intimate familiarity with the solutions that the others have been suggesting as alternatives for what you are working on. I think the "glorified resource agent" analogy is particularly descriptive. :) Hopefully others will chime on on the other thread about the message bus/shared key storageso we can take this feature to the next level! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130212/65a76a41/attachment-0001.html From msc at freeswitch.org Tue Feb 12 21:42:11 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 12 Feb 2013 10:42:11 -0800 Subject: [Freeswitch-users] High Availability Cluster Module for FreeSWITCH In-Reply-To: References: Message-ID: On Tue, Feb 12, 2013 at 10:32 AM, Vik Killa wrote: > > > > I would like to see that statement put to the test. I would like to see > > someone attempt to put Corosync (or whatever existing solution) to use as > > the HA engine in a FS cluster. I suspect the amount of work involved is > way > > more than has been suggested. If it really were that easy ("easy" being > > relative compared to what people perceive Eliot is doing) then I suspect > it > > would have been done already. > > Hey MC, > http://wiki.freeswitch.org/wiki/Enterprise_deployment_with_Corosync > ^ I wrote that months ago. Pretty easy :) > > How many failure scenarios does that handle? -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130212/1ebc1f40/attachment.html From mario_fs at mgtech.com Tue Feb 12 21:43:07 2013 From: mario_fs at mgtech.com (Mario G) Date: Tue, 12 Feb 2013 10:43:07 -0800 Subject: [Freeswitch-users] Can local_stream MOH have a time limit/max duration? In-Reply-To: References: <219BCB63-3A5A-419F-B374-BDD45FDE3FAB@mgtech.com> Message-ID: I tried a few things with transfer_ringaback but no dice. The wiki also states transfer_ringback is only for after the call is answered, use ringback for early media, which is what I need it for. i looked at source code but did not find anything that would allow limiting moh duration, bummer. Thanks. Mario G On Feb 11, 2013, at 12:57 PM, Michael Collins wrote: > For #2 use transfer_ringback channel variable and set it to MOH and it should work the same way, i.e. instead of ringing the caller will hear music. > > -MC > > On Fri, Feb 8, 2013 at 8:55 AM, Mario G wrote: > I looked into chime-ins, but it only chimes in at fixed intervals, and it was not flexible enough for true customization. I could not find anything else on the wiki that would play moh (or any other file) for a specific time. See my previous post for a details of how it works and what I wanted to do. Thanks. > Mario G > > On Feb 8, 2013, at 5:48 AM, Vik Killa wrote: > > > I don't quite understand what you are trying to do here. > > mod_local_stream plays audio in a loop constantly with chime-ins > > (announcements) set to play periodically. If you need something to > > play from the beginning for a certain amount of time, you need to use > > something other than mod_local_stream. > > > > > >> On Wed, Feb 6, 2013 at 9:04 AM, Mario G wrote: > >>> > >>> Could not find anything in the wiki and this may not be possible but... I > >>> have early media with instructions, is has voice, ring, voice, music. I want > >>> to change the ring to music but I could not find a way to limit the time the > >>> music plays. See the second example, I would like to limit the first moh to > >>> a maximum time. If not possible, is this a "bounty" enhancement? Thanks for > >>> any help. > >>> Mario G > >>> > >>> > >>> Currently: > >>> >>> data="ringback=file_string://${lua_ring1}!tone_stream://${us-ring};loops=3!file_string://${lua_ring2}!local_stream://moh"/> > >>> > >>> Would Like (add something to limit the first moh): > >>> >>> data="ringback=file_string://${lua_ring1}!local_stream://moh!file_string://${lua_ring2}!local_stream://moh"/> > >>> > >>> > >>> ______________________________________________________________________ > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130212/7d3db45a/attachment.html From egable+freeswitch at gmail.com Tue Feb 12 21:45:33 2013 From: egable+freeswitch at gmail.com (Eliot Gable) Date: Tue, 12 Feb 2013 13:45:33 -0500 Subject: [Freeswitch-users] High Availability Cluster Module for FreeSWITCH In-Reply-To: References: Message-ID: On Tue, Feb 12, 2013 at 1:32 PM, Vik Killa wrote: > > > > I would like to see that statement put to the test. I would like to see > > someone attempt to put Corosync (or whatever existing solution) to use as > > the HA engine in a FS cluster. I suspect the amount of work involved is > way > > more than has been suggested. If it really were that easy ("easy" being > > relative compared to what people perceive Eliot is doing) then I suspect > it > > would have been done already. > > Hey MC, > http://wiki.freeswitch.org/wiki/Enterprise_deployment_with_Corosync > ^ I wrote that months ago. Pretty easy :) > ip link set arp off Watch your cluster sit there like an idiot while your traffic fails and you spend hours trying to find the cause. ethtool -s speed 10 duplex half Watch your cluster sit there like an idiot while your traffic fails and you spend hours trying to find the cause. Rinse and repeat with literally dozens upon dozens of other things you could do where your cluster just sits there broken and you pull your hair out. Still seem simple? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130212/c06a483f/attachment.html From avi at avimarcus.net Tue Feb 12 21:53:34 2013 From: avi at avimarcus.net (Avi Marcus) Date: Tue, 12 Feb 2013 20:53:34 +0200 Subject: [Freeswitch-users] Nibblebill database question In-Reply-To: References: Message-ID: All modules that touch the DB (I think!) have their own configuration that allows you pick the database. mod_nibblebill does indeed has a param name="db_dsn" option so yes, it will honor that. -Avi On Tue, Feb 12, 2013 at 8:11 PM, David Villasmil < david.villasmil.work at gmail.com> wrote: > Hello guys, > > I have created 2 dsns, one for freeswitch core "freeswitch" and one for > nibblebill ("whatever"). > Is it possible to use different dsns in core and nibblebill? or does > nibblebill uses the same dsn as the core?? > > thanks > > David > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130212/9e802ac0/attachment-0001.html From david.villasmil.work at gmail.com Tue Feb 12 22:00:31 2013 From: david.villasmil.work at gmail.com (David Villasmil) Date: Tue, 12 Feb 2013 20:00:31 +0100 Subject: [Freeswitch-users] Nibblebill database question In-Reply-To: References: Message-ID: Yes, it does have the option, but it doesn't seem to be using a different one... I must've missed something... On Tue, Feb 12, 2013 at 7:53 PM, Avi Marcus wrote: > All modules that touch the DB (I think!) have their own configuration that > allows you pick the database. > mod_nibblebill does indeed has a param name="db_dsn" option so yes, it > will honor that. > > -Avi > > On Tue, Feb 12, 2013 at 8:11 PM, David Villasmil < > david.villasmil.work at gmail.com> wrote: > >> Hello guys, >> >> I have created 2 dsns, one for freeswitch core "freeswitch" and one for >> nibblebill ("whatever"). >> Is it possible to use different dsns in core and nibblebill? or does >> nibblebill uses the same dsn as the core?? >> >> thanks >> >> David >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130212/c0c9d690/attachment.html From david.villasmil.work at gmail.com Tue Feb 12 22:02:44 2013 From: david.villasmil.work at gmail.com (David Villasmil) Date: Tue, 12 Feb 2013 20:02:44 +0100 Subject: [Freeswitch-users] Nibblebill database question In-Reply-To: References: Message-ID: It is strange, core is connecting perfectly, but nibblebill, either with the db used by core or the one i want doesn't seem to be connecting... Is there anyway to check whether it is? Thanks for your help Avi David On Tue, Feb 12, 2013 at 8:00 PM, David Villasmil < david.villasmil.work at gmail.com> wrote: > Yes, it does have the option, but it doesn't seem to be using a different > one... I must've missed something... > > > On Tue, Feb 12, 2013 at 7:53 PM, Avi Marcus wrote: > >> All modules that touch the DB (I think!) have their own configuration >> that allows you pick the database. >> mod_nibblebill does indeed has a param name="db_dsn" option so yes, it >> will honor that. >> >> -Avi >> >> On Tue, Feb 12, 2013 at 8:11 PM, David Villasmil < >> david.villasmil.work at gmail.com> wrote: >> >>> Hello guys, >>> >>> I have created 2 dsns, one for freeswitch core "freeswitch" and one for >>> nibblebill ("whatever"). >>> Is it possible to use different dsns in core and nibblebill? or does >>> nibblebill uses the same dsn as the core?? >>> >>> thanks >>> >>> David >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130212/2fe4fa6f/attachment.html From avi at avimarcus.net Tue Feb 12 22:04:52 2013 From: avi at avimarcus.net (Avi Marcus) Date: Tue, 12 Feb 2013 21:04:52 +0200 Subject: [Freeswitch-users] Nibblebill database question In-Reply-To: References: Message-ID: Last I saw, if you load/reload mod_nibblebill, it logs which DSN it's using to fs_cli / freeswitch.log. -Avi On Tue, Feb 12, 2013 at 9:00 PM, David Villasmil < david.villasmil.work at gmail.com> wrote: > Yes, it does have the option, but it doesn't seem to be using a different > one... I must've missed something... > > > On Tue, Feb 12, 2013 at 7:53 PM, Avi Marcus wrote: > >> All modules that touch the DB (I think!) have their own configuration >> that allows you pick the database. >> mod_nibblebill does indeed has a param name="db_dsn" option so yes, it >> will honor that. >> >> -Avi >> >> On Tue, Feb 12, 2013 at 8:11 PM, David Villasmil < >> david.villasmil.work at gmail.com> wrote: >> >>> Hello guys, >>> >>> I have created 2 dsns, one for freeswitch core "freeswitch" and one for >>> nibblebill ("whatever"). >>> Is it possible to use different dsns in core and nibblebill? or does >>> nibblebill uses the same dsn as the core?? >>> >>> thanks >>> >>> David >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130212/d7da7e4a/attachment-0001.html From vbvbrj at gmail.com Tue Feb 12 22:16:37 2013 From: vbvbrj at gmail.com (Mimiko) Date: Tue, 12 Feb 2013 21:16:37 +0200 Subject: [Freeswitch-users] Video conference Message-ID: <511A9515.5020103@gmail.com> Hi. What is the minimum requirements for a server to handle video conference with 40+ users on BigBlueButton software? Or is there other solutions which integrates with FS? -- Mimiko desu. From mike at jerris.com Tue Feb 12 22:18:53 2013 From: mike at jerris.com (Michael Jerris) Date: Tue, 12 Feb 2013 14:18:53 -0500 Subject: [Freeswitch-users] (no subject) In-Reply-To: <511A7F04.9010704@gmail.com> References: <1360647260499-7587257.post@n2.nabble.com> <511A7F04.9010704@gmail.com> Message-ID: I agree On Feb 12, 2013, at 12:42 PM, Joao Cesar S. Pereira wrote: > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 krice at freeswitch.org Tue Feb 12 22:22:19 2013 From: krice at freeswitch.org (Ken Rice) Date: Tue, 12 Feb 2013 13:22:19 -0600 Subject: [Freeswitch-users] Video conference In-Reply-To: <511A9515.5020103@gmail.com> Message-ID: That's more of a question for the BigBlueButton guys... On 2/12/13 1:16 PM, "Mimiko" wrote: > Hi. > > What is the minimum requirements for a server to handle video conference > with 40+ users on BigBlueButton software? Or is there other solutions > which integrates with FS? -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch From david.villasmil.work at gmail.com Tue Feb 12 22:23:40 2013 From: david.villasmil.work at gmail.com (David Villasmil) Date: Tue, 12 Feb 2013 20:23:40 +0100 Subject: [Freeswitch-users] Nibblebill database question In-Reply-To: References: Message-ID: No, there's nothing related to db: 2013-02-12 19:23:04.823909 [CONSOLE] switch_loadable_module.c:1348 Successfully Loaded [mod_nibblebill] 2013-02-12 19:23:04.823909 [NOTICE] switch_loadable_module.c:254 Adding Application 'nibblebill' 2013-02-12 19:23:04.823909 [NOTICE] switch_loadable_module.c:298 Adding API Function 'nibblebill' On Tue, Feb 12, 2013 at 8:04 PM, Avi Marcus wrote: > Last I saw, if you load/reload mod_nibblebill, it logs which DSN it's > using to fs_cli / freeswitch.log. > > -Avi > > On Tue, Feb 12, 2013 at 9:00 PM, David Villasmil < > david.villasmil.work at gmail.com> wrote: > >> Yes, it does have the option, but it doesn't seem to be using a different >> one... I must've missed something... >> >> >> On Tue, Feb 12, 2013 at 7:53 PM, Avi Marcus wrote: >> >>> All modules that touch the DB (I think!) have their own configuration >>> that allows you pick the database. >>> mod_nibblebill does indeed has a param name="db_dsn" option so yes, it >>> will honor that. >>> >>> -Avi >>> >>> On Tue, Feb 12, 2013 at 8:11 PM, David Villasmil < >>> david.villasmil.work at gmail.com> wrote: >>> >>>> Hello guys, >>>> >>>> I have created 2 dsns, one for freeswitch core "freeswitch" and one for >>>> nibblebill ("whatever"). >>>> Is it possible to use different dsns in core and nibblebill? or does >>>> nibblebill uses the same dsn as the core?? >>>> >>>> thanks >>>> >>>> David >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130212/37b63d1d/attachment.html From vbvbrj at gmail.com Tue Feb 12 22:31:13 2013 From: vbvbrj at gmail.com (Mimiko) Date: Tue, 12 Feb 2013 21:31:13 +0200 Subject: [Freeswitch-users] Video conference In-Reply-To: References: Message-ID: <511A9881.7070803@gmail.com> On 12.02.2013 21:22, Ken Rice wrote: > That's more of a question for the BigBlueButton guys... > > > On 2/12/13 1:16 PM, "Mimiko" wrote: > >> Hi. >> >> What is the minimum requirements for a server to handle video conference >> with 40+ users on BigBlueButton software? Or is there other solutions >> which integrates with FS? > Yes, I know. I asked in case if someone here used it. And maybe used other solution. :) -- Mimiko desu. From david.villasmil.work at gmail.com Tue Feb 12 22:33:25 2013 From: david.villasmil.work at gmail.com (David Villasmil) Date: Tue, 12 Feb 2013 20:33:25 +0100 Subject: [Freeswitch-users] Nibblebill database question In-Reply-To: References: Message-ID: Hello, Is uncommenting the "custom sql" lines mandatory?? it doesn't seem so, i do see nibblebill trying to do it's job: 2013-02-12 19:32:07.784156 [DEBUG] mod_nibblebill.c:351 Doing update query [UPDATE sip_users SET balance=balance-0.027683 WHERE id='1'] 2013-02-12 19:32:07.784156 [CRIT] mod_nibblebill.c:535 Failed to log to database! 2013-02-12 19:32:07.784156 [DEBUG] mod_nibblebill.c:378 Doing lookup query [SELECT balance AS nibble_balance FROM sip_users WHERE id='1'] 2013-02-12 19:32:07.784156 [ERR] mod_nibblebill.c:380 Error running this query: [SELECT balance AS nibble_balance FROM sip_users WHERE id='1'] On Tue, Feb 12, 2013 at 8:23 PM, David Villasmil < david.villasmil.work at gmail.com> wrote: > No, there's nothing related to db: > > 2013-02-12 19:23:04.823909 [CONSOLE] switch_loadable_module.c:1348 > Successfully Loaded [mod_nibblebill] > 2013-02-12 19:23:04.823909 [NOTICE] switch_loadable_module.c:254 Adding > Application 'nibblebill' > 2013-02-12 19:23:04.823909 [NOTICE] switch_loadable_module.c:298 Adding > API Function 'nibblebill' > > > > > On Tue, Feb 12, 2013 at 8:04 PM, Avi Marcus wrote: > >> Last I saw, if you load/reload mod_nibblebill, it logs which DSN it's >> using to fs_cli / freeswitch.log. >> >> -Avi >> >> On Tue, Feb 12, 2013 at 9:00 PM, David Villasmil < >> david.villasmil.work at gmail.com> wrote: >> >>> Yes, it does have the option, but it doesn't seem to be using a >>> different one... I must've missed something... >>> >>> >>> On Tue, Feb 12, 2013 at 7:53 PM, Avi Marcus wrote: >>> >>>> All modules that touch the DB (I think!) have their own configuration >>>> that allows you pick the database. >>>> mod_nibblebill does indeed has a param name="db_dsn" option so yes, it >>>> will honor that. >>>> >>>> -Avi >>>> >>>> On Tue, Feb 12, 2013 at 8:11 PM, David Villasmil < >>>> david.villasmil.work at gmail.com> wrote: >>>> >>>>> Hello guys, >>>>> >>>>> I have created 2 dsns, one for freeswitch core "freeswitch" and one >>>>> for nibblebill ("whatever"). >>>>> Is it possible to use different dsns in core and nibblebill? or does >>>>> nibblebill uses the same dsn as the core?? >>>>> >>>>> thanks >>>>> >>>>> David >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> 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 >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130212/de4064e6/attachment-0001.html From gozdal at gmail.com Tue Feb 12 23:05:50 2013 From: gozdal at gmail.com (Marcin Gozdalik) Date: Tue, 12 Feb 2013 21:05:50 +0100 Subject: [Freeswitch-users] High Availability Cluster Module for FreeSWITCH In-Reply-To: References: Message-ID: 2013/2/12 Michael Collins : > >> building business around FS ;) Jokes aside - if somebody wants to >> configure FS in HA evironment I don't think it is much of an obstacle >> for him to configure Corosync as already he has to have some kind of >> DB failover as well, redundant switches, power, etc. > > > I would like to see that statement put to the test. I would like to see > someone attempt to put Corosync (or whatever existing solution) to use as > the HA engine in a FS cluster. I suspect the amount of work involved is way > more than has been suggested. If it really were that easy ("easy" being > relative compared to what people perceive Eliot is doing) then I suspect it > would have been done already. If there was a readily available resource agent for FS it is "easy" to setup Corosync (if you know hat you are doing). I didn't setup Corosync to handle FS, I have OpenSIPS as SBC before FS which handles HA. Of course it is not ideal, as in case of a problem current calls are lost. From drk at drkngs.net Tue Feb 12 23:21:28 2013 From: drk at drkngs.net (Dave R. Kompel) Date: Tue, 12 Feb 2013 12:21:28 -0800 Subject: [Freeswitch-users] =?iso-8859-1?q?High_Availability_Cluster_Modul?= =?iso-8859-1?q?e_for=09FreeSWITCH?= In-Reply-To: Message-ID: <20130212202128.3c0aaa08@mail.tritonwest.net> Tony, I'm glad you're thinking this way. I've been looking into this myself, and I think we both got the same idea. I've been playing around with how to hook up the eventing system to external Service Bus implmentations as a starting point. Is there a chance we could all plan a meeting on the FS conference (out of band from the wednsday meetings) to maybe discuss this with the interested ppl? --Dave _____ From: Anthony Minessale [mailto:anthony.minessale at gmail.com] To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] Sent: Mon, 11 Feb 2013 13:30:25 -0800 Subject: Re: [Freeswitch-users] High Availability Cluster Module for FreeSWITCH I believe some more planning and thought needs to go into this before proceeding with code. I recommend a continued discussion. My point of view is we should specifically review the way FS does the message exchange since its the cornerstone of many other possible cluster scenarios besides just fault tolerance. Some are starting to build this externally and I think the real answer is a more clean abstraction between the framework for many FS communicating with each other and the business logic. So some of what people like plivo and 2600hz do externally would be better served as part of FS at the comms level and then still separate the logic. That way everyone can benefit from the low level code. On Mon, Feb 11, 2013 at 12:37 PM, Henry Huang wrote: I would choose to go with Elliot's idea of having the HA being build within the FS itself under the condition that it's a easy adaptation comparing to having to use and struggling with Pacemaker and Corosync. They can literally take months if someone is starting from scratch. And it would be nice to have this kind of capability to go against those brand name solutions. On Mon, Feb 11, 2013 at 6:06 AM, Eliot Gable wrote: On Mon, Feb 11, 2013 at 7:36 AM, Marcin Gozdalik wrote: +1 I do not doubt mod_ha is necessary inside of FS and it may be better/simpler than writing Pacemaker resource agent, but writing yet-another-cluster-communication-engine is IMHO the wrong way to go and using Corosync for communication will give a lot of value from mature codebase. I understand what you are saying, but what I am trying to get across is that I am not writing yet-another-cluster-communication-engine. All I am really doing is combining a multicast messaging API written by Tony and the event API in FS to broadcast existing state information between multiple FS nodes, as well as adding a tiny amount of logic on top of that to coordinate call fail over and recovery. That's probably a little over-simplified, but it gets the point across. The network communication code is already in FS and well tested. The event system is already in FS and well tested. I have already written the code to the point that it parses the configuration files and starts sending heartbeats out all of the interfaces configured. I have also already written a lot of the code that deals with the state transitions. All I am talking about doing is implementing a tiny little finite state machine. It's a pretty trivial programming task. In fact, I think it was covered in my first year at Carnegie Mellon University. Of course, I had already figured out how to write such things in high school, I just did not know what it was called at that point. My point is, that this is not yet-another-cluster-communication-engine. It is a very specific and small finite state machine designed solely with the goal in mind of making FS have just enough information to coordinate call fail over internally. If I recall correctly, a lot of people also said writing yet-another-VoIP-server was a waste of time, but now we have FreeSWITCH, and it was obviously worth the effort. And I am not even trying to do something as complex as that. If you think this is yet-another-cluster-communication-engine, you are missing the point. It is not. It never will be. Look at Sonus, Genband, Broadsoft, Veraz, etc. All the big-name carrier-grade telecom providers have a built-in solution for automatic call fail over. The only way FreeSWITCH will ever compete with such solutions is if it also has that feature. Pacemaker and Corosync are overkill just to get FS to handle single node failures and provide call recovery. It took me a full 3 months of working with them every day to really understand how to deploy them properly in conjunction with FreeSWITCH and Postgres to provide a carrier-grade hot-standby solution which was robust enough to handle 99% of the failures I could throw at it. Granted, this was back when the configuration still needed to be written by hand in XML and prior the existence of any resource agent for FreeSWITCH. But, even with those changes, deploying Pacemaker and Corosync is not a simple task. If that is the requirement for FS to have HA, it will never truly stand a chance against commercial offerings. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130212/99276ec9/attachment.html From dvl36.ripe.nick at gmail.com Tue Feb 12 23:24:16 2013 From: dvl36.ripe.nick at gmail.com (Dmitry Lysenko) Date: Tue, 12 Feb 2013 22:24:16 +0200 Subject: [Freeswitch-users] best freeswitch timer In-Reply-To: <1360685096.22409.768.camel@luna.madrid.commsmundi.com> References: <1360685096.22409.768.camel@luna.madrid.commsmundi.com> Message-ID: Hello, You should try each of timers on your system, as it depends of hardware,kernel version and kernel config too. On my ARM (Marvell Kirkwood at 1.2Ghz) with 3.2.38 PREEMPT RT kernel and "freeswitch -rt" best results, according to "timer_test 20 200", I get using mod_timerfd (+/- 1-3mks with rare spikes of 10-20mks), posix timers slightly worse. Soft timer working not very precise on my system. Sometimes it give periodic spikes of ~=150-200mks. I tried(hard enough) to tune softtimer with different switch.conf timing options, but this options is not well documented, so without success. P.S. Sorry for my English. 2013/2/12 Fran?ois Delawarde > Hello, > > Could someone knowing the different FS timers clarify the difference > between those: > > - default core softtimer > - core heavy timer (does this option still exists?) > - posix timer > - timerfd > > Can any of those provide a significant performance improvement, or is it > just not worth switching from the default? > > In any case, which one would be the recommended/best one under Linux > with a relatively recent kernel (>3.0)? > > Thanks, > Fran?ois. > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130212/232edc04/attachment.html From Chad.Engler at patlive.com Tue Feb 12 23:42:52 2013 From: Chad.Engler at patlive.com (Chad Engler) Date: Tue, 12 Feb 2013 15:42:52 -0500 Subject: [Freeswitch-users] mod_pocketsphinx dialplan example Message-ID: I'm looking for a good example of doing speech detection in a dialplan, and I can't find one. Right now I am trying: But after enabling speech detection it just hangs up. How can I let the caller respond and collect/use the result from a dialplan? Thanks, Chad Engler Telephony Programmer PATLive 1.800.775.7790 x746 chad.engler at patlive.com Hosted Communications | Friendly Service www.patlive.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130212/aefb7fb2/attachment-0001.html From drk at drkngs.net Tue Feb 12 23:42:51 2013 From: drk at drkngs.net (Dave R. Kompel) Date: Tue, 12 Feb 2013 12:42:51 -0800 Subject: [Freeswitch-users] FreeSWITCH Message Bus / Shared Key Value Store In-Reply-To: Message-ID: <20130212204251.7066d944@mail.tritonwest.net> I've done a few experments with using both Redis, and the evil "Microsoft Azure Service bus" (the server on prem based version) to extend the eventing system to have global PUB/SUB. This way things like registrations, and Limit stuff could be made global. I'm looking for a way, in my carrier switch implmentation, to implment both HA Failover and Scaleout clustering. --Dave _____ From: Eliot Gable [mailto:egable+freeswitch at gmail.com] To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] Sent: Tue, 12 Feb 2013 05:49:13 -0800 Subject: [Freeswitch-users] FreeSWITCH Message Bus / Shared Key Value Store Tony and Mike and I had a discussion last night about FreeSWITCH with regards to implementing some form of core message bus or shared key-value store. We discussed a few different options, but did not really settle on anything. If you are writing modules or using FreeSWITCH in a multi-node setting, please share what features / functionality you would like to see implemented in this regard, how you would use it, and why you want to see the specific mechanism of your choice rather than some alternative. Also, please consider and mention whether "cluster awareness" is something that factors into your use case. By this, I mean having each FS node have some idea about the state / status of each other node in terms of taking calls vs acting as a standby or slave node, etc. -- Eliot Gable -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130212/76c73513/attachment.html From sdevoy at bizfocused.com Tue Feb 12 23:44:29 2013 From: sdevoy at bizfocused.com (Sean Devoy) Date: Tue, 12 Feb 2013 15:44:29 -0500 Subject: [Freeswitch-users] new site, cant call OUT! Message-ID: <076e01ce0961$c14511d0$43cf3570$@bizfocused.com> Hi All, (Git head build last week on centos 5.n) This is a new one for me. Calls IN to the new customer with Cisco phones behind NAT are working, but not outbound! I have not been to the site yet. I am told that they can call out once or twice then have to reboot the phone to dial out. Dialing in continues to work! one person says the phone says "Call merged" another says it say "Request merged" on the phone display! I did see in the logs, that when it works, we get: 2013-02-12 11:09:26.383576 [NOTICE] switch_channel.c:968 New Channel sofia/external/105@ [51884b2d-6aa2-47fe-b687-8604a4e39a7a] 2013-02-12 11:09:26.383576 [DEBUG] switch_core_session.c:975 Send signal sofia/external/105@ [BREAK] 2013-02-12 11:09:26.383576 [DEBUG] switch_core_session.c:975 Send signal sofia/external/105@ [BREAK] 2013-02-12 11:09:26.383576 [DEBUG] switch_core_state_machine.c:415 (sofia/external/105@) Running State Change CS_NEW 2013-02-12 11:09:26.383576 [DEBUG] switch_core_state_machine.c:433 (sofia/external/105@) State NEW 2013-02-12 11:09:26.413581 [DEBUG] switch_core_session.c:975 Send signal sofia/external/105@ [BREAK] 2013-02-12 11:09:26.413581 [DEBUG] sofia.c:1750 detaching session 51884b2d-6aa2-47fe-b687-8604a4e39a7a 2013-02-12 11:09:26.413581 [WARNING] sofia_reg.c:1502 SIP auth challenge (INVITE) on sofia profile 'external' for [3014040914@] from ip 2013-02-12 11:09:26.493574 [DEBUG] sofia.c:1842 Re-attaching to session 51884b2d-6aa2-47fe-b687-8604a4e39a7a And when it fails: 2013-02-12 13:32:26.363343 [NOTICE] switch_channel.c:968 New Channel sofia/external/220@ [91e9eb64-253c-4604-bdfa-4dd18baae18d] 2013-02-12 13:32:26.363343 [DEBUG] switch_core_session.c:975 Send signal sofia/external/220@ [BREAK] 2013-02-12 13:32:26.363343 [DEBUG] switch_core_session.c:975 Send signal sofia/external/220@ [BREAK] 2013-02-12 13:32:26.363343 [DEBUG] switch_core_state_machine.c:415 (sofia/external/220@) Running State Change CS_NEW 2013-02-12 13:32:26.363343 [DEBUG] switch_core_state_machine.c:433 (sofia/external/220@) State NEW 2013-02-12 13:32:26.383344 [DEBUG] switch_core_session.c:975 Send signal sofia/external/220@ [BREAK] 2013-02-12 13:32:26.383344 [DEBUG] sofia.c:1750 detaching session 91e9eb64-253c-4604-bdfa-4dd18baae18d 2013-02-12 13:32:26.383344 [WARNING] sofia_reg.c:1502 SIP auth challenge (INVITE) on sofia profile 'external' for [4104934408@] from ip 2 2013-02-12 13:32:36.363085 [WARNING] switch_core_state_machine.c:514 91e9eb64-253c-4604-bdfa-4dd18baae18d sofia/external/220@ Abandoned 2013-02-12 13:32:36.363085 [DEBUG] switch_channel.c:2994 (sofia/external/220@) Callstate Change DOWN -> HANGUP Can anyone shed light on what's happening "at that particular juncture" and how I may go about debugging it? I was thinking FS was not finding a connection through NAT to phone, but then why do incoming calls work? Thanks in advance, Sean -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130212/716149c7/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 70 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130212/716149c7/attachment.gif From avi at avimarcus.net Tue Feb 12 23:47:42 2013 From: avi at avimarcus.net (Avi Marcus) Date: Tue, 12 Feb 2013 22:47:42 +0200 Subject: [Freeswitch-users] mod_pocketsphinx dialplan example In-Reply-To: References: Message-ID: http://wiki.freeswitch.org/wiki/Mod_pocketsphinx describes the Pizza Demo a bit. -Avi On Tue, Feb 12, 2013 at 10:42 PM, Chad Engler wrote: > I'm looking for a good example of doing speech detection in a dialplan, > and I can't find one.**** > > ** ** > > Right now I am trying:**** > > ** ** > > **** > > **** > > **** > > expression="true">**** > > **** > > **** > > > **** > > **** > > **** > > **** > > **** > > ** ** > > But after enabling speech detection it just hangs up. How can I let the > caller respond and collect/use the result from a dialplan?**** > > ** ** > > Thanks,**** > > ** ** > > *Chad Engler***** > > Telephony Programmer**** > > PATLive**** > > 1.800.775.7790 x746**** > > chad.engler at patlive.com**** > > ** ** > > Hosted Communications | Friendly Service www.patlive.com**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130212/f1338cb5/attachment-0001.html From drk at drkngs.net Tue Feb 12 23:51:44 2013 From: drk at drkngs.net (Dave R. Kompel) Date: Tue, 12 Feb 2013 12:51:44 -0800 Subject: [Freeswitch-users] mod_pocketsphinx dialplan example In-Reply-To: Message-ID: <20130212205144.da8a0190@mail.tritonwest.net> You can't do it all in XML, you need to use scripting or a module. You need to respond to the input events for the session. Look at the Pizza example. --Dave _____ From: Chad Engler [mailto:Chad.Engler at patlive.com] To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] Sent: Tue, 12 Feb 2013 12:42:52 -0800 Subject: [Freeswitch-users] mod_pocketsphinx dialplan example I'm looking for a good example of doing speech detection in a dialplan, and I can't find one. Right now I am trying: But after enabling speech detection it just hangs up. How can I let the caller respond and collect/use the result from a dialplan? Thanks, Chad Engler Telephony Programmer PATLive 1.800.775.7790 x746 chad.engler at patlive.com Hosted Communications | Friendly Service www.patlive.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130212/332743e6/attachment.html From krice at freeswitch.org Tue Feb 12 23:57:37 2013 From: krice at freeswitch.org (Ken Rice) Date: Tue, 12 Feb 2013 14:57:37 -0600 Subject: [Freeswitch-users] new site, cant call OUT! In-Reply-To: <076e01ce0961$c14511d0$43cf3570$@bizfocused.com> Message-ID: Sounds like a possible NAT issue... But the only way to know that is look at a sip trace and see whats going on that way and maybe even looking from the phones perspective On 2/12/13 2:44 PM, "Sean Devoy" wrote: > > Hi All, > > (Git head build last week on centos 5.n) > > This is a new one for me. Calls IN to the new customer with Cisco phones > behind NAT are working, but not outbound! > > I have not been to the site yet. I am told that they can call out once or > twice then have to reboot the phone to dial out. Dialing in continues to > work! one person says the phone says ?Call merged? another says it say > ?Request merged? on the phone display! > > I did see in the logs, that when it works, we get: > 2013-02-12 11:09:26.383576 [NOTICE] switch_channel.c:968 New Channel > sofia/external/105@ [51884b2d-6aa2-47fe-b687-8604a4e39a7a] > 2013-02-12 11:09:26.383576 [DEBUG] switch_core_session.c:975 Send signal > sofia/external/105@ [BREAK] > 2013-02-12 11:09:26.383576 [DEBUG] switch_core_session.c:975 Send signal > sofia/external/105@ [BREAK] > 2013-02-12 11:09:26.383576 [DEBUG] switch_core_state_machine.c:415 > (sofia/external/105@) Running State Change CS_NEW > 2013-02-12 11:09:26.383576 [DEBUG] switch_core_state_machine.c:433 > (sofia/external/105@) State NEW > 2013-02-12 11:09:26.413581 [DEBUG] switch_core_session.c:975 Send signal > sofia/external/105@ [BREAK] > 2013-02-12 11:09:26.413581 [DEBUG] sofia.c:1750 detaching session > 51884b2d-6aa2-47fe-b687-8604a4e39a7a > 2013-02-12 11:09:26.413581 [WARNING] sofia_reg.c:1502 SIP auth challenge > (INVITE) on sofia profile 'external' for [3014040914@] from > ip > 2013-02-12 11:09:26.493574 [DEBUG] sofia.c:1842 Re-attaching to session > 51884b2d-6aa2-47fe-b687-8604a4e39a7a > > And when it fails: > 2013-02-12 13:32:26.363343 [NOTICE] switch_channel.c:968 New Channel > sofia/external/220@ [91e9eb64-253c-4604-bdfa-4dd18baae18d] > 2013-02-12 13:32:26.363343 [DEBUG] switch_core_session.c:975 Send signal > sofia/external/220@ [BREAK] > 2013-02-12 13:32:26.363343 [DEBUG] switch_core_session.c:975 Send signal > sofia/external/220@ [BREAK] > 2013-02-12 13:32:26.363343 [DEBUG] switch_core_state_machine.c:415 > (sofia/external/220@) Running State Change CS_NEW > 2013-02-12 13:32:26.363343 [DEBUG] switch_core_state_machine.c:433 > (sofia/external/220@) State NEW > 2013-02-12 13:32:26.383344 [DEBUG] switch_core_session.c:975 Send signal > sofia/external/220@ [BREAK] > 2013-02-12 13:32:26.383344 [DEBUG] sofia.c:1750 detaching session > 91e9eb64-253c-4604-bdfa-4dd18baae18d > 2013-02-12 13:32:26.383344 [WARNING] sofia_reg.c:1502 SIP auth challenge > (INVITE) on sofia profile 'external' for [4104934408@] from > ip 2 > 2013-02-12 13:32:36.363085 [WARNING] switch_core_state_machine.c:514 > 91e9eb64-253c-4604-bdfa-4dd18baae18d sofia/external/220@ > Abandoned > 2013-02-12 13:32:36.363085 [DEBUG] switch_channel.c:2994 > (sofia/external/220@) Callstate Change DOWN -> HANGUP > > Can anyone shed light on what?s happening ?at that particular juncture? and > how I may go about debugging it? > > I was thinking FS was not finding a connection through NAT to phone, but then > why do incoming calls work? > > Thanks in advance, > Sean > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130212/e349b44d/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 70 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130212/e349b44d/attachment.gif From Chad.Engler at patlive.com Wed Feb 13 00:02:36 2013 From: Chad.Engler at patlive.com (Chad Engler) Date: Tue, 12 Feb 2013 16:02:36 -0500 Subject: [Freeswitch-users] mod_pocketsphinx dialplan example In-Reply-To: <20130212205144.da8a0190@mail.tritonwest.net> References: <20130212205144.da8a0190@mail.tritonwest.net> Message-ID: I saw the pizza example, and I understand how to do it through ESL; just was wondering how to do it with pure XML. If you can't that is fine I can use ESL, thanks for the help. -Chad From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Dave R. Kompel Sent: Tuesday, February 12, 2013 3:52 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] mod_pocketsphinx dialplan example You can't do it all in XML, you need to use scripting or a module. You need to respond to the input events for the session. Look at the Pizza example. --Dave ________________________________ From: Chad Engler [mailto:Chad.Engler at patlive.com] To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] Sent: Tue, 12 Feb 2013 12:42:52 -0800 Subject: [Freeswitch-users] mod_pocketsphinx dialplan example I'm looking for a good example of doing speech detection in a dialplan, and I can't find one. Right now I am trying: But after enabling speech detection it just hangs up. How can I let the caller respond and collect/use the result from a dialplan? Thanks, Chad Engler Telephony Programmer PATLive 1.800.775.7790 x746 chad.engler at patlive.com Hosted Communications | Friendly Service www.patlive.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130212/5c1eee59/attachment-0001.html From msc at freeswitch.org Wed Feb 13 00:18:24 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 12 Feb 2013 13:18:24 -0800 Subject: [Freeswitch-users] High Availability Cluster Module for FreeSWITCH In-Reply-To: References: Message-ID: > If there was a readily available resource agent for FS it is "easy" to > setup Corosync (if you know hat you are doing). I didn't setup > Corosync to handle FS, I have OpenSIPS as SBC before FS which handles > HA. Of course it is not ideal, as in case of a problem current calls > are lost. > Which is precisely the point - this is *HIGH* availability we're talking about, not "kinda quickly recovering from a problem but it's okay to lose the calls that were in progress as long as our VoIP comes back up within 5-10 seconds." We're talking about the Holy Grail of VoIP here - a fault-tolerant cluster of VoIP servers that can keep calls up and running even if the primary server goes down. Note that "primary server goes down" includes literally hundreds of scenarios from physical (power, network cable unplugged, HDD crash, etc.) to logical (user misconfiguration of FreeSWITCH or LAN settings, router/firewall changes, service provider goofups, FreeSWITCH core dump, a single sofia profile locked up but other FS is okay, etc.) I'll bet Eliot can name dozens of them off the top of his head. ;) I have a sneaking suspicion that not everyone in this thread is thinking the same thing when we say "HA". Eliot's proposed solution encompasses fault scenarios that some of us haven't even dreamed of. I am open to someone getting P + C to do everything that Eliot's mod_ha_cluster can do but I suspect it isn't realistic. If you can prove me wrong on that then please by all means do so. -MC -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130212/9cacd032/attachment.html From msc at freeswitch.org Wed Feb 13 00:19:40 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 12 Feb 2013 13:19:40 -0800 Subject: [Freeswitch-users] High Availability Cluster Module for FreeSWITCH In-Reply-To: <20130212202128.3c0aaa08@mail.tritonwest.net> References: <20130212202128.3c0aaa08@mail.tritonwest.net> Message-ID: Yes, on the meeting tomorrow we'll gauge the interest level of the participants and then you can figure out when to meet. Feel free to use the public conf server outside of our normally scheduled times. -MC On Tue, Feb 12, 2013 at 12:21 PM, Dave R. Kompel wrote: > ** > Tony, > > I'm glad you're thinking this way. I've been looking into this myself, and > I think we both got the same idea. I've been playing around with how to > hook up the eventing system to external Service Bus implmentations as a > starting point. Is there a chance we could all plan a meeting on the FS > conference (out of band from the wednsday meetings) to maybe discuss this > with the interested ppl? > > --Dave > > ------------------------------ > *From:* Anthony Minessale [mailto:anthony.minessale at gmail.com] > *To:* FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] > *Sent:* Mon, 11 Feb 2013 13:30:25 -0800 > *Subject:* Re: [Freeswitch-users] High Availability Cluster Module for > FreeSWITCH > > > I believe some more planning and thought needs to go into this before > proceeding with code. > I recommend a continued discussion. My point of view is we should > specifically review the way FS does the message exchange since its the > cornerstone of many other possible cluster scenarios besides > just fault tolerance. Some are starting to build this externally and I > think the real answer is a more clean abstraction between the framework for > many FS communicating with each other and the business logic. So some of > what people like plivo and 2600hz do externally would be better served as > part of FS at the comms level and then still separate the logic. That way > everyone can benefit from the low level code. > > > > > On Mon, Feb 11, 2013 at 12:37 PM, Henry Huang wrote: > >> I would choose to go with Elliot's idea of having the HA being build >> within the FS itself under the condition that it's a easy adaptation >> comparing to having to use and struggling with Pacemaker and Corosync. They >> can literally take months if someone is starting from scratch. >> >> And it would be nice to have this kind of capability to go against those >> brand name solutions. >> >> >> On Mon, Feb 11, 2013 at 6:06 AM, Eliot Gable < >> egable+freeswitch at gmail.com> wrote: >> >>> On Mon, Feb 11, 2013 at 7:36 AM, Marcin Gozdalik wrote: >>> >>>> +1 >>>> >>>> I do not doubt mod_ha is necessary inside of FS and it may be >>>> better/simpler than writing Pacemaker resource agent, but writing >>>> yet-another-cluster-communication-engine is IMHO the wrong way to go >>>> and using Corosync for communication will give a lot of value from >>>> mature codebase. >>>> >>>> >>> I understand what you are saying, but what I am trying to get across is >>> that I am not writing yet-another-cluster-communication-engine. All I am >>> really doing is combining a multicast messaging API written by Tony and the >>> event API in FS to broadcast existing state information between multiple FS >>> nodes, as well as adding a tiny amount of logic on top of that to >>> coordinate call fail over and recovery. That's probably a little >>> over-simplified, but it gets the point across. The network communication >>> code is already in FS and well tested. The event system is already in FS >>> and well tested. I have already written the code to the point that it >>> parses the configuration files and starts sending heartbeats out all of the >>> interfaces configured. I have also already written a lot of the code that >>> deals with the state transitions. All I am talking about doing is >>> implementing a tiny little finite state machine. It's a pretty trivial >>> programming task. In fact, I think it was covered in my first year at >>> Carnegie Mellon University. Of course, I had already figured out how to >>> write such things in high school, I just did not know what it was called at >>> that point. My point is, that this is not >>> yet-another-cluster-communication-engine. It is a very specific and small >>> finite state machine designed solely with the goal in mind of making FS >>> have just enough information to coordinate call fail over internally. If I >>> recall correctly, a lot of people also said writing yet-another-VoIP-server >>> was a waste of time, but now we have FreeSWITCH, and it was obviously worth >>> the effort. And I am not even trying to do something as complex as that. If >>> you think this is yet-another-cluster-communication-engine, you are missing >>> the point. It is not. It never will be. >>> >>> Look at Sonus, Genband, Broadsoft, Veraz, etc. All the big-name >>> carrier-grade telecom providers have a built-in solution for automatic call >>> fail over. The only way FreeSWITCH will ever compete with such solutions is >>> if it also has that feature. Pacemaker and Corosync are overkill just to >>> get FS to handle single node failures and provide call recovery. It took me >>> a full 3 months of working with them every day to really understand how to >>> deploy them properly in conjunction with FreeSWITCH and Postgres to provide >>> a carrier-grade hot-standby solution which was robust enough to handle 99% >>> of the failures I could throw at it. Granted, this was back when the >>> configuration still needed to be written by hand in XML and prior the >>> existence of any resource agent for FreeSWITCH. But, even with those >>> changes, deploying Pacemaker and Corosync is not a simple task. If that is >>> the requirement for FS to have HA, it will never truly stand a chance >>> against commercial offerings. >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130212/7c4fc7b0/attachment-0001.html From msc at freeswitch.org Wed Feb 13 00:33:44 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 12 Feb 2013 13:33:44 -0800 Subject: [Freeswitch-users] Video conference In-Reply-To: <511A9881.7070803@gmail.com> References: <511A9881.7070803@gmail.com> Message-ID: BBB is actually pretty awesome. It does have a lot of stuff going on underneath the hood besides FreeSWITCH. I would recommend a quad-core at an absolute minimum with at least 16GB RAM. (Hey RAM is cheap so stock up!) If you're going to be using the screen sharing stuff in BBB then beef up the CPU power - more cores and faster processors. The BBB guys can definitely give you more details, but this should get you going. On a personal note I just want to say that after talking to the BBB guys and seeing what all they've done I am really impressed. They've glued together a lot of open source giblets to make a really cool product and they make it OSS to boot. (You can also get premium support, etc. if you need it.) Have fun! -MC On Tue, Feb 12, 2013 at 11:31 AM, Mimiko wrote: > On 12.02.2013 21:22, Ken Rice wrote: > > That's more of a question for the BigBlueButton guys... > > > > > > On 2/12/13 1:16 PM, "Mimiko" wrote: > > > >> Hi. > >> > >> What is the minimum requirements for a server to handle video conference > >> with 40+ users on BigBlueButton software? Or is there other solutions > >> which integrates with FS? > > > > Yes, I know. I asked in case if someone here used it. And maybe used > other solution. :) > > -- > Mimiko desu. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130212/71b08c92/attachment.html From gozdal at gmail.com Wed Feb 13 00:33:43 2013 From: gozdal at gmail.com (Marcin Gozdalik) Date: Tue, 12 Feb 2013 22:33:43 +0100 Subject: [Freeswitch-users] High Availability Cluster Module for FreeSWITCH In-Reply-To: References: Message-ID: 2013/2/12 Michael Collins : > >> If there was a readily available resource agent for FS it is "easy" to >> setup Corosync (if you know hat you are doing). I didn't setup >> Corosync to handle FS, I have OpenSIPS as SBC before FS which handles >> HA. Of course it is not ideal, as in case of a problem current calls >> are lost. > > > Which is precisely the point - this is *HIGH* availability we're talking > about, not "kinda quickly recovering from a problem but it's okay to lose > the calls that were in progress as long as our VoIP comes back up within > 5-10 seconds." > > We're talking about the Holy Grail of VoIP here - a fault-tolerant cluster > of VoIP servers that can keep calls up and running even if the primary > server goes down. Note that "primary server goes down" includes literally > hundreds of scenarios from physical (power, network cable unplugged, HDD > crash, etc.) to logical (user misconfiguration of FreeSWITCH or LAN > settings, router/firewall changes, service provider goofups, FreeSWITCH core > dump, a single sofia profile locked up but other FS is okay, etc.) I'll bet > Eliot can name dozens of them off the top of his head. ;) > > I have a sneaking suspicion that not everyone in this thread is thinking the > same thing when we say "HA". Eliot's proposed solution encompasses fault > scenarios that some of us haven't even dreamed of. I am open to someone > getting P + C to do everything that Eliot's mod_ha_cluster can do but I > suspect it isn't realistic. If you can prove me wrong on that then please by > all means do so. The Holy Grail you're describing is probably impossible to achieve (even Google sometimes serves you 500, just press reload to be routed to another server :). Maybe this discussion could continue to prepare some kind of a blueprint describing the predicted failure scenarios and how the proposed solution would deal with them? There are several solutions best suited for different classes of availability you're willing to tolerate. -- Marcin Gozdalik From msc at freeswitch.org Wed Feb 13 00:37:30 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 12 Feb 2013 13:37:30 -0800 Subject: [Freeswitch-users] Different music for Hold and Transfer (internal SIP-functions) In-Reply-To: References: Message-ID: I believe you can use transfer_ringback variable to set different sound/MOH for when an answered call is transferred. The regular MOH will still be the same. -MC On Fri, Feb 8, 2013 at 7:18 AM, Michael Huber wrote: > I am using some regular sip-phones and the internal sip-functions for > HOLD and TRANSFER. > > I do NOT use the applications ?transfer/att_xfer? and ?soft_hold?. The > phone is a ?Siemens OpenStage 60 G?, but this fact should not be important. > > My problem is now that I can?t have different music for transfer and hold. > If I would use the freeswitch-applications ?transfer/att_xfer? and > ?soft_hold? then I would be able to set the variables ?hold_music? and > ?temp_hold_music?. > > Any idea to achieve this? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130212/58244515/attachment.html From ben at langfeld.co.uk Wed Feb 13 01:02:07 2013 From: ben at langfeld.co.uk (Ben Langfeld) Date: Tue, 12 Feb 2013 22:02:07 +0000 Subject: [Freeswitch-users] High Availability Cluster Module for FreeSWITCH In-Reply-To: References: Message-ID: Marcin, have you used commercial platforms that do this kind of live-call failover? They work very well. "impossible", certainly not. Regards, Ben Langfeld On 12 February 2013 21:33, Marcin Gozdalik wrote: > 2013/2/12 Michael Collins : > > > >> If there was a readily available resource agent for FS it is "easy" to > >> setup Corosync (if you know hat you are doing). I didn't setup > >> Corosync to handle FS, I have OpenSIPS as SBC before FS which handles > >> HA. Of course it is not ideal, as in case of a problem current calls > >> are lost. > > > > > > Which is precisely the point - this is *HIGH* availability we're talking > > about, not "kinda quickly recovering from a problem but it's okay to lose > > the calls that were in progress as long as our VoIP comes back up within > > 5-10 seconds." > > > > We're talking about the Holy Grail of VoIP here - a fault-tolerant > cluster > > of VoIP servers that can keep calls up and running even if the primary > > server goes down. Note that "primary server goes down" includes literally > > hundreds of scenarios from physical (power, network cable unplugged, HDD > > crash, etc.) to logical (user misconfiguration of FreeSWITCH or LAN > > settings, router/firewall changes, service provider goofups, FreeSWITCH > core > > dump, a single sofia profile locked up but other FS is okay, etc.) I'll > bet > > Eliot can name dozens of them off the top of his head. ;) > > > > I have a sneaking suspicion that not everyone in this thread is thinking > the > > same thing when we say "HA". Eliot's proposed solution encompasses fault > > scenarios that some of us haven't even dreamed of. I am open to someone > > getting P + C to do everything that Eliot's mod_ha_cluster can do but I > > suspect it isn't realistic. If you can prove me wrong on that then > please by > > all means do so. > > The Holy Grail you're describing is probably impossible to achieve > (even Google sometimes serves you 500, just press reload to be routed > to another server :). > Maybe this discussion could continue to prepare some kind of a > blueprint describing the predicted failure scenarios and how the > proposed solution would deal with them? There are several solutions > best suited for different classes of availability you're willing to > tolerate. > > > -- > Marcin Gozdalik > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130212/47729c44/attachment-0001.html From ritzalam at gmail.com Wed Feb 13 01:10:41 2013 From: ritzalam at gmail.com (Richard Alam) Date: Tue, 12 Feb 2013 17:10:41 -0500 Subject: [Freeswitch-users] Video conference In-Reply-To: References: <511A9881.7070803@gmail.com> Message-ID: On Tue, Feb 12, 2013 at 4:33 PM, Michael Collins wrote: > BBB is actually pretty awesome. It does have a lot of stuff going on > underneath the hood besides FreeSWITCH. I would recommend a quad-core at an > absolute minimum with at least 16GB RAM. (Hey RAM is cheap so stock up!) If > you're going to be using the screen sharing stuff in BBB then beef up the > CPU power - more cores and faster processors. The BBB guys can definitely > give you more details, but this should get you going. > > On a personal note I just want to say that after talking to the BBB guys > and seeing what all they've done I am really impressed. They've glued > together a lot of open source giblets to make a really cool product and > they make it OSS to boot. (You can also get premium support, etc. if you > need it.) > > Have fun! > Many thanks for the kind words Michael. Without FreeSWITCH, I don't think we'll have as great success as we have now. Voice is the most important component in real-time collaboration systems and it's one of the part we don't worry about as FS is solid. Users can tolerate bad video (they just turn the webcam off) but if the audio hiccups, users start to complain. And being able to allow users to call in using the phone is a big plus. Mimiko, we don't use FS for video but only for audio. For video, we use Red5. We'll most like be experimenting with it soon. If you have more questions, you can try searching our FAQ at https://code.google.com/p/bigbluebutton/wiki/FAQ Or join our mailing list. Richad > > -MC > > On Tue, Feb 12, 2013 at 11:31 AM, Mimiko wrote: > >> On 12.02.2013 21:22, Ken Rice wrote: >> > That's more of a question for the BigBlueButton guys... >> > >> > >> > On 2/12/13 1:16 PM, "Mimiko" wrote: >> > >> >> Hi. >> >> >> >> What is the minimum requirements for a server to handle video >> conference >> >> with 40+ users on BigBlueButton software? Or is there other solutions >> >> which integrates with FS? >> > >> >> Yes, I know. I asked in case if someone here used it. And maybe used >> other solution. :) >> >> -- >> Mimiko desu. >> >> > > -- ----- BigBlueButton Developer http://www.bigbluebutton.org http://code.google.com/p/bigbluebutton -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130212/87dbd852/attachment.html From gozdal at gmail.com Wed Feb 13 01:21:00 2013 From: gozdal at gmail.com (Marcin Gozdalik) Date: Tue, 12 Feb 2013 23:21:00 +0100 Subject: [Freeswitch-users] High Availability Cluster Module for FreeSWITCH In-Reply-To: References: Message-ID: I did some, I got some anecdotes from people running them. It certainly is possible to handle *some* failure cases, but there are some *other* cases that will be not handled correctly. Personally I've observed OpenSIPS that run out of some internal memory and anyhow you wanted to monitor it it will reply it is alive. It will even route your simple call scenarios well! Unfortunately the "real" calls were *usually* timing out, but always. Sorry, but I can't imagine how you can automatically handle that, i.e. discover that this node is "bad" and fail-over to some other. Even if you do that correctly your monitoring tool can check that this node is operating perfectly well (because after all the traffic is diverted from the faulty node it begins to work well) and will want to move the traffic back. By "impossible" I mean handling all such gray areas. Certainly if you'll power down the box or unplug Ethernet it is possible to migrate live calls to some other box. I'd just like to have HA that works everytime and everywhere and try to poke some holes in reasoning in hope it'll help to reach better solution. 2013/2/12 Ben Langfeld : > Marcin, have you used commercial platforms that do this kind of live-call > failover? They work very well. "impossible", certainly not. > > Regards, > Ben Langfeld > > > On 12 February 2013 21:33, Marcin Gozdalik wrote: >> >> 2013/2/12 Michael Collins : >> > >> >> If there was a readily available resource agent for FS it is "easy" to >> >> setup Corosync (if you know hat you are doing). I didn't setup >> >> Corosync to handle FS, I have OpenSIPS as SBC before FS which handles >> >> HA. Of course it is not ideal, as in case of a problem current calls >> >> are lost. >> > >> > >> > Which is precisely the point - this is *HIGH* availability we're talking >> > about, not "kinda quickly recovering from a problem but it's okay to >> > lose >> > the calls that were in progress as long as our VoIP comes back up within >> > 5-10 seconds." >> > >> > We're talking about the Holy Grail of VoIP here - a fault-tolerant >> > cluster >> > of VoIP servers that can keep calls up and running even if the primary >> > server goes down. Note that "primary server goes down" includes >> > literally >> > hundreds of scenarios from physical (power, network cable unplugged, HDD >> > crash, etc.) to logical (user misconfiguration of FreeSWITCH or LAN >> > settings, router/firewall changes, service provider goofups, FreeSWITCH >> > core >> > dump, a single sofia profile locked up but other FS is okay, etc.) I'll >> > bet >> > Eliot can name dozens of them off the top of his head. ;) >> > >> > I have a sneaking suspicion that not everyone in this thread is thinking >> > the >> > same thing when we say "HA". Eliot's proposed solution encompasses fault >> > scenarios that some of us haven't even dreamed of. I am open to someone >> > getting P + C to do everything that Eliot's mod_ha_cluster can do but I >> > suspect it isn't realistic. If you can prove me wrong on that then >> > please by >> > all means do so. >> >> The Holy Grail you're describing is probably impossible to achieve >> (even Google sometimes serves you 500, just press reload to be routed >> to another server :). >> Maybe this discussion could continue to prepare some kind of a >> blueprint describing the predicted failure scenarios and how the >> proposed solution would deal with them? There are several solutions >> best suited for different classes of availability you're willing to >> tolerate. >> >> >> -- >> Marcin Gozdalik >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -- Marcin Gozdalik From jaybinks at gmail.com Wed Feb 13 01:48:44 2013 From: jaybinks at gmail.com (jay binks) Date: Wed, 13 Feb 2013 08:48:44 +1000 Subject: [Freeswitch-users] best freeswitch timer In-Reply-To: References: <1360685096.22409.768.camel@luna.madrid.commsmundi.com> Message-ID: so does : freeswitch at default> show timer type,name,ikey timer,soft,CORE_SOFTTIMER_MODULE 1 total. on FreeSWITCH Version 1.2.6+git~20130104T154559Z~a4247651ca (git a424765 2013-01-04 15:45:59Z) indicate it will be using timerfd calls or is this the old softtimer ? On 13 February 2013 02:21, Ken Rice wrote: > The default one is the best one on linux with kernels starting around > 2.6.32... Anything after that and you are good to go... > > The other timers are left around for other platforms > > K > > > > On 2/12/13 10:04 AM, "Fran?ois Delawarde" > wrote: > >> Hello, > > Could someone knowing the different FS timers clarify the >> difference > between those: > > - default core softtimer > - core heavy timer (does >> this option still exists?) > - posix timer > - timerfd > > Can any of those provide a >> significant performance improvement, or is it > just not worth switching from >> the default? > > In any case, which one would be the recommended/best one under >> Linux > with a relatively recent kernel >> (>3.0)? > > Thanks, > Fran?ois. > > > > ________________________________________________ >> _________________________ > Professional FreeSWITCH Consulting >> Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > FreeSW >> ITCH-powered IP PBX: The CudaTel Communication >> Server > > > Official FreeSWITCH >> Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon. >> com > > FreeSWITCH-users mailing >> list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman >> /listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/opt >> ions/freeswitch-users > http://www.freeswitch.org > > > -- > Ken > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > irc.freenode.net #freeswitch > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -- Sincerely Jay From dvl36.ripe.nick at gmail.com Wed Feb 13 02:03:54 2013 From: dvl36.ripe.nick at gmail.com (Dmitry Lysenko) Date: Wed, 13 Feb 2013 01:03:54 +0200 Subject: [Freeswitch-users] best freeswitch timer In-Reply-To: References: <1360685096.22409.768.camel@luna.madrid.commsmundi.com> Message-ID: Hello, That indicate that none of timer modules are loaded. Default "soft" timer used. If you want to try, you should load mod_timerfd/mod_posix_timer manualy or edit modules.conf.xml. 2013/2/13 jay binks > so does : > > freeswitch at default> show timer > type,name,ikey > timer,soft,CORE_SOFTTIMER_MODULE > > 1 total. > > on FreeSWITCH Version 1.2.6+git~20130104T154559Z~a4247651ca (git > a424765 2013-01-04 15:45:59Z) > > indicate it will be using timerfd calls or is this the old softtimer ? > > On 13 February 2013 02:21, Ken Rice wrote: > > The default one is the best one on linux with kernels starting around > > 2.6.32... Anything after that and you are good to go... > > > > The other timers are left around for other platforms > > > > K > > > > > > > > On 2/12/13 10:04 AM, "Fran?ois Delawarde" > > wrote: > > > >> Hello, > > > > Could someone knowing the different FS timers clarify the > >> difference > > between those: > > > > - default core softtimer > > - core heavy timer (does > >> this option still exists?) > > - posix timer > > - timerfd > > > > Can any of those provide a > >> significant performance improvement, or is it > > just not worth switching from > >> the default? > > > > In any case, which one would be the recommended/best one under > >> Linux > > with a relatively recent kernel > >> (>3.0)? > > > > Thanks, > > Fran?ois. > > > > > > > > ________________________________________________ > >> _________________________ > > Professional FreeSWITCH Consulting > >> Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > FreeSW > >> ITCH-powered IP PBX: The CudaTel Communication > >> Server > > > > > > Official FreeSWITCH > >> Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon. > >> com > > > > FreeSWITCH-users mailing > >> list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman > >> /listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/opt > >> ions/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > > Ken > > http://www.FreeSWITCH.org > > http://www.ClueCon.com > > http://www.OSTAG.org > > irc.freenode.net #freeswitch > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > 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 > > > > -- > Sincerely > > Jay > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130213/d74bdddd/attachment-0001.html From egable+freeswitch at gmail.com Wed Feb 13 02:18:49 2013 From: egable+freeswitch at gmail.com (Eliot Gable) Date: Tue, 12 Feb 2013 18:18:49 -0500 Subject: [Freeswitch-users] High Availability Cluster Module for FreeSWITCH In-Reply-To: References: Message-ID: On Tue, Feb 12, 2013 at 5:21 PM, Marcin Gozdalik wrote: > I did some, I got some anecdotes from people running them. It > certainly is possible to handle *some* failure cases, but there are > some *other* cases that will be not handled correctly. > > I have actually used Sonus, Genband, both of which seamlessly and gracefully handle failures. In fact, I cannot think of a single time where either of them failed to migrate calls over to an alternative system when the one handling calls had even one thing go wrong which prevented them from handling calls properly. And, as I mentioned earlier, I have also used P + C to build a multi-city, 6-node FS cluster (three iterations of it, actually) which could do essentially the same thing. While I don't have the resource agent I wrote for that cluster which handled the vast majority of failure conditions (including the two I mentioned earlier where you turn off ARP on the interface or set the speed/duplex wrong), I could fairly easily update my newly written resource agent to cover such scenarios (if someone wanted to pay for my time to do it). > Personally I've observed OpenSIPS that run out of some internal memory > and anyhow you wanted to monitor it it will reply it is alive. It will > even route your simple call scenarios well! Unfortunately the "real" > calls were *usually* timing out, but always. Sorry, but I can't > imagine how you can automatically handle that, i.e. discover that this > node is "bad" and fail-over to some other. Even if you do that > correctly your monitoring tool can check that this node is operating > perfectly well (because after all the traffic is diverted from the > faulty node it begins to work well) and will want to move the traffic > back. > Just because you cannot imagine how it would work, it does not mean everyone else has the same limitations. You are consistently referring to a node (let's call it node A) telling one or more other nodes (B, C, whatever) about its state and ability to handle calls. You are taking the wrong approach entirely. You cannot have a failed node reliably report it's state! What you can do is have Node A broadcast its state many times per second. Nodes B, C, etc, all listen for that state. They keep track of the state and when they *stop seeing* Node A, they know *something* failed. It's that simple. At this point, it is up to those nodes to determine *what* failed. They have to do things like check whether *they* failed, check whether the *network* failed, etc. There are very simple and sane ways they can do each of those things. Eventually, they can conclude that *Node A* failed. They can do this determination very quickly. Think microseconds or maybe 1 - 2 milliseconds quickly. At this point, nobody tells them what to do. They know already what they need to do. They have already pre-negotiated what will happen in the event of a failure, or they have very specific programmatic instructions on what to do, and so they act immediately. > By "impossible" I mean handling all such gray areas. Certainly if > you'll power down the box or unplug Ethernet it is possible to migrate > live calls to some other box. > There are no such "gray" areas. That is just a fantasy you have. Everything in computing is black or white, true or false. If you don't know, you test and become certain. If you are prevented from accurately testing because (for example) you cannot see the node at all anymore or interact with it in any way, you assume the worst and nuke the box (STONITH). Let's say the FS box runs out of memory. Great! I designed mod_ha_cluster to cause FS to segfault if it runs out of memory. Heartbeats stopped, other node takes over. No gray area. Wait. Did the IP get removed from the box? No? Don't know? STONITH. Did that fail? Seriously? You deployed the system wrong; don't blame me for your mistakes. Did the hard drive go away? Great! I have a test for that and a way to tell the other nodes I need to be STONITH'd if I cannot reboot myself. Did FS deadlock? Great! No more heartbeats. Other node takes over. STONITH. Did a module in FS take an event from the eventing thread and get stuck spinning, never to return? Great! No more heartbeats. Other node takes over. STONITH. Did a module in FS launch 128 threads, all of which want to use 100% CPU? Great! Untimely heartbeat delivery, other node takes over. STONITH. Did your dual-router network have the connection between the two routers go down leaving you with a split network? Great! If you have that secondary network I talked about, it's all properly detected and handled for you! If not, well, don't blame me for your failures. Did someone slap a firewall rule on the box and we suddenly cannot accept SIP messages? Great! One of the other nodes in the cluster will be sending us test SIP traffic on occasion and when we see it doesn't work anymore, we shut down and another node takes over. > I'd just like to have HA that works everytime and everywhere and try > So does everyone else. That is why I want to write mod_ha_cluster. Because what is there right now is overly complex, difficult to configure and test, and does not and cannot catch all of the possible ways in which a FS system can fail. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130212/23082431/attachment.html From david.villasmil.work at gmail.com Wed Feb 13 02:20:25 2013 From: david.villasmil.work at gmail.com (David Villasmil) Date: Wed, 13 Feb 2013 00:20:25 +0100 Subject: [Freeswitch-users] Nibblebill database question In-Reply-To: References: Message-ID: ok, some more info: dialplan: On Tue, Feb 12, 2013 at 8:33 PM, David Villasmil < david.villasmil.work at gmail.com> wrote: > Hello, > > Is uncommenting the "custom sql" lines mandatory?? it doesn't seem so, i > do see nibblebill trying to do it's job: > > 2013-02-12 19:32:07.784156 [DEBUG] mod_nibblebill.c:351 Doing update query > [UPDATE sip_users SET balance=balance-0.027683 WHERE id='1'] > 2013-02-12 19:32:07.784156 [CRIT] mod_nibblebill.c:535 Failed to log to > database! > 2013-02-12 19:32:07.784156 [DEBUG] mod_nibblebill.c:378 Doing lookup query > [SELECT balance AS nibble_balance FROM sip_users WHERE id='1'] > 2013-02-12 19:32:07.784156 [ERR] mod_nibblebill.c:380 Error running this > query: [SELECT balance AS nibble_balance FROM sip_users WHERE id='1'] > > > On Tue, Feb 12, 2013 at 8:23 PM, David Villasmil < > david.villasmil.work at gmail.com> wrote: > >> No, there's nothing related to db: >> >> 2013-02-12 19:23:04.823909 [CONSOLE] switch_loadable_module.c:1348 >> Successfully Loaded [mod_nibblebill] >> 2013-02-12 19:23:04.823909 [NOTICE] switch_loadable_module.c:254 Adding >> Application 'nibblebill' >> 2013-02-12 19:23:04.823909 [NOTICE] switch_loadable_module.c:298 Adding >> API Function 'nibblebill' >> >> >> >> >> On Tue, Feb 12, 2013 at 8:04 PM, Avi Marcus wrote: >> >>> Last I saw, if you load/reload mod_nibblebill, it logs which DSN it's >>> using to fs_cli / freeswitch.log. >>> >>> -Avi >>> >>> On Tue, Feb 12, 2013 at 9:00 PM, David Villasmil < >>> david.villasmil.work at gmail.com> wrote: >>> >>>> Yes, it does have the option, but it doesn't seem to be using a >>>> different one... I must've missed something... >>>> >>>> >>>> On Tue, Feb 12, 2013 at 7:53 PM, Avi Marcus wrote: >>>> >>>>> All modules that touch the DB (I think!) have their own configuration >>>>> that allows you pick the database. >>>>> mod_nibblebill does indeed has a param name="db_dsn" option so yes, >>>>> it will honor that. >>>>> >>>>> -Avi >>>>> >>>>> On Tue, Feb 12, 2013 at 8:11 PM, David Villasmil < >>>>> david.villasmil.work at gmail.com> wrote: >>>>> >>>>>> Hello guys, >>>>>> >>>>>> I have created 2 dsns, one for freeswitch core "freeswitch" and one >>>>>> for nibblebill ("whatever"). >>>>>> Is it possible to use different dsns in core and nibblebill? or does >>>>>> nibblebill uses the same dsn as the core?? >>>>>> >>>>>> thanks >>>>>> >>>>>> David >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> 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 >>>>>> >>>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> 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 >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130213/49fe3f90/attachment-0001.html From david.villasmil.work at gmail.com Wed Feb 13 02:26:48 2013 From: david.villasmil.work at gmail.com (David Villasmil) Date: Wed, 13 Feb 2013 00:26:48 +0100 Subject: [Freeswitch-users] Nibblebill database question In-Reply-To: References: Message-ID: ok, some more info: /etc/odbc.ini cat /etc/odbc.ini [freeswitch] Driver = /usr/lib/odbc/libmyodbc.so SERVER = localhost PORT = 3306 DATABASE = freeswitch OPTION = 67108864 USER = root PASSWORD = dv092171 [mydb] Driver = /usr/lib/odbc/libmyodbc.so SERVER = localhost PORT = 3306 DATABASE = viking OPTION = 67108864 USER = root PASSWORD = mypass switch.conf.xml nibblebill:
dialplan:
call goes out fine... no billing though: 2013-02-12 19:21:49.844277 [DEBUG] mod_nibblebill.c:504 22 seconds passed since last bill time of 2013-02-12 19:21:27 2013-02-12 19:21:49.844277 [DEBUG] mod_nibblebill.c:521 Billing $0.018983 to 1 (Call: d442dc6b-a3a7-423a-8450-598d17ca43b3 / 0.000000 so far) 2013-02-12 19:21:49.844277 [DEBUG] mod_nibblebill.c:351 Doing update query [UPDATE sip_users SET balance=balance-0.018983 WHERE ='1'] 2013-02-12 19:21:49.844277 [CRIT] mod_nibblebill.c:535 Failed to log to database! 2013-02-12 19:21:49.844277 [DEBUG] mod_nibblebill.c:378 Doing lookup query [SELECT balance AS nibble_balance FROM sip_users WHERE ='1'] 2013-02-12 19:21:49.844277 [ERR] mod_nibblebill.c:380 Error running this query: [SELECT balance AS nibble_balance FROM sip_users WHERE ='1'] the select go fine if executed "by hand" Thanks for your help! David On Tue, Feb 12, 2013 at 8:33 PM, David Villasmil < david.villasmil.work at gmail.com> wrote: > Hello, > > Is uncommenting the "custom sql" lines mandatory?? it doesn't seem so, i > do see nibblebill trying to do it's job: > > 2013-02-12 19:32:07.784156 [DEBUG] mod_nibblebill.c:351 Doing update query > [UPDATE sip_users SET balance=balance-0.027683 WHERE id='1'] > 2013-02-12 19:32:07.784156 [CRIT] mod_nibblebill.c:535 Failed to log to > database! > 2013-02-12 19:32:07.784156 [DEBUG] mod_nibblebill.c:378 Doing lookup query > [SELECT balance AS nibble_balance FROM sip_users WHERE id='1'] > 2013-02-12 19:32:07.784156 [ERR] mod_nibblebill.c:380 Error running this > query: [SELECT balance AS nibble_balance FROM sip_users WHERE id='1'] > > > On Tue, Feb 12, 2013 at 8:23 PM, David Villasmil < > david.villasmil.work at gmail.com> wrote: > >> No, there's nothing related to db: >> >> 2013-02-12 19:23:04.823909 [CONSOLE] switch_loadable_module.c:1348 >> Successfully Loaded [mod_nibblebill] >> 2013-02-12 19:23:04.823909 [NOTICE] switch_loadable_module.c:254 Adding >> Application 'nibblebill' >> 2013-02-12 19:23:04.823909 [NOTICE] switch_loadable_module.c:298 Adding >> API Function 'nibblebill' >> >> >> >> >> On Tue, Feb 12, 2013 at 8:04 PM, Avi Marcus wrote: >> >>> Last I saw, if you load/reload mod_nibblebill, it logs which DSN it's >>> using to fs_cli / freeswitch.log. >>> >>> -Avi >>> >>> On Tue, Feb 12, 2013 at 9:00 PM, David Villasmil < >>> david.villasmil.work at gmail.com> wrote: >>> >>>> Yes, it does have the option, but it doesn't seem to be using a >>>> different one... I must've missed something... >>>> >>>> >>>> On Tue, Feb 12, 2013 at 7:53 PM, Avi Marcus wrote: >>>> >>>>> All modules that touch the DB (I think!) have their own configuration >>>>> that allows you pick the database. >>>>> mod_nibblebill does indeed has a param name="db_dsn" option so yes, >>>>> it will honor that. >>>>> >>>>> -Avi >>>>> >>>>> On Tue, Feb 12, 2013 at 8:11 PM, David Villasmil < >>>>> david.villasmil.work at gmail.com> wrote: >>>>> >>>>>> Hello guys, >>>>>> >>>>>> I have created 2 dsns, one for freeswitch core "freeswitch" and one >>>>>> for nibblebill ("whatever"). >>>>>> Is it possible to use different dsns in core and nibblebill? or does >>>>>> nibblebill uses the same dsn as the core?? >>>>>> >>>>>> thanks >>>>>> >>>>>> David >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> 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 >>>>>> >>>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> 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 >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130213/c7d6e985/attachment.html From avi at avimarcus.net Wed Feb 13 02:59:03 2013 From: avi at avimarcus.net (Avi Marcus) Date: Wed, 13 Feb 2013 01:59:03 +0200 Subject: [Freeswitch-users] High Availability Cluster Module for FreeSWITCH In-Reply-To: References: Message-ID: Eliot, for those of us not running our own network, which datacenters/colos offer multiple networks/nics? e.g. amazon, rackspace, linode, softlayer..? It's not something I recall seeing mentioned... -Avi On Wed, Feb 13, 2013 at 1:18 AM, Eliot Gable wrote: > On Tue, Feb 12, 2013 at 5:21 PM, Marcin Gozdalik wrote: > >> I did some, I got some anecdotes from people running them. It >> certainly is possible to handle *some* failure cases, but there are >> some *other* cases that will be not handled correctly. >> >> > I have actually used Sonus, Genband, both of which seamlessly and > gracefully handle failures. In fact, I cannot think of a single time where > either of them failed to migrate calls over to an alternative system when > the one handling calls had even one thing go wrong which prevented them > from handling calls properly. And, as I mentioned earlier, I have also used > P + C to build a multi-city, 6-node FS cluster (three iterations of it, > actually) which could do essentially the same thing. While I don't have the > resource agent I wrote for that cluster which handled the vast majority of > failure conditions (including the two I mentioned earlier where you turn > off ARP on the interface or set the speed/duplex wrong), I could fairly > easily update my newly written resource agent to cover such scenarios (if > someone wanted to pay for my time to do it). > > >> Personally I've observed OpenSIPS that run out of some internal memory >> and anyhow you wanted to monitor it it will reply it is alive. It will >> even route your simple call scenarios well! Unfortunately the "real" >> calls were *usually* timing out, but always. Sorry, but I can't >> imagine how you can automatically handle that, i.e. discover that this >> node is "bad" and fail-over to some other. Even if you do that >> correctly your monitoring tool can check that this node is operating >> perfectly well (because after all the traffic is diverted from the >> faulty node it begins to work well) and will want to move the traffic >> back. >> > > Just because you cannot imagine how it would work, it does not mean > everyone else has the same limitations. You are consistently referring to a > node (let's call it node A) telling one or more other nodes (B, C, > whatever) about its state and ability to handle calls. You are taking the > wrong approach entirely. You cannot have a failed node reliably report it's > state! What you can do is have Node A broadcast its state many times per > second. Nodes B, C, etc, all listen for that state. They keep track of the > state and when they *stop seeing* Node A, they know *something* failed. > It's that simple. > > At this point, it is up to those nodes to determine *what* failed. They > have to do things like check whether *they* failed, check whether the > *network* failed, etc. There are very simple and sane ways they can do each > of those things. Eventually, they can conclude that *Node A* failed. They > can do this determination very quickly. Think microseconds or maybe 1 - 2 > milliseconds quickly. At this point, nobody tells them what to do. They > know already what they need to do. They have already pre-negotiated what > will happen in the event of a failure, or they have very > specific programmatic instructions on what to do, and so they act > immediately. > > >> By "impossible" I mean handling all such gray areas. Certainly if >> you'll power down the box or unplug Ethernet it is possible to migrate >> live calls to some other box. >> > > There are no such "gray" areas. That is just a fantasy you have. > Everything in computing is black or white, true or false. If you don't > know, you test and become certain. If you are prevented from accurately > testing because (for example) you cannot see the node at all anymore or > interact with it in any way, you assume the worst and nuke the box > (STONITH). > > Let's say the FS box runs out of memory. Great! I designed mod_ha_cluster > to cause FS to segfault if it runs out of memory. Heartbeats stopped, other > node takes over. No gray area. Wait. Did the IP get removed from the box? > No? Don't know? STONITH. Did that fail? Seriously? You deployed the system > wrong; don't blame me for your mistakes. > > Did the hard drive go away? Great! I have a test for that and a way to > tell the other nodes I need to be STONITH'd if I cannot reboot myself. > > Did FS deadlock? Great! No more heartbeats. Other node takes over. STONITH. > > Did a module in FS take an event from the eventing thread and get stuck > spinning, never to return? Great! No more heartbeats. Other node takes > over. STONITH. > > Did a module in FS launch 128 threads, all of which want to use 100% CPU? > Great! Untimely heartbeat delivery, other node takes over. STONITH. > > Did your dual-router network have the connection between the two routers > go down leaving you with a split network? Great! If you have that secondary > network I talked about, it's all properly detected and handled for you! If > not, well, don't blame me for your failures. > > Did someone slap a firewall rule on the box and we suddenly cannot accept > SIP messages? Great! One of the other nodes in the cluster will be sending > us test SIP traffic on occasion and when we see it doesn't work anymore, we > shut down and another node takes over. > > >> I'd just like to have HA that works everytime and everywhere and try >> > > So does everyone else. That is why I want to write mod_ha_cluster. Because > what is there right now is overly complex, difficult to configure and test, > and does not and cannot catch all of the possible ways in which a FS system > can fail. > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130213/906d14e0/attachment-0001.html From msc at freeswitch.org Wed Feb 13 03:12:41 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 12 Feb 2013 16:12:41 -0800 Subject: [Freeswitch-users] High Availability Cluster Module for FreeSWITCH In-Reply-To: References: Message-ID: On Tue, Feb 12, 2013 at 3:18 PM, Eliot Gable wrote: > On Tue, Feb 12, 2013 at 5:21 PM, Marcin Gozdalik wrote: > >> I did some, I got some anecdotes from people running them. It >> certainly is possible to handle *some* failure cases, but there are >> some *other* cases that will be not handled correctly. >> >> > I have actually used Sonus, Genband, both of which seamlessly and > gracefully handle failures. In fact, I cannot think of a single time where > either of them failed to migrate calls over to an alternative system when > the one handling calls had even one thing go wrong which prevented them > from handling calls properly. And, as I mentioned earlier, I have also used > P + C to build a multi-city, 6-node FS cluster (three iterations of it, > actually) which could do essentially the same thing. While I don't have the > resource agent I wrote for that cluster which handled the vast majority of > failure conditions (including the two I mentioned earlier where you turn > off ARP on the interface or set the speed/duplex wrong), I could fairly > easily update my newly written resource agent to cover such scenarios (if > someone wanted to pay for my time to do it). > > >> Personally I've observed OpenSIPS that run out of some internal memory >> and anyhow you wanted to monitor it it will reply it is alive. It will >> even route your simple call scenarios well! Unfortunately the "real" >> calls were *usually* timing out, but always. Sorry, but I can't >> imagine how you can automatically handle that, i.e. discover that this >> node is "bad" and fail-over to some other. Even if you do that >> correctly your monitoring tool can check that this node is operating >> perfectly well (because after all the traffic is diverted from the >> faulty node it begins to work well) and will want to move the traffic >> back. >> > > Just because you cannot imagine how it would work, it does not mean > everyone else has the same limitations. You are consistently referring to a > node (let's call it node A) telling one or more other nodes (B, C, > whatever) about its state and ability to handle calls. You are taking the > wrong approach entirely. You cannot have a failed node reliably report it's > state! What you can do is have Node A broadcast its state many times per > second. Nodes B, C, etc, all listen for that state. They keep track of the > state and when they *stop seeing* Node A, they know *something* failed. > It's that simple. > > At this point, it is up to those nodes to determine *what* failed. They > have to do things like check whether *they* failed, check whether the > *network* failed, etc. There are very simple and sane ways they can do each > of those things. Eventually, they can conclude that *Node A* failed. They > can do this determination very quickly. Think microseconds or maybe 1 - 2 > milliseconds quickly. At this point, nobody tells them what to do. They > know already what they need to do. They have already pre-negotiated what > will happen in the event of a failure, or they have very > specific programmatic instructions on what to do, and so they act > immediately. > > >> By "impossible" I mean handling all such gray areas. Certainly if >> you'll power down the box or unplug Ethernet it is possible to migrate >> live calls to some other box. >> > > There are no such "gray" areas. That is just a fantasy you have. > Everything in computing is black or white, true or false. If you don't > know, you test and become certain. If you are prevented from accurately > testing because (for example) you cannot see the node at all anymore or > interact with it in any way, you assume the worst and nuke the box > (STONITH). > > Let's say the FS box runs out of memory. Great! I designed mod_ha_cluster > to cause FS to segfault if it runs out of memory. Heartbeats stopped, other > node takes over. No gray area. Wait. Did the IP get removed from the box? > No? Don't know? STONITH. Did that fail? Seriously? You deployed the system > wrong; don't blame me for your mistakes. > > Did the hard drive go away? Great! I have a test for that and a way to > tell the other nodes I need to be STONITH'd if I cannot reboot myself. > > Did FS deadlock? Great! No more heartbeats. Other node takes over. STONITH. > > Did a module in FS take an event from the eventing thread and get stuck > spinning, never to return? Great! No more heartbeats. Other node takes > over. STONITH. > > Did a module in FS launch 128 threads, all of which want to use 100% CPU? > Great! Untimely heartbeat delivery, other node takes over. STONITH. > > Did your dual-router network have the connection between the two routers > go down leaving you with a split network? Great! If you have that secondary > network I talked about, it's all properly detected and handled for you! If > not, well, don't blame me for your failures. > > Did someone slap a firewall rule on the box and we suddenly cannot accept > SIP messages? Great! One of the other nodes in the cluster will be sending > us test SIP traffic on occasion and when we see it doesn't work anymore, we > shut down and another node takes over. > > >> I'd just like to have HA that works everytime and everywhere and try >> > > So does everyone else. That is why I want to write mod_ha_cluster. Because > what is there right now is overly complex, difficult to configure and test, > and does not and cannot catch all of the possible ways in which a FS system > can fail. > > > > +1 again. If you have reservations about what Eliot is doing then great - offer a valid alternative or ask a valid question. However, let's not make assumptions about what can and cannot be accomplished when the discussion is barely 3 days old. Eliot has answered a lot of questions here and has also done a good job of eliminating the FUD while demystifying what happens in a mod_ha_cluster. Be sure to join us tomorrow on the FreeSWITCH conference calland we'll talk about how to move forward and who wants to be part of the discussion. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130212/370b7284/attachment.html From msc at freeswitch.org Wed Feb 13 03:30:17 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 12 Feb 2013 16:30:17 -0800 Subject: [Freeswitch-users] how to end conference after initiator hangs up (dialout style) In-Reply-To: <5116D31F.8030807@junsun.net> References: <5116D31F.8030807@junsun.net> Message-ID: You could have the initiator put into the conference room via the originate command, in which case you could set flags: originate sofia/internal/$myNumber at x.x.x.x:5060 &conference("conference_test++flags{endconf}") -MC On Sat, Feb 9, 2013 at 2:52 PM, Jun Sun wrote: > > I have set up a conferencing call node using dialout style. That is, I > use the following commands to bridge everyone into a conference: > > conference $myRoom dial sofia/internal/$myNumber at x.x.x.x:5060 $myNumber > conference_test"; > conference $myRoom dial sofia/internal/$theirNumber1 at x.x.x.x:5060 > $myNumber conference_test"; > conference $myRoom dial sofia/internal/$theirNumber2 at x.x.x.x:5060 > $myNumber conference_test"; > ... > > Right now things are working fine except for one thing. I like to end > the conference (i.e., all legs are disconnected) when the initiator > ($myNumber) hangs up. But it is not doing that (understandably so!). > > How can I achieve that? > > I searched around and it seems most conf call controls only apply to > dial-in style conference calls. > > Thanks for any pointers! > > Jun > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130212/070023e1/attachment.html From krice at freeswitch.org Wed Feb 13 03:36:48 2013 From: krice at freeswitch.org (Ken Rice) Date: Tue, 12 Feb 2013 18:36:48 -0600 Subject: [Freeswitch-users] best freeswitch timer In-Reply-To: Message-ID: we don?t need all the previous messages on this thread What you need to know is that if you are using a kernel with timerfd support you do not need to do anything to use it, the Core will figure it out automagically and use it. This means main line kernels 2.6.22 and later and you are good to go. Doing a show timer on your FS console will reveal something like this... This is showing us the CORE_SOFTTIME_MODULE is being used for timing. The Core Timer uses timerfd freeswitch at devnode>show timer type,name,ikey timer,soft,CORE_SOFTTIMER_MODULE 1 total. If your kernel does not have timerfd support, then the old core soft timer is used... I?ll look at a small patch so you can know for sure, but, if your kernel supports timerfd and you see CORE_SOFTTIMER_MODULE as the timer, you are using timerfd K -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130212/59e85f40/attachment.html From sdevoy at bizfocused.com Wed Feb 13 04:19:12 2013 From: sdevoy at bizfocused.com (Sean Devoy) Date: Tue, 12 Feb 2013 20:19:12 -0500 Subject: [Freeswitch-users] new site, cant call OUT! In-Reply-To: References: <076e01ce0961$c14511d0$43cf3570$@bizfocused.com> Message-ID: <08f401ce0988$21b97e90$652c7bb0$@bizfocused.com> Thanks Ken, I captured a siptrace and started pocking around. Then I got logged into the local lan (logmein). The cisco router has multiple local ip addresses I can login into - odd. Then I noticed the Cisco (which is in Gateway mode) has a WAN address of 10.1.10.10!!! I poked around and found the Comcast Business Router on the other side of the Cisco. Yippee DOUBLE NAT. I chose to just say "NO." Thanks for the answer. Sean From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice Sent: Tuesday, February 12, 2013 3:58 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] new site, cant call OUT! Sounds like a possible NAT issue... But the only way to know that is look at a sip trace and see whats going on that way and maybe even looking from the phones perspective On 2/12/13 2:44 PM, "Sean Devoy" wrote: Hi All, (Git head build last week on centos 5.n) This is a new one for me. Calls IN to the new customer with Cisco phones behind NAT are working, but not outbound! I have not been to the site yet. I am told that they can call out once or twice then have to reboot the phone to dial out. Dialing in continues to work! one person says the phone says "Call merged" another says it say "Request merged" on the phone display! I did see in the logs, that when it works, we get: 2013-02-12 11:09:26.383576 [NOTICE] switch_channel.c:968 New Channel sofia/external/105@ [51884b2d-6aa2-47fe-b687-8604a4e39a7a] 2013-02-12 11:09:26.383576 [DEBUG] switch_core_session.c:975 Send signal sofia/external/105@ [BREAK] 2013-02-12 11:09:26.383576 [DEBUG] switch_core_session.c:975 Send signal sofia/external/105@ [BREAK] 2013-02-12 11:09:26.383576 [DEBUG] switch_core_state_machine.c:415 (sofia/external/105@) Running State Change CS_NEW 2013-02-12 11:09:26.383576 [DEBUG] switch_core_state_machine.c:433 (sofia/external/105@) State NEW 2013-02-12 11:09:26.413581 [DEBUG] switch_core_session.c:975 Send signal sofia/external/105@ [BREAK] 2013-02-12 11:09:26.413581 [DEBUG] sofia.c:1750 detaching session 51884b2d-6aa2-47fe-b687-8604a4e39a7a 2013-02-12 11:09:26.413581 [WARNING] sofia_reg.c:1502 SIP auth challenge (INVITE) on sofia profile 'external' for [3014040914@] from ip 2013-02-12 11:09:26.493574 [DEBUG] sofia.c:1842 Re-attaching to session 51884b2d-6aa2-47fe-b687-8604a4e39a7a And when it fails: 2013-02-12 13:32:26.363343 [NOTICE] switch_channel.c:968 New Channel sofia/external/220@ [91e9eb64-253c-4604-bdfa-4dd18baae18d] 2013-02-12 13:32:26.363343 [DEBUG] switch_core_session.c:975 Send signal sofia/external/220@ [BREAK] 2013-02-12 13:32:26.363343 [DEBUG] switch_core_session.c:975 Send signal sofia/external/220@ [BREAK] 2013-02-12 13:32:26.363343 [DEBUG] switch_core_state_machine.c:415 (sofia/external/220@) Running State Change CS_NEW 2013-02-12 13:32:26.363343 [DEBUG] switch_core_state_machine.c:433 (sofia/external/220@) State NEW 2013-02-12 13:32:26.383344 [DEBUG] switch_core_session.c:975 Send signal sofia/external/220@ [BREAK] 2013-02-12 13:32:26.383344 [DEBUG] sofia.c:1750 detaching session 91e9eb64-253c-4604-bdfa-4dd18baae18d 2013-02-12 13:32:26.383344 [WARNING] sofia_reg.c:1502 SIP auth challenge (INVITE) on sofia profile 'external' for [4104934408@] from ip 2 2013-02-12 13:32:36.363085 [WARNING] switch_core_state_machine.c:514 91e9eb64-253c-4604-bdfa-4dd18baae18d sofia/external/220@ Abandoned 2013-02-12 13:32:36.363085 [DEBUG] switch_channel.c:2994 (sofia/external/220@) Callstate Change DOWN -> HANGUP Can anyone shed light on what's happening "at that particular juncture" and how I may go about debugging it? I was thinking FS was not finding a connection through NAT to phone, but then why do incoming calls work? Thanks in advance, Sean _____ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130212/410ae0d2/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 70 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130212/410ae0d2/attachment-0001.gif From jmesquita at freeswitch.org Wed Feb 13 04:48:02 2013 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Tue, 12 Feb 2013 22:48:02 -0300 Subject: [Freeswitch-users] FreeSWITCH Message Bus / Shared Key Value Store In-Reply-To: <20130212204251.7066d944@mail.tritonwest.net> References: <20130212204251.7066d944@mail.tritonwest.net> Message-ID: I have used ZeroMQ in the past for this sorts of things but it really won't be able to detect failures really fast. It is not made for this. Maybe we can gather the requirements for such message bus? Zmq for example provides you with this cool interface to build messaging protocols on top of it but it does not provide reliability when it comes to endpoint to endpoint connection without a heartbeat implemented on the user end. Can this be used for FS as well? Anyhow, just throwing some ideas... Jo?o Mesquita FreeSWITCH? Solutions On Tue, Feb 12, 2013 at 5:42 PM, Dave R. Kompel wrote: > ** > I've done a few experments with using both Redis, and the evil "Microsoft > Azure Service bus" (the server on prem based version) to extend the > eventing system to have global PUB/SUB. This way things like registrations, > and Limit stuff could be made global. > > I'm looking for a way, in my carrier switch implmentation, to implment > both HA Failover and Scaleout clustering. > > --Dave > > ------------------------------ > *From:* Eliot Gable [mailto:egable+freeswitch at gmail.com] > *To:* FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] > *Sent:* Tue, 12 Feb 2013 05:49:13 -0800 > *Subject:* [Freeswitch-users] FreeSWITCH Message Bus / Shared Key Value > Store > > > Tony and Mike and I had a discussion last night about FreeSWITCH with > regards to implementing some form of core message bus or shared key-value > store. We discussed a few different options, but did not really settle on > anything. If you are writing modules or using FreeSWITCH in a multi-node > setting, please share what features / functionality you would like to see > implemented in this regard, how you would use it, and why you want to see > the specific mechanism of your choice rather than some alternative. Also, > please consider and mention whether "cluster awareness" is something that > factors into your use case. By this, I mean having each FS node have some > idea about the state / status of each other node in terms of taking calls > vs acting as a standby or slave node, etc. > > -- > Eliot Gable > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130212/540d0429/attachment.html From egable+freeswitch at gmail.com Wed Feb 13 04:57:06 2013 From: egable+freeswitch at gmail.com (Eliot Gable) Date: Tue, 12 Feb 2013 20:57:06 -0500 Subject: [Freeswitch-users] High Availability Cluster Module for FreeSWITCH In-Reply-To: References: Message-ID: On Tue, Feb 12, 2013 at 6:59 PM, Avi Marcus wrote: > Eliot, for those of us not running our own network, which > datacenters/colos offer multiple networks/nics? > e.g. amazon, rackspace, linode, softlayer..? > > It's not something I recall seeing mentioned... > > -Avi > Honestly, I have never utilized such services. I have always worked for / with companies which run their own network. Regardless of what datacenter / colo you use, you need to ensure you have redundant communications between your nodes. This is true for *every* HA deployment. The way Sonus and other providers do it is to use a physical chassis which provides the redundant communication mechanism. When you do not have such a chassis, you must use a separate network connection. When you have only two nodes, it's easy to do a crossover cable between them. However, if you have lots of nodes, you need an entirely separate network. That separate network only needs to facilitate communication between nodes. It does not need Internet connectivity. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130212/8f170bdb/attachment.html From covici at ccs.covici.com Wed Feb 13 05:01:59 2013 From: covici at ccs.covici.com (covici at ccs.covici.com) Date: Tue, 12 Feb 2013 21:01:59 -0500 Subject: [Freeswitch-users] best freeswitch timer In-Reply-To: References: Message-ID: <22257.1360720919@ccs.covici.com> Mine says timer_soft instead of what you have -- I have certainly a recent kernel, how do I get timerfb going? Ken Rice wrote: > we don?t need all the previous messages on this thread > > What you need to know is that if you are using a kernel with timerfd support > you do not need to do anything to use it, the Core will figure it out > automagically and use it. This means main line kernels 2.6.22 and later and > you are good to go. > > Doing a show timer on your FS console will reveal something like this... > This is showing us the CORE_SOFTTIME_MODULE is being used for timing. The > Core Timer uses timerfd > > freeswitch at devnode>show timer > > type,name,ikey > timer,soft,CORE_SOFTTIMER_MODULE > > 1 total. > > If your kernel does not have timerfd support, then the old core soft timer > is used... > > I?ll look at a small patch so you can know for sure, but, if your kernel > supports timerfd and you see CORE_SOFTTIMER_MODULE as the timer, you are > using timerfd > > K > > ---------------------------------------------------- > Alternatives: > > ---------------------------------------------------- > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici at ccs.covici.com From covici at ccs.covici.com Wed Feb 13 05:07:57 2013 From: covici at ccs.covici.com (covici at ccs.covici.com) Date: Tue, 12 Feb 2013 21:07:57 -0500 Subject: [Freeswitch-users] best freeswitch timer In-Reply-To: <22257.1360720919@ccs.covici.com> References: <22257.1360720919@ccs.covici.com> Message-ID: <23538.1360721277@ccs.covici.com> Never mind, I already seem to have the correct show timer -- just a brain far. covici at ccs.covici.com wrote: > Mine says timer_soft instead of what you have -- I have certainly a > recent kernel, how do I get timerfb going? > > Ken Rice wrote: > > > we don?t need all the previous messages on this thread > > > > What you need to know is that if you are using a kernel with timerfd support > > you do not need to do anything to use it, the Core will figure it out > > automagically and use it. This means main line kernels 2.6.22 and later and > > you are good to go. > > > > Doing a show timer on your FS console will reveal something like this... > > This is showing us the CORE_SOFTTIME_MODULE is being used for timing. The > > Core Timer uses timerfd > > > > freeswitch at devnode>show timer > > > > type,name,ikey > > timer,soft,CORE_SOFTTIMER_MODULE > > > > 1 total. > > > > If your kernel does not have timerfd support, then the old core soft timer > > is used... > > > > I?ll look at a small patch so you can know for sure, but, if your kernel > > supports timerfd and you see CORE_SOFTTIMER_MODULE as the timer, you are > > using timerfd > > > > K > > > > ---------------------------------------------------- > > Alternatives: > > > > ---------------------------------------------------- > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > 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 > > -- > Your life is like a penny. You're going to lose it. The question is: > How do > you spend it? > > John Covici > covici at ccs.covici.com > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici at ccs.covici.com From dvl36.ripe.nick at gmail.com Wed Feb 13 05:31:14 2013 From: dvl36.ripe.nick at gmail.com (Dmitry Lysenko) Date: Wed, 13 Feb 2013 04:31:14 +0200 Subject: [Freeswitch-users] best freeswitch timer In-Reply-To: References: Message-ID: <1360722674.19143.21.camel@Dmitry-AOA150.private> ? ???, 12/02/2013 ? 18:36 -0600, Ken Rice ?????: > What you need to know is that if you are using a kernel with timerfd > support you do not need to do anything to use it, the Core will figure > it out automagically and use it. This means main line kernels 2.6.22 > and later and you are good to go. This automagic is really magic. But softtimer does not work well on my system. :( Kernel is latest (3.2.38-rt56 PREEMPT RT) and has all modern timing features as timerfd, clock_nanosleep, HR timer(1ns reported, 470ns measured by cyclictest -R),etc. --- switch.conf.xml --- --- running freeswitch with -rp. freeswitch at internal> timer_test 20 200 Avg: 19.999ms Total Time: 3999.985ms 2013-02-13 03:46:05.326648 [CONSOLE] mod_commands.c:827 Timer Test: 1 sleep 20 20204 2013-02-13 03:46:05.346428 [CONSOLE] mod_commands.c:827 Timer Test: 2 sleep 20 19779 2013-02-13 03:46:05.366429 [CONSOLE] mod_commands.c:827 Timer Test: 3 sleep 20 20001 2013-02-13 03:46:05.386427 [CONSOLE] mod_commands.c:827 Timer Test: 4 sleep 20 19999 2013-02-13 03:46:05.406427 [CONSOLE] mod_commands.c:827 Timer Test: 5 sleep 20 19998 2013-02-13 03:46:05.426644 [CONSOLE] mod_commands.c:827 Timer Test: 6 sleep 20 20218 2013-02-13 03:46:05.446428 [CONSOLE] mod_commands.c:827 Timer Test: 7 sleep 20 19783 2013-02-13 03:46:05.466428 [CONSOLE] mod_commands.c:827 Timer Test: 8 sleep 20 20000 2013-02-13 03:46:05.486428 [CONSOLE] mod_commands.c:827 Timer Test: 9 sleep 20 20000 2013-02-13 03:46:05.506428 [CONSOLE] mod_commands.c:827 Timer Test: 10 sleep 20 20001 2013-02-13 03:46:05.526644 [CONSOLE] mod_commands.c:827 Timer Test: 11 sleep 20 20216 ... ------ freeswitch at internal> timer_test 20 200 timerfd Avg: 19.999ms Total Time: 3999.983ms 2013-02-13 03:48:02.726442 [CONSOLE] mod_commands.c:827 Timer Test: 1 sleep 20 19985 2013-02-13 03:48:02.746437 [CONSOLE] mod_commands.c:827 Timer Test: 2 sleep 20 20000 2013-02-13 03:48:02.766442 [CONSOLE] mod_commands.c:827 Timer Test: 3 sleep 20 20003 2013-02-13 03:48:02.786433 [CONSOLE] mod_commands.c:827 Timer Test: 4 sleep 20 19998 2013-02-13 03:48:02.806433 [CONSOLE] mod_commands.c:827 Timer Test: 5 sleep 20 19997 2013-02-13 03:48:02.826433 [CONSOLE] mod_commands.c:827 Timer Test: 6 sleep 20 20000 2013-02-13 03:48:02.846432 [CONSOLE] mod_commands.c:827 Timer Test: 7 sleep 20 19999 2013-02-13 03:48:02.866432 [CONSOLE] mod_commands.c:827 Timer Test: 8 sleep 20 20000 2013-02-13 03:48:02.886432 [CONSOLE] mod_commands.c:827 Timer Test: 9 sleep 20 20002 2013-02-13 03:48:02.906432 [CONSOLE] mod_commands.c:827 Timer Test: 10 sleep 20 20000 2013-02-13 03:48:02.926434 [CONSOLE] mod_commands.c:827 Timer Test: 11 sleep 20 20000 ... ----- Tried many timing switch.conf options, but always spikes ~=200mks on soft timer. What I can do to make soft timer more precise? Thanks. From rupa at rupa.com Wed Feb 13 05:35:10 2013 From: rupa at rupa.com (Rupa Schomaker) Date: Tue, 12 Feb 2013 20:35:10 -0600 Subject: [Freeswitch-users] Nibblebill database question In-Reply-To: References: Message-ID: On Tue, Feb 12, 2013 at 5:26 PM, David Villasmil < david.villasmil.work at gmail.com> wrote: > call goes out fine... no billing though: > > > 2013-02-12 19:21:49.844277 [DEBUG] mod_nibblebill.c:504 22 seconds passed > since last bill time of 2013-02-12 19:21:27 > 2013-02-12 19:21:49.844277 [DEBUG] mod_nibblebill.c:521 Billing $0.018983 > to 1 (Call: d442dc6b-a3a7-423a-8450-598d17ca43b3 / 0.000000 so far) > 2013-02-12 19:21:49.844277 [DEBUG] mod_nibblebill.c:351 Doing update query > [UPDATE sip_users SET balance=balance-0.018983 WHERE ='1'] > 2013-02-12 19:21:49.844277 [CRIT] mod_nibblebill.c:535 Failed to log to > database! > 2013-02-12 19:21:49.844277 [DEBUG] mod_nibblebill.c:378 Doing lookup query > [SELECT balance AS nibble_balance FROM sip_users WHERE ='1'] > 2013-02-12 19:21:49.844277 [ERR] mod_nibblebill.c:380 Error running this > query: [SELECT balance AS nibble_balance FROM sip_users WHERE ='1'] > > > the select go fine if executed "by hand" > None of those SQL statements are valid. (hint: WHERE = '1' is no good). Been a while since I hacked/worked with nibblebill, but I'm pretty sure the 'id' field isn't not being filled in the sql for some reason. -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130212/ec492a20/attachment-0001.html From c.barthes4 at free.fr Wed Feb 13 08:14:22 2013 From: c.barthes4 at free.fr (cbarthes35) Date: Tue, 12 Feb 2013 21:14:22 -0800 (PST) Subject: [Freeswitch-users] (no subject) In-Reply-To: References: <1360647260499-7587257.post@n2.nabble.com> <511A7F04.9010704@gmail.com> Message-ID: <1360732462985-7587331.post@n2.nabble.com> Hello everybody, Thanks for answering my question. you mean this module is not operational ??!!?? Why is it mentionned in the features list of freeswitch.org ? Otherwise, would it be possible to go hrough a working module like chan_sccp_b in the asterisk world ? Bye Chris. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/skinny-SCCP-dialing-is-impossible-tp7587257p7587331.html Sent from the freeswitch-users mailing list archive at Nabble.com. From krice at freeswitch.org Wed Feb 13 08:29:11 2013 From: krice at freeswitch.org (Ken Rice) Date: Tue, 12 Feb 2013 23:29:11 -0600 Subject: [Freeswitch-users] (no subject) In-Reply-To: <1360732462985-7587331.post@n2.nabble.com> Message-ID: I could be wrong, you should check Jira for possible open issues also On 2/12/13 11:14 PM, "cbarthes35" wrote: > Hello everybody, > > Thanks for answering my question. > > you mean this module is not operational ??!!?? > > Why is it mentionned in the features list of freeswitch.org ? > > Otherwise, would it be possible to go hrough a working module like > chan_sccp_b in the asterisk world ? > > Bye > > Chris. > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/skinny-SCCP-dialing-is-impossibl > e-tp7587257p7587331.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch From POlsson at enghouse.com Wed Feb 13 10:02:12 2013 From: POlsson at enghouse.com (Peter Olsson) Date: Wed, 13 Feb 2013 07:02:12 +0000 Subject: [Freeswitch-users] best freeswitch timer In-Reply-To: <1360722674.19143.21.camel@Dmitry-AOA150.private> References: , <1360722674.19143.21.camel@Dmitry-AOA150.private> Message-ID: <1A89F457-BDAE-418C-9316-AED00FDBE230@visionutveckling.se> I wouldn't worry too much about spikes of 200usec, do you have an actual problem with audio quality? /Peter 13 feb 2013 kl. 03:35 skrev "Dmitry Lysenko" : > ? ???, 12/02/2013 ? 18:36 -0600, Ken Rice ?????: >> What you need to know is that if you are using a kernel with timerfd >> support you do not need to do anything to use it, the Core will figure >> it out automagically and use it. This means main line kernels 2.6.22 >> and later and you are good to go. > > This automagic is really magic. But softtimer does not work well on my > system. :( > Kernel is latest (3.2.38-rt56 PREEMPT RT) and has all modern timing > features as timerfd, clock_nanosleep, HR timer(1ns reported, 470ns > measured by cyclictest -R),etc. > --- switch.conf.xml --- > > --- > running freeswitch with -rp. > > freeswitch at internal> timer_test 20 200 > Avg: 19.999ms Total Time: 3999.985ms > 2013-02-13 03:46:05.326648 [CONSOLE] mod_commands.c:827 Timer Test: 1 > sleep 20 20204 > 2013-02-13 03:46:05.346428 [CONSOLE] mod_commands.c:827 Timer Test: 2 > sleep 20 19779 > 2013-02-13 03:46:05.366429 [CONSOLE] mod_commands.c:827 Timer Test: 3 > sleep 20 20001 > 2013-02-13 03:46:05.386427 [CONSOLE] mod_commands.c:827 Timer Test: 4 > sleep 20 19999 > 2013-02-13 03:46:05.406427 [CONSOLE] mod_commands.c:827 Timer Test: 5 > sleep 20 19998 > 2013-02-13 03:46:05.426644 [CONSOLE] mod_commands.c:827 Timer Test: 6 > sleep 20 20218 > 2013-02-13 03:46:05.446428 [CONSOLE] mod_commands.c:827 Timer Test: 7 > sleep 20 19783 > 2013-02-13 03:46:05.466428 [CONSOLE] mod_commands.c:827 Timer Test: 8 > sleep 20 20000 > 2013-02-13 03:46:05.486428 [CONSOLE] mod_commands.c:827 Timer Test: 9 > sleep 20 20000 > 2013-02-13 03:46:05.506428 [CONSOLE] mod_commands.c:827 Timer Test: 10 > sleep 20 20001 > 2013-02-13 03:46:05.526644 [CONSOLE] mod_commands.c:827 Timer Test: 11 > sleep 20 20216 > ... > ------ > freeswitch at internal> timer_test 20 200 timerfd > Avg: 19.999ms Total Time: 3999.983ms > > 2013-02-13 03:48:02.726442 [CONSOLE] mod_commands.c:827 Timer Test: 1 > sleep 20 19985 > 2013-02-13 03:48:02.746437 [CONSOLE] mod_commands.c:827 Timer Test: 2 > sleep 20 20000 > 2013-02-13 03:48:02.766442 [CONSOLE] mod_commands.c:827 Timer Test: 3 > sleep 20 20003 > 2013-02-13 03:48:02.786433 [CONSOLE] mod_commands.c:827 Timer Test: 4 > sleep 20 19998 > 2013-02-13 03:48:02.806433 [CONSOLE] mod_commands.c:827 Timer Test: 5 > sleep 20 19997 > 2013-02-13 03:48:02.826433 [CONSOLE] mod_commands.c:827 Timer Test: 6 > sleep 20 20000 > 2013-02-13 03:48:02.846432 [CONSOLE] mod_commands.c:827 Timer Test: 7 > sleep 20 19999 > 2013-02-13 03:48:02.866432 [CONSOLE] mod_commands.c:827 Timer Test: 8 > sleep 20 20000 > 2013-02-13 03:48:02.886432 [CONSOLE] mod_commands.c:827 Timer Test: 9 > sleep 20 20002 > 2013-02-13 03:48:02.906432 [CONSOLE] mod_commands.c:827 Timer Test: 10 > sleep 20 20000 > 2013-02-13 03:48:02.926434 [CONSOLE] mod_commands.c:827 Timer Test: 11 > sleep 20 20000 > ... > ----- > Tried many timing switch.conf options, but always spikes ~=200mks on > soft timer. What I can do to make soft timer more precise? > Thanks. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > !DSPAM:511af76f32761034681629! > From itsusama at gmail.com Wed Feb 13 13:24:58 2013 From: itsusama at gmail.com (Usama Zaidi) Date: Wed, 13 Feb 2013 15:24:58 +0500 Subject: [Freeswitch-users] ESL Mod_Conference Observations Message-ID: Hi, Just wanted to know why don't I get any events on start/stop speaking conference joins/leaves if I do something like this: $con->events("plain","all"); $con->events("plain","CUSTOM conference::maintenance"); $con->sendRecv("filter Unique-ID $confUUID"); Instead I've to do something like this: $con->events("plain","all"); $con->events("plain","CUSTOM conference::maintenance"); $con->sendRecv("filter Unique-ID $confmember1"); $con->sendRecv("filter Unique-ID $confmember2"); $con->sendRecv("filter Unique-ID $confmember3"); . . . Which is useless in case you want to check if a user is entering/leaving a particular conference in a single inbound ESL script... What I'm doing instead now is something on these lines: $con->events("plain","all"); $con->sendRecv("filter Unique-ID $CONFNAME"); i//psuedocode if event equals PRESENCE_IN and presence-call-direction equals inbound then { execute API command conference list confname parse uuid from members add a filter update the hashtable for existing confmembers } //once we have the uuid filter added we'll get to know if that user leaves, so the logic for removing the filter is straight forward if event equals CUSTOM and subtype is CONFERENCE::MAINTANANCE { if event action=del-member remove filter for UUID } I don't get ANY event if I'm filtering on conference UUID... I think that needs to be fixed, because its mod_conference which is throwing all those events. Regards. -- I'd love to change the world, but they wont gimme the source code to it -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130213/781b33a6/attachment.html From steveayre at gmail.com Wed Feb 13 13:31:29 2013 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 13 Feb 2013 10:31:29 +0000 Subject: [Freeswitch-users] best freeswitch timer In-Reply-To: References: <1360685096.22409.768.camel@luna.madrid.commsmundi.com> Message-ID: 'show timer' shows all loaded timer modules. That shows none are loaded. CORE_SOFTTIMER_MODULE is the soft timer built into the FS core... it may or may not use timerfd, depending on your system. You pick which timer to actually use in your configuration files... for example the rtp-timer-name Sofia profile param. Use the timer_test command to see which timer actually works best for you (ie most accurate, least jitter). http://wiki.freeswitch.org/wiki/Mod_commands#timer_test -Steve On 12 February 2013 22:48, jay binks wrote: > so does : > > freeswitch at default> show timer > type,name,ikey > timer,soft,CORE_SOFTTIMER_MODULE > > 1 total. > > on FreeSWITCH Version 1.2.6+git~20130104T154559Z~a4247651ca (git > a424765 2013-01-04 15:45:59Z) > > indicate it will be using timerfd calls or is this the old softtimer ? > > On 13 February 2013 02:21, Ken Rice wrote: > > The default one is the best one on linux with kernels starting around > > 2.6.32... Anything after that and you are good to go... > > > > The other timers are left around for other platforms > > > > K > > > > > > > > On 2/12/13 10:04 AM, "Fran?ois Delawarde" > > wrote: > > > >> Hello, > > > > Could someone knowing the different FS timers clarify the > >> difference > > between those: > > > > - default core softtimer > > - core heavy timer (does > >> this option still exists?) > > - posix timer > > - timerfd > > > > Can any of those provide a > >> significant performance improvement, or is it > > just not worth switching from > >> the default? > > > > In any case, which one would be the recommended/best one under > >> Linux > > with a relatively recent kernel > >> (>3.0)? > > > > Thanks, > > Fran?ois. > > > > > > > > ________________________________________________ > >> _________________________ > > Professional FreeSWITCH Consulting > >> Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > FreeSW > >> ITCH-powered IP PBX: The CudaTel Communication > >> Server > > > > > > Official FreeSWITCH > >> Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon. > >> com > > > > FreeSWITCH-users mailing > >> list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman > >> /listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/opt > >> ions/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > > Ken > > http://www.FreeSWITCH.org > > http://www.ClueCon.com > > http://www.OSTAG.org > > irc.freenode.net #freeswitch > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > 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 > > > > -- > Sincerely > > Jay > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130213/b71de884/attachment-0001.html From nathandownes at hotmail.com Wed Feb 13 13:42:37 2013 From: nathandownes at hotmail.com (Mr Nathan Downes) Date: Wed, 13 Feb 2013 21:42:37 +1100 Subject: [Freeswitch-users] Door intercom/gate controller Message-ID: Hi list, I have a gate controller that only understands RFC 2833 but it calls endpoints that can only provide inband DTMF, I can't seem to get the digits recognised by the gate controller to trigger the relay using start_dtmf_generate, as they are coming back to the a-leg? Rather than out the b-leg.. It works fine when I call a SPA502g as it will provide RFC 2833.. is there some trickery I can achieve this? Or am I just doing it wrong?? J -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130213/570f84fd/attachment.html From steveayre at gmail.com Wed Feb 13 13:47:43 2013 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 13 Feb 2013 10:47:43 +0000 Subject: [Freeswitch-users] High Availability Cluster Module for FreeSWITCH In-Reply-To: References: Message-ID: This is something where you'll need to get a physical rack and install your own setup. It's too specialised for you to find anyone offering it as standard. You're probably going to want them to be pretty close together connected via a dedicated LAN anyway - otherwise packet loss / routing loops tin/ latency / jitter will mean your heartbeats start failing. Similarly you don't want anything stealing your CPU away preventing you from sending heartbeats at the correct time, so VPS options like Amazon AWS are not going to be appropriate either. And you're going to want STONITH support too (eg via power sockets that're controllable via IP) - it's unlikely a dedicated server from someone like rackspace will offer that and shutting down virtual machine instances aren't reliable enough for STONITH (the guest could be down because of a fault on the host which might leave you unable to shutdown the guest). -Steve On 12 February 2013 23:59, Avi Marcus wrote: > Eliot, for those of us not running our own network, which > datacenters/colos offer multiple networks/nics? > e.g. amazon, rackspace, linode, softlayer..? > > It's not something I recall seeing mentioned... > > -Avi > > On Wed, Feb 13, 2013 at 1:18 AM, Eliot Gable wrote: > >> On Tue, Feb 12, 2013 at 5:21 PM, Marcin Gozdalik wrote: >> >>> I did some, I got some anecdotes from people running them. It >>> certainly is possible to handle *some* failure cases, but there are >>> some *other* cases that will be not handled correctly. >>> >>> >> I have actually used Sonus, Genband, both of which seamlessly and >> gracefully handle failures. In fact, I cannot think of a single time where >> either of them failed to migrate calls over to an alternative system when >> the one handling calls had even one thing go wrong which prevented them >> from handling calls properly. And, as I mentioned earlier, I have also used >> P + C to build a multi-city, 6-node FS cluster (three iterations of it, >> actually) which could do essentially the same thing. While I don't have the >> resource agent I wrote for that cluster which handled the vast majority of >> failure conditions (including the two I mentioned earlier where you turn >> off ARP on the interface or set the speed/duplex wrong), I could fairly >> easily update my newly written resource agent to cover such scenarios (if >> someone wanted to pay for my time to do it). >> >> >>> Personally I've observed OpenSIPS that run out of some internal memory >>> and anyhow you wanted to monitor it it will reply it is alive. It will >>> even route your simple call scenarios well! Unfortunately the "real" >>> calls were *usually* timing out, but always. Sorry, but I can't >>> imagine how you can automatically handle that, i.e. discover that this >>> node is "bad" and fail-over to some other. Even if you do that >>> correctly your monitoring tool can check that this node is operating >>> perfectly well (because after all the traffic is diverted from the >>> faulty node it begins to work well) and will want to move the traffic >>> back. >>> >> >> Just because you cannot imagine how it would work, it does not mean >> everyone else has the same limitations. You are consistently referring to a >> node (let's call it node A) telling one or more other nodes (B, C, >> whatever) about its state and ability to handle calls. You are taking the >> wrong approach entirely. You cannot have a failed node reliably report it's >> state! What you can do is have Node A broadcast its state many times per >> second. Nodes B, C, etc, all listen for that state. They keep track of the >> state and when they *stop seeing* Node A, they know *something* failed. >> It's that simple. >> >> At this point, it is up to those nodes to determine *what* failed. They >> have to do things like check whether *they* failed, check whether the >> *network* failed, etc. There are very simple and sane ways they can do each >> of those things. Eventually, they can conclude that *Node A* failed. They >> can do this determination very quickly. Think microseconds or maybe 1 - 2 >> milliseconds quickly. At this point, nobody tells them what to do. They >> know already what they need to do. They have already pre-negotiated what >> will happen in the event of a failure, or they have very >> specific programmatic instructions on what to do, and so they act >> immediately. >> >> >>> By "impossible" I mean handling all such gray areas. Certainly if >>> you'll power down the box or unplug Ethernet it is possible to migrate >>> live calls to some other box. >>> >> >> There are no such "gray" areas. That is just a fantasy you have. >> Everything in computing is black or white, true or false. If you don't >> know, you test and become certain. If you are prevented from accurately >> testing because (for example) you cannot see the node at all anymore or >> interact with it in any way, you assume the worst and nuke the box >> (STONITH). >> >> Let's say the FS box runs out of memory. Great! I designed mod_ha_cluster >> to cause FS to segfault if it runs out of memory. Heartbeats stopped, other >> node takes over. No gray area. Wait. Did the IP get removed from the box? >> No? Don't know? STONITH. Did that fail? Seriously? You deployed the system >> wrong; don't blame me for your mistakes. >> >> Did the hard drive go away? Great! I have a test for that and a way to >> tell the other nodes I need to be STONITH'd if I cannot reboot myself. >> >> Did FS deadlock? Great! No more heartbeats. Other node takes over. >> STONITH. >> >> Did a module in FS take an event from the eventing thread and get stuck >> spinning, never to return? Great! No more heartbeats. Other node takes >> over. STONITH. >> >> Did a module in FS launch 128 threads, all of which want to use 100% CPU? >> Great! Untimely heartbeat delivery, other node takes over. STONITH. >> >> Did your dual-router network have the connection between the two routers >> go down leaving you with a split network? Great! If you have that secondary >> network I talked about, it's all properly detected and handled for you! If >> not, well, don't blame me for your failures. >> >> Did someone slap a firewall rule on the box and we suddenly cannot accept >> SIP messages? Great! One of the other nodes in the cluster will be sending >> us test SIP traffic on occasion and when we see it doesn't work anymore, we >> shut down and another node takes over. >> >> >>> I'd just like to have HA that works everytime and everywhere and try >>> >> >> So does everyone else. That is why I want to write mod_ha_cluster. >> Because what is there right now is overly complex, difficult to configure >> and test, and does not and cannot catch all of the possible ways in which a >> FS system can fail. >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130213/214cc0b0/attachment.html From david.villasmil.work at gmail.com Wed Feb 13 13:48:56 2013 From: david.villasmil.work at gmail.com (David Villasmil) Date: Wed, 13 Feb 2013 11:48:56 +0100 Subject: [Freeswitch-users] Nibblebill database question In-Reply-To: References: Message-ID: Hello Rupa, the query if ok, the user is there: mysql> select * from sip_users; +----+----------+----------+----------------+------------+--------------+--------------+ | id | username | password | rateplan | balance | domain | max_channels | +----+----------+----------+----------------+------------+--------------+--------------+ | 1 | 1002 | 1234 | retail_default | 100.000000 | 192.168.1.44 | 1 | +----+----------+----------+----------------+------------+--------------+--------------+ 1 row in set (0.00 sec) and the actual sql executed by nibblebill: mysql> SELECT balance AS nibble_balance FROM sip_users WHERE id='1'; +----------------+ | nibble_balance | +----------------+ | 100.000000 | +----------------+ 1 row in set (0.00 sec) so that's ok.. any othe ideas? Thanks a lot for your help! David On Wed, Feb 13, 2013 at 3:35 AM, Rupa Schomaker wrote: > > On Tue, Feb 12, 2013 at 5:26 PM, David Villasmil < > david.villasmil.work at gmail.com> wrote: > >> call goes out fine... no billing though: >> >> >> 2013-02-12 19:21:49.844277 [DEBUG] mod_nibblebill.c:504 22 seconds passed >> since last bill time of 2013-02-12 19:21:27 >> 2013-02-12 19:21:49.844277 [DEBUG] mod_nibblebill.c:521 Billing $0.018983 >> to 1 (Call: d442dc6b-a3a7-423a-8450-598d17ca43b3 / 0.000000 so far) >> 2013-02-12 19:21:49.844277 [DEBUG] mod_nibblebill.c:351 Doing update query >> [UPDATE sip_users SET balance=balance-0.018983 WHERE ='1'] >> 2013-02-12 19:21:49.844277 [CRIT] mod_nibblebill.c:535 Failed to log to >> database! >> 2013-02-12 19:21:49.844277 [DEBUG] mod_nibblebill.c:378 Doing lookup query >> [SELECT balance AS nibble_balance FROM sip_users WHERE ='1'] >> 2013-02-12 19:21:49.844277 [ERR] mod_nibblebill.c:380 Error running this >> query: [SELECT balance AS nibble_balance FROM sip_users WHERE ='1'] >> >> >> the select go fine if executed "by hand" >> > > None of those SQL statements are valid. (hint: WHERE = '1' is no good). > Been a while since I hacked/worked with nibblebill, but I'm pretty sure > the 'id' field isn't not being filled in the sql for some reason. > > > -- > -Rupa > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130213/3f2bbaf6/attachment-0001.html From itsusama at gmail.com Wed Feb 13 14:05:56 2013 From: itsusama at gmail.com (Usama Zaidi) Date: Wed, 13 Feb 2013 16:05:56 +0500 Subject: [Freeswitch-users] ESL Mod_Conference Observations Message-ID: Hey, Looks like I found a solution to this one as well, and turns out ESL is MUCH more powerful then I thought, I've updated the wiki page for filters if anyone wants to have a peek http://wiki.freeswitch.org/wiki/Mod_event_socket#filter Short answer, filter Conference-Unique-ID $ConfUUID On Wed, Feb 13, 2013 at 3:32 PM, < freeswitch-users-request at lists.freeswitch.org> wrote: > ---------- Forwarded message ---------- > From: Usama Zaidi > To: freeswitch-users at lists.freeswitch.org > Cc: > Date: Wed, 13 Feb 2013 15:24:58 +0500 > Subject: [Freeswitch-users] ESL Mod_Conference Observations > Hi, > > Just wanted to know why don't I get any events on start/stop speaking > conference joins/leaves if I do something like this: > > $con->events("plain","all"); > $con->events("plain","CUSTOM conference::maintenance"); > $con->sendRecv("filter Unique-ID $confUUID"); > > Instead I've to do something like this: > > $con->events("plain","all"); > $con->events("plain","CUSTOM conference::maintenance"); > $con->sendRecv("filter Unique-ID $confmember1"); > $con->sendRecv("filter Unique-ID $confmember2"); > $con->sendRecv("filter Unique-ID $confmember3"); > . > . > . > > Which is useless in case you want to check if a user is entering/leaving a > particular conference in a single inbound ESL script... > > What I'm doing instead now is something on these lines: > > $con->events("plain","all"); > $con->sendRecv("filter Unique-ID $CONFNAME"); > i//psuedocode > if event equals PRESENCE_IN and presence-call-direction equals inbound then > { > execute API command conference list confname > parse uuid from members add a filter update the hashtable for existing > confmembers > } > //once we have the uuid filter added we'll get to know if that user > leaves, so the logic for removing the filter is straight forward > if event equals CUSTOM and subtype is CONFERENCE::MAINTANANCE > { > if event action=del-member remove filter for UUID > } > > I don't get ANY event if I'm filtering on conference UUID... I think that > needs to be fixed, because its mod_conference which is throwing all those > events. > > Regards. > > -- > I'd love to change the world, but they wont gimme the source code to it > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130213/5557b806/attachment.html From gozdal at gmail.com Wed Feb 13 14:16:16 2013 From: gozdal at gmail.com (Marcin Gozdalik) Date: Wed, 13 Feb 2013 12:16:16 +0100 Subject: [Freeswitch-users] High Availability Cluster Module for FreeSWITCH Message-ID: 2013/2/13 Eliot Gable : > On Tue, Feb 12, 2013 at 5:21 PM, Marcin Gozdalik wrote: > What you can do is have Node A broadcast its state many times per second. > Nodes B, C, etc, all listen for that state. They keep track of the state and > when they *stop seeing* Node A, they know *something* failed. It's that > simple. I'm not arguing with that, I'm arguing something slightly different: - sometimes B and C can't agree whether A is up or not (B sees A, C does not see A) - B and C have to agree on who will take over from A - when B and C on who takes over from A (let it be B) it might possibly happen that *during* takeover B goes down and C must be able to recognize this situation and take over both from A and B Also it is a challenge to define what it actually means that a node is down. Most of the situations are clear enough, but some are tricky to check. All this are *solvable* problems, I just didn't see how those are addressed, but probably I wasn't looking hard enough. >> By "impossible" I mean handling all such gray areas. Certainly if >> you'll power down the box or unplug Ethernet it is possible to migrate >> live calls to some other box. > > > There are no such "gray" areas. That is just a fantasy you have. Everything > in computing is black or white, true or false. If you don't know, you test > and become certain. If you are prevented from accurately testing because > (for example) you cannot see the node at all anymore or interact with it in > any way, you assume the worst and nuke the box (STONITH). It's not so simple in computing - perfectly deterministic programs can exhibit chaotic behavior. The "gray" area I'm writing about is that the tests are always somewhat inaccurate, i.e. all tests tell OK, yet the software being tested does not perform its functions for the end user. > Let's say the FS box runs out of memory. Great! I designed mod_ha_cluster to [snip] > shut down and another node takes over. I agree with what you've written. I'm nitpicking that sometimes you get a timely heartbeat and yet the service in question (be it FS or any other software) does not function properly for one reason or another. You make another test to e.g. check if it responds to SIP, yet still there are real conditions under which SIP works and yet real calls can't get through (or at least some of them). Consider configuration de-synchronization, AAA subsystem malfunctioning, one node of DB going down. They are not core FS elements but IMGO when speaking of competing with commercial offerings all have to be taken into account. >> I'd just like to have HA that works everytime and everywhere and try > > > So does everyone else. That is why I want to write mod_ha_cluster. Because > what is there right now is overly complex, difficult to configure and test, > and does not and cannot catch all of the possible ways in which a FS system > can fail. Frankly I'd love to get this conversation going into more constructive area than just pure speculations. What about some kind of blueprint that would describe how such a module could work and what scenarios were considered and how such a module would deal with them? -- Marcin Gozdalik From avi at avimarcus.net Wed Feb 13 14:58:16 2013 From: avi at avimarcus.net (Avi Marcus) Date: Wed, 13 Feb 2013 13:58:16 +0200 Subject: [Freeswitch-users] Nibblebill database question In-Reply-To: References: Message-ID: Your logs showed the queries run were: UPDATE sip_users SET balance=balance-0.018983 WHERE ='1' SELECT balance AS nibble_balance FROM sip_users WHERE ='1' You just tried now.. "WHERE *id*='1' " -- they are not the same query... -Avi On Wed, Feb 13, 2013 at 12:48 PM, David Villasmil < david.villasmil.work at gmail.com> wrote: > Hello Rupa, > > the query if ok, the user is there: > > mysql> select * from sip_users; > > +----+----------+----------+----------------+------------+--------------+--------------+ > | id | username | password | rateplan | balance | domain | > max_channels | > > +----+----------+----------+----------------+------------+--------------+--------------+ > | 1 | 1002 | 1234 | retail_default | 100.000000 | 192.168.1.44 | > 1 | > > +----+----------+----------+----------------+------------+--------------+--------------+ > 1 row in set (0.00 sec) > > > and the actual sql executed by nibblebill: > > mysql> SELECT balance AS nibble_balance FROM sip_users WHERE id='1'; > +----------------+ > | nibble_balance | > +----------------+ > | 100.000000 | > +----------------+ > 1 row in set (0.00 sec) > > > so that's ok.. > > > any othe ideas? > > Thanks a lot for your help! > > > David > > > > On Wed, Feb 13, 2013 at 3:35 AM, Rupa Schomaker wrote: > >> >> On Tue, Feb 12, 2013 at 5:26 PM, David Villasmil < >> david.villasmil.work at gmail.com> wrote: >> >>> call goes out fine... no billing though: >>> >>> >>> 2013-02-12 19:21:49.844277 [DEBUG] mod_nibblebill.c:504 22 seconds >>> passed since last bill time of 2013-02-12 19:21:27 >>> 2013-02-12 19:21:49.844277 [DEBUG] mod_nibblebill.c:521 Billing >>> $0.018983 to 1 (Call: d442dc6b-a3a7-423a-8450-598d17ca43b3 / 0.000000 so >>> far) >>> 2013-02-12 19:21:49.844277 [DEBUG] mod_nibblebill.c:351 Doing update >>> query >>> [UPDATE sip_users SET balance=balance-0.018983 WHERE ='1'] >>> 2013-02-12 19:21:49.844277 [CRIT] mod_nibblebill.c:535 Failed to log to >>> database! >>> 2013-02-12 19:21:49.844277 [DEBUG] mod_nibblebill.c:378 Doing lookup >>> query >>> [SELECT balance AS nibble_balance FROM sip_users WHERE ='1'] >>> 2013-02-12 19:21:49.844277 [ERR] mod_nibblebill.c:380 Error running this >>> query: [SELECT balance AS nibble_balance FROM sip_users WHERE ='1'] >>> >>> >>> the select go fine if executed "by hand" >>> >> >> None of those SQL statements are valid. (hint: WHERE = '1' is no good). >> Been a while since I hacked/worked with nibblebill, but I'm pretty sure >> the 'id' field isn't not being filled in the sql for some reason. >> >> >> -- >> -Rupa >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130213/457254a9/attachment.html From martyn at magiccow.co.uk Wed Feb 13 15:26:29 2013 From: martyn at magiccow.co.uk (Martyn Davies) Date: Wed, 13 Feb 2013 12:26:29 +0000 Subject: [Freeswitch-users] Conference moderator via the ESL API? In-Reply-To: References: Message-ID: Thanks, that works. On 11 February 2013 22:52, Michael Collins wrote: > You can set the moderator flag on a member prior to his/her entry into the > conference, but not after they've entered. Instead of using conference dial > you could do something like this: > > originate user/1000 &conference(confname++flags{moderator}) > or > originate sofia/gateway/18005551212 &conference(confname++flags{moderator}) > > Those are how you'd type them at fs_cli. You can also use event socket/ESL > and the api (or bgapi) event socket command to send the originate. > > -MC > > On Sat, Feb 9, 2013 at 1:01 AM, Martyn Davies wrote: > >> Is there a way to switch the moderator flag on/off for members of an >> active conference, and can this be done from the API? >> >> Also, can the moderator flag be switched on at "conference dial..."? The >> only way I can see to control mod status seems to be to define two profiles >> in the config. >> >> Regards, >> Martyn >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130213/6a0f37f5/attachment-0001.html From steveayre at gmail.com Wed Feb 13 15:27:37 2013 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 13 Feb 2013 12:27:37 +0000 Subject: [Freeswitch-users] ooH323 vs mod_h323 In-Reply-To: <267C8398-6CA3-442E-B973-BE096D728B28@freeswitch.org> References: <2BD0DC60-D428-4323-A020-D9AD20E76553@freeswitch.org> <4C2EA249.4070704@coppice.org> <267C8398-6CA3-442E-B973-BE096D728B28@freeswitch.org> Message-ID: I was just looking at this library to play with the idea of writing a mod_ooh323c. It looks like the library may have forked away from the version used in Asterisk though? The latest version on the website ( http://www.obj-sys.com/telephony-objective.shtml) is 0.9.4, but cannot hangup with a Q.931 hangup cause. The CVS repository ( sourceforge.net/projects/ooh323c) doesn't seem up-to-date since that's only holding 0.9.0. The version in asterisk has been patched to add a q931cause parameter to ooHangCall so it can - but it seems to be based off 'v0.8.3m' and I don't know if that's before the FLOSS exception was added. So does anyone more familiar with the library know whether the repository had been moved elsewhere, or whether there's a more recent version somewhere else, and whether the one bundled in Asterisk would be usable? -Steve On 5 July 2010 15:28, Brian West wrote: > It has a FLOSS exception in the GPL license. > > /b > > On Jul 2, 2010, at 9:36 PM, Steve Underwood wrote: > > > On 07/03/2010 04:28 AM, Brian West wrote: > >> DOH... btw ooh323 is now license compatible with FreeSWITCH if anyone > wants to write one. > >> > >> /b > >> > > Where did you see that? The current ooh323c tarball still says GPL 2. > > > > Steve > > > _______________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130213/638df4b9/attachment.html From gmaruzz at gmail.com Wed Feb 13 17:07:01 2013 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Wed, 13 Feb 2013 15:07:01 +0100 Subject: [Freeswitch-users] High Availability Cluster Module for FreeSWITCH In-Reply-To: References: Message-ID: I'm very interested too. Lets the ball roll. -giovanni On Wed, Feb 13, 2013 at 12:16 PM, Marcin Gozdalik wrote: Frankly I'd love to get this conversation going into more constructive > area than just pure speculations. What about some kind of blueprint > that would describe how such a module could work and what scenarios > were considered and how such a module would deal with them? > > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130213/dc2c715c/attachment.html From jsun at junsun.net Wed Feb 13 10:02:12 2013 From: jsun at junsun.net (Jun Sun) Date: Tue, 12 Feb 2013 23:02:12 -0800 Subject: [Freeswitch-users] how to end conference after initiator hangs up (dialout style) In-Reply-To: References: <5116D31F.8030807@junsun.net> Message-ID: <511B3A74.1010903@junsun.net> Thanks, Michael. It works! However, an unintended side effect is that "originate" command returns much earlier than the "conference" command. "Conference" command waits until the initiator picks up phone before dialing others (possibly in background with "bgdial"). "originate" command would return almost immediately and start dialing others without waiting for the initiator picking up phone. Is there a way to force "originate" command to delay the return until the endpoint picking up the phone? I'm sure all this is possible to events and scripts. But I just hope there is a shortcut here. Cheers. Jun On 2/12/2013 4:30 PM, Michael Collins wrote: > You could have the initiator put into the conference room via the > originate command, in which case you could set flags: > > originate sofia/internal/$myNumber at x.x.x.x:5060 > &conference("conference_test++flags{endconf}") > > -MC > > On Sat, Feb 9, 2013 at 2:52 PM, Jun Sun > wrote: > > > I have set up a conferencing call node using dialout style. That is, I > use the following commands to bridge everyone into a conference: > > conference $myRoom dial sofia/internal/$myNumber at x.x.x.x:5060 $myNumber > conference_test"; > conference $myRoom dial sofia/internal/$theirNumber1 at x.x.x.x:5060 > $myNumber conference_test"; > conference $myRoom dial sofia/internal/$theirNumber2 at x.x.x.x:5060 > $myNumber conference_test"; > ... > > Right now things are working fine except for one thing. I like to end > the conference (i.e., all legs are disconnected) when the initiator > ($myNumber) hangs up. But it is not doing that (understandably so!). > > How can I achieve that? > > I searched around and it seems most conf call controls only apply to > dial-in style conference calls. > > Thanks for any pointers! > > Jun > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 krice at freeswitch.org Wed Feb 13 17:55:33 2013 From: krice at freeswitch.org (Ken Rice) Date: Wed, 13 Feb 2013 08:55:33 -0600 Subject: [Freeswitch-users] best freeswitch timer In-Reply-To: Message-ID: Actually it defaults to using timerfd if its enabled in your kernel.... There should be no debate on that, I reviewed that code with tony... If you don?t have a timerfd enabled kernel it uses the old default soft timer... No you cant tell the difference in there currently with just a show timer command, however that?s likely to change... Are you running like centos6 or debian squeeze with a stock kernel? Then you likely have timerfd and are using it unless you changed the configs to do something else... Theres always a lot of debate around this, but the fact is, unless you are seeing crappy audio issues, you probably shouldn?t mess with the timers On 2/13/13 4:31 AM, "Steven Ayre" wrote: > 'show timer' shows all loaded timer modules. That shows none are loaded. > > CORE_SOFTTIMER_MODULE is the soft timer built into the FS core... it may or > may not use timerfd, depending on your system. > > You pick which timer to actually use in your configuration files... for > example the?rtp-timer-name Sofia profile param. > > Use the timer_test command to see which timer actually works best for you (ie > most accurate, least jitter). > http://wiki.freeswitch.org/wiki/Mod_commands#timer_test > > -Steve > > > > On 12 February 2013 22:48, jay binks wrote: >> so does : >> >> freeswitch at default> show timer >> type,name,ikey >> timer,soft,CORE_SOFTTIMER_MODULE >> >> 1 total. >> >> on FreeSWITCH Version 1.2.6+git~20130104T154559Z~a4247651ca (git >> a424765 2013-01-04 15:45:59Z) >> >> indicate it will be using timerfd calls or is this the old softtimer ? >> >> On 13 February 2013 02:21, Ken Rice wrote: >>> > The default one is the best one on linux with kernels starting around >>> > 2.6.32... Anything after that and you are good to go... >>> > >>> > The other timers are left around for other platforms >>> > >>> > K >>> > >>> > >>> > >>> > On 2/12/13 10:04 AM, "Fran?ois Delawarde" >>> > wrote: >>> > >>>> >> Hello, >>> > >>> > Could someone knowing the different FS timers clarify the >>>> >> difference >>> > between those: >>> > >>> > - default core softtimer >>> > - core heavy timer (does >>>> >> this option still exists?) >>> > - posix timer >>> > - timerfd >>> > >>> > Can any of those provide a >>>> >> significant performance improvement, or is it >>> > just not worth switching from >>>> >> the default? >>> > >>> > In any case, which one would be the recommended/best one under >>>> >> Linux >>> > with a relatively recent kernel >>>> >> (>3.0)? >>> > >>> > Thanks, >>> > Fran?ois. >>> > >>> > >>> > >>> > ________________________________________________ >>>> >> _________________________ >>> > Professional FreeSWITCH Consulting >>>> >> Services: >>> > consulting at freeswitch.org >>> > http://www.freeswitchsolutions.com >>> > >>> > FreeSW >>>> >> ITCH-powered IP PBX: The CudaTel Communication >>>> >> Server >>> > >>> > >>> > Official FreeSWITCH >>>> >> Sites >>> > http://www.freeswitch.org >>> > http://wiki.freeswitch.org >>> > http://www.cluecon. >>>> >> com >>> > >>> > FreeSWITCH-users mailing >>>> >> list >>> > FreeSWITCH-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman >>>> >> /listinfo/freeswitch-users >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/opt >>>> >> ions/freeswitch-users >>> > http://www.freeswitch.org >>> > >>> > >>> > -- >>> > Ken >>> > http://www.FreeSWITCH.org >>> > http://www.ClueCon.com >>> > http://www.OSTAG.org >>> > irc.freenode.net #freeswitch >>> > >>> > >>> > >>> > _________________________________________________________________________ >>> > Professional FreeSWITCH Consulting Services: >>> > consulting at freeswitch.org >>> > http://www.freeswitchsolutions.com >>> > >>> > >>> > >>> > >>> > Official FreeSWITCH Sites >>> > http://www.freeswitch.org >>> > http://wiki.freeswitch.org >>> > http://www.cluecon.com >>> > >>> > 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 >> >> >> >> -- >> Sincerely >> >> Jay >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130213/701f7a6f/attachment-0001.html From nasida at live.ru Wed Feb 13 18:17:25 2013 From: nasida at live.ru (Yuriy Nasida) Date: Wed, 13 Feb 2013 19:17:25 +0400 Subject: [Freeswitch-users] 486 Busy Here with Reason: Q.850; cause=16; text="NORMAL_CLEARING" Message-ID: Hi guys, I see that my FS sends sip message "486 Busy Here" with Reason: Q.850; cause=16;text="NORMAL_CLEARING". Why 16 and "NORMAL_CLEARING ? Probably it should be 17 "user busy". One of my carrier play a "fast busy" (number not available) and blames Q.850 field. Any advice are welcome.Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130213/216b9a52/attachment.html From kbdfck at gmail.com Wed Feb 13 18:28:24 2013 From: kbdfck at gmail.com (Dmitry Sytchev) Date: Wed, 13 Feb 2013 19:28:24 +0400 Subject: [Freeswitch-users] 486 Busy Here with Reason: Q.850; cause=16; text="NORMAL_CLEARING" In-Reply-To: References: Message-ID: I observe similar behavior on my 1.3.0+git~20121012T011602Z~ff7def219f (git ff7def2 2012-10-12 01:16:02Z). Depending on upstream/downstream SIP stack implementation this leads to unpredictable interpretation of cause codes. 2013/2/13 Yuriy Nasida > Hi guys, > > I see that my FS sends sip message "486 Busy Here" with Reason: Q.850; > cause=16;text="NORMAL_CLEARING". Why 16 and "NORMAL_CLEARING ? Probably it > should be 17 "user busy". One of my carrier play a "fast busy" (number not > available) and blames Q.850 field. > > Any advice are welcome. > Thanks. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Best regards, Dmitry Sytchev, IT Engineer -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130213/68bdb44d/attachment.html From david.villasmil.work at gmail.com Wed Feb 13 18:33:43 2013 From: david.villasmil.work at gmail.com (David Villasmil) Date: Wed, 13 Feb 2013 16:33:43 +0100 Subject: [Freeswitch-users] Nibblebill database question In-Reply-To: References: Message-ID: hello Avi, that's right, but nevermind that, that's the output when I was trying something else: 2013-02-12 19:32:07.784156 [INFO] mod_nibblebill.c:498 Beginning new billing on 58e6762b-c049-470e-84f7-519a815ab6f0 2013-02-12 19:32:07.784156 [CRIT] mod_nibblebill.c:535 Failed to log to database! 2013-02-12 19:32:07.784156 [ERR] mod_nibblebill.c:380 Error running this query: [SELECT balance AS nibble_balance FROM sip_users WHERE id='1'] On Wed, Feb 13, 2013 at 12:58 PM, Avi Marcus wrote: > Your logs showed the queries run were: > > UPDATE sip_users SET balance=balance-0.018983 WHERE ='1' > SELECT balance AS nibble_balance FROM sip_users WHERE ='1' > > You just tried now.. "WHERE *id*='1' " -- they are not the same query... > > -Avi > > > > On Wed, Feb 13, 2013 at 12:48 PM, David Villasmil < > david.villasmil.work at gmail.com> wrote: > >> Hello Rupa, >> >> the query if ok, the user is there: >> >> mysql> select * from sip_users; >> >> +----+----------+----------+----------------+------------+--------------+--------------+ >> | id | username | password | rateplan | balance | domain | >> max_channels | >> >> +----+----------+----------+----------------+------------+--------------+--------------+ >> | 1 | 1002 | 1234 | retail_default | 100.000000 | 192.168.1.44 | >> 1 | >> >> +----+----------+----------+----------------+------------+--------------+--------------+ >> 1 row in set (0.00 sec) >> >> >> and the actual sql executed by nibblebill: >> >> mysql> SELECT balance AS nibble_balance FROM sip_users WHERE id='1'; >> +----------------+ >> | nibble_balance | >> +----------------+ >> | 100.000000 | >> +----------------+ >> 1 row in set (0.00 sec) >> >> >> so that's ok.. >> >> >> any othe ideas? >> >> Thanks a lot for your help! >> >> >> David >> >> >> >> On Wed, Feb 13, 2013 at 3:35 AM, Rupa Schomaker wrote: >> >>> >>> On Tue, Feb 12, 2013 at 5:26 PM, David Villasmil < >>> david.villasmil.work at gmail.com> wrote: >>> >>>> call goes out fine... no billing though: >>>> >>>> >>>> 2013-02-12 19:21:49.844277 [DEBUG] mod_nibblebill.c:504 22 seconds >>>> passed since last bill time of 2013-02-12 19:21:27 >>>> 2013-02-12 19:21:49.844277 [DEBUG] mod_nibblebill.c:521 Billing >>>> $0.018983 to 1 (Call: d442dc6b-a3a7-423a-8450-598d17ca43b3 / 0.000000 so >>>> far) >>>> 2013-02-12 19:21:49.844277 [DEBUG] mod_nibblebill.c:351 Doing update >>>> query >>>> [UPDATE sip_users SET balance=balance-0.018983 WHERE ='1'] >>>> 2013-02-12 19:21:49.844277 [CRIT] mod_nibblebill.c:535 Failed to log to >>>> database! >>>> 2013-02-12 19:21:49.844277 [DEBUG] mod_nibblebill.c:378 Doing lookup >>>> query >>>> [SELECT balance AS nibble_balance FROM sip_users WHERE ='1'] >>>> 2013-02-12 19:21:49.844277 [ERR] mod_nibblebill.c:380 Error running >>>> this query: [SELECT balance AS nibble_balance FROM sip_users WHERE ='1'] >>>> >>>> >>>> the select go fine if executed "by hand" >>>> >>> >>> None of those SQL statements are valid. (hint: WHERE = '1' is no good). >>> Been a while since I hacked/worked with nibblebill, but I'm pretty sure >>> the 'id' field isn't not being filled in the sql for some reason. >>> >>> >>> -- >>> -Rupa >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130213/d332609e/attachment.html From x.liu at hw.ac.uk Wed Feb 13 18:44:24 2013 From: x.liu at hw.ac.uk (x.liu) Date: Wed, 13 Feb 2013 15:44:24 +0000 Subject: [Freeswitch-users] issues in simultaneous speech recognition and TTS via mod_unimrcp In-Reply-To: References: Message-ID: <511BB4D8.70505@hw.ac.uk> Hello, I am trying to do simultaneous speech recognition and TTS via mod_unimrcp using detect_speech and speak. What I want to achieve is to enable a user to say something like "shut up" to stop TTS rather than using standard barge-in to stop TTS. When I spoke something during playing a sentence, I noticed there were some recognition results from FS log, but my ESL app didn't get any DETECTED_SPEECH or BEGIN_SPEAKING events. Has any body had similar experiences before? From Arsen Chaloyan's comments at 11 July 2011: "If I'm not mistaken, neither Asterisk nor FreeSWITCH are currently capable to play a TTS prompt while making a recognition. However, I don't see any conceptual problem here. This is somewhat not implemented yet for those open source projects. Other UniMRCP based implementations do this job very well." I am wondering if the current mod_unimrcp does not support concurrent recognition and tts or if I am not using it properly? Thanks! Xing ----- Sunday Times Scottish University of the Year 2011-2013 Top in the UK for student experience Fourth university in the UK and top in Scotland (National Student Survey 2012) We invite research leaders and ambitious early career researchers to join us in leading and driving research in key inter-disciplinary themes. Please see www.hw.ac.uk/researchleaders for further information and how to apply. Heriot-Watt University is a Scottish charity registered under charity number SC000278. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130213/d14756a8/attachment-0001.html From chris at gonumina.com Wed Feb 13 19:02:37 2013 From: chris at gonumina.com (Chris Ferreira) Date: Wed, 13 Feb 2013 11:02:37 -0500 Subject: [Freeswitch-users] Best current ATA and Setup for Faxing Message-ID: Hi Folks, We had previously been working with the Cisco SPA112 for T.38 Faxing through Freeswitch and out to Flowroute. At the time we didn't get it working 100% perfectly. I now have one of these SPA112's that I need to replace because someone fried it by plugging in a much larger power supply by accident. Is my scenario above still a good starting point for Faxing or should I be looking at another ATA or provider other than Flowroute? Also, one more question, and this may be better suited to the business list (let me know if I should move it there), but we are looking for a part time as needed remote or local Freeswitch Admin / Engineer / Tech. Our previous person is MIA. Thanks Everyone! -Chris -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130213/85da7346/attachment.html From jnvines at gmail.com Wed Feb 13 19:14:50 2013 From: jnvines at gmail.com (Nick Vines) Date: Wed, 13 Feb 2013 08:14:50 -0800 Subject: [Freeswitch-users] Best current ATA and Setup for Faxing In-Reply-To: References: Message-ID: <0B7A549A-767A-4DA6-B732-9999DC6B1B48@gmail.com> I havent used the cisco atas, but I've had good success with the grand stream ht502s and ht702s. My preferred faxing carrier is also flowroute. Nick On Feb 13, 2013, at 8:02 AM, Chris Ferreira wrote: > Hi Folks, > > > We had previously been working with the Cisco SPA112 for T.38 Faxing through Freeswitch and out to Flowroute. At the time we didn't get it working 100% perfectly. > > > I now have one of these SPA112's that I need to replace because someone fried it by plugging in a much larger power supply by accident. > > > Is my scenario above still a good starting point for Faxing or should I be looking at another ATA or provider other than Flowroute? > > > > > > Also, one more question, and this may be better suited to the business list (let me know if I should move it there), but we are looking for a part time as needed remote or local Freeswitch Admin / Engineer / Tech. Our previous person is MIA. > > > > > Thanks Everyone! > > > > -Chris > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 levend.sayar at karel.com.tr Wed Feb 13 20:13:49 2013 From: levend.sayar at karel.com.tr (Levend Sayar) Date: Wed, 13 Feb 2013 17:13:49 +0000 Subject: [Freeswitch-users] SRTP disabling In-Reply-To: References: Message-ID: Hi again. I checked the FS code and i see that rpm i am using is built with ZRTP disabled. There is only one place that "sip_secure_media" on the code. That is on switch_rtp.c #ifdef ENABLE_ZRTP if (zrtp_on) { switch_rtp_t *master_rtp_session = NULL; int initiator = 0; const char *zrtp_enabled = switch_channel_get_variable(channel, "zrtp_secure_media"); const char *srtp_enabled = switch_channel_get_variable(channel, "sip_secure_media"); So since ENABLE_ZRTP is 0, i don't have chance to use "sip_secure_media" variable. Is there any other variable that i can use and make sofia module not to choose SRTP ? _lvnd_ {^_^} On Thu, 2013-02-07 at 13:54 +0000, Levend Sayar wrote: But the very same phone calls another phone and talk with RTP, not SRTP if the peer does not accept SRTP Here is the SDP offer by the same phone v=0 o=- 20186 20186 IN IP4 192.168.173.69 s=SDP data c=IN IP4 192.168.173.69 t=0 0 m=audio 11782 RTP/SAVP 0 8 18 9 101 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:NzFmYjdiMjk1OTY2ODQwYzExZjM0ZmE2NGM0YWMw a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:M2MxMTE2OWFjOGY2ZjEwADEzZmZkNzAxNjRlMzFm a=crypto:3 F8_128_HMAC_SHA1_80 inline:NjkzZDg2Mjk0ZTkxMjg1YzdmYjFiNjRlMmFhNGFm a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:9 G722/8000 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=ptime:20 a=sendrecv And here is the SDP answer sent by the other phone v=0 o=- 20029 20029 IN IP4 192.168.173.65 s=SDP data c=IN IP4 192.168.173.65 t=0 0 m=audio 11794 RTP/SAVP 0 101 a=rtpmap:0 PCMU/8000 a=sendrecv a=ptime:20 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 -- _lvnd_ {^_^} On Thu, 2013-02-07 at 13:37 +0000, Steven Ayre wrote: m=audio 11780 RTP/SAVP 0 8 18 9 101 RTP/SAVP means SRTP is mandatory. You need to reconfigure the phone. If the phone sends RTP/AVP then that means plain RTP, and RTP/AVP with a a=crypto attribute means SRTP is optional. -Steve On 7 February 2013 13:26, Levend Sayar > wrote: Below is the SDP offer sent by the phone. v=0 o=- 20185 20185 IN IP4 192.168.173.69 s=SDP data c=IN IP4 192.168.173.69 t=0 0 m=audio 11780 RTP/SAVP 0 8 18 9 101 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:MTM2MjVhMGI1NDZjYmRjADU5NWVjNGVkNTNlYzA1 a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:YmExYmZhNQAzN2ZjNDgzYTRkNGU2ZjFiN2Q0MmE3 a=crypto:3 F8_128_HMAC_SHA1_80 inline:N2Q2NTRiYQAxZjA3MWY3ZjI1YTI5NjIyM2FjODYw a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:9 G722/8000 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=ptime:20 a=sendrecv And below is the SDP answer sent by FS v=0 o=FreeSWITCH 1360230601 1360230602 IN IP4 192.168.169.114 s=FreeSWITCH c=IN IP4 192.168.169.114 t=0 0 m=audio 12532 RTP/SAVP 9 101 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:l8v0R64H7CP0vEx9j0Ycdbob8bgMCpLDppWGy7Dy _lvnd_ {^_^} On Thu, 2013-02-07 at 13:09 +0000, Steven Ayre wrote: What I mean is you'll see two separate m=audio lines within the callee's SDP, one for 'RTP/AVP' and one for 'SRTP/AVP'. If there is no m=audio line for RTP/AVP the caller won't know of a port that's expecting RTP. So if the callee only sends SRTP/AVP the caller can't send RTP. Can you show us the SDP being sent by the phone? -Steve On 7 February 2013 11:01, Levend Sayar > wrote: Thanx Steven. Caller makes the offer for SDP but callee chooses whatever it wants. So caller can offer SRTP but callee can prefer not to talk encrypted. In our case I want FS to choose non secure media. Phone will offer SRTP on the conference call but FS must prefer RTP, not SRTP. _lvnd_ {^_^} On 7 ?ub 2013, at 11:13, "Steven Ayre" > wrote: It's also going to rely on the phone actually offering RTP/AVP as well as SRTP/AVP in their SDP - without that there'd be nowhere to send insecure RTP. -Steve On 6 February 2013 16:09, Levend Sayar > wrote: Thanks Daniel for the reply. I tried But did not work. Upon your reply I also tried But did not work either. I am doing something wrong ? _lvnd_ {^_^} On 6 ?ub 2013, at 18:00, "Daniel Ivanov" > wrote: Of course you can. Just set the secure_media var to false and you will be srtp-free in sip. On Feb 5, 2013 6:06 PM, "Levend Sayar" > wrote: Hi all. I am using FS as a conference server. Some of my phones are using SRTP , some of them not. Both type of phone can join a conference. FS can talk to each peer with SRTP or not depending on the phone itself. My question: Is it possible to disable SRTP on FS ? I suppose if i can disable SRTP, FS will talk without SRTP with each phone whether they are using SRTP or not. TIA plain text document attachment (ATT00001) _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130213/5812bf6c/attachment-0001.html From msc at freeswitch.org Wed Feb 13 20:14:53 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 13 Feb 2013 09:14:53 -0800 Subject: [Freeswitch-users] FreeSWITCH Community Conference Call Today Message-ID: Hello folks, Today's conference call agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2013_02_13 I would like to talk about the interesting HA thread from the mailing list. Depending on the interest level there we will talk about that for the conference topic. I really want to know how people feel about this. Links to the ML threads are on the agenda page under "Items Needing Discussion." Thanks! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130213/3bf56d44/attachment.html From steveayre at gmail.com Wed Feb 13 20:19:29 2013 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 13 Feb 2013 17:19:29 +0000 Subject: [Freeswitch-users] 486 Busy Here with Reason: Q.850; cause=16; text="NORMAL_CLEARING" In-Reply-To: References: Message-ID: Please post a debug-level log of the call. How are you hanging up the call? If with the 'hangup' app what parameters do you use? In particular if you bridge and it fails with 486 then hangup then the SIP code sent may be from the last bridge cause while the Q.850 cause is the one given to hangup. If that's the case try setting sip_ignore_remote_cause=true before the bridge and it'll only send the Q.850->SIP mapping (https://tools.ietf.org/html/rfc3398). Note there's no SIP code mapping for 16 though. If hanging up without answering the call use a different cause. -Steve On 13 February 2013 15:17, Yuriy Nasida wrote: > Hi guys, > > I see that my FS sends sip message "486 Busy Here" with Reason: Q.850; > cause=16;text="NORMAL_CLEARING". Why 16 and "NORMAL_CLEARING ? Probably it > should be 17 "user busy". One of my carrier play a "fast busy" (number not > available) and blames Q.850 field. > > Any advice are welcome. > Thanks. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130213/e75f0e31/attachment.html From steveayre at gmail.com Wed Feb 13 20:33:08 2013 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 13 Feb 2013 17:33:08 +0000 Subject: [Freeswitch-users] Nibblebill database question In-Reply-To: References: Message-ID: How are you setting the DSN? The param name mentioned earlier is wrong. Reading the source the correct parameter is "odbc-dsn". Looks like the wiki documentation is incorrect. -Steve On 13 February 2013 15:33, David Villasmil wrote: > hello Avi, > > that's right, but nevermind that, that's the output when I was trying > something else: > > 2013-02-12 19:32:07.784156 [INFO] mod_nibblebill.c:498 Beginning new > billing on 58e6762b-c049-470e-84f7-519a815ab6f0 > 2013-02-12 19:32:07.784156 [CRIT] mod_nibblebill.c:535 Failed to log to > database! > 2013-02-12 19:32:07.784156 [ERR] mod_nibblebill.c:380 Error running this > query: [SELECT balance AS nibble_balance FROM sip_users WHERE id='1'] > > > On Wed, Feb 13, 2013 at 12:58 PM, Avi Marcus wrote: > >> Your logs showed the queries run were: >> >> UPDATE sip_users SET balance=balance-0.018983 WHERE ='1' >> SELECT balance AS nibble_balance FROM sip_users WHERE ='1' >> >> You just tried now.. "WHERE *id*='1' " -- they are not the same query... >> >> -Avi >> >> >> >> On Wed, Feb 13, 2013 at 12:48 PM, David Villasmil < >> david.villasmil.work at gmail.com> wrote: >> >>> Hello Rupa, >>> >>> the query if ok, the user is there: >>> >>> mysql> select * from sip_users; >>> >>> +----+----------+----------+----------------+------------+--------------+--------------+ >>> | id | username | password | rateplan | balance | domain >>> | max_channels | >>> >>> +----+----------+----------+----------------+------------+--------------+--------------+ >>> | 1 | 1002 | 1234 | retail_default | 100.000000 | 192.168.1.44 >>> | 1 | >>> >>> +----+----------+----------+----------------+------------+--------------+--------------+ >>> 1 row in set (0.00 sec) >>> >>> >>> and the actual sql executed by nibblebill: >>> >>> mysql> SELECT balance AS nibble_balance FROM sip_users WHERE id='1'; >>> +----------------+ >>> | nibble_balance | >>> +----------------+ >>> | 100.000000 | >>> +----------------+ >>> 1 row in set (0.00 sec) >>> >>> >>> so that's ok.. >>> >>> >>> any othe ideas? >>> >>> Thanks a lot for your help! >>> >>> >>> David >>> >>> >>> >>> On Wed, Feb 13, 2013 at 3:35 AM, Rupa Schomaker wrote: >>> >>>> >>>> On Tue, Feb 12, 2013 at 5:26 PM, David Villasmil < >>>> david.villasmil.work at gmail.com> wrote: >>>> >>>>> call goes out fine... no billing though: >>>>> >>>>> >>>>> 2013-02-12 19:21:49.844277 [DEBUG] mod_nibblebill.c:504 22 seconds >>>>> passed since last bill time of 2013-02-12 19:21:27 >>>>> 2013-02-12 19:21:49.844277 [DEBUG] mod_nibblebill.c:521 Billing >>>>> $0.018983 to 1 (Call: d442dc6b-a3a7-423a-8450-598d17ca43b3 / 0.000000 so >>>>> far) >>>>> 2013-02-12 19:21:49.844277 [DEBUG] mod_nibblebill.c:351 Doing update >>>>> query >>>>> [UPDATE sip_users SET balance=balance-0.018983 WHERE ='1'] >>>>> 2013-02-12 19:21:49.844277 [CRIT] mod_nibblebill.c:535 Failed to log >>>>> to database! >>>>> 2013-02-12 19:21:49.844277 [DEBUG] mod_nibblebill.c:378 Doing lookup >>>>> query >>>>> [SELECT balance AS nibble_balance FROM sip_users WHERE ='1'] >>>>> 2013-02-12 19:21:49.844277 [ERR] mod_nibblebill.c:380 Error running >>>>> this query: [SELECT balance AS nibble_balance FROM sip_users WHERE ='1'] >>>>> >>>>> >>>>> the select go fine if executed "by hand" >>>>> >>>> >>>> None of those SQL statements are valid. (hint: WHERE = '1' is no >>>> good). Been a while since I hacked/worked with nibblebill, but I'm >>>> pretty sure the 'id' field isn't not being filled in the sql for some >>>> reason. >>>> >>>> >>>> -- >>>> -Rupa >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130213/4019e281/attachment-0001.html From steveayre at gmail.com Wed Feb 13 20:37:47 2013 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 13 Feb 2013 17:37:47 +0000 Subject: [Freeswitch-users] Nibblebill database question In-Reply-To: References: Message-ID: Looks like this difference is brought in by the modifications to support the generalised DSN interface. odbc-dsn would be consistent with all other modules, so I guess that's why it's happened. But... The HEAD of v1.2.stable understands db_dsn but not odbc-dsn. The HEAD of master understands odbc-dsn but not db_dsn. This is a configuration breaking behaviour, which doesn't seem to have been documented! At the very least it'd make sense for the master branch to understand both odbc-dsn and db_dsn for backwards compatibility. I'll file a Jira with a patch (it's a trivial fix) in a moment. -Steve On 13 February 2013 17:33, Steven Ayre wrote: > How are you setting the DSN? > > The param name mentioned earlier is wrong. Reading the source the correct > parameter is "odbc-dsn". Looks like the wiki documentation is incorrect. > > -Steve > > > > > On 13 February 2013 15:33, David Villasmil > wrote: > >> hello Avi, >> >> that's right, but nevermind that, that's the output when I was trying >> something else: >> >> 2013-02-12 19:32:07.784156 [INFO] mod_nibblebill.c:498 Beginning new >> billing on 58e6762b-c049-470e-84f7-519a815ab6f0 >> 2013-02-12 19:32:07.784156 [CRIT] mod_nibblebill.c:535 Failed to log to >> database! >> 2013-02-12 19:32:07.784156 [ERR] mod_nibblebill.c:380 Error running this >> query: [SELECT balance AS nibble_balance FROM sip_users WHERE id='1'] >> >> >> On Wed, Feb 13, 2013 at 12:58 PM, Avi Marcus wrote: >> >>> Your logs showed the queries run were: >>> >>> UPDATE sip_users SET balance=balance-0.018983 WHERE ='1' >>> SELECT balance AS nibble_balance FROM sip_users WHERE ='1' >>> >>> You just tried now.. "WHERE *id*='1' " -- they are not the same >>> query... >>> >>> -Avi >>> >>> >>> >>> On Wed, Feb 13, 2013 at 12:48 PM, David Villasmil < >>> david.villasmil.work at gmail.com> wrote: >>> >>>> Hello Rupa, >>>> >>>> the query if ok, the user is there: >>>> >>>> mysql> select * from sip_users; >>>> >>>> +----+----------+----------+----------------+------------+--------------+--------------+ >>>> | id | username | password | rateplan | balance | domain >>>> | max_channels | >>>> >>>> +----+----------+----------+----------------+------------+--------------+--------------+ >>>> | 1 | 1002 | 1234 | retail_default | 100.000000 | 192.168.1.44 >>>> | 1 | >>>> >>>> +----+----------+----------+----------------+------------+--------------+--------------+ >>>> 1 row in set (0.00 sec) >>>> >>>> >>>> and the actual sql executed by nibblebill: >>>> >>>> mysql> SELECT balance AS nibble_balance FROM sip_users WHERE id='1'; >>>> +----------------+ >>>> | nibble_balance | >>>> +----------------+ >>>> | 100.000000 | >>>> +----------------+ >>>> 1 row in set (0.00 sec) >>>> >>>> >>>> so that's ok.. >>>> >>>> >>>> any othe ideas? >>>> >>>> Thanks a lot for your help! >>>> >>>> >>>> David >>>> >>>> >>>> >>>> On Wed, Feb 13, 2013 at 3:35 AM, Rupa Schomaker wrote: >>>> >>>>> >>>>> On Tue, Feb 12, 2013 at 5:26 PM, David Villasmil < >>>>> david.villasmil.work at gmail.com> wrote: >>>>> >>>>>> call goes out fine... no billing though: >>>>>> >>>>>> >>>>>> 2013-02-12 19:21:49.844277 [DEBUG] mod_nibblebill.c:504 22 seconds >>>>>> passed since last bill time of 2013-02-12 19:21:27 >>>>>> 2013-02-12 19:21:49.844277 [DEBUG] mod_nibblebill.c:521 Billing >>>>>> $0.018983 to 1 (Call: d442dc6b-a3a7-423a-8450-598d17ca43b3 / 0.000000 so >>>>>> far) >>>>>> 2013-02-12 19:21:49.844277 [DEBUG] mod_nibblebill.c:351 Doing update >>>>>> query >>>>>> [UPDATE sip_users SET balance=balance-0.018983 WHERE ='1'] >>>>>> 2013-02-12 19:21:49.844277 [CRIT] mod_nibblebill.c:535 Failed to log >>>>>> to database! >>>>>> 2013-02-12 19:21:49.844277 [DEBUG] mod_nibblebill.c:378 Doing lookup >>>>>> query >>>>>> [SELECT balance AS nibble_balance FROM sip_users WHERE ='1'] >>>>>> 2013-02-12 19:21:49.844277 [ERR] mod_nibblebill.c:380 Error running >>>>>> this query: [SELECT balance AS nibble_balance FROM sip_users WHERE ='1'] >>>>>> >>>>>> >>>>>> the select go fine if executed "by hand" >>>>>> >>>>> >>>>> None of those SQL statements are valid. (hint: WHERE = '1' is no >>>>> good). Been a while since I hacked/worked with nibblebill, but I'm >>>>> pretty sure the 'id' field isn't not being filled in the sql for some >>>>> reason. >>>>> >>>>> >>>>> -- >>>>> -Rupa >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> 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 >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130213/3faa0cc8/attachment.html From steveayre at gmail.com Wed Feb 13 20:40:33 2013 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 13 Feb 2013 17:40:33 +0000 Subject: [Freeswitch-users] Nibblebill database question In-Reply-To: References: Message-ID: Specifically it's introduced by commit 5c176e8 which was resolving FS-4810, which is indeed adding support for the the generalised DSN interface. http://fisheye.freeswitch.org/changelog/freeswitch.git?showid=5c176e889d8cc2da7efb18a89782407ee5524a74 http://jira.freeswitch.org/browse/FS-4810 -Steve On 13 February 2013 17:37, Steven Ayre wrote: > Looks like this difference is brought in by the modifications to support > the generalised DSN interface. odbc-dsn would be consistent with all other > modules, so I guess that's why it's happened. > > But... > The HEAD of v1.2.stable understands db_dsn but not odbc-dsn. > The HEAD of master understands odbc-dsn but not db_dsn. > > This is a configuration breaking behaviour, which doesn't seem to have > been documented! > > At the very least it'd make sense for the master branch to understand both > odbc-dsn and db_dsn for backwards compatibility. I'll file a Jira with a > patch (it's a trivial fix) in a moment. > > -Steve > > > > > > > On 13 February 2013 17:33, Steven Ayre wrote: > >> How are you setting the DSN? >> >> The param name mentioned earlier is wrong. Reading the source the correct >> parameter is "odbc-dsn". Looks like the wiki documentation is incorrect. >> >> -Steve >> >> >> >> >> On 13 February 2013 15:33, David Villasmil < >> david.villasmil.work at gmail.com> wrote: >> >>> hello Avi, >>> >>> that's right, but nevermind that, that's the output when I was trying >>> something else: >>> >>> 2013-02-12 19:32:07.784156 [INFO] mod_nibblebill.c:498 Beginning new >>> billing on 58e6762b-c049-470e-84f7-519a815ab6f0 >>> 2013-02-12 19:32:07.784156 [CRIT] mod_nibblebill.c:535 Failed to log to >>> database! >>> 2013-02-12 19:32:07.784156 [ERR] mod_nibblebill.c:380 Error running this >>> query: [SELECT balance AS nibble_balance FROM sip_users WHERE id='1'] >>> >>> >>> On Wed, Feb 13, 2013 at 12:58 PM, Avi Marcus wrote: >>> >>>> Your logs showed the queries run were: >>>> >>>> UPDATE sip_users SET balance=balance-0.018983 WHERE ='1' >>>> SELECT balance AS nibble_balance FROM sip_users WHERE ='1' >>>> >>>> You just tried now.. "WHERE *id*='1' " -- they are not the same >>>> query... >>>> >>>> -Avi >>>> >>>> >>>> >>>> On Wed, Feb 13, 2013 at 12:48 PM, David Villasmil < >>>> david.villasmil.work at gmail.com> wrote: >>>> >>>>> Hello Rupa, >>>>> >>>>> the query if ok, the user is there: >>>>> >>>>> mysql> select * from sip_users; >>>>> >>>>> +----+----------+----------+----------------+------------+--------------+--------------+ >>>>> | id | username | password | rateplan | balance | domain >>>>> | max_channels | >>>>> >>>>> +----+----------+----------+----------------+------------+--------------+--------------+ >>>>> | 1 | 1002 | 1234 | retail_default | 100.000000 | >>>>> 192.168.1.44 | 1 | >>>>> >>>>> +----+----------+----------+----------------+------------+--------------+--------------+ >>>>> 1 row in set (0.00 sec) >>>>> >>>>> >>>>> and the actual sql executed by nibblebill: >>>>> >>>>> mysql> SELECT balance AS nibble_balance FROM sip_users WHERE id='1'; >>>>> +----------------+ >>>>> | nibble_balance | >>>>> +----------------+ >>>>> | 100.000000 | >>>>> +----------------+ >>>>> 1 row in set (0.00 sec) >>>>> >>>>> >>>>> so that's ok.. >>>>> >>>>> >>>>> any othe ideas? >>>>> >>>>> Thanks a lot for your help! >>>>> >>>>> >>>>> David >>>>> >>>>> >>>>> >>>>> On Wed, Feb 13, 2013 at 3:35 AM, Rupa Schomaker wrote: >>>>> >>>>>> >>>>>> On Tue, Feb 12, 2013 at 5:26 PM, David Villasmil < >>>>>> david.villasmil.work at gmail.com> wrote: >>>>>> >>>>>>> call goes out fine... no billing though: >>>>>>> >>>>>>> >>>>>>> 2013-02-12 19:21:49.844277 [DEBUG] mod_nibblebill.c:504 22 seconds >>>>>>> passed since last bill time of 2013-02-12 19:21:27 >>>>>>> 2013-02-12 19:21:49.844277 [DEBUG] mod_nibblebill.c:521 Billing >>>>>>> $0.018983 to 1 (Call: d442dc6b-a3a7-423a-8450-598d17ca43b3 / 0.000000 so >>>>>>> far) >>>>>>> 2013-02-12 19:21:49.844277 [DEBUG] mod_nibblebill.c:351 Doing update >>>>>>> query >>>>>>> [UPDATE sip_users SET balance=balance-0.018983 WHERE ='1'] >>>>>>> 2013-02-12 19:21:49.844277 [CRIT] mod_nibblebill.c:535 Failed to log >>>>>>> to database! >>>>>>> 2013-02-12 19:21:49.844277 [DEBUG] mod_nibblebill.c:378 Doing lookup >>>>>>> query >>>>>>> [SELECT balance AS nibble_balance FROM sip_users WHERE ='1'] >>>>>>> 2013-02-12 19:21:49.844277 [ERR] mod_nibblebill.c:380 Error running >>>>>>> this query: [SELECT balance AS nibble_balance FROM sip_users WHERE ='1'] >>>>>>> >>>>>>> >>>>>>> the select go fine if executed "by hand" >>>>>>> >>>>>> >>>>>> None of those SQL statements are valid. (hint: WHERE = '1' is no >>>>>> good). Been a while since I hacked/worked with nibblebill, but I'm >>>>>> pretty sure the 'id' field isn't not being filled in the sql for some >>>>>> reason. >>>>>> >>>>>> >>>>>> -- >>>>>> -Rupa >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> 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 >>>>>> >>>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> 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 >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130213/9a7872e6/attachment-0001.html From steveayre at gmail.com Wed Feb 13 20:46:34 2013 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 13 Feb 2013 17:46:34 +0000 Subject: [Freeswitch-users] Nibblebill database question In-Reply-To: References: Message-ID: http://jira.freeswitch.org/browse/FS-5100 On 13 February 2013 17:40, Steven Ayre wrote: > Specifically it's introduced by commit 5c176e8 which was resolving > FS-4810, which is indeed adding support for the the generalised DSN > interface. > > > http://fisheye.freeswitch.org/changelog/freeswitch.git?showid=5c176e889d8cc2da7efb18a89782407ee5524a74 > http://jira.freeswitch.org/browse/FS-4810 > > -Steve > > > > On 13 February 2013 17:37, Steven Ayre wrote: > >> Looks like this difference is brought in by the modifications to support >> the generalised DSN interface. odbc-dsn would be consistent with all other >> modules, so I guess that's why it's happened. >> >> But... >> The HEAD of v1.2.stable understands db_dsn but not odbc-dsn. >> The HEAD of master understands odbc-dsn but not db_dsn. >> >> This is a configuration breaking behaviour, which doesn't seem to have >> been documented! >> >> At the very least it'd make sense for the master branch to understand >> both odbc-dsn and db_dsn for backwards compatibility. I'll file a Jira with >> a patch (it's a trivial fix) in a moment. >> >> -Steve >> >> >> >> >> >> >> On 13 February 2013 17:33, Steven Ayre wrote: >> >>> How are you setting the DSN? >>> >>> The param name mentioned earlier is wrong. Reading the source the >>> correct parameter is "odbc-dsn". Looks like the wiki documentation is >>> incorrect. >>> >>> -Steve >>> >>> >>> >>> >>> On 13 February 2013 15:33, David Villasmil < >>> david.villasmil.work at gmail.com> wrote: >>> >>>> hello Avi, >>>> >>>> that's right, but nevermind that, that's the output when I was trying >>>> something else: >>>> >>>> 2013-02-12 19:32:07.784156 [INFO] mod_nibblebill.c:498 Beginning new >>>> billing on 58e6762b-c049-470e-84f7-519a815ab6f0 >>>> 2013-02-12 19:32:07.784156 [CRIT] mod_nibblebill.c:535 Failed to log to >>>> database! >>>> 2013-02-12 19:32:07.784156 [ERR] mod_nibblebill.c:380 Error running >>>> this query: [SELECT balance AS nibble_balance FROM sip_users WHERE id='1'] >>>> >>>> >>>> On Wed, Feb 13, 2013 at 12:58 PM, Avi Marcus wrote: >>>> >>>>> Your logs showed the queries run were: >>>>> >>>>> UPDATE sip_users SET balance=balance-0.018983 WHERE ='1' >>>>> SELECT balance AS nibble_balance FROM sip_users WHERE ='1' >>>>> >>>>> You just tried now.. "WHERE *id*='1' " -- they are not the same >>>>> query... >>>>> >>>>> -Avi >>>>> >>>>> >>>>> >>>>> On Wed, Feb 13, 2013 at 12:48 PM, David Villasmil < >>>>> david.villasmil.work at gmail.com> wrote: >>>>> >>>>>> Hello Rupa, >>>>>> >>>>>> the query if ok, the user is there: >>>>>> >>>>>> mysql> select * from sip_users; >>>>>> >>>>>> +----+----------+----------+----------------+------------+--------------+--------------+ >>>>>> | id | username | password | rateplan | balance | domain >>>>>> | max_channels | >>>>>> >>>>>> +----+----------+----------+----------------+------------+--------------+--------------+ >>>>>> | 1 | 1002 | 1234 | retail_default | 100.000000 | >>>>>> 192.168.1.44 | 1 | >>>>>> >>>>>> +----+----------+----------+----------------+------------+--------------+--------------+ >>>>>> 1 row in set (0.00 sec) >>>>>> >>>>>> >>>>>> and the actual sql executed by nibblebill: >>>>>> >>>>>> mysql> SELECT balance AS nibble_balance FROM sip_users WHERE id='1'; >>>>>> +----------------+ >>>>>> | nibble_balance | >>>>>> +----------------+ >>>>>> | 100.000000 | >>>>>> +----------------+ >>>>>> 1 row in set (0.00 sec) >>>>>> >>>>>> >>>>>> so that's ok.. >>>>>> >>>>>> >>>>>> any othe ideas? >>>>>> >>>>>> Thanks a lot for your help! >>>>>> >>>>>> >>>>>> David >>>>>> >>>>>> >>>>>> >>>>>> On Wed, Feb 13, 2013 at 3:35 AM, Rupa Schomaker wrote: >>>>>> >>>>>>> >>>>>>> On Tue, Feb 12, 2013 at 5:26 PM, David Villasmil < >>>>>>> david.villasmil.work at gmail.com> wrote: >>>>>>> >>>>>>>> call goes out fine... no billing though: >>>>>>>> >>>>>>>> >>>>>>>> 2013-02-12 19:21:49.844277 [DEBUG] mod_nibblebill.c:504 22 seconds >>>>>>>> passed since last bill time of 2013-02-12 19:21:27 >>>>>>>> 2013-02-12 19:21:49.844277 [DEBUG] mod_nibblebill.c:521 Billing >>>>>>>> $0.018983 to 1 (Call: d442dc6b-a3a7-423a-8450-598d17ca43b3 / 0.000000 so >>>>>>>> far) >>>>>>>> 2013-02-12 19:21:49.844277 [DEBUG] mod_nibblebill.c:351 Doing >>>>>>>> update query >>>>>>>> [UPDATE sip_users SET balance=balance-0.018983 WHERE ='1'] >>>>>>>> 2013-02-12 19:21:49.844277 [CRIT] mod_nibblebill.c:535 Failed to >>>>>>>> log to database! >>>>>>>> 2013-02-12 19:21:49.844277 [DEBUG] mod_nibblebill.c:378 Doing >>>>>>>> lookup query >>>>>>>> [SELECT balance AS nibble_balance FROM sip_users WHERE ='1'] >>>>>>>> 2013-02-12 19:21:49.844277 [ERR] mod_nibblebill.c:380 Error running >>>>>>>> this query: [SELECT balance AS nibble_balance FROM sip_users WHERE ='1'] >>>>>>>> >>>>>>>> >>>>>>>> the select go fine if executed "by hand" >>>>>>>> >>>>>>> >>>>>>> None of those SQL statements are valid. (hint: WHERE = '1' is no >>>>>>> good). Been a while since I hacked/worked with nibblebill, but I'm >>>>>>> pretty sure the 'id' field isn't not being filled in the sql for some >>>>>>> reason. >>>>>>> >>>>>>> >>>>>>> -- >>>>>>> -Rupa >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://wiki.freeswitch.org >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> 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 >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> 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 >>>>>> >>>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> 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 >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>>> >>>> >>> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130213/8e89125b/attachment.html From david.villasmil.work at gmail.com Wed Feb 13 20:50:20 2013 From: david.villasmil.work at gmail.com (David Villasmil) Date: Wed, 13 Feb 2013 18:50:20 +0100 Subject: [Freeswitch-users] Nibblebill database question In-Reply-To: References: Message-ID: Hello Steven, Great! At least I now see it took the dsn when loading: 2013-02-13 17:44:47.623905 [INFO] mod_nibblebill.c:197 odbc_dsn is viking 2013-02-13 17:44:47.623905 [DEBUG] mod_nibblebill.c:242 dsn is "viking" I don't know whether it actually makes the connection, because I am still seeing an error, it complaints the table doesn't exists, though it does: 2013-02-13 17:45:56.903600 [INFO] mod_nibblebill.c:498 Beginning new billing on 214e85c3-6286-4aa1-a8bc-5dda6ad0fd44 2013-02-13 17:45:56.903600 [ERR] switch_core_sqldb.c:1124 SQL ERR: [UPDATE sip_users SET balance=balance-0.044817 WHERE id='1'] no such table: sip_users 2013-02-13 17:45:56.903600 [CRIT] mod_nibblebill.c:535 Failed to log to database! 2013-02-13 17:45:56.903600 [ERR] switch_core_sqldb.c:1124 SQL ERR: [SELECT balance AS nibble_balance FROM sip_users WHERE id='1'] no such table: sip_users 2013-02-13 17:45:56.903600 [ERR] mod_nibblebill.c:380 Error running this query: [SELECT balance AS nibble_balance FROM sip_users WHERE id='1'] On Wed, Feb 13, 2013 at 6:40 PM, Steven Ayre wrote: > Specifically it's introduced by commit 5c176e8 which was resolving > FS-4810, which is indeed adding support for the the generalised DSN > interface. > > > http://fisheye.freeswitch.org/changelog/freeswitch.git?showid=5c176e889d8cc2da7efb18a89782407ee5524a74 > http://jira.freeswitch.org/browse/FS-4810 > > -Steve > > > > On 13 February 2013 17:37, Steven Ayre wrote: > >> Looks like this difference is brought in by the modifications to support >> the generalised DSN interface. odbc-dsn would be consistent with all other >> modules, so I guess that's why it's happened. >> >> But... >> The HEAD of v1.2.stable understands db_dsn but not odbc-dsn. >> The HEAD of master understands odbc-dsn but not db_dsn. >> >> This is a configuration breaking behaviour, which doesn't seem to have >> been documented! >> >> At the very least it'd make sense for the master branch to understand >> both odbc-dsn and db_dsn for backwards compatibility. I'll file a Jira with >> a patch (it's a trivial fix) in a moment. >> >> -Steve >> >> >> >> >> >> >> On 13 February 2013 17:33, Steven Ayre wrote: >> >>> How are you setting the DSN? >>> >>> The param name mentioned earlier is wrong. Reading the source the >>> correct parameter is "odbc-dsn". Looks like the wiki documentation is >>> incorrect. >>> >>> -Steve >>> >>> >>> >>> >>> On 13 February 2013 15:33, David Villasmil < >>> david.villasmil.work at gmail.com> wrote: >>> >>>> hello Avi, >>>> >>>> that's right, but nevermind that, that's the output when I was trying >>>> something else: >>>> >>>> 2013-02-12 19:32:07.784156 [INFO] mod_nibblebill.c:498 Beginning new >>>> billing on 58e6762b-c049-470e-84f7-519a815ab6f0 >>>> 2013-02-12 19:32:07.784156 [CRIT] mod_nibblebill.c:535 Failed to log to >>>> database! >>>> 2013-02-12 19:32:07.784156 [ERR] mod_nibblebill.c:380 Error running >>>> this query: [SELECT balance AS nibble_balance FROM sip_users WHERE id='1'] >>>> >>>> >>>> On Wed, Feb 13, 2013 at 12:58 PM, Avi Marcus wrote: >>>> >>>>> Your logs showed the queries run were: >>>>> >>>>> UPDATE sip_users SET balance=balance-0.018983 WHERE ='1' >>>>> SELECT balance AS nibble_balance FROM sip_users WHERE ='1' >>>>> >>>>> You just tried now.. "WHERE *id*='1' " -- they are not the same >>>>> query... >>>>> >>>>> -Avi >>>>> >>>>> >>>>> >>>>> On Wed, Feb 13, 2013 at 12:48 PM, David Villasmil < >>>>> david.villasmil.work at gmail.com> wrote: >>>>> >>>>>> Hello Rupa, >>>>>> >>>>>> the query if ok, the user is there: >>>>>> >>>>>> mysql> select * from sip_users; >>>>>> >>>>>> +----+----------+----------+----------------+------------+--------------+--------------+ >>>>>> | id | username | password | rateplan | balance | domain >>>>>> | max_channels | >>>>>> >>>>>> +----+----------+----------+----------------+------------+--------------+--------------+ >>>>>> | 1 | 1002 | 1234 | retail_default | 100.000000 | >>>>>> 192.168.1.44 | 1 | >>>>>> >>>>>> +----+----------+----------+----------------+------------+--------------+--------------+ >>>>>> 1 row in set (0.00 sec) >>>>>> >>>>>> >>>>>> and the actual sql executed by nibblebill: >>>>>> >>>>>> mysql> SELECT balance AS nibble_balance FROM sip_users WHERE id='1'; >>>>>> +----------------+ >>>>>> | nibble_balance | >>>>>> +----------------+ >>>>>> | 100.000000 | >>>>>> +----------------+ >>>>>> 1 row in set (0.00 sec) >>>>>> >>>>>> >>>>>> so that's ok.. >>>>>> >>>>>> >>>>>> any othe ideas? >>>>>> >>>>>> Thanks a lot for your help! >>>>>> >>>>>> >>>>>> David >>>>>> >>>>>> >>>>>> >>>>>> On Wed, Feb 13, 2013 at 3:35 AM, Rupa Schomaker wrote: >>>>>> >>>>>>> >>>>>>> On Tue, Feb 12, 2013 at 5:26 PM, David Villasmil < >>>>>>> david.villasmil.work at gmail.com> wrote: >>>>>>> >>>>>>>> call goes out fine... no billing though: >>>>>>>> >>>>>>>> >>>>>>>> 2013-02-12 19:21:49.844277 [DEBUG] mod_nibblebill.c:504 22 seconds >>>>>>>> passed since last bill time of 2013-02-12 19:21:27 >>>>>>>> 2013-02-12 19:21:49.844277 [DEBUG] mod_nibblebill.c:521 Billing >>>>>>>> $0.018983 to 1 (Call: d442dc6b-a3a7-423a-8450-598d17ca43b3 / 0.000000 so >>>>>>>> far) >>>>>>>> 2013-02-12 19:21:49.844277 [DEBUG] mod_nibblebill.c:351 Doing >>>>>>>> update query >>>>>>>> [UPDATE sip_users SET balance=balance-0.018983 WHERE ='1'] >>>>>>>> 2013-02-12 19:21:49.844277 [CRIT] mod_nibblebill.c:535 Failed to >>>>>>>> log to database! >>>>>>>> 2013-02-12 19:21:49.844277 [DEBUG] mod_nibblebill.c:378 Doing >>>>>>>> lookup query >>>>>>>> [SELECT balance AS nibble_balance FROM sip_users WHERE ='1'] >>>>>>>> 2013-02-12 19:21:49.844277 [ERR] mod_nibblebill.c:380 Error running >>>>>>>> this query: [SELECT balance AS nibble_balance FROM sip_users WHERE ='1'] >>>>>>>> >>>>>>>> >>>>>>>> the select go fine if executed "by hand" >>>>>>>> >>>>>>> >>>>>>> None of those SQL statements are valid. (hint: WHERE = '1' is no >>>>>>> good). Been a while since I hacked/worked with nibblebill, but I'm >>>>>>> pretty sure the 'id' field isn't not being filled in the sql for some >>>>>>> reason. >>>>>>> >>>>>>> >>>>>>> -- >>>>>>> -Rupa >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://wiki.freeswitch.org >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> 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 >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> 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 >>>>>> >>>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> 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 >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>>> >>>> >>> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130213/b9421008/attachment-0001.html From david.villasmil.work at gmail.com Wed Feb 13 20:51:17 2013 From: david.villasmil.work at gmail.com (David Villasmil) Date: Wed, 13 Feb 2013 18:51:17 +0100 Subject: [Freeswitch-users] Nibblebill database question In-Reply-To: References: Message-ID: And i can connect from the cli: root at lamp ~# isql viking +---------------------------------------+ | Connected! | | | | sql-statement | | help [tablename] | | quit | | | +---------------------------------------+ SQL> strange... On Wed, Feb 13, 2013 at 6:46 PM, Steven Ayre wrote: > http://jira.freeswitch.org/browse/FS-5100 > > > On 13 February 2013 17:40, Steven Ayre wrote: > >> Specifically it's introduced by commit 5c176e8 which was resolving >> FS-4810, which is indeed adding support for the the generalised DSN >> interface. >> >> >> http://fisheye.freeswitch.org/changelog/freeswitch.git?showid=5c176e889d8cc2da7efb18a89782407ee5524a74 >> http://jira.freeswitch.org/browse/FS-4810 >> >> -Steve >> >> >> >> On 13 February 2013 17:37, Steven Ayre wrote: >> >>> Looks like this difference is brought in by the modifications to support >>> the generalised DSN interface. odbc-dsn would be consistent with all other >>> modules, so I guess that's why it's happened. >>> >>> But... >>> The HEAD of v1.2.stable understands db_dsn but not odbc-dsn. >>> The HEAD of master understands odbc-dsn but not db_dsn. >>> >>> This is a configuration breaking behaviour, which doesn't seem to have >>> been documented! >>> >>> At the very least it'd make sense for the master branch to understand >>> both odbc-dsn and db_dsn for backwards compatibility. I'll file a Jira with >>> a patch (it's a trivial fix) in a moment. >>> >>> -Steve >>> >>> >>> >>> >>> >>> >>> On 13 February 2013 17:33, Steven Ayre wrote: >>> >>>> How are you setting the DSN? >>>> >>>> The param name mentioned earlier is wrong. Reading the source the >>>> correct parameter is "odbc-dsn". Looks like the wiki documentation is >>>> incorrect. >>>> >>>> -Steve >>>> >>>> >>>> >>>> >>>> On 13 February 2013 15:33, David Villasmil < >>>> david.villasmil.work at gmail.com> wrote: >>>> >>>>> hello Avi, >>>>> >>>>> that's right, but nevermind that, that's the output when I was trying >>>>> something else: >>>>> >>>>> 2013-02-12 19:32:07.784156 [INFO] mod_nibblebill.c:498 Beginning new >>>>> billing on 58e6762b-c049-470e-84f7-519a815ab6f0 >>>>> 2013-02-12 19:32:07.784156 [CRIT] mod_nibblebill.c:535 Failed to log >>>>> to database! >>>>> 2013-02-12 19:32:07.784156 [ERR] mod_nibblebill.c:380 Error running >>>>> this query: [SELECT balance AS nibble_balance FROM sip_users WHERE id='1'] >>>>> >>>>> >>>>> On Wed, Feb 13, 2013 at 12:58 PM, Avi Marcus wrote: >>>>> >>>>>> Your logs showed the queries run were: >>>>>> >>>>>> UPDATE sip_users SET balance=balance-0.018983 WHERE ='1' >>>>>> SELECT balance AS nibble_balance FROM sip_users WHERE ='1' >>>>>> >>>>>> You just tried now.. "WHERE *id*='1' " -- they are not the same >>>>>> query... >>>>>> >>>>>> -Avi >>>>>> >>>>>> >>>>>> >>>>>> On Wed, Feb 13, 2013 at 12:48 PM, David Villasmil < >>>>>> david.villasmil.work at gmail.com> wrote: >>>>>> >>>>>>> Hello Rupa, >>>>>>> >>>>>>> the query if ok, the user is there: >>>>>>> >>>>>>> mysql> select * from sip_users; >>>>>>> >>>>>>> +----+----------+----------+----------------+------------+--------------+--------------+ >>>>>>> | id | username | password | rateplan | balance | domain >>>>>>> | max_channels | >>>>>>> >>>>>>> +----+----------+----------+----------------+------------+--------------+--------------+ >>>>>>> | 1 | 1002 | 1234 | retail_default | 100.000000 | >>>>>>> 192.168.1.44 | 1 | >>>>>>> >>>>>>> +----+----------+----------+----------------+------------+--------------+--------------+ >>>>>>> 1 row in set (0.00 sec) >>>>>>> >>>>>>> >>>>>>> and the actual sql executed by nibblebill: >>>>>>> >>>>>>> mysql> SELECT balance AS nibble_balance FROM sip_users WHERE id='1'; >>>>>>> +----------------+ >>>>>>> | nibble_balance | >>>>>>> +----------------+ >>>>>>> | 100.000000 | >>>>>>> +----------------+ >>>>>>> 1 row in set (0.00 sec) >>>>>>> >>>>>>> >>>>>>> so that's ok.. >>>>>>> >>>>>>> >>>>>>> any othe ideas? >>>>>>> >>>>>>> Thanks a lot for your help! >>>>>>> >>>>>>> >>>>>>> David >>>>>>> >>>>>>> >>>>>>> >>>>>>> On Wed, Feb 13, 2013 at 3:35 AM, Rupa Schomaker wrote: >>>>>>> >>>>>>>> >>>>>>>> On Tue, Feb 12, 2013 at 5:26 PM, David Villasmil < >>>>>>>> david.villasmil.work at gmail.com> wrote: >>>>>>>> >>>>>>>>> call goes out fine... no billing though: >>>>>>>>> >>>>>>>>> >>>>>>>>> 2013-02-12 19:21:49.844277 [DEBUG] mod_nibblebill.c:504 22 seconds >>>>>>>>> passed since last bill time of 2013-02-12 19:21:27 >>>>>>>>> 2013-02-12 19:21:49.844277 [DEBUG] mod_nibblebill.c:521 Billing >>>>>>>>> $0.018983 to 1 (Call: d442dc6b-a3a7-423a-8450-598d17ca43b3 / 0.000000 so >>>>>>>>> far) >>>>>>>>> 2013-02-12 19:21:49.844277 [DEBUG] mod_nibblebill.c:351 Doing >>>>>>>>> update query >>>>>>>>> [UPDATE sip_users SET balance=balance-0.018983 WHERE ='1'] >>>>>>>>> 2013-02-12 19:21:49.844277 [CRIT] mod_nibblebill.c:535 Failed to >>>>>>>>> log to database! >>>>>>>>> 2013-02-12 19:21:49.844277 [DEBUG] mod_nibblebill.c:378 Doing >>>>>>>>> lookup query >>>>>>>>> [SELECT balance AS nibble_balance FROM sip_users WHERE ='1'] >>>>>>>>> 2013-02-12 19:21:49.844277 [ERR] mod_nibblebill.c:380 Error >>>>>>>>> running this query: [SELECT balance AS nibble_balance FROM sip_users WHERE >>>>>>>>> ='1'] >>>>>>>>> >>>>>>>>> >>>>>>>>> the select go fine if executed "by hand" >>>>>>>>> >>>>>>>> >>>>>>>> None of those SQL statements are valid. (hint: WHERE = '1' is no >>>>>>>> good). Been a while since I hacked/worked with nibblebill, but I'm >>>>>>>> pretty sure the 'id' field isn't not being filled in the sql for some >>>>>>>> reason. >>>>>>>> >>>>>>>> >>>>>>>> -- >>>>>>>> -Rupa >>>>>>>> >>>>>>>> >>>>>>>> _________________________________________________________________________ >>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>> consulting at freeswitch.org >>>>>>>> http://www.freeswitchsolutions.com >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> Official FreeSWITCH Sites >>>>>>>> http://www.freeswitch.org >>>>>>>> http://wiki.freeswitch.org >>>>>>>> http://www.cluecon.com >>>>>>>> >>>>>>>> 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 >>>>>>>> >>>>>>>> >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://wiki.freeswitch.org >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> 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 >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> 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 >>>>>> >>>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> 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 >>>>> >>>>> >>>> >>> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130213/686be59e/attachment-0001.html From steveayre at gmail.com Wed Feb 13 20:51:08 2013 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 13 Feb 2013 17:51:08 +0000 Subject: [Freeswitch-users] SRTP disabling In-Reply-To: References: Message-ID: sip_secure_media also covers SRTP without ZRTP. It's actually used elsewhere too, but aliased. src/mod/endpoints/mod_sofia/mod_sofia.h:115:#define SOFIA_SECURE_MEDIA_VARIABLE "sip_secure_media" Search the source tree for SOFIA_SECURE_MEDIA_VARIABLE and you'll see it's used in a number of other places too. -Steve On 13 February 2013 17:13, Levend Sayar wrote: > ** > Hi again. > > I checked the FS code and i see that rpm i am using is built with ZRTP > disabled. > There is only one place that "sip_secure_media" on the code. That is on > switch_rtp.c > > #ifdef ENABLE_ZRTP > if (zrtp_on) { > switch_rtp_t *master_rtp_session = NULL > > int initiator = 0; > const char *zrtp_enabled = switch_channel_get_variable(channel, > "zrtp_secure_media"); > const char *srtp_enabled = switch_channel_get_variable(channel, > "sip_secure_media"); > > > So since ENABLE_ZRTP is 0, i don't have chance to use "sip_secure_media" > variable. > > Is there any other variable that i can use and make sofia module not to > choose SRTP ? > > > _lvnd_ > {^_^} > > > On Thu, 2013-02-07 at 13:54 +0000, Levend Sayar wrote: > > But the very same phone calls another phone and talk with RTP, not SRTP if > the peer does not accept SRTP > > Here is the SDP offer by the same phone > > v=0 > o=- 20186 20186 IN IP4 192.168.173.69 > s=SDP data > c=IN IP4 192.168.173.69 > t=0 0 > m=audio 11782 RTP/SAVP 0 8 18 9 101 > a=crypto:1 AES_CM_128_HMAC_SHA1_80 > inline:NzFmYjdiMjk1OTY2ODQwYzExZjM0ZmE2NGM0YWMw > a=crypto:2 AES_CM_128_HMAC_SHA1_32 > inline:M2MxMTE2OWFjOGY2ZjEwADEzZmZkNzAxNjRlMzFm > a=crypto:3 F8_128_HMAC_SHA1_80 > inline:NjkzZDg2Mjk0ZTkxMjg1YzdmYjFiNjRlMmFhNGFm > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:9 G722/8000 > a=fmtp:101 0-15 > a=rtpmap:101 telephone-event/8000 > a=ptime:20 > a=sendrecv > > And here is the SDP answer sent by the other phone > > v=0 > o=- 20029 20029 IN IP4 192.168.173.65 > s=SDP data > c=IN IP4 192.168.173.65 > t=0 0 > m=audio 11794 RTP/SAVP 0 101 > a=rtpmap:0 PCMU/8000 > a=sendrecv > a=ptime:20 > a=fmtp:101 0-15 > a=rtpmap:101 telephone-event/8000 > > > > > -- > > > _lvnd_ > {^_^} > > > > > > > On Thu, 2013-02-07 at 13:37 +0000, Steven Ayre wrote: > > m=audio 11780 RTP/SAVP 0 8 18 9 101 > > > RTP/SAVP means SRTP is mandatory. You need to reconfigure the phone. > > > If the phone sends RTP/AVP then that means plain RTP, and RTP/AVP with a > a=crypto attribute means SRTP is optional. > > > -Steve > > > > > On 7 February 2013 13:26, Levend Sayar wrote: > > Below is the SDP offer sent by the phone. > > v=0 > o=- 20185 20185 IN IP4 192.168.173.69 > s=SDP data > c=IN IP4 192.168.173.69 > t=0 0 > m=audio 11780 RTP/SAVP 0 8 18 9 101 > a=crypto:1 AES_CM_128_HMAC_SHA1_80 > inline:MTM2MjVhMGI1NDZjYmRjADU5NWVjNGVkNTNlYzA1 > a=crypto:2 AES_CM_128_HMAC_SHA1_32 > inline:YmExYmZhNQAzN2ZjNDgzYTRkNGU2ZjFiN2Q0MmE3 > a=crypto:3 F8_128_HMAC_SHA1_80 > inline:N2Q2NTRiYQAxZjA3MWY3ZjI1YTI5NjIyM2FjODYw > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:9 G722/8000 > a=fmtp:101 0-15 > a=rtpmap:101 telephone-event/8000 > a=ptime:20 > a=sendrecv > > > > And below is the SDP answer sent by FS > > v=0 > o=FreeSWITCH 1360230601 1360230602 IN IP4 192.168.169.114 > s=FreeSWITCH > c=IN IP4 192.168.169.114 > t=0 0 > m=audio 12532 RTP/SAVP 9 101 > a=rtpmap:9 G722/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=crypto:1 AES_CM_128_HMAC_SHA1_80 > inline:l8v0R64H7CP0vEx9j0Ycdbob8bgMCpLDppWGy7Dy > > > > > _lvnd_ > {^_^} > > > > > > > > On Thu, 2013-02-07 at 13:09 +0000, Steven Ayre wrote: > > What I mean is you'll see two separate m=audio lines within the callee's > SDP, one for 'RTP/AVP' and one for 'SRTP/AVP'. If there is no m=audio line > for RTP/AVP the caller won't know of a port that's expecting RTP. So if the > callee only sends SRTP/AVP the caller can't send RTP. > > > Can you show us the SDP being sent by the phone? > > > -Steve > > > > > > On 7 February 2013 11:01, Levend Sayar wrote: > > Thanx Steven. > > > Caller makes the offer for SDP but callee chooses whatever it wants. So > caller can offer SRTP but callee can prefer not to talk encrypted. In our > case I want FS to choose non secure media. > Phone will offer SRTP on the conference call but FS must prefer RTP, not > SRTP. > > > > > _lvnd_ > {^_^} > > > > > > > On 7 ?ub 2013, at 11:13, "Steven Ayre" wrote: > > > It's also going to rely on the phone actually offering RTP/AVP as well as > SRTP/AVP in their SDP - without that there'd be nowhere to send insecure > RTP. > > > -Steve > > > > > On 6 February 2013 16:09, Levend Sayar wrote: > > Thanks Daniel for the reply. > > > I tried > > > > > > But did not work. Upon your reply I also tried > > > > > > But did not work either. I am doing something wrong ? > > > > > _lvnd_ > {^_^} > > > > > > > On 6 ?ub 2013, at 18:00, "Daniel Ivanov" wrote: > > > > Of course you can. Just set the secure_media var to false and you will be > srtp-free in sip. > > On Feb 5, 2013 6:06 PM, "Levend Sayar" wrote: > > Hi all. > > I am using FS as a conference server. Some of my phones are using SRTP , > some of them not. Both type of phone can > join a conference. FS can talk to each peer with SRTP or not depending on > the phone itself. > > My question: > > Is it possible to disable SRTP on FS ? > > I suppose if i can disable SRTP, FS will talk without SRTP with each phone > whether they are using SRTP or not. > > TIA > > > plain text document attachment (ATT00001) > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130213/4ea4df61/attachment.html From steveayre at gmail.com Wed Feb 13 20:53:03 2013 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 13 Feb 2013 17:53:03 +0000 Subject: [Freeswitch-users] SRTP disabling In-Reply-To: References: Message-ID: Nevertheless it's not being negotiated and this is possibly a bug in the phone. Adding a workaround in FS would decrease security because the phone is explicitly saying it'll only accept SRTP so sending plain RTP wouldn't be wise. I assume both phones are the same make? -Steve On 7 February 2013 13:54, Levend Sayar wrote: > ** > But the very same phone calls another phone and talk with RTP, not SRTP if > the peer does not accept SRTP > > Here is the SDP offer by the same phone > > v=0 > o=- 20186 20186 IN IP4 192.168.173.69 > > s=SDP data > c=IN IP4 192.168.173.69 > t=0 0 > m=audio 11782 RTP/SAVP 0 8 18 9 101 > a=crypto:1 AES_CM_128_HMAC_SHA1_80 > inline:NzFmYjdiMjk1OTY2ODQwYzExZjM0ZmE2NGM0YWMw > a=crypto:2 AES_CM_128_HMAC_SHA1_32 > inline:M2MxMTE2OWFjOGY2ZjEwADEzZmZkNzAxNjRlMzFm > a=crypto:3 F8_128_HMAC_SHA1_80 > inline:NjkzZDg2Mjk0ZTkxMjg1YzdmYjFiNjRlMmFhNGFm > > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:9 G722/8000 > a=fmtp:101 0-15 > a=rtpmap:101 telephone-event/8000 > a=ptime:20 > a=sendrecv > > And here is the SDP answer sent by the other phone > > v=0 > o=- 20029 20029 IN IP4 192.168.173.65 > s=SDP data > c=IN IP4 192.168.173.65 > t=0 0 > m=audio 11794 RTP/SAVP 0 101 > a=rtpmap:0 PCMU/8000 > a=sendrecv > a=ptime:20 > > a=fmtp:101 0-15 > a=rtpmap:101 telephone-event/8000 > > > > > -- > ?yi ?al??malar. > > _lvnd_ > {^_^} > > KAREL`UCAP > > > > On Thu, 2013-02-07 at 13:37 +0000, Steven Ayre wrote: > > m=audio 11780 RTP/SAVP 0 8 18 9 101 > > > > RTP/SAVP means SRTP is mandatory. You need to reconfigure the phone. > > > > If the phone sends RTP/AVP then that means plain RTP, and RTP/AVP with a > a=crypto attribute means SRTP is optional. > > > > -Steve > > > > > > On 7 February 2013 13:26, Levend Sayar > wrote: > > Below is the SDP offer sent by the phone. > > v=0 > o=- 20185 20185 IN IP4 192.168.173.69 > s=SDP data > c=IN IP4 192.168.173.69 > t=0 0 > m=audio 11780 RTP/SAVP 0 8 18 9 101 > a=crypto:1 AES_CM_128_HMAC_SHA1_80 > inline:MTM2MjVhMGI1NDZjYmRjADU5NWVjNGVkNTNlYzA1 > a=crypto:2 AES_CM_128_HMAC_SHA1_32 > inline:YmExYmZhNQAzN2ZjNDgzYTRkNGU2ZjFiN2Q0MmE3 > a=crypto:3 F8_128_HMAC_SHA1_80 > inline:N2Q2NTRiYQAxZjA3MWY3ZjI1YTI5NjIyM2FjODYw > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:9 G722/8000 > a=fmtp:101 0-15 > a=rtpmap:101 telephone-event/8000 > a=ptime:20 > a=sendrecv > > > > And below is the SDP answer sent by FS > > v=0 > o=FreeSWITCH 1360230601 1360230602 IN IP4 192.168.169.114 > s=FreeSWITCH > c=IN IP4 192.168.169.114 > t=0 0 > m=audio 12532 RTP/SAVP 9 101 > a=rtpmap:9 G722/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=crypto:1 AES_CM_128_HMAC_SHA1_80 > inline:l8v0R64H7CP0vEx9j0Ycdbob8bgMCpLDppWGy7Dy > > > > > -- > ?yi ?al??malar. > > _lvnd_ > {^_^} > > KAREL`UCAP > > > > > On Thu, 2013-02-07 at 13:09 +0000, Steven Ayre wrote: > > What I mean is you'll see two separate m=audio lines within the callee's > SDP, one for 'RTP/AVP' and one for 'SRTP/AVP'. If there is no m=audio line > for RTP/AVP the caller won't know of a port that's expecting RTP. So if the > callee only sends SRTP/AVP the caller can't send RTP. > > > Can you show us the SDP being sent by the phone? > > > -Steve > > > > > > On 7 February 2013 11:01, Levend Sayar wrote: > > Thanx Steven. > > > Caller makes the offer for SDP but callee chooses whatever it wants. So > caller can offer SRTP but callee can prefer not to talk encrypted. In our > case I want FS to choose non secure media. > Phone will offer SRTP on the conference call but FS must prefer RTP, not > SRTP. > > > ?yi ?al??malar. > > > _lvnd_ > {^_^} > > > KAREL`UCAP > > > > On 7 ?ub 2013, at 11:13, "Steven Ayre" wrote: > > > It's also going to rely on the phone actually offering RTP/AVP as well as > SRTP/AVP in their SDP - without that there'd be nowhere to send insecure > RTP. > > > -Steve > > > > > On 6 February 2013 16:09, Levend Sayar wrote: > > Thanks Daniel for the reply. > > > I tried > > > > > > But did not work. Upon your reply I also tried > > > > > > But did not work either. I am doing something wrong ? > > ?yi ?al??malar. > > > _lvnd_ > {^_^} > > > KAREL`UCAP > > > > On 6 ?ub 2013, at 18:00, "Daniel Ivanov" wrote: > > > > Of course you can. Just set the secure_media var to false and you will be > srtp-free in sip. > > On Feb 5, 2013 6:06 PM, "Levend Sayar" wrote: > > Hi all. > > I am using FS as a conference server. Some of my phones are using SRTP , > some of them not. Both type of phone can > join a conference. FS can talk to each peer with SRTP or not depending on > the phone itself. > > My question: > > Is it possible to disable SRTP on FS ? > > I suppose if i can disable SRTP, FS will talk without SRTP with each phone > whether they are using SRTP or not. > > TIA > > > -- > > _lvnd_ > {^_^} > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > plain text document attachment (ATT00001) > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > plain text document attachment (ATT00001) > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130213/46c208b9/attachment-0001.html From steveayre at gmail.com Wed Feb 13 20:59:23 2013 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 13 Feb 2013 17:59:23 +0000 Subject: [Freeswitch-users] SRTP disabling In-Reply-To: References: Message-ID: If the phone is Polycom or Linksys have you read the notes on http://wiki.freeswitch.org/wiki/SRTP? -Steve On 13 February 2013 17:53, Steven Ayre wrote: > Nevertheless it's not being negotiated and this is possibly a bug in the > phone. Adding a workaround in FS would decrease security because the phone > is explicitly saying it'll only accept SRTP so sending plain RTP wouldn't > be wise. > > I assume both phones are the same make? > > -Steve > > > > > On 7 February 2013 13:54, Levend Sayar wrote: > >> ** >> But the very same phone calls another phone and talk with RTP, not SRTP >> if the peer does not accept SRTP >> >> Here is the SDP offer by the same phone >> >> v=0 >> o=- 20186 20186 IN IP4 192.168.173.69 >> >> s=SDP data >> c=IN IP4 192.168.173.69 >> t=0 0 >> m=audio 11782 RTP/SAVP 0 8 18 9 101 >> a=crypto:1 AES_CM_128_HMAC_SHA1_80 >> inline:NzFmYjdiMjk1OTY2ODQwYzExZjM0ZmE2NGM0YWMw >> a=crypto:2 AES_CM_128_HMAC_SHA1_32 >> inline:M2MxMTE2OWFjOGY2ZjEwADEzZmZkNzAxNjRlMzFm >> a=crypto:3 F8_128_HMAC_SHA1_80 >> inline:NjkzZDg2Mjk0ZTkxMjg1YzdmYjFiNjRlMmFhNGFm >> >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:18 G729/8000 >> a=fmtp:18 annexb=no >> a=rtpmap:9 G722/8000 >> a=fmtp:101 0-15 >> a=rtpmap:101 telephone-event/8000 >> a=ptime:20 >> a=sendrecv >> >> And here is the SDP answer sent by the other phone >> >> v=0 >> o=- 20029 20029 IN IP4 192.168.173.65 >> s=SDP data >> c=IN IP4 192.168.173.65 >> t=0 0 >> m=audio 11794 RTP/SAVP 0 101 >> a=rtpmap:0 PCMU/8000 >> a=sendrecv >> a=ptime:20 >> >> a=fmtp:101 0-15 >> a=rtpmap:101 telephone-event/8000 >> >> >> >> >> -- >> ?yi ?al??malar. >> >> _lvnd_ >> {^_^} >> >> KAREL`UCAP >> >> >> >> On Thu, 2013-02-07 at 13:37 +0000, Steven Ayre wrote: >> >> m=audio 11780 RTP/SAVP 0 8 18 9 101 >> >> >> >> RTP/SAVP means SRTP is mandatory. You need to reconfigure the phone. >> >> >> >> If the phone sends RTP/AVP then that means plain RTP, and RTP/AVP with >> a a=crypto attribute means SRTP is optional. >> >> >> >> -Steve >> >> >> >> >> >> On 7 February 2013 13:26, Levend Sayar >> wrote: >> >> Below is the SDP offer sent by the phone. >> >> v=0 >> o=- 20185 20185 IN IP4 192.168.173.69 >> s=SDP data >> c=IN IP4 192.168.173.69 >> t=0 0 >> m=audio 11780 RTP/SAVP 0 8 18 9 101 >> a=crypto:1 AES_CM_128_HMAC_SHA1_80 >> inline:MTM2MjVhMGI1NDZjYmRjADU5NWVjNGVkNTNlYzA1 >> a=crypto:2 AES_CM_128_HMAC_SHA1_32 >> inline:YmExYmZhNQAzN2ZjNDgzYTRkNGU2ZjFiN2Q0MmE3 >> a=crypto:3 F8_128_HMAC_SHA1_80 >> inline:N2Q2NTRiYQAxZjA3MWY3ZjI1YTI5NjIyM2FjODYw >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:18 G729/8000 >> a=fmtp:18 annexb=no >> a=rtpmap:9 G722/8000 >> a=fmtp:101 0-15 >> a=rtpmap:101 telephone-event/8000 >> a=ptime:20 >> a=sendrecv >> >> >> >> And below is the SDP answer sent by FS >> >> v=0 >> o=FreeSWITCH 1360230601 1360230602 IN IP4 192.168.169.114 >> s=FreeSWITCH >> c=IN IP4 192.168.169.114 >> t=0 0 >> m=audio 12532 RTP/SAVP 9 101 >> a=rtpmap:9 G722/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=silenceSupp:off - - - - >> a=ptime:20 >> a=crypto:1 AES_CM_128_HMAC_SHA1_80 >> inline:l8v0R64H7CP0vEx9j0Ycdbob8bgMCpLDppWGy7Dy >> >> >> >> >> -- >> ?yi ?al??malar. >> >> _lvnd_ >> {^_^} >> >> KAREL`UCAP >> >> >> >> >> On Thu, 2013-02-07 at 13:09 +0000, Steven Ayre wrote: >> >> What I mean is you'll see two separate m=audio lines within the callee's >> SDP, one for 'RTP/AVP' and one for 'SRTP/AVP'. If there is no m=audio line >> for RTP/AVP the caller won't know of a port that's expecting RTP. So if the >> callee only sends SRTP/AVP the caller can't send RTP. >> >> >> Can you show us the SDP being sent by the phone? >> >> >> -Steve >> >> >> >> >> >> On 7 February 2013 11:01, Levend Sayar wrote: >> >> Thanx Steven. >> >> >> Caller makes the offer for SDP but callee chooses whatever it wants. So >> caller can offer SRTP but callee can prefer not to talk encrypted. In our >> case I want FS to choose non secure media. >> Phone will offer SRTP on the conference call but FS must prefer RTP, not >> SRTP. >> >> >> ?yi ?al??malar. >> >> >> _lvnd_ >> {^_^} >> >> >> KAREL`UCAP >> >> >> >> On 7 ?ub 2013, at 11:13, "Steven Ayre" wrote: >> >> >> It's also going to rely on the phone actually offering RTP/AVP as well as >> SRTP/AVP in their SDP - without that there'd be nowhere to send insecure >> RTP. >> >> >> -Steve >> >> >> >> >> On 6 February 2013 16:09, Levend Sayar wrote: >> >> Thanks Daniel for the reply. >> >> >> I tried >> >> >> >> >> >> But did not work. Upon your reply I also tried >> >> >> >> >> >> But did not work either. I am doing something wrong ? >> >> ?yi ?al??malar. >> >> >> _lvnd_ >> {^_^} >> >> >> KAREL`UCAP >> >> >> >> On 6 ?ub 2013, at 18:00, "Daniel Ivanov" wrote: >> >> >> >> Of course you can. Just set the secure_media var to false and you will be >> srtp-free in sip. >> >> On Feb 5, 2013 6:06 PM, "Levend Sayar" wrote: >> >> Hi all. >> >> I am using FS as a conference server. Some of my phones are using SRTP , >> some of them not. Both type of phone can >> join a conference. FS can talk to each peer with SRTP or not depending on >> the phone itself. >> >> My question: >> >> Is it possible to disable SRTP on FS ? >> >> I suppose if i can disable SRTP, FS will talk without SRTP with each >> phone whether they are using SRTP or not. >> >> TIA >> >> >> -- >> >> _lvnd_ >> {^_^} >> >> >> >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> >> >> plain text document attachment (ATT00001) >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >> >> >> >> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com >> >> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> >> >> plain text document attachment (ATT00001) >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >> >> >> >> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com >> >> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130213/1e82acb3/attachment-0001.html From dvl36.ripe.nick at gmail.com Wed Feb 13 21:00:09 2013 From: dvl36.ripe.nick at gmail.com (Dmitry Lysenko) Date: Wed, 13 Feb 2013 20:00:09 +0200 Subject: [Freeswitch-users] best freeswitch timer In-Reply-To: <1A89F457-BDAE-418C-9316-AED00FDBE230@visionutveckling.se> References: , <1360722674.19143.21.camel@Dmitry-AOA150.private> <1A89F457-BDAE-418C-9316-AED00FDBE230@visionutveckling.se> Message-ID: <1360778409.21568.2.camel@Dmitry-AOA150.private> ? ???, 13/02/2013 ? 07:02 +0000, Peter Olsson ?????: > I wouldn't worry too much about spikes of 200usec, do you have an actual problem with audio quality? > > /Peter Yes, I experienced problems with audio quality. I use mod_gsmopen, it work with hardware GSM modems via USB bus, and using timers in channel_read_frame() function. If work on reading of PCM frame will be done too late, this frame may be dropped, if too early - CNG frame may be generated. Adding a buffer increases the already large audio delay. It seems I resolved this problem by switching to realtime kernel, adding Flush() function to libctb serial library, and patching mod_gsmopen AUDIO_SYNC accordingly. From a.venugopan at mundio.com Wed Feb 13 21:01:32 2013 From: a.venugopan at mundio.com (Archana Venugopan) Date: Wed, 13 Feb 2013 18:01:32 +0000 Subject: [Freeswitch-users] DB close in lua Message-ID: <592A9CF93E12394E8472A6CC66E66BF23591E0@Mail-Kilo.squay.com> Hi, Am trying close DB connection in lua script(dbh:close()) in freeswitch. But I am getting this error 2013-02-13 16:43:41.327172 [ERR] mod_lua.cpp:102 /usr/local/freeswitch/scripts/directory.lua:66: attempt to call method 'close' (a nil value) stack traceback: Can anyone please let me know why my lua is not recognising the close function? Is there anyother ways to close my DB connection in lua? Thanks. Regards, Archana -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130213/0eff0335/attachment.html From avi at avimarcus.net Wed Feb 13 21:07:26 2013 From: avi at avimarcus.net (Avi Marcus) Date: Wed, 13 Feb 2013 20:07:26 +0200 Subject: [Freeswitch-users] DB close in lua In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF23591E0@Mail-Kilo.squay.com> References: <592A9CF93E12394E8472A6CC66E66BF23591E0@Mail-Kilo.squay.com> Message-ID: Wiki say: *dbh:release()* (optional) releases the handle back to the pool so it can be re-used by another thread. This is also automatically done when the dbh goes out of scope and is garbage collected (for example when your script returns). -Avi Marcus On Wed, Feb 13, 2013 at 8:01 PM, Archana Venugopan wrote: > Hi,**** > > ** ** > > Am trying close DB connection in lua script(dbh:close()) in freeswitch. > But I am getting this error**** > > ** ** > > 2013-02-13 16:43:41.327172 [ERR] mod_lua.cpp:102 > /usr/local/freeswitch/scripts/directory.lua:66: attempt to call method > 'close' (a nil value)**** > > stack traceback:**** > > ** ** > > Can anyone please let me know why my lua is not recognising the close > function? Is there anyother ways to close my DB connection in lua? Thanks. > **** > > ** ** > > Regards,**** > > Archana**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130213/52ea7647/attachment.html From steveayre at gmail.com Wed Feb 13 21:07:10 2013 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 13 Feb 2013 18:07:10 +0000 Subject: [Freeswitch-users] Nibblebill database question In-Reply-To: References: Message-ID: Do the SQL queries work ok within isql? On 13 February 2013 17:51, David Villasmil wrote: > And i can connect from the cli: > > root at lamp ~# isql viking > +---------------------------------------+ > | Connected! | > | | > | sql-statement | > | help [tablename] | > | quit | > | | > +---------------------------------------+ > SQL> > > strange... > > > On Wed, Feb 13, 2013 at 6:46 PM, Steven Ayre wrote: > >> http://jira.freeswitch.org/browse/FS-5100 >> >> >> On 13 February 2013 17:40, Steven Ayre wrote: >> >>> Specifically it's introduced by commit 5c176e8 which was resolving >>> FS-4810, which is indeed adding support for the the generalised DSN >>> interface. >>> >>> >>> http://fisheye.freeswitch.org/changelog/freeswitch.git?showid=5c176e889d8cc2da7efb18a89782407ee5524a74 >>> http://jira.freeswitch.org/browse/FS-4810 >>> >>> -Steve >>> >>> >>> >>> On 13 February 2013 17:37, Steven Ayre wrote: >>> >>>> Looks like this difference is brought in by the modifications to >>>> support the generalised DSN interface. odbc-dsn would be consistent with >>>> all other modules, so I guess that's why it's happened. >>>> >>>> But... >>>> The HEAD of v1.2.stable understands db_dsn but not odbc-dsn. >>>> The HEAD of master understands odbc-dsn but not db_dsn. >>>> >>>> This is a configuration breaking behaviour, which doesn't seem to have >>>> been documented! >>>> >>>> At the very least it'd make sense for the master branch to understand >>>> both odbc-dsn and db_dsn for backwards compatibility. I'll file a Jira with >>>> a patch (it's a trivial fix) in a moment. >>>> >>>> -Steve >>>> >>>> >>>> >>>> >>>> >>>> >>>> On 13 February 2013 17:33, Steven Ayre wrote: >>>> >>>>> How are you setting the DSN? >>>>> >>>>> The param name mentioned earlier is wrong. Reading the source the >>>>> correct parameter is "odbc-dsn". Looks like the wiki documentation is >>>>> incorrect. >>>>> >>>>> -Steve >>>>> >>>>> >>>>> >>>>> >>>>> On 13 February 2013 15:33, David Villasmil < >>>>> david.villasmil.work at gmail.com> wrote: >>>>> >>>>>> hello Avi, >>>>>> >>>>>> that's right, but nevermind that, that's the output when I was trying >>>>>> something else: >>>>>> >>>>>> 2013-02-12 19:32:07.784156 [INFO] mod_nibblebill.c:498 Beginning new >>>>>> billing on 58e6762b-c049-470e-84f7-519a815ab6f0 >>>>>> 2013-02-12 19:32:07.784156 [CRIT] mod_nibblebill.c:535 Failed to log >>>>>> to database! >>>>>> 2013-02-12 19:32:07.784156 [ERR] mod_nibblebill.c:380 Error running >>>>>> this query: [SELECT balance AS nibble_balance FROM sip_users WHERE id='1'] >>>>>> >>>>>> >>>>>> On Wed, Feb 13, 2013 at 12:58 PM, Avi Marcus wrote: >>>>>> >>>>>>> Your logs showed the queries run were: >>>>>>> >>>>>>> UPDATE sip_users SET balance=balance-0.018983 WHERE ='1' >>>>>>> SELECT balance AS nibble_balance FROM sip_users WHERE ='1' >>>>>>> >>>>>>> You just tried now.. "WHERE *id*='1' " -- they are not the same >>>>>>> query... >>>>>>> >>>>>>> -Avi >>>>>>> >>>>>>> >>>>>>> >>>>>>> On Wed, Feb 13, 2013 at 12:48 PM, David Villasmil < >>>>>>> david.villasmil.work at gmail.com> wrote: >>>>>>> >>>>>>>> Hello Rupa, >>>>>>>> >>>>>>>> the query if ok, the user is there: >>>>>>>> >>>>>>>> mysql> select * from sip_users; >>>>>>>> >>>>>>>> +----+----------+----------+----------------+------------+--------------+--------------+ >>>>>>>> | id | username | password | rateplan | balance | domain >>>>>>>> | max_channels | >>>>>>>> >>>>>>>> +----+----------+----------+----------------+------------+--------------+--------------+ >>>>>>>> | 1 | 1002 | 1234 | retail_default | 100.000000 | >>>>>>>> 192.168.1.44 | 1 | >>>>>>>> >>>>>>>> +----+----------+----------+----------------+------------+--------------+--------------+ >>>>>>>> 1 row in set (0.00 sec) >>>>>>>> >>>>>>>> >>>>>>>> and the actual sql executed by nibblebill: >>>>>>>> >>>>>>>> mysql> SELECT balance AS nibble_balance FROM sip_users WHERE id='1'; >>>>>>>> +----------------+ >>>>>>>> | nibble_balance | >>>>>>>> +----------------+ >>>>>>>> | 100.000000 | >>>>>>>> +----------------+ >>>>>>>> 1 row in set (0.00 sec) >>>>>>>> >>>>>>>> >>>>>>>> so that's ok.. >>>>>>>> >>>>>>>> >>>>>>>> any othe ideas? >>>>>>>> >>>>>>>> Thanks a lot for your help! >>>>>>>> >>>>>>>> >>>>>>>> David >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> On Wed, Feb 13, 2013 at 3:35 AM, Rupa Schomaker wrote: >>>>>>>> >>>>>>>>> >>>>>>>>> On Tue, Feb 12, 2013 at 5:26 PM, David Villasmil < >>>>>>>>> david.villasmil.work at gmail.com> wrote: >>>>>>>>> >>>>>>>>>> call goes out fine... no billing though: >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> 2013-02-12 19:21:49.844277 [DEBUG] mod_nibblebill.c:504 22 >>>>>>>>>> seconds passed since last bill time of 2013-02-12 19:21:27 >>>>>>>>>> 2013-02-12 19:21:49.844277 [DEBUG] mod_nibblebill.c:521 Billing >>>>>>>>>> $0.018983 to 1 (Call: d442dc6b-a3a7-423a-8450-598d17ca43b3 / 0.000000 so >>>>>>>>>> far) >>>>>>>>>> 2013-02-12 19:21:49.844277 [DEBUG] mod_nibblebill.c:351 Doing >>>>>>>>>> update query >>>>>>>>>> [UPDATE sip_users SET balance=balance-0.018983 WHERE ='1'] >>>>>>>>>> 2013-02-12 19:21:49.844277 [CRIT] mod_nibblebill.c:535 Failed to >>>>>>>>>> log to database! >>>>>>>>>> 2013-02-12 19:21:49.844277 [DEBUG] mod_nibblebill.c:378 Doing >>>>>>>>>> lookup query >>>>>>>>>> [SELECT balance AS nibble_balance FROM sip_users WHERE ='1'] >>>>>>>>>> 2013-02-12 19:21:49.844277 [ERR] mod_nibblebill.c:380 Error >>>>>>>>>> running this query: [SELECT balance AS nibble_balance FROM sip_users WHERE >>>>>>>>>> ='1'] >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> the select go fine if executed "by hand" >>>>>>>>>> >>>>>>>>> >>>>>>>>> None of those SQL statements are valid. (hint: WHERE = '1' is no >>>>>>>>> good). Been a while since I hacked/worked with nibblebill, but I'm >>>>>>>>> pretty sure the 'id' field isn't not being filled in the sql for some >>>>>>>>> reason. >>>>>>>>> >>>>>>>>> >>>>>>>>> -- >>>>>>>>> -Rupa >>>>>>>>> >>>>>>>>> >>>>>>>>> _________________________________________________________________________ >>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>> consulting at freeswitch.org >>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> Official FreeSWITCH Sites >>>>>>>>> http://www.freeswitch.org >>>>>>>>> http://wiki.freeswitch.org >>>>>>>>> http://www.cluecon.com >>>>>>>>> >>>>>>>>> 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 >>>>>>>>> >>>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> _________________________________________________________________________ >>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>> consulting at freeswitch.org >>>>>>>> http://www.freeswitchsolutions.com >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> Official FreeSWITCH Sites >>>>>>>> http://www.freeswitch.org >>>>>>>> http://wiki.freeswitch.org >>>>>>>> http://www.cluecon.com >>>>>>>> >>>>>>>> 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 >>>>>>>> >>>>>>>> >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://wiki.freeswitch.org >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> 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 >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> 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 >>>>>> >>>>>> >>>>> >>>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130213/6a225da2/attachment-0001.html From kris at kriskinc.com Wed Feb 13 21:16:58 2013 From: kris at kriskinc.com (Kristian Kielhofner) Date: Wed, 13 Feb 2013 13:16:58 -0500 Subject: [Freeswitch-users] best freeswitch timer In-Reply-To: References: Message-ID: Keep in mind you also need a recent glibc that supports timerfd. On Wed, Feb 13, 2013 at 9:55 AM, Ken Rice wrote: > Actually it defaults to using timerfd if its enabled in your kernel.... > There should be no debate on that, I reviewed that code with tony... > > If you don?t have a timerfd enabled kernel it uses the old default soft > timer... No you cant tell the difference in there currently with just a show > timer command, however that?s likely to change... > > Are you running like centos6 or debian squeeze with a stock kernel? Then you > likely have timerfd and are using it unless you changed the configs to do > something else... > > Theres always a lot of debate around this, but the fact is, unless you are > seeing crappy audio issues, you probably shouldn?t mess with the timers > > > > > On 2/13/13 4:31 AM, "Steven Ayre" wrote: > > 'show timer' shows all loaded timer modules. That shows none are loaded. > > CORE_SOFTTIMER_MODULE is the soft timer built into the FS core... it may or > may not use timerfd, depending on your system. > > You pick which timer to actually use in your configuration files... for > example the rtp-timer-name Sofia profile param. > > Use the timer_test command to see which timer actually works best for you > (ie most accurate, least jitter). > http://wiki.freeswitch.org/wiki/Mod_commands#timer_test > > -Steve > > > > On 12 February 2013 22:48, jay binks wrote: > > so does : > > freeswitch at default> show timer > type,name,ikey > timer,soft,CORE_SOFTTIMER_MODULE > > 1 total. > > on FreeSWITCH Version 1.2.6+git~20130104T154559Z~a4247651ca (git > a424765 2013-01-04 15:45:59Z) > > indicate it will be using timerfd calls or is this the old softtimer ? > > On 13 February 2013 02:21, Ken Rice wrote: >> The default one is the best one on linux with kernels starting around >> 2.6.32... Anything after that and you are good to go... >> >> The other timers are left around for other platforms >> >> K >> >> >> >> On 2/12/13 10:04 AM, "Fran?ois Delawarde" >> wrote: >> >>> Hello, >> >> Could someone knowing the different FS timers clarify the >>> difference >> between those: >> >> - default core softtimer >> - core heavy timer (does >>> this option still exists?) >> - posix timer >> - timerfd >> >> Can any of those provide a >>> significant performance improvement, or is it >> just not worth switching from >>> the default? >> >> In any case, which one would be the recommended/best one under >>> Linux >> with a relatively recent kernel >>> (>3.0)? >> >> Thanks, >> Fran?ois. >> >> >> >> ________________________________________________ >>> _________________________ >> Professional FreeSWITCH Consulting >>> Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSW >>> ITCH-powered IP PBX: The CudaTel Communication >>> Server >> >> >> Official FreeSWITCH >>> Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon. >>> com >> >> FreeSWITCH-users mailing >>> list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman >>> /listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/opt >>> ions/freeswitch-users >> http://www.freeswitch.org >> >> >> -- >> Ken >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> irc.freenode.net #freeswitch > >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 > > > > -- > Sincerely > > Jay > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > ________________________________ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > -- > Ken > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > irc.freenode.net #freeswitch > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -- Kristian Kielhofner From steveayre at gmail.com Wed Feb 13 21:23:04 2013 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 13 Feb 2013 18:23:04 +0000 Subject: [Freeswitch-users] best freeswitch timer In-Reply-To: References: Message-ID: +1 Debian Lenny had timerfd in the kernel but not in glibc. There's an option --enable-timerfd-wrapper for such systems since there timerfd can be used via syscall. Squeeze and newer have it in both (so no need for the wrapper). -Steve On 13 February 2013 18:16, Kristian Kielhofner wrote: > Keep in mind you also need a recent glibc that supports timerfd. > > On Wed, Feb 13, 2013 at 9:55 AM, Ken Rice wrote: > > Actually it defaults to using timerfd if its enabled in your kernel.... > > There should be no debate on that, I reviewed that code with tony... > > > > If you don?t have a timerfd enabled kernel it uses the old default soft > > timer... No you cant tell the difference in there currently with just a > show > > timer command, however that?s likely to change... > > > > Are you running like centos6 or debian squeeze with a stock kernel? Then > you > > likely have timerfd and are using it unless you changed the configs to do > > something else... > > > > Theres always a lot of debate around this, but the fact is, unless you > are > > seeing crappy audio issues, you probably shouldn?t mess with the timers > > > > > > > > > > On 2/13/13 4:31 AM, "Steven Ayre" wrote: > > > > 'show timer' shows all loaded timer modules. That shows none are loaded. > > > > CORE_SOFTTIMER_MODULE is the soft timer built into the FS core... it may > or > > may not use timerfd, depending on your system. > > > > You pick which timer to actually use in your configuration files... for > > example the rtp-timer-name Sofia profile param. > > > > Use the timer_test command to see which timer actually works best for you > > (ie most accurate, least jitter). > > http://wiki.freeswitch.org/wiki/Mod_commands#timer_test > > > > -Steve > > > > > > > > On 12 February 2013 22:48, jay binks wrote: > > > > so does : > > > > freeswitch at default> show timer > > type,name,ikey > > timer,soft,CORE_SOFTTIMER_MODULE > > > > 1 total. > > > > on FreeSWITCH Version 1.2.6+git~20130104T154559Z~a4247651ca (git > > a424765 2013-01-04 15:45:59Z) > > > > indicate it will be using timerfd calls or is this the old softtimer ? > > > > On 13 February 2013 02:21, Ken Rice wrote: > >> The default one is the best one on linux with kernels starting around > >> 2.6.32... Anything after that and you are good to go... > >> > >> The other timers are left around for other platforms > >> > >> K > >> > >> > >> > >> On 2/12/13 10:04 AM, "Fran?ois Delawarde" > > >> wrote: > >> > >>> Hello, > >> > >> Could someone knowing the different FS timers clarify the > >>> difference > >> between those: > >> > >> - default core softtimer > >> - core heavy timer (does > >>> this option still exists?) > >> - posix timer > >> - timerfd > >> > >> Can any of those provide a > >>> significant performance improvement, or is it > >> just not worth switching from > >>> the default? > >> > >> In any case, which one would be the recommended/best one under > >>> Linux > >> with a relatively recent kernel > >>> (>3.0)? > >> > >> Thanks, > >> Fran?ois. > >> > >> > >> > >> ________________________________________________ > >>> _________________________ > >> Professional FreeSWITCH Consulting > >>> Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> FreeSW > >>> ITCH-powered IP PBX: The CudaTel Communication > >>> Server > >> > >> > >> Official FreeSWITCH > >>> Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon. > >>> com > >> > >> FreeSWITCH-users mailing > >>> list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman > >>> /listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/opt > >>> ions/freeswitch-users > >> http://www.freeswitch.org > >> > >> > >> -- > >> Ken > >> http://www.FreeSWITCH.org > >> http://www.ClueCon.com > >> http://www.OSTAG.org > >> irc.freenode.net #freeswitch > > > >> > >> > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> 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 > > > > > > > > -- > > Sincerely > > > > Jay > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > 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 > > > > > > > > ________________________________ > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > 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 > > > > > > -- > > Ken > > http://www.FreeSWITCH.org > > http://www.ClueCon.com > > http://www.OSTAG.org > > irc.freenode.net #freeswitch > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > 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 > > > > > > -- > Kristian Kielhofner > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130213/d18eabf0/attachment-0001.html From krice at freeswitch.org Wed Feb 13 21:40:57 2013 From: krice at freeswitch.org (Ken Rice) Date: Wed, 13 Feb 2013 12:40:57 -0600 Subject: [Freeswitch-users] best freeswitch timer In-Reply-To: Message-ID: Hey Kris, Do you know the version of glibc where they added that in? On 2/13/13 12:16 PM, "Kristian Kielhofner" wrote: > Keep in mind you also need a recent glibc that supports timerfd. > > On Wed, Feb 13, 2013 at 9:55 AM, Ken Rice wrote: >> Actually it defaults to using timerfd if its enabled in your kernel.... >> There should be no debate on that, I reviewed that code with tony... >> >> If you don?t have a timerfd enabled kernel it uses the old default soft >> timer... No you cant tell the difference in there currently with just a show >> timer command, however that?s likely to change... >> >> Are you running like centos6 or debian squeeze with a stock kernel? Then you >> likely have timerfd and are using it unless you changed the configs to do >> something else... >> >> Theres always a lot of debate around this, but the fact is, unless you are >> seeing crappy audio issues, you probably shouldn?t mess with the timers >> >> >> >> >> On 2/13/13 4:31 AM, "Steven Ayre" wrote: >> >> 'show timer' shows all loaded timer modules. That shows none are loaded. >> >> CORE_SOFTTIMER_MODULE is the soft timer built into the FS core... it may or >> may not use timerfd, depending on your system. >> >> You pick which timer to actually use in your configuration files... for >> example the rtp-timer-name Sofia profile param. >> >> Use the timer_test command to see which timer actually works best for you >> (ie most accurate, least jitter). >> http://wiki.freeswitch.org/wiki/Mod_commands#timer_test >> >> -Steve >> >> >> >> On 12 February 2013 22:48, jay binks wrote: >> >> so does : >> >> freeswitch at default> show timer >> type,name,ikey >> timer,soft,CORE_SOFTTIMER_MODULE >> >> 1 total. >> >> on FreeSWITCH Version 1.2.6+git~20130104T154559Z~a4247651ca (git >> a424765 2013-01-04 15:45:59Z) >> >> indicate it will be using timerfd calls or is this the old softtimer ? >> >> On 13 February 2013 02:21, Ken Rice wrote: >>> The default one is the best one on linux with kernels starting around >>> 2.6.32... Anything after that and you are good to go... >>> >>> The other timers are left around for other platforms >>> >>> K >>> >>> >>> >>> On 2/12/13 10:04 AM, "Fran?ois Delawarde" >>> wrote: >>> >>>> Hello, >>> >>> Could someone knowing the different FS timers clarify the >>>> difference >>> between those: >>> >>> - default core softtimer >>> - core heavy timer (does >>>> this option still exists?) >>> - posix timer >>> - timerfd >>> >>> Can any of those provide a >>>> significant performance improvement, or is it >>> just not worth switching from >>>> the default? >>> >>> In any case, which one would be the recommended/best one under >>>> Linux >>> with a relatively recent kernel >>>> (>3.0)? >>> >>> Thanks, >>> Fran?ois. >>> >>> >>> >>> ________________________________________________ >>>> _________________________ >>> Professional FreeSWITCH Consulting >>>> Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> FreeSW >>>> ITCH-powered IP PBX: The CudaTel Communication >>>> Server >>> >>> >>> Official FreeSWITCH >>>> Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon. >>>> com >>> >>> FreeSWITCH-users mailing >>>> list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman >>>> /listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/opt >>>> ions/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> -- >>> Ken >>> http://www.FreeSWITCH.org >>> http://www.ClueCon.com >>> http://www.OSTAG.org >>> irc.freenode.net #freeswitch >> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >> >> >> >> -- >> Sincerely >> >> Jay >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> >> >> ________________________________ >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> >> -- >> Ken >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> irc.freenode.net #freeswitch >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> > > -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch From steveayre at gmail.com Wed Feb 13 22:04:49 2013 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 13 Feb 2013 19:04:49 +0000 Subject: [Freeswitch-users] best freeswitch timer In-Reply-To: References: Message-ID: http://www.kernel.org/doc/man-pages/online/pages/man2/timerfd_create.2.html "These system calls are available on Linux since kernel 2.6.25. Library support is provided by glibc since version 2.8." On 13 February 2013 18:40, Ken Rice wrote: > Hey Kris, > > Do you know the version of glibc where they added that in? > > > On 2/13/13 12:16 PM, "Kristian Kielhofner" wrote: > > > Keep in mind you also need a recent glibc that supports timerfd. > > > > On Wed, Feb 13, 2013 at 9:55 AM, Ken Rice wrote: > >> Actually it defaults to using timerfd if its enabled in your kernel.... > >> There should be no debate on that, I reviewed that code with tony... > >> > >> If you don?t have a timerfd enabled kernel it uses the old default soft > >> timer... No you cant tell the difference in there currently with just a > show > >> timer command, however that?s likely to change... > >> > >> Are you running like centos6 or debian squeeze with a stock kernel? > Then you > >> likely have timerfd and are using it unless you changed the configs to > do > >> something else... > >> > >> Theres always a lot of debate around this, but the fact is, unless you > are > >> seeing crappy audio issues, you probably shouldn?t mess with the timers > >> > >> > >> > >> > >> On 2/13/13 4:31 AM, "Steven Ayre" wrote: > >> > >> 'show timer' shows all loaded timer modules. That shows none are loaded. > >> > >> CORE_SOFTTIMER_MODULE is the soft timer built into the FS core... it > may or > >> may not use timerfd, depending on your system. > >> > >> You pick which timer to actually use in your configuration files... for > >> example the rtp-timer-name Sofia profile param. > >> > >> Use the timer_test command to see which timer actually works best for > you > >> (ie most accurate, least jitter). > >> http://wiki.freeswitch.org/wiki/Mod_commands#timer_test > >> > >> -Steve > >> > >> > >> > >> On 12 February 2013 22:48, jay binks wrote: > >> > >> so does : > >> > >> freeswitch at default> show timer > >> type,name,ikey > >> timer,soft,CORE_SOFTTIMER_MODULE > >> > >> 1 total. > >> > >> on FreeSWITCH Version 1.2.6+git~20130104T154559Z~a4247651ca (git > >> a424765 2013-01-04 15:45:59Z) > >> > >> indicate it will be using timerfd calls or is this the old softtimer ? > >> > >> On 13 February 2013 02:21, Ken Rice wrote: > >>> The default one is the best one on linux with kernels starting around > >>> 2.6.32... Anything after that and you are good to go... > >>> > >>> The other timers are left around for other platforms > >>> > >>> K > >>> > >>> > >>> > >>> On 2/12/13 10:04 AM, "Fran?ois Delawarde" < > fdelawarde at wirelessmundi.com> > >>> wrote: > >>> > >>>> Hello, > >>> > >>> Could someone knowing the different FS timers clarify the > >>>> difference > >>> between those: > >>> > >>> - default core softtimer > >>> - core heavy timer (does > >>>> this option still exists?) > >>> - posix timer > >>> - timerfd > >>> > >>> Can any of those provide a > >>>> significant performance improvement, or is it > >>> just not worth switching from > >>>> the default? > >>> > >>> In any case, which one would be the recommended/best one under > >>>> Linux > >>> with a relatively recent kernel > >>>> (>3.0)? > >>> > >>> Thanks, > >>> Fran?ois. > >>> > >>> > >>> > >>> ________________________________________________ > >>>> _________________________ > >>> Professional FreeSWITCH Consulting > >>>> Services: > >>> consulting at freeswitch.org > >>> http://www.freeswitchsolutions.com > >>> > >>> FreeSW > >>>> ITCH-powered IP PBX: The CudaTel Communication > >>>> Server > >>> > >>> > >>> Official FreeSWITCH > >>>> Sites > >>> http://www.freeswitch.org > >>> http://wiki.freeswitch.org > >>> http://www.cluecon. > >>>> com > >>> > >>> FreeSWITCH-users mailing > >>>> list > >>> FreeSWITCH-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman > >>>> /listinfo/freeswitch-users > >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/opt > >>>> ions/freeswitch-users > >>> http://www.freeswitch.org > >>> > >>> > >>> -- > >>> Ken > >>> http://www.FreeSWITCH.org > >>> http://www.ClueCon.com > >>> http://www.OSTAG.org > >>> irc.freenode.net #freeswitch > >> > >>> > >>> > >>> > >>> > _________________________________________________________________________ > >>> Professional FreeSWITCH Consulting Services: > >>> consulting at freeswitch.org > >>> http://www.freeswitchsolutions.com > >>> > >>> > >>> > >>> > >>> Official FreeSWITCH Sites > >>> http://www.freeswitch.org > >>> http://wiki.freeswitch.org > >>> http://www.cluecon.com > >>> > >>> 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 > >> > >> > >> > >> -- > >> Sincerely > >> > >> Jay > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> 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 > >> > >> > >> > >> ________________________________ > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> 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 > >> > >> > >> -- > >> Ken > >> http://www.FreeSWITCH.org > >> http://www.ClueCon.com > >> http://www.OSTAG.org > >> irc.freenode.net #freeswitch > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> 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 > >> > > > > > > -- > Ken > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > irc.freenode.net #freeswitch > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130213/3c6f02e7/attachment-0001.html From david.villasmil.work at gmail.com Wed Feb 13 22:07:35 2013 From: david.villasmil.work at gmail.com (David Villasmil) Date: Wed, 13 Feb 2013 20:07:35 +0100 Subject: [Freeswitch-users] Nibblebill database question In-Reply-To: References: Message-ID: Yeah, everything works within isql: root at lamp ~# isql viking +---------------------------------------+ | Connected! | | | | sql-statement | | help [tablename] | | quit | | | +---------------------------------------+ SQL> SELECT balance AS nibble_balance FROM sip_users WHERE id='1' +---------------+ | nibble_balance| +---------------+ | 100.000000 | +---------------+ SQLRowCount returns 1 1 rows fetched SQL> UPDATE sip_users SET balance=balance-0.044817 WHERE id='1' SQLRowCount returns 1 SQL> SELECT balance AS nibble_balance FROM sip_users WHERE id='1' +---------------+ | nibble_balance| +---------------+ | 99.955183 | +---------------+ SQLRowCount returns 1 1 rows fetched SQL> and freeswitch core is also using odbc (although a differen dsn) David On Wed, Feb 13, 2013 at 7:07 PM, Steven Ayre wrote: > Do the SQL queries work ok within isql? > > > > On 13 February 2013 17:51, David Villasmil > wrote: > >> And i can connect from the cli: >> >> root at lamp ~# isql viking >> +---------------------------------------+ >> | Connected! | >> | | >> | sql-statement | >> | help [tablename] | >> | quit | >> | | >> +---------------------------------------+ >> SQL> >> >> strange... >> >> >> On Wed, Feb 13, 2013 at 6:46 PM, Steven Ayre wrote: >> >>> http://jira.freeswitch.org/browse/FS-5100 >>> >>> >>> On 13 February 2013 17:40, Steven Ayre wrote: >>> >>>> Specifically it's introduced by commit 5c176e8 which was resolving >>>> FS-4810, which is indeed adding support for the the generalised DSN >>>> interface. >>>> >>>> >>>> http://fisheye.freeswitch.org/changelog/freeswitch.git?showid=5c176e889d8cc2da7efb18a89782407ee5524a74 >>>> http://jira.freeswitch.org/browse/FS-4810 >>>> >>>> -Steve >>>> >>>> >>>> >>>> On 13 February 2013 17:37, Steven Ayre wrote: >>>> >>>>> Looks like this difference is brought in by the modifications to >>>>> support the generalised DSN interface. odbc-dsn would be consistent with >>>>> all other modules, so I guess that's why it's happened. >>>>> >>>>> But... >>>>> The HEAD of v1.2.stable understands db_dsn but not odbc-dsn. >>>>> The HEAD of master understands odbc-dsn but not db_dsn. >>>>> >>>>> This is a configuration breaking behaviour, which doesn't seem to have >>>>> been documented! >>>>> >>>>> At the very least it'd make sense for the master branch to understand >>>>> both odbc-dsn and db_dsn for backwards compatibility. I'll file a Jira with >>>>> a patch (it's a trivial fix) in a moment. >>>>> >>>>> -Steve >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> On 13 February 2013 17:33, Steven Ayre wrote: >>>>> >>>>>> How are you setting the DSN? >>>>>> >>>>>> The param name mentioned earlier is wrong. Reading the source the >>>>>> correct parameter is "odbc-dsn". Looks like the wiki documentation is >>>>>> incorrect. >>>>>> >>>>>> -Steve >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> On 13 February 2013 15:33, David Villasmil < >>>>>> david.villasmil.work at gmail.com> wrote: >>>>>> >>>>>>> hello Avi, >>>>>>> >>>>>>> that's right, but nevermind that, that's the output when I was >>>>>>> trying something else: >>>>>>> >>>>>>> 2013-02-12 19:32:07.784156 [INFO] mod_nibblebill.c:498 Beginning new >>>>>>> billing on 58e6762b-c049-470e-84f7-519a815ab6f0 >>>>>>> 2013-02-12 19:32:07.784156 [CRIT] mod_nibblebill.c:535 Failed to log >>>>>>> to database! >>>>>>> 2013-02-12 19:32:07.784156 [ERR] mod_nibblebill.c:380 Error running >>>>>>> this query: [SELECT balance AS nibble_balance FROM sip_users WHERE id='1'] >>>>>>> >>>>>>> >>>>>>> On Wed, Feb 13, 2013 at 12:58 PM, Avi Marcus wrote: >>>>>>> >>>>>>>> Your logs showed the queries run were: >>>>>>>> >>>>>>>> UPDATE sip_users SET balance=balance-0.018983 WHERE ='1' >>>>>>>> SELECT balance AS nibble_balance FROM sip_users WHERE ='1' >>>>>>>> >>>>>>>> You just tried now.. "WHERE *id*='1' " -- they are not the same >>>>>>>> query... >>>>>>>> >>>>>>>> -Avi >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> On Wed, Feb 13, 2013 at 12:48 PM, David Villasmil < >>>>>>>> david.villasmil.work at gmail.com> wrote: >>>>>>>> >>>>>>>>> Hello Rupa, >>>>>>>>> >>>>>>>>> the query if ok, the user is there: >>>>>>>>> >>>>>>>>> mysql> select * from sip_users; >>>>>>>>> >>>>>>>>> +----+----------+----------+----------------+------------+--------------+--------------+ >>>>>>>>> | id | username | password | rateplan | balance | domain >>>>>>>>> | max_channels | >>>>>>>>> >>>>>>>>> +----+----------+----------+----------------+------------+--------------+--------------+ >>>>>>>>> | 1 | 1002 | 1234 | retail_default | 100.000000 | >>>>>>>>> 192.168.1.44 | 1 | >>>>>>>>> >>>>>>>>> +----+----------+----------+----------------+------------+--------------+--------------+ >>>>>>>>> 1 row in set (0.00 sec) >>>>>>>>> >>>>>>>>> >>>>>>>>> and the actual sql executed by nibblebill: >>>>>>>>> >>>>>>>>> mysql> SELECT balance AS nibble_balance FROM sip_users WHERE >>>>>>>>> id='1'; >>>>>>>>> +----------------+ >>>>>>>>> | nibble_balance | >>>>>>>>> +----------------+ >>>>>>>>> | 100.000000 | >>>>>>>>> +----------------+ >>>>>>>>> 1 row in set (0.00 sec) >>>>>>>>> >>>>>>>>> >>>>>>>>> so that's ok.. >>>>>>>>> >>>>>>>>> >>>>>>>>> any othe ideas? >>>>>>>>> >>>>>>>>> Thanks a lot for your help! >>>>>>>>> >>>>>>>>> >>>>>>>>> David >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> On Wed, Feb 13, 2013 at 3:35 AM, Rupa Schomaker wrote: >>>>>>>>> >>>>>>>>>> >>>>>>>>>> On Tue, Feb 12, 2013 at 5:26 PM, David Villasmil < >>>>>>>>>> david.villasmil.work at gmail.com> wrote: >>>>>>>>>> >>>>>>>>>>> call goes out fine... no billing though: >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> 2013-02-12 19:21:49.844277 [DEBUG] mod_nibblebill.c:504 22 >>>>>>>>>>> seconds passed since last bill time of 2013-02-12 19:21:27 >>>>>>>>>>> 2013-02-12 19:21:49.844277 [DEBUG] mod_nibblebill.c:521 Billing >>>>>>>>>>> $0.018983 to 1 (Call: d442dc6b-a3a7-423a-8450-598d17ca43b3 / 0.000000 so >>>>>>>>>>> far) >>>>>>>>>>> 2013-02-12 19:21:49.844277 [DEBUG] mod_nibblebill.c:351 Doing >>>>>>>>>>> update query >>>>>>>>>>> [UPDATE sip_users SET balance=balance-0.018983 WHERE ='1'] >>>>>>>>>>> 2013-02-12 19:21:49.844277 [CRIT] mod_nibblebill.c:535 Failed to >>>>>>>>>>> log to database! >>>>>>>>>>> 2013-02-12 19:21:49.844277 [DEBUG] mod_nibblebill.c:378 Doing >>>>>>>>>>> lookup query >>>>>>>>>>> [SELECT balance AS nibble_balance FROM sip_users WHERE ='1'] >>>>>>>>>>> 2013-02-12 19:21:49.844277 [ERR] mod_nibblebill.c:380 Error >>>>>>>>>>> running this query: [SELECT balance AS nibble_balance FROM sip_users WHERE >>>>>>>>>>> ='1'] >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> the select go fine if executed "by hand" >>>>>>>>>>> >>>>>>>>>> >>>>>>>>>> None of those SQL statements are valid. (hint: WHERE = '1' is no >>>>>>>>>> good). Been a while since I hacked/worked with nibblebill, but I'm >>>>>>>>>> pretty sure the 'id' field isn't not being filled in the sql for some >>>>>>>>>> reason. >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> -- >>>>>>>>>> -Rupa >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> _________________________________________________________________________ >>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>> consulting at freeswitch.org >>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>> http://www.freeswitch.org >>>>>>>>>> http://wiki.freeswitch.org >>>>>>>>>> http://www.cluecon.com >>>>>>>>>> >>>>>>>>>> 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 >>>>>>>>>> >>>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> _________________________________________________________________________ >>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>> consulting at freeswitch.org >>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> Official FreeSWITCH Sites >>>>>>>>> http://www.freeswitch.org >>>>>>>>> http://wiki.freeswitch.org >>>>>>>>> http://www.cluecon.com >>>>>>>>> >>>>>>>>> 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 >>>>>>>>> >>>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> _________________________________________________________________________ >>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>> consulting at freeswitch.org >>>>>>>> http://www.freeswitchsolutions.com >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> Official FreeSWITCH Sites >>>>>>>> http://www.freeswitch.org >>>>>>>> http://wiki.freeswitch.org >>>>>>>> http://www.cluecon.com >>>>>>>> >>>>>>>> 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 >>>>>>>> >>>>>>>> >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://wiki.freeswitch.org >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> 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 >>>>>>> >>>>>>> >>>>>> >>>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130213/e4568e19/attachment-0001.html From steveayre at gmail.com Wed Feb 13 22:35:16 2013 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 13 Feb 2013 19:35:16 +0000 Subject: [Freeswitch-users] Nibblebill database question In-Reply-To: References: Message-ID: Is "viking" the actual DSN you're using? If so that format won't work any longer since the generalized interface was implemented. name:user:pass still works but plain name no longer works*.* See http://wiki.freeswitch.org/wiki/Release_Notes#odbc-dsn You're best off updating it to odbc://viking -Steve On 13 February 2013 19:07, David Villasmil wrote: > Yeah, everything works within isql: > > root at lamp ~# isql viking > +---------------------------------------+ > | Connected! | > | | > | sql-statement | > | help [tablename] | > | quit | > | | > +---------------------------------------+ > SQL> SELECT balance AS nibble_balance FROM sip_users WHERE id='1' > +---------------+ > | nibble_balance| > +---------------+ > | 100.000000 | > +---------------+ > SQLRowCount returns 1 > 1 rows fetched > SQL> UPDATE sip_users SET balance=balance-0.044817 WHERE id='1' > SQLRowCount returns 1 > SQL> SELECT balance AS nibble_balance FROM sip_users WHERE id='1' > +---------------+ > | nibble_balance| > +---------------+ > | 99.955183 | > +---------------+ > SQLRowCount returns 1 > 1 rows fetched > SQL> > > and freeswitch core is also using odbc (although a differen dsn) > > David > > > On Wed, Feb 13, 2013 at 7:07 PM, Steven Ayre wrote: > >> Do the SQL queries work ok within isql? >> >> >> >> On 13 February 2013 17:51, David Villasmil < >> david.villasmil.work at gmail.com> wrote: >> >>> And i can connect from the cli: >>> >>> root at lamp ~# isql viking >>> +---------------------------------------+ >>> | Connected! | >>> | | >>> | sql-statement | >>> | help [tablename] | >>> | quit | >>> | | >>> +---------------------------------------+ >>> SQL> >>> >>> strange... >>> >>> >>> On Wed, Feb 13, 2013 at 6:46 PM, Steven Ayre wrote: >>> >>>> http://jira.freeswitch.org/browse/FS-5100 >>>> >>>> >>>> On 13 February 2013 17:40, Steven Ayre wrote: >>>> >>>>> Specifically it's introduced by commit 5c176e8 which was resolving >>>>> FS-4810, which is indeed adding support for the the generalised DSN >>>>> interface. >>>>> >>>>> >>>>> http://fisheye.freeswitch.org/changelog/freeswitch.git?showid=5c176e889d8cc2da7efb18a89782407ee5524a74 >>>>> http://jira.freeswitch.org/browse/FS-4810 >>>>> >>>>> -Steve >>>>> >>>>> >>>>> >>>>> On 13 February 2013 17:37, Steven Ayre wrote: >>>>> >>>>>> Looks like this difference is brought in by the modifications to >>>>>> support the generalised DSN interface. odbc-dsn would be consistent with >>>>>> all other modules, so I guess that's why it's happened. >>>>>> >>>>>> But... >>>>>> The HEAD of v1.2.stable understands db_dsn but not odbc-dsn. >>>>>> The HEAD of master understands odbc-dsn but not db_dsn. >>>>>> >>>>>> This is a configuration breaking behaviour, which doesn't seem to >>>>>> have been documented! >>>>>> >>>>>> At the very least it'd make sense for the master branch to understand >>>>>> both odbc-dsn and db_dsn for backwards compatibility. I'll file a Jira with >>>>>> a patch (it's a trivial fix) in a moment. >>>>>> >>>>>> -Steve >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> On 13 February 2013 17:33, Steven Ayre wrote: >>>>>> >>>>>>> How are you setting the DSN? >>>>>>> >>>>>>> The param name mentioned earlier is wrong. Reading the source the >>>>>>> correct parameter is "odbc-dsn". Looks like the wiki documentation is >>>>>>> incorrect. >>>>>>> >>>>>>> -Steve >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> On 13 February 2013 15:33, David Villasmil < >>>>>>> david.villasmil.work at gmail.com> wrote: >>>>>>> >>>>>>>> hello Avi, >>>>>>>> >>>>>>>> that's right, but nevermind that, that's the output when I was >>>>>>>> trying something else: >>>>>>>> >>>>>>>> 2013-02-12 19:32:07.784156 [INFO] mod_nibblebill.c:498 Beginning >>>>>>>> new billing on 58e6762b-c049-470e-84f7-519a815ab6f0 >>>>>>>> 2013-02-12 19:32:07.784156 [CRIT] mod_nibblebill.c:535 Failed to >>>>>>>> log to database! >>>>>>>> 2013-02-12 19:32:07.784156 [ERR] mod_nibblebill.c:380 Error running >>>>>>>> this query: [SELECT balance AS nibble_balance FROM sip_users WHERE id='1'] >>>>>>>> >>>>>>>> >>>>>>>> On Wed, Feb 13, 2013 at 12:58 PM, Avi Marcus wrote: >>>>>>>> >>>>>>>>> Your logs showed the queries run were: >>>>>>>>> >>>>>>>>> UPDATE sip_users SET balance=balance-0.018983 WHERE ='1' >>>>>>>>> SELECT balance AS nibble_balance FROM sip_users WHERE ='1' >>>>>>>>> >>>>>>>>> You just tried now.. "WHERE *id*='1' " -- they are not the same >>>>>>>>> query... >>>>>>>>> >>>>>>>>> -Avi >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> On Wed, Feb 13, 2013 at 12:48 PM, David Villasmil < >>>>>>>>> david.villasmil.work at gmail.com> wrote: >>>>>>>>> >>>>>>>>>> Hello Rupa, >>>>>>>>>> >>>>>>>>>> the query if ok, the user is there: >>>>>>>>>> >>>>>>>>>> mysql> select * from sip_users; >>>>>>>>>> >>>>>>>>>> +----+----------+----------+----------------+------------+--------------+--------------+ >>>>>>>>>> | id | username | password | rateplan | balance | domain >>>>>>>>>> | max_channels | >>>>>>>>>> >>>>>>>>>> +----+----------+----------+----------------+------------+--------------+--------------+ >>>>>>>>>> | 1 | 1002 | 1234 | retail_default | 100.000000 | >>>>>>>>>> 192.168.1.44 | 1 | >>>>>>>>>> >>>>>>>>>> +----+----------+----------+----------------+------------+--------------+--------------+ >>>>>>>>>> 1 row in set (0.00 sec) >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> and the actual sql executed by nibblebill: >>>>>>>>>> >>>>>>>>>> mysql> SELECT balance AS nibble_balance FROM sip_users WHERE >>>>>>>>>> id='1'; >>>>>>>>>> +----------------+ >>>>>>>>>> | nibble_balance | >>>>>>>>>> +----------------+ >>>>>>>>>> | 100.000000 | >>>>>>>>>> +----------------+ >>>>>>>>>> 1 row in set (0.00 sec) >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> so that's ok.. >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> any othe ideas? >>>>>>>>>> >>>>>>>>>> Thanks a lot for your help! >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> David >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> On Wed, Feb 13, 2013 at 3:35 AM, Rupa Schomaker wrote: >>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> On Tue, Feb 12, 2013 at 5:26 PM, David Villasmil < >>>>>>>>>>> david.villasmil.work at gmail.com> wrote: >>>>>>>>>>> >>>>>>>>>>>> call goes out fine... no billing though: >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> 2013-02-12 19:21:49.844277 [DEBUG] mod_nibblebill.c:504 22 >>>>>>>>>>>> seconds passed since last bill time of 2013-02-12 19:21:27 >>>>>>>>>>>> 2013-02-12 19:21:49.844277 [DEBUG] mod_nibblebill.c:521 Billing >>>>>>>>>>>> $0.018983 to 1 (Call: d442dc6b-a3a7-423a-8450-598d17ca43b3 / 0.000000 so >>>>>>>>>>>> far) >>>>>>>>>>>> 2013-02-12 19:21:49.844277 [DEBUG] mod_nibblebill.c:351 Doing >>>>>>>>>>>> update query >>>>>>>>>>>> [UPDATE sip_users SET balance=balance-0.018983 WHERE ='1'] >>>>>>>>>>>> 2013-02-12 19:21:49.844277 [CRIT] mod_nibblebill.c:535 Failed >>>>>>>>>>>> to log to database! >>>>>>>>>>>> 2013-02-12 19:21:49.844277 [DEBUG] mod_nibblebill.c:378 Doing >>>>>>>>>>>> lookup query >>>>>>>>>>>> [SELECT balance AS nibble_balance FROM sip_users WHERE ='1'] >>>>>>>>>>>> 2013-02-12 19:21:49.844277 [ERR] mod_nibblebill.c:380 Error >>>>>>>>>>>> running this query: [SELECT balance AS nibble_balance FROM sip_users WHERE >>>>>>>>>>>> ='1'] >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> the select go fine if executed "by hand" >>>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> None of those SQL statements are valid. (hint: WHERE = '1' is >>>>>>>>>>> no good). Been a while since I hacked/worked with nibblebill, but I'm >>>>>>>>>>> pretty sure the 'id' field isn't not being filled in the sql for some >>>>>>>>>>> reason. >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> -- >>>>>>>>>>> -Rupa >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> _________________________________________________________________________ >>>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>>> consulting at freeswitch.org >>>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>> http://wiki.freeswitch.org >>>>>>>>>>> http://www.cluecon.com >>>>>>>>>>> >>>>>>>>>>> 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 >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> _________________________________________________________________________ >>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>> consulting at freeswitch.org >>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>> http://www.freeswitch.org >>>>>>>>>> http://wiki.freeswitch.org >>>>>>>>>> http://www.cluecon.com >>>>>>>>>> >>>>>>>>>> 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 >>>>>>>>>> >>>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> _________________________________________________________________________ >>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>> consulting at freeswitch.org >>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> Official FreeSWITCH Sites >>>>>>>>> http://www.freeswitch.org >>>>>>>>> http://wiki.freeswitch.org >>>>>>>>> http://www.cluecon.com >>>>>>>>> >>>>>>>>> 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 >>>>>>>>> >>>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> _________________________________________________________________________ >>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>> consulting at freeswitch.org >>>>>>>> http://www.freeswitchsolutions.com >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> Official FreeSWITCH Sites >>>>>>>> http://www.freeswitch.org >>>>>>>> http://wiki.freeswitch.org >>>>>>>> http://www.cluecon.com >>>>>>>> >>>>>>>> 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 >>>>>>>> >>>>>>>> >>>>>>> >>>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130213/b4912adc/attachment-0001.html From gassaad at emassembly.com Wed Feb 13 22:47:50 2013 From: gassaad at emassembly.com (George Assaad) Date: Wed, 13 Feb 2013 14:47:50 -0500 Subject: [Freeswitch-users] Best current ATA and Setup for Faxing In-Reply-To: <0B7A549A-767A-4DA6-B732-9999DC6B1B48@gmail.com> References: <0B7A549A-767A-4DA6-B732-9999DC6B1B48@gmail.com> Message-ID: Can you please share with us the setting for fax, if you don't mind. On Feb 13, 2013 11:17 AM, "Nick Vines" wrote: > I havent used the cisco atas, but I've had good success with the grand > stream ht502s and ht702s. My preferred faxing carrier is also flowroute. > > Nick > > On Feb 13, 2013, at 8:02 AM, Chris Ferreira wrote: > > > Hi Folks, > > > > > > We had previously been working with the Cisco SPA112 for T.38 Faxing > through Freeswitch and out to Flowroute. At the time we didn't get it > working 100% perfectly. > > > > > > I now have one of these SPA112's that I need to replace because someone > fried it by plugging in a much larger power supply by accident. > > > > > > Is my scenario above still a good starting point for Faxing or should I > be looking at another ATA or provider other than Flowroute? > > > > > > > > > > > > Also, one more question, and this may be better suited to the business > list (let me know if I should move it there), but we are looking for a part > time as needed remote or local Freeswitch Admin / Engineer / Tech. Our > previous person is MIA. > > > > > > > > > > Thanks Everyone! > > > > > > > > -Chris > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130213/03738126/attachment.html From david.villasmil.work at gmail.com Wed Feb 13 22:49:58 2013 From: david.villasmil.work at gmail.com (David Villasmil) Date: Wed, 13 Feb 2013 20:49:58 +0100 Subject: [Freeswitch-users] Nibblebill database question In-Reply-To: References: Message-ID: <6F00484B-1F8C-44F4-A9D4-9B8D5097E78A@gmail.com> Aha! That must it, i'll test and let you know... David --- David Villasmil On Feb 13, 2013, at 20:35, Steven Ayre wrote: > Is "viking" the actual DSN you're using? If so that format won't work any longer since the generalized interface was implemented. name:user:pass still works but plain name no longer works. > > See http://wiki.freeswitch.org/wiki/Release_Notes#odbc-dsn > > You're best off updating it to odbc://viking > > -Steve > > > > > On 13 February 2013 19:07, David Villasmil wrote: >> Yeah, everything works within isql: >> >> root at lamp ~# isql viking >> +---------------------------------------+ >> | Connected! | >> | | >> | sql-statement | >> | help [tablename] | >> | quit | >> | | >> +---------------------------------------+ >> SQL> SELECT balance AS nibble_balance FROM sip_users WHERE id='1' >> +---------------+ >> | nibble_balance| >> +---------------+ >> | 100.000000 | >> +---------------+ >> SQLRowCount returns 1 >> 1 rows fetched >> SQL> UPDATE sip_users SET balance=balance-0.044817 WHERE id='1' >> SQLRowCount returns 1 >> SQL> SELECT balance AS nibble_balance FROM sip_users WHERE id='1' >> +---------------+ >> | nibble_balance| >> +---------------+ >> | 99.955183 | >> +---------------+ >> SQLRowCount returns 1 >> 1 rows fetched >> SQL> >> >> and freeswitch core is also using odbc (although a differen dsn) >> >> David >> >> >> On Wed, Feb 13, 2013 at 7:07 PM, Steven Ayre wrote: >>> Do the SQL queries work ok within isql? >>> >>> >>> >>> On 13 February 2013 17:51, David Villasmil wrote: >>>> And i can connect from the cli: >>>> >>>> root at lamp ~# isql viking >>>> +---------------------------------------+ >>>> | Connected! | >>>> | | >>>> | sql-statement | >>>> | help [tablename] | >>>> | quit | >>>> | | >>>> +---------------------------------------+ >>>> SQL> >>>> >>>> strange... >>>> >>>> >>>> On Wed, Feb 13, 2013 at 6:46 PM, Steven Ayre wrote: >>>>> http://jira.freeswitch.org/browse/FS-5100 >>>>> >>>>> >>>>> On 13 February 2013 17:40, Steven Ayre wrote: >>>>>> Specifically it's introduced by commit 5c176e8 which was resolving FS-4810, which is indeed adding support for the the generalised DSN interface. >>>>>> >>>>>> http://fisheye.freeswitch.org/changelog/freeswitch.git?showid=5c176e889d8cc2da7efb18a89782407ee5524a74 >>>>>> http://jira.freeswitch.org/browse/FS-4810 >>>>>> >>>>>> -Steve >>>>>> >>>>>> >>>>>> >>>>>> On 13 February 2013 17:37, Steven Ayre wrote: >>>>>>> Looks like this difference is brought in by the modifications to support the generalised DSN interface. odbc-dsn would be consistent with all other modules, so I guess that's why it's happened. >>>>>>> >>>>>>> But... >>>>>>> The HEAD of v1.2.stable understands db_dsn but not odbc-dsn. >>>>>>> The HEAD of master understands odbc-dsn but not db_dsn. >>>>>>> >>>>>>> This is a configuration breaking behaviour, which doesn't seem to have been documented! >>>>>>> >>>>>>> At the very least it'd make sense for the master branch to understand both odbc-dsn and db_dsn for backwards compatibility. I'll file a Jira with a patch (it's a trivial fix) in a moment. >>>>>>> >>>>>>> -Steve >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> On 13 February 2013 17:33, Steven Ayre wrote: >>>>>>>> How are you setting the DSN? >>>>>>>> >>>>>>>> The param name mentioned earlier is wrong. Reading the source the correct parameter is "odbc-dsn". Looks like the wiki documentation is incorrect. >>>>>>>> >>>>>>>> -Steve >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> On 13 February 2013 15:33, David Villasmil wrote: >>>>>>>>> hello Avi, >>>>>>>>> >>>>>>>>> that's right, but nevermind that, that's the output when I was trying something else: >>>>>>>>> >>>>>>>>> 2013-02-12 19:32:07.784156 [INFO] mod_nibblebill.c:498 Beginning new billing on 58e6762b-c049-470e-84f7-519a815ab6f0 >>>>>>>>> 2013-02-12 19:32:07.784156 [CRIT] mod_nibblebill.c:535 Failed to log to database! >>>>>>>>> 2013-02-12 19:32:07.784156 [ERR] mod_nibblebill.c:380 Error running this query: [SELECT balance AS nibble_balance FROM sip_users WHERE id='1'] >>>>>>>>> >>>>>>>>> >>>>>>>>> On Wed, Feb 13, 2013 at 12:58 PM, Avi Marcus wrote: >>>>>>>>>> Your logs showed the queries run were: >>>>>>>>>> >>>>>>>>>> UPDATE sip_users SET balance=balance-0.018983 WHERE ='1' >>>>>>>>>> SELECT balance AS nibble_balance FROM sip_users WHERE ='1' >>>>>>>>>> >>>>>>>>>> You just tried now.. "WHERE id='1' " -- they are not the same query... >>>>>>>>>> >>>>>>>>>> -Avi >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> On Wed, Feb 13, 2013 at 12:48 PM, David Villasmil wrote: >>>>>>>>>>> Hello Rupa, >>>>>>>>>>> >>>>>>>>>>> the query if ok, the user is there: >>>>>>>>>>> >>>>>>>>>>> mysql> select * from sip_users; >>>>>>>>>>> +----+----------+----------+----------------+------------+--------------+--------------+ >>>>>>>>>>> | id | username | password | rateplan | balance | domain | max_channels | >>>>>>>>>>> +----+----------+----------+----------------+------------+--------------+--------------+ >>>>>>>>>>> | 1 | 1002 | 1234 | retail_default | 100.000000 | 192.168.1.44 | 1 | >>>>>>>>>>> +----+----------+----------+----------------+------------+--------------+--------------+ >>>>>>>>>>> 1 row in set (0.00 sec) >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> and the actual sql executed by nibblebill: >>>>>>>>>>> >>>>>>>>>>> mysql> SELECT balance AS nibble_balance FROM sip_users WHERE id='1'; >>>>>>>>>>> +----------------+ >>>>>>>>>>> | nibble_balance | >>>>>>>>>>> +----------------+ >>>>>>>>>>> | 100.000000 | >>>>>>>>>>> +----------------+ >>>>>>>>>>> 1 row in set (0.00 sec) >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> so that's ok.. >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> any othe ideas? >>>>>>>>>>> >>>>>>>>>>> Thanks a lot for your help! >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> David >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> On Wed, Feb 13, 2013 at 3:35 AM, Rupa Schomaker wrote: >>>>>>>>>>>> >>>>>>>>>>>> On Tue, Feb 12, 2013 at 5:26 PM, David Villasmil wrote: >>>>>>>>>>>>> call goes out fine... no billing though: >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> 2013-02-12 19:21:49.844277 [DEBUG] mod_nibblebill.c:504 22 seconds passed since last bill time of 2013-02-12 19:21:27 >>>>>>>>>>>>> 2013-02-12 19:21:49.844277 [DEBUG] mod_nibblebill.c:521 Billing $0.018983 to 1 (Call: d442dc6b-a3a7-423a-8450-598d17ca43b3 / 0.000000 so far) >>>>>>>>>>>>> 2013-02-12 19:21:49.844277 [DEBUG] mod_nibblebill.c:351 Doing update query >>>>>>>>>>>>> [UPDATE sip_users SET balance=balance-0.018983 WHERE ='1'] >>>>>>>>>>>>> 2013-02-12 19:21:49.844277 [CRIT] mod_nibblebill.c:535 Failed to log to database! >>>>>>>>>>>>> 2013-02-12 19:21:49.844277 [DEBUG] mod_nibblebill.c:378 Doing lookup query >>>>>>>>>>>>> [SELECT balance AS nibble_balance FROM sip_users WHERE ='1'] >>>>>>>>>>>>> 2013-02-12 19:21:49.844277 [ERR] mod_nibblebill.c:380 Error running this query: [SELECT balance AS nibble_balance FROM sip_users WHERE ='1'] >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> the select go fine if executed "by hand" >>>>>>>>>>>> >>>>>>>>>>>> None of those SQL statements are valid. (hint: WHERE = '1' is no good). Been a while since I hacked/worked with nibblebill, but I'm pretty sure the 'id' field isn't not being filled in the sql for some reason. >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> -- >>>>>>>>>>>> -Rupa >>>>>>>>>>>> >>>>>>>>>>>> _________________________________________________________________________ >>>>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>>>> consulting at freeswitch.org >>>>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>> http://wiki.freeswitch.org >>>>>>>>>>>> http://www.cluecon.com >>>>>>>>>>>> >>>>>>>>>>>> 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 >>>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> _________________________________________________________________________ >>>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>>> consulting at freeswitch.org >>>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>> http://wiki.freeswitch.org >>>>>>>>>>> http://www.cluecon.com >>>>>>>>>>> >>>>>>>>>>> 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 >>>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> _________________________________________________________________________ >>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>> consulting at freeswitch.org >>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>> http://www.freeswitch.org >>>>>>>>>> http://wiki.freeswitch.org >>>>>>>>>> http://www.cluecon.com >>>>>>>>>> >>>>>>>>>> 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 >>>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> _________________________________________________________________________ >>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>> consulting at freeswitch.org >>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> Official FreeSWITCH Sites >>>>>>>>> http://www.freeswitch.org >>>>>>>>> http://wiki.freeswitch.org >>>>>>>>> http://www.cluecon.com >>>>>>>>> >>>>>>>>> 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 >>>>>>>>> >>>>>>>> >>>>>>> >>>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> 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 >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130213/598cc097/attachment-0001.html From david.villasmil.work at gmail.com Wed Feb 13 22:57:46 2013 From: david.villasmil.work at gmail.com (David Villasmil) Date: Wed, 13 Feb 2013 20:57:46 +0100 Subject: [Freeswitch-users] Nibblebill database question In-Reply-To: <6F00484B-1F8C-44F4-A9D4-9B8D5097E78A@gmail.com> References: <6F00484B-1F8C-44F4-A9D4-9B8D5097E78A@gmail.com> Message-ID: It worked! finally! That happend when one doesn't read the release notes :( it updated the balance, BUT I still got and error, balance was 100: 2013-02-13 19:55:09.903784 [DEBUG] mod_nibblebill.c:351 Doing update query [UPDATE sip_users SET balance=balance-0.022267 WHERE id='1'] 2013-02-13 19:55:09.903784 [CRIT] mod_nibblebill.c:535 Failed to log to database! <----- error? 2013-02-13 19:55:09.903784 [DEBUG] mod_nibblebill.c:378 Doing lookup query [SELECT balance AS nibble_balance FROM sip_users WHERE id='1'] 2013-02-13 19:55:09.923320 [DEBUG] mod_nibblebill.c:386 Retrieved current balance for account 1 (balance = 99.977733) So it did the update fine, but stil got errors somehow... On Wed, Feb 13, 2013 at 8:49 PM, David Villasmil < david.villasmil.work at gmail.com> wrote: > Aha! > > That must it, i'll test and let you know... > > David > > --- > > David Villasmil > > > On Feb 13, 2013, at 20:35, Steven Ayre wrote: > > Is "viking" the actual DSN you're using? If so that format won't work any > longer since the generalized interface was implemented. name:user:pass > still works but plain name no longer works*.* > > See http://wiki.freeswitch.org/wiki/Release_Notes#odbc-dsn > > You're best off updating it to odbc://viking > > -Steve > > > > > On 13 February 2013 19:07, David Villasmil > wrote: > >> Yeah, everything works within isql: >> >> root at lamp ~# isql viking >> +---------------------------------------+ >> | Connected! | >> | | >> | sql-statement | >> | help [tablename] | >> | quit | >> | | >> +---------------------------------------+ >> SQL> SELECT balance AS nibble_balance FROM sip_users WHERE id='1' >> +---------------+ >> | nibble_balance| >> +---------------+ >> | 100.000000 | >> +---------------+ >> SQLRowCount returns 1 >> 1 rows fetched >> SQL> UPDATE sip_users SET balance=balance-0.044817 WHERE id='1' >> SQLRowCount returns 1 >> SQL> SELECT balance AS nibble_balance FROM sip_users WHERE id='1' >> +---------------+ >> | nibble_balance| >> +---------------+ >> | 99.955183 | >> +---------------+ >> SQLRowCount returns 1 >> 1 rows fetched >> SQL> >> >> and freeswitch core is also using odbc (although a differen dsn) >> >> David >> >> >> On Wed, Feb 13, 2013 at 7:07 PM, Steven Ayre wrote: >> >>> Do the SQL queries work ok within isql? >>> >>> >>> >>> On 13 February 2013 17:51, David Villasmil < >>> david.villasmil.work at gmail.com> wrote: >>> >>>> And i can connect from the cli: >>>> >>>> root at lamp ~# isql viking >>>> +---------------------------------------+ >>>> | Connected! | >>>> | | >>>> | sql-statement | >>>> | help [tablename] | >>>> | quit | >>>> | | >>>> +---------------------------------------+ >>>> SQL> >>>> >>>> strange... >>>> >>>> >>>> On Wed, Feb 13, 2013 at 6:46 PM, Steven Ayre wrote: >>>> >>>>> http://jira.freeswitch.org/browse/FS-5100 >>>>> >>>>> >>>>> On 13 February 2013 17:40, Steven Ayre wrote: >>>>> >>>>>> Specifically it's introduced by commit 5c176e8 which was resolving >>>>>> FS-4810, which is indeed adding support for the the generalised DSN >>>>>> interface. >>>>>> >>>>>> >>>>>> http://fisheye.freeswitch.org/changelog/freeswitch.git?showid=5c176e889d8cc2da7efb18a89782407ee5524a74 >>>>>> http://jira.freeswitch.org/browse/FS-4810 >>>>>> >>>>>> -Steve >>>>>> >>>>>> >>>>>> >>>>>> On 13 February 2013 17:37, Steven Ayre wrote: >>>>>> >>>>>>> Looks like this difference is brought in by the modifications to >>>>>>> support the generalised DSN interface. odbc-dsn would be consistent with >>>>>>> all other modules, so I guess that's why it's happened. >>>>>>> >>>>>>> But... >>>>>>> The HEAD of v1.2.stable understands db_dsn but not odbc-dsn. >>>>>>> The HEAD of master understands odbc-dsn but not db_dsn. >>>>>>> >>>>>>> This is a configuration breaking behaviour, which doesn't seem to >>>>>>> have been documented! >>>>>>> >>>>>>> At the very least it'd make sense for the master branch to >>>>>>> understand both odbc-dsn and db_dsn for backwards compatibility. I'll file >>>>>>> a Jira with a patch (it's a trivial fix) in a moment. >>>>>>> >>>>>>> -Steve >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> On 13 February 2013 17:33, Steven Ayre wrote: >>>>>>> >>>>>>>> How are you setting the DSN? >>>>>>>> >>>>>>>> The param name mentioned earlier is wrong. Reading the source the >>>>>>>> correct parameter is "odbc-dsn". Looks like the wiki documentation is >>>>>>>> incorrect. >>>>>>>> >>>>>>>> -Steve >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> On 13 February 2013 15:33, David Villasmil < >>>>>>>> david.villasmil.work at gmail.com> wrote: >>>>>>>> >>>>>>>>> hello Avi, >>>>>>>>> >>>>>>>>> that's right, but nevermind that, that's the output when I was >>>>>>>>> trying something else: >>>>>>>>> >>>>>>>>> 2013-02-12 19:32:07.784156 [INFO] mod_nibblebill.c:498 Beginning >>>>>>>>> new billing on 58e6762b-c049-470e-84f7-519a815ab6f0 >>>>>>>>> 2013-02-12 19:32:07.784156 [CRIT] mod_nibblebill.c:535 Failed to >>>>>>>>> log to database! >>>>>>>>> 2013-02-12 19:32:07.784156 [ERR] mod_nibblebill.c:380 Error >>>>>>>>> running this query: [SELECT balance AS nibble_balance FROM sip_users WHERE >>>>>>>>> id='1'] >>>>>>>>> >>>>>>>>> >>>>>>>>> On Wed, Feb 13, 2013 at 12:58 PM, Avi Marcus wrote: >>>>>>>>> >>>>>>>>>> Your logs showed the queries run were: >>>>>>>>>> >>>>>>>>>> UPDATE sip_users SET balance=balance-0.018983 WHERE ='1' >>>>>>>>>> SELECT balance AS nibble_balance FROM sip_users WHERE ='1' >>>>>>>>>> >>>>>>>>>> You just tried now.. "WHERE *id*='1' " -- they are not the same >>>>>>>>>> query... >>>>>>>>>> >>>>>>>>>> -Avi >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> On Wed, Feb 13, 2013 at 12:48 PM, David Villasmil < >>>>>>>>>> david.villasmil.work at gmail.com> wrote: >>>>>>>>>> >>>>>>>>>>> Hello Rupa, >>>>>>>>>>> >>>>>>>>>>> the query if ok, the user is there: >>>>>>>>>>> >>>>>>>>>>> mysql> select * from sip_users; >>>>>>>>>>> >>>>>>>>>>> +----+----------+----------+----------------+------------+--------------+--------------+ >>>>>>>>>>> | id | username | password | rateplan | balance | >>>>>>>>>>> domain | max_channels | >>>>>>>>>>> >>>>>>>>>>> +----+----------+----------+----------------+------------+--------------+--------------+ >>>>>>>>>>> | 1 | 1002 | 1234 | retail_default | 100.000000 | >>>>>>>>>>> 192.168.1.44 | 1 | >>>>>>>>>>> >>>>>>>>>>> +----+----------+----------+----------------+------------+--------------+--------------+ >>>>>>>>>>> 1 row in set (0.00 sec) >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> and the actual sql executed by nibblebill: >>>>>>>>>>> >>>>>>>>>>> mysql> SELECT balance AS nibble_balance FROM sip_users WHERE >>>>>>>>>>> id='1'; >>>>>>>>>>> +----------------+ >>>>>>>>>>> | nibble_balance | >>>>>>>>>>> +----------------+ >>>>>>>>>>> | 100.000000 | >>>>>>>>>>> +----------------+ >>>>>>>>>>> 1 row in set (0.00 sec) >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> so that's ok.. >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> any othe ideas? >>>>>>>>>>> >>>>>>>>>>> Thanks a lot for your help! >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> David >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> On Wed, Feb 13, 2013 at 3:35 AM, Rupa Schomaker wrote: >>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> On Tue, Feb 12, 2013 at 5:26 PM, David Villasmil < >>>>>>>>>>>> david.villasmil.work at gmail.com> wrote: >>>>>>>>>>>> >>>>>>>>>>>>> call goes out fine... no billing though: >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> 2013-02-12 19:21:49.844277 [DEBUG] mod_nibblebill.c:504 22 >>>>>>>>>>>>> seconds passed since last bill time of 2013-02-12 19:21:27 >>>>>>>>>>>>> 2013-02-12 19:21:49.844277 [DEBUG] mod_nibblebill.c:521 >>>>>>>>>>>>> Billing $0.018983 to 1 (Call: d442dc6b-a3a7-423a-8450-598d17ca43b3 / >>>>>>>>>>>>> 0.000000 so far) >>>>>>>>>>>>> 2013-02-12 19:21:49.844277 [DEBUG] mod_nibblebill.c:351 Doing >>>>>>>>>>>>> update query >>>>>>>>>>>>> [UPDATE sip_users SET balance=balance-0.018983 WHERE ='1'] >>>>>>>>>>>>> 2013-02-12 19:21:49.844277 [CRIT] mod_nibblebill.c:535 Failed >>>>>>>>>>>>> to log to database! >>>>>>>>>>>>> 2013-02-12 19:21:49.844277 [DEBUG] mod_nibblebill.c:378 Doing >>>>>>>>>>>>> lookup query >>>>>>>>>>>>> [SELECT balance AS nibble_balance FROM sip_users WHERE ='1'] >>>>>>>>>>>>> 2013-02-12 19:21:49.844277 [ERR] mod_nibblebill.c:380 Error >>>>>>>>>>>>> running this query: [SELECT balance AS nibble_balance FROM sip_users WHERE >>>>>>>>>>>>> ='1'] >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> the select go fine if executed "by hand" >>>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> None of those SQL statements are valid. (hint: WHERE = '1' is >>>>>>>>>>>> no good). Been a while since I hacked/worked with nibblebill, but I'm >>>>>>>>>>>> pretty sure the 'id' field isn't not being filled in the sql for some >>>>>>>>>>>> reason. >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> -- >>>>>>>>>>>> -Rupa >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> _________________________________________________________________________ >>>>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>>>> consulting at freeswitch.org >>>>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>> http://wiki.freeswitch.org >>>>>>>>>>>> http://www.cluecon.com >>>>>>>>>>>> >>>>>>>>>>>> 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 >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> _________________________________________________________________________ >>>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>>> consulting at freeswitch.org >>>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>> http://wiki.freeswitch.org >>>>>>>>>>> http://www.cluecon.com >>>>>>>>>>> >>>>>>>>>>> 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 >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> _________________________________________________________________________ >>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>> consulting at freeswitch.org >>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>> http://www.freeswitch.org >>>>>>>>>> http://wiki.freeswitch.org >>>>>>>>>> http://www.cluecon.com >>>>>>>>>> >>>>>>>>>> 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 >>>>>>>>>> >>>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> _________________________________________________________________________ >>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>> consulting at freeswitch.org >>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> Official FreeSWITCH Sites >>>>>>>>> http://www.freeswitch.org >>>>>>>>> http://wiki.freeswitch.org >>>>>>>>> http://www.cluecon.com >>>>>>>>> >>>>>>>>> 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 >>>>>>>>> >>>>>>>>> >>>>>>>> >>>>>>> >>>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> 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 >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130213/ee3b324f/attachment-0001.html From david.villasmil.work at gmail.com Wed Feb 13 22:59:54 2013 From: david.villasmil.work at gmail.com (David Villasmil) Date: Wed, 13 Feb 2013 20:59:54 +0100 Subject: [Freeswitch-users] Nibblebill database update In-Reply-To: <8724844.crljyV7uo8@sos> References: <5091F32B.2010302@xofap.com> <509374A4.1090203@bestnet.kharkov.ua> <50937971.4090809@xofap.com> <8724844.crljyV7uo8@sos> Message-ID: Hello, I was having the same problem, the solution is to use: where "dsn_name" is obviously the dsn inside the /etc/odbc.ini file previously tested with "isql" good luck! David On Fri, Nov 2, 2012 at 11:25 AM, Sergey Okhapkin wrote: > What is the structure of "accounts" table? > > On Friday 02 November 2012 14:42:41 William Alianto wrote: > > Thanks for pointing that out. I think I missed that part when I edited > > the config. > > > > On 11/02/2012 02:22 PM, Evgeniy Movlyan wrote: > > > Seems you need to uncomment custom SQL-queries and rewrite it to > > > according to your needs. > > > > > > 02.11.2012 05:54, William Alianto ?????: > > >> Hi, > > >> > > >> This is my nibblebill.conf.xml > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> On 11/01/2012 02:56 PM, Evgeniy Movlyan wrote: > > >>> Hello. > > >>> Show please your nibblebill.conf.xml. > > >>> > > >>> 01.11.2012 05:57, William Alianto ?????: > > >>>> Hi, > > >>>> > > >>>> I was trying to use nibblebill as billing control of my FS. I have > > >>>> configured the odbc and the nibblebill configuration. When I tried > to > > >>>> call, the query looks ok. But when I hanged up the call, it seems > like > > >>>> there is something wrong with the database connection, since it's > not > > >>>> updating the database at all. I got error message on CLI > > >>>> > > >>>> 2012-11-01 10:51:00.559472 [ERR] switch_odbc.c:494 ERR: [UPDATE > > >>>> accounts > > >>>> SET usage=age-0.476000 WHERE user='1001'] > > >>>> [STATE: 42000 CODE 1064 ERROR: [unixODBC][MySQL][ODBC 5.1 > > >>>> Driver][mysqld-5.5.24-0ubuntu0.12.04.1]You have an error in your SQL > > >>>> syntax; check the manual that corresponds to your MySQL server > version > > >>>> for the right syntax to use near 'usage=age-0.476000 WHERE > > >>>> user=001'' at line 1 > > >>>> ] > > >>>> 2012-11-01 10:51:00.559472 [ERR] mod_nibblebill.c:343 ERR: [UPDATE > > >>>> accounts SET usage=age-0.476000 WHERE user='1001'] > > >>>> [] > > >>>> 2012-11-01 10:51:00.559472 [CRIT] mod_nibblebill.c:542 Failed to > > >>>> log to > > >>>> database! > > >>>> 2012-11-01 10:51:00.559472 [DEBUG] mod_nibblebill.c:383 Doing lookup > > >>>> query > > >>>> [SELECT usage AS nibble_balance FROM accounts WHERE user=001'] > > >>>> 2012-11-01 10:51:00.559472 [ERR] mod_nibblebill.c:385 ERR: [SELECT > > >>>> usage > > >>>> AS nibble_balance FROM accounts WHERE user=001'] > > >>>> [STATE: 42000 CODE 1064 ERROR: [unixODBC][MySQL][ODBC 5.1 > > >>>> Driver][mysqld-5.5.24-0ubuntu0.12.04.1]You have an error in your SQL > > >>>> syntax; check the manual that corresponds to your MySQL server > version > > >>>> for the right syntax to use near 'usage AS nibble_balance FROM > > >>>> accounts > > >>>> WHERE user=001'' at line 1 > > >>>> ] > > >>>> 2012-11-01 10:51:00.559472 [ERR] mod_nibblebill.c:386 Error running > > >>>> this > > >>>> query: [SELECT usage AS nibble_balance FROM accounts WHERE > user=001'] > > >>>> > > >>>> Could anybody help me solve this issue? > > >>>> > > >>>> Regards > > >>>> > > >>>> > _______________________________________________________________________ > > >>>> __ > > >>>> > > >>>> > > >>>> Professional FreeSWITCH Consulting Services: > > >>>> consulting at freeswitch.org > > >>>> http://www.freeswitchsolutions.com > > >>>> > > >>>> > > >>>> > > >>>> > > >>>> Official FreeSWITCH Sites > > >>>> http://www.freeswitch.org > > >>>> http://wiki.freeswitch.org > > >>>> http://www.cluecon.com > > >>>> > > >>>> 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-user > > >>>> s > > >>>> > > >>>> http://www.freeswitch.org > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130213/e527e386/attachment.html From steveayre at gmail.com Wed Feb 13 23:06:04 2013 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 13 Feb 2013 20:06:04 +0000 Subject: [Freeswitch-users] ooH323 vs mod_h323 In-Reply-To: References: <2BD0DC60-D428-4323-A020-D9AD20E76553@freeswitch.org> <4C2EA249.4070704@coppice.org> <267C8398-6CA3-442E-B973-BE096D728B28@freeswitch.org> Message-ID: Indeed comparing the COPYING files, the Asterisk one lacks the FLOSS exception. So that version isn't usable, and neither is copying their patches into the 0.9.4 version since they'd be GPL-only. So while 0.9.4 would be usable, it'd lack some useful functionality (Q.931 causes are just the one I spotted from a 5 minute glance through the API). -Steve On 13 February 2013 12:27, Steven Ayre wrote: > I was just looking at this library to play with the idea of writing a > mod_ooh323c. It looks like the library may have forked away from the > version used in Asterisk though? > > The latest version on the website ( > http://www.obj-sys.com/telephony-objective.shtml) is 0.9.4, but cannot > hangup with a Q.931 hangup cause. The CVS repository ( > sourceforge.net/projects/ooh323c) doesn't seem up-to-date since that's > only holding 0.9.0. > > The version in asterisk has been patched to add a q931cause parameter to > ooHangCall so it can - but it seems to be based off 'v0.8.3m' and I don't > know if that's before the FLOSS exception was added. > > So does anyone more familiar with the library know whether the repository > had been moved elsewhere, or whether there's a more recent version > somewhere else, and whether the one bundled in Asterisk would be usable? > > -Steve > > > > > On 5 July 2010 15:28, Brian West wrote: > >> It has a FLOSS exception in the GPL license. >> >> /b >> >> On Jul 2, 2010, at 9:36 PM, Steve Underwood wrote: >> >> > On 07/03/2010 04:28 AM, Brian West wrote: >> >> DOH... btw ooh323 is now license compatible with FreeSWITCH if anyone >> wants to write one. >> >> >> >> /b >> >> >> > Where did you see that? The current ooh323c tarball still says GPL 2. >> > >> > Steve >> >> >> _______________________________________________ >> 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 >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130213/90658187/attachment-0001.html From steveayre at gmail.com Wed Feb 13 23:13:19 2013 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 13 Feb 2013 20:13:19 +0000 Subject: [Freeswitch-users] Nibblebill database question In-Reply-To: References: <6F00484B-1F8C-44F4-A9D4-9B8D5097E78A@gmail.com> Message-ID: It's probably not a very well known page. I created it when the odbc-dsn change broke my own system on an upgrade (I too store user/password in odbc.ini not the DSN). It lacks any earlier breaking changes, but seemed a good idea to add now we have a stable branch.e It's linked to on the Wiki front page and the download & install guide, but if people are used to upgrading from the CLI they probably never even read the Wiki each time they update. :o) There's also http://wiki.freeswitch.org/wiki/Changelog_Summary, but the Release Notes page is a nice place to highlight changes that might actually break stuff. The odsn-dsn change is rather innocuously mentioned in the changelog as merely 'core: Add Postgres core db support (r:0c1180d5)'. -Steve On 13 February 2013 19:57, David Villasmil wrote: > It worked! finally! That happend when one doesn't read the release notes :( > > it updated the balance, BUT I still got and error, balance was 100: > > 2013-02-13 19:55:09.903784 [DEBUG] mod_nibblebill.c:351 Doing update query > [UPDATE sip_users SET balance=balance-0.022267 WHERE id='1'] > 2013-02-13 19:55:09.903784 [CRIT] mod_nibblebill.c:535 Failed to log to > database! <----- error? > 2013-02-13 19:55:09.903784 [DEBUG] mod_nibblebill.c:378 Doing lookup query > [SELECT balance AS nibble_balance FROM sip_users WHERE id='1'] > 2013-02-13 19:55:09.923320 [DEBUG] mod_nibblebill.c:386 Retrieved current > balance for account 1 (balance = 99.977733) > > So it did the update fine, but stil got errors somehow... > > > On Wed, Feb 13, 2013 at 8:49 PM, David Villasmil < > david.villasmil.work at gmail.com> wrote: > >> Aha! >> >> That must it, i'll test and let you know... >> >> David >> >> --- >> >> David Villasmil >> >> >> On Feb 13, 2013, at 20:35, Steven Ayre wrote: >> >> Is "viking" the actual DSN you're using? If so that format won't work any >> longer since the generalized interface was implemented. name:user:pass >> still works but plain name no longer works*.* >> >> See http://wiki.freeswitch.org/wiki/Release_Notes#odbc-dsn >> >> You're best off updating it to odbc://viking >> >> -Steve >> >> >> >> >> On 13 February 2013 19:07, David Villasmil < >> david.villasmil.work at gmail.com> wrote: >> >>> Yeah, everything works within isql: >>> >>> root at lamp ~# isql viking >>> +---------------------------------------+ >>> | Connected! | >>> | | >>> | sql-statement | >>> | help [tablename] | >>> | quit | >>> | | >>> +---------------------------------------+ >>> SQL> SELECT balance AS nibble_balance FROM sip_users WHERE id='1' >>> +---------------+ >>> | nibble_balance| >>> +---------------+ >>> | 100.000000 | >>> +---------------+ >>> SQLRowCount returns 1 >>> 1 rows fetched >>> SQL> UPDATE sip_users SET balance=balance-0.044817 WHERE id='1' >>> SQLRowCount returns 1 >>> SQL> SELECT balance AS nibble_balance FROM sip_users WHERE id='1' >>> +---------------+ >>> | nibble_balance| >>> +---------------+ >>> | 99.955183 | >>> +---------------+ >>> SQLRowCount returns 1 >>> 1 rows fetched >>> SQL> >>> >>> and freeswitch core is also using odbc (although a differen dsn) >>> >>> David >>> >>> >>> On Wed, Feb 13, 2013 at 7:07 PM, Steven Ayre wrote: >>> >>>> Do the SQL queries work ok within isql? >>>> >>>> >>>> >>>> On 13 February 2013 17:51, David Villasmil < >>>> david.villasmil.work at gmail.com> wrote: >>>> >>>>> And i can connect from the cli: >>>>> >>>>> root at lamp ~# isql viking >>>>> +---------------------------------------+ >>>>> | Connected! | >>>>> | | >>>>> | sql-statement | >>>>> | help [tablename] | >>>>> | quit | >>>>> | | >>>>> +---------------------------------------+ >>>>> SQL> >>>>> >>>>> strange... >>>>> >>>>> >>>>> On Wed, Feb 13, 2013 at 6:46 PM, Steven Ayre wrote: >>>>> >>>>>> http://jira.freeswitch.org/browse/FS-5100 >>>>>> >>>>>> >>>>>> On 13 February 2013 17:40, Steven Ayre wrote: >>>>>> >>>>>>> Specifically it's introduced by commit 5c176e8 which was resolving >>>>>>> FS-4810, which is indeed adding support for the the generalised DSN >>>>>>> interface. >>>>>>> >>>>>>> >>>>>>> http://fisheye.freeswitch.org/changelog/freeswitch.git?showid=5c176e889d8cc2da7efb18a89782407ee5524a74 >>>>>>> http://jira.freeswitch.org/browse/FS-4810 >>>>>>> >>>>>>> -Steve >>>>>>> >>>>>>> >>>>>>> >>>>>>> On 13 February 2013 17:37, Steven Ayre wrote: >>>>>>> >>>>>>>> Looks like this difference is brought in by the modifications to >>>>>>>> support the generalised DSN interface. odbc-dsn would be consistent with >>>>>>>> all other modules, so I guess that's why it's happened. >>>>>>>> >>>>>>>> But... >>>>>>>> The HEAD of v1.2.stable understands db_dsn but not odbc-dsn. >>>>>>>> The HEAD of master understands odbc-dsn but not db_dsn. >>>>>>>> >>>>>>>> This is a configuration breaking behaviour, which doesn't seem to >>>>>>>> have been documented! >>>>>>>> >>>>>>>> At the very least it'd make sense for the master branch to >>>>>>>> understand both odbc-dsn and db_dsn for backwards compatibility. I'll file >>>>>>>> a Jira with a patch (it's a trivial fix) in a moment. >>>>>>>> >>>>>>>> -Steve >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> On 13 February 2013 17:33, Steven Ayre wrote: >>>>>>>> >>>>>>>>> How are you setting the DSN? >>>>>>>>> >>>>>>>>> The param name mentioned earlier is wrong. Reading the source the >>>>>>>>> correct parameter is "odbc-dsn". Looks like the wiki documentation is >>>>>>>>> incorrect. >>>>>>>>> >>>>>>>>> -Steve >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> On 13 February 2013 15:33, David Villasmil < >>>>>>>>> david.villasmil.work at gmail.com> wrote: >>>>>>>>> >>>>>>>>>> hello Avi, >>>>>>>>>> >>>>>>>>>> that's right, but nevermind that, that's the output when I was >>>>>>>>>> trying something else: >>>>>>>>>> >>>>>>>>>> 2013-02-12 19:32:07.784156 [INFO] mod_nibblebill.c:498 Beginning >>>>>>>>>> new billing on 58e6762b-c049-470e-84f7-519a815ab6f0 >>>>>>>>>> 2013-02-12 19:32:07.784156 [CRIT] mod_nibblebill.c:535 Failed to >>>>>>>>>> log to database! >>>>>>>>>> 2013-02-12 19:32:07.784156 [ERR] mod_nibblebill.c:380 Error >>>>>>>>>> running this query: [SELECT balance AS nibble_balance FROM sip_users WHERE >>>>>>>>>> id='1'] >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> On Wed, Feb 13, 2013 at 12:58 PM, Avi Marcus wrote: >>>>>>>>>> >>>>>>>>>>> Your logs showed the queries run were: >>>>>>>>>>> >>>>>>>>>>> UPDATE sip_users SET balance=balance-0.018983 WHERE ='1' >>>>>>>>>>> SELECT balance AS nibble_balance FROM sip_users WHERE ='1' >>>>>>>>>>> >>>>>>>>>>> You just tried now.. "WHERE *id*='1' " -- they are not the >>>>>>>>>>> same query... >>>>>>>>>>> >>>>>>>>>>> -Avi >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> On Wed, Feb 13, 2013 at 12:48 PM, David Villasmil < >>>>>>>>>>> david.villasmil.work at gmail.com> wrote: >>>>>>>>>>> >>>>>>>>>>>> Hello Rupa, >>>>>>>>>>>> >>>>>>>>>>>> the query if ok, the user is there: >>>>>>>>>>>> >>>>>>>>>>>> mysql> select * from sip_users; >>>>>>>>>>>> >>>>>>>>>>>> +----+----------+----------+----------------+------------+--------------+--------------+ >>>>>>>>>>>> | id | username | password | rateplan | balance | >>>>>>>>>>>> domain | max_channels | >>>>>>>>>>>> >>>>>>>>>>>> +----+----------+----------+----------------+------------+--------------+--------------+ >>>>>>>>>>>> | 1 | 1002 | 1234 | retail_default | 100.000000 | >>>>>>>>>>>> 192.168.1.44 | 1 | >>>>>>>>>>>> >>>>>>>>>>>> +----+----------+----------+----------------+------------+--------------+--------------+ >>>>>>>>>>>> 1 row in set (0.00 sec) >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> and the actual sql executed by nibblebill: >>>>>>>>>>>> >>>>>>>>>>>> mysql> SELECT balance AS nibble_balance FROM sip_users WHERE >>>>>>>>>>>> id='1'; >>>>>>>>>>>> +----------------+ >>>>>>>>>>>> | nibble_balance | >>>>>>>>>>>> +----------------+ >>>>>>>>>>>> | 100.000000 | >>>>>>>>>>>> +----------------+ >>>>>>>>>>>> 1 row in set (0.00 sec) >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> so that's ok.. >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> any othe ideas? >>>>>>>>>>>> >>>>>>>>>>>> Thanks a lot for your help! >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> David >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> On Wed, Feb 13, 2013 at 3:35 AM, Rupa Schomaker wrote: >>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> On Tue, Feb 12, 2013 at 5:26 PM, David Villasmil < >>>>>>>>>>>>> david.villasmil.work at gmail.com> wrote: >>>>>>>>>>>>> >>>>>>>>>>>>>> call goes out fine... no billing though: >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> 2013-02-12 19:21:49.844277 [DEBUG] mod_nibblebill.c:504 22 >>>>>>>>>>>>>> seconds passed since last bill time of 2013-02-12 19:21:27 >>>>>>>>>>>>>> 2013-02-12 19:21:49.844277 [DEBUG] mod_nibblebill.c:521 >>>>>>>>>>>>>> Billing $0.018983 to 1 (Call: d442dc6b-a3a7-423a-8450-598d17ca43b3 / >>>>>>>>>>>>>> 0.000000 so far) >>>>>>>>>>>>>> 2013-02-12 19:21:49.844277 [DEBUG] mod_nibblebill.c:351 Doing >>>>>>>>>>>>>> update query >>>>>>>>>>>>>> [UPDATE sip_users SET balance=balance-0.018983 WHERE ='1'] >>>>>>>>>>>>>> 2013-02-12 19:21:49.844277 [CRIT] mod_nibblebill.c:535 Failed >>>>>>>>>>>>>> to log to database! >>>>>>>>>>>>>> 2013-02-12 19:21:49.844277 [DEBUG] mod_nibblebill.c:378 Doing >>>>>>>>>>>>>> lookup query >>>>>>>>>>>>>> [SELECT balance AS nibble_balance FROM sip_users WHERE ='1'] >>>>>>>>>>>>>> 2013-02-12 19:21:49.844277 [ERR] mod_nibblebill.c:380 Error >>>>>>>>>>>>>> running this query: [SELECT balance AS nibble_balance FROM sip_users WHERE >>>>>>>>>>>>>> ='1'] >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> the select go fine if executed "by hand" >>>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> None of those SQL statements are valid. (hint: WHERE = '1' is >>>>>>>>>>>>> no good). Been a while since I hacked/worked with nibblebill, but I'm >>>>>>>>>>>>> pretty sure the 'id' field isn't not being filled in the sql for some >>>>>>>>>>>>> reason. >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> -- >>>>>>>>>>>>> -Rupa >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> _________________________________________________________________________ >>>>>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>>>>> consulting at freeswitch.org >>>>>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>>> http://wiki.freeswitch.org >>>>>>>>>>>>> http://www.cluecon.com >>>>>>>>>>>>> >>>>>>>>>>>>> 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 >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> _________________________________________________________________________ >>>>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>>>> consulting at freeswitch.org >>>>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>> http://wiki.freeswitch.org >>>>>>>>>>>> http://www.cluecon.com >>>>>>>>>>>> >>>>>>>>>>>> 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 >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> _________________________________________________________________________ >>>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>>> consulting at freeswitch.org >>>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>> http://wiki.freeswitch.org >>>>>>>>>>> http://www.cluecon.com >>>>>>>>>>> >>>>>>>>>>> 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 >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> _________________________________________________________________________ >>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>> consulting at freeswitch.org >>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>> http://www.freeswitch.org >>>>>>>>>> http://wiki.freeswitch.org >>>>>>>>>> http://www.cluecon.com >>>>>>>>>> >>>>>>>>>> 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 >>>>>>>>>> >>>>>>>>>> >>>>>>>>> >>>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> 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 >>>>>> >>>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> 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 >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130213/1e8ea610/attachment-0001.html From steveayre at gmail.com Wed Feb 13 23:19:15 2013 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 13 Feb 2013 20:19:15 +0000 Subject: [Freeswitch-users] Nibblebill database update In-Reply-To: References: <5091F32B.2010302@xofap.com> <509374A4.1090203@bestnet.kharkov.ua> <50937971.4090809@xofap.com> <8724844.crljyV7uo8@sos> Message-ID: [STATE: 42000 CODE 1064 ERROR: [unixODBC][MySQL][ODBC 5.1 Driver][mysqld-5.5.24-0ubuntu0.12.04.1]You have an error in your SQL syntax; check the manual that corresponds to your MySQL server version for the right syntax to use near 'usage=usage-0.476000 WHERE user='1001'' at line 1 ] This error is coming from MySQL itself, which indicates it is actually using the ODBC connection. This isn't the same as David's issue. The error message starts at the first character that failed the SQL parsing. That's "usage". That happens to be on the list of MySQL reserved words which will give an error in 5.5 (which wouldn't have happened in 5.1 or earlier). http://dev.mysql.com/doc/refman/5.5/en/reserved-words.html You either need to change your DB structure to avoid using the reserved words (best) or quote the column name in backticks (`). Try setting: -Steve On 13 February 2013 19:59, David Villasmil wrote: > Hello, > > I was having the same problem, the solution is to use: > > > > where "dsn_name" is obviously the dsn inside the /etc/odbc.ini file > previously tested with "isql" > > > good luck! > > David > > > On Fri, Nov 2, 2012 at 11:25 AM, Sergey Okhapkin > wrote: > >> What is the structure of "accounts" table? >> >> On Friday 02 November 2012 14:42:41 William Alianto wrote: >> > Thanks for pointing that out. I think I missed that part when I edited >> > the config. >> > >> > On 11/02/2012 02:22 PM, Evgeniy Movlyan wrote: >> > > Seems you need to uncomment custom SQL-queries and rewrite it to >> > > according to your needs. >> > > >> > > 02.11.2012 05:54, William Alianto ?????: >> > >> Hi, >> > >> >> > >> This is my nibblebill.conf.xml >> > >> >> > >> >> > >> >> > >> >> > >> >> > >> >> > >> >> > >> >> > >> >> > >> >> > >> >> > >> >> > >> >> > >> >> > >> >> > >> >> > >> >> > >> >> > >> >> > >> >> > >> >> > >> >> > >> >> > >> >> > >> >> > >> >> > >> >> > >> >> > >> >> > >> >> > >> >> > >> >> > >> >> > >> >> > >> >> > >> >> > >> >> > >> >> > >> >> > >> >> > >> On 11/01/2012 02:56 PM, Evgeniy Movlyan wrote: >> > >>> Hello. >> > >>> Show please your nibblebill.conf.xml. >> > >>> >> > >>> 01.11.2012 05:57, William Alianto ?????: >> > >>>> Hi, >> > >>>> >> > >>>> I was trying to use nibblebill as billing control of my FS. I have >> > >>>> configured the odbc and the nibblebill configuration. When I tried >> to >> > >>>> call, the query looks ok. But when I hanged up the call, it seems >> like >> > >>>> there is something wrong with the database connection, since it's >> not >> > >>>> updating the database at all. I got error message on CLI >> > >>>> >> > >>>> 2012-11-01 10:51:00.559472 [ERR] switch_odbc.c:494 ERR: [UPDATE >> > >>>> accounts >> > >>>> SET usage=age-0.476000 WHERE user='1001'] >> > >>>> [STATE: 42000 CODE 1064 ERROR: [unixODBC][MySQL][ODBC 5.1 >> > >>>> Driver][mysqld-5.5.24-0ubuntu0.12.04.1]You have an error in your >> SQL >> > >>>> syntax; check the manual that corresponds to your MySQL server >> version >> > >>>> for the right syntax to use near 'usage=age-0.476000 WHERE >> > >>>> user=001'' at line 1 >> > >>>> ] >> > >>>> 2012-11-01 10:51:00.559472 [ERR] mod_nibblebill.c:343 ERR: [UPDATE >> > >>>> accounts SET usage=age-0.476000 WHERE user='1001'] >> > >>>> [] >> > >>>> 2012-11-01 10:51:00.559472 [CRIT] mod_nibblebill.c:542 Failed to >> > >>>> log to >> > >>>> database! >> > >>>> 2012-11-01 10:51:00.559472 [DEBUG] mod_nibblebill.c:383 Doing >> lookup >> > >>>> query >> > >>>> [SELECT usage AS nibble_balance FROM accounts WHERE user=001'] >> > >>>> 2012-11-01 10:51:00.559472 [ERR] mod_nibblebill.c:385 ERR: [SELECT >> > >>>> usage >> > >>>> AS nibble_balance FROM accounts WHERE user=001'] >> > >>>> [STATE: 42000 CODE 1064 ERROR: [unixODBC][MySQL][ODBC 5.1 >> > >>>> Driver][mysqld-5.5.24-0ubuntu0.12.04.1]You have an error in your >> SQL >> > >>>> syntax; check the manual that corresponds to your MySQL server >> version >> > >>>> for the right syntax to use near 'usage AS nibble_balance FROM >> > >>>> accounts >> > >>>> WHERE user=001'' at line 1 >> > >>>> ] >> > >>>> 2012-11-01 10:51:00.559472 [ERR] mod_nibblebill.c:386 Error running >> > >>>> this >> > >>>> query: [SELECT usage AS nibble_balance FROM accounts WHERE >> user=001'] >> > >>>> >> > >>>> Could anybody help me solve this issue? >> > >>>> >> > >>>> Regards >> > >>>> >> > >>>> >> _______________________________________________________________________ >> > >>>> __ >> > >>>> >> > >>>> >> > >>>> Professional FreeSWITCH Consulting Services: >> > >>>> consulting at freeswitch.org >> > >>>> http://www.freeswitchsolutions.com >> > >>>> >> > >>>> >> > >>>> >> > >>>> >> > >>>> Official FreeSWITCH Sites >> > >>>> http://www.freeswitch.org >> > >>>> http://wiki.freeswitch.org >> > >>>> http://www.cluecon.com >> > >>>> >> > >>>> 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-user >> > >>>> s >> > >>>> >> > >>>> http://www.freeswitch.org >> > >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > 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 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130213/e9cd35f4/attachment.html From chris at gonumina.com Thu Feb 14 00:23:40 2013 From: chris at gonumina.com (Chris Ferreira) Date: Wed, 13 Feb 2013 16:23:40 -0500 Subject: [Freeswitch-users] Best current ATA and Setup for Faxing In-Reply-To: References: <0B7A549A-767A-4DA6-B732-9999DC6B1B48@gmail.com> Message-ID: Well, this ATA died, and I don't have a backup of it's config when we were using it for Fax testing. We went through a number of tweaks and firmware upgrades. This is months ago. I'm more looking for whats the best starting point for a new ATA. At the time it seemed like it was the SPA112, but that was 6 months ago. Just wondering if this is still a good starting point, or if I should be checking out a new ATA or provider. Thanks, -Chris On Wed, Feb 13, 2013 at 2:47 PM, George Assaad wrote: > Can you please share with us the setting for fax, if you don't mind. > On Feb 13, 2013 11:17 AM, "Nick Vines" wrote: > >> I havent used the cisco atas, but I've had good success with the grand >> stream ht502s and ht702s. My preferred faxing carrier is also flowroute. >> >> Nick >> >> On Feb 13, 2013, at 8:02 AM, Chris Ferreira wrote: >> >> > Hi Folks, >> > >> > >> > We had previously been working with the Cisco SPA112 for T.38 Faxing >> through Freeswitch and out to Flowroute. At the time we didn't get it >> working 100% perfectly. >> > >> > >> > I now have one of these SPA112's that I need to replace because someone >> fried it by plugging in a much larger power supply by accident. >> > >> > >> > Is my scenario above still a good starting point for Faxing or should I >> be looking at another ATA or provider other than Flowroute? >> > >> > >> > >> > >> > >> > Also, one more question, and this may be better suited to the business >> list (let me know if I should move it there), but we are looking for a part >> time as needed remote or local Freeswitch Admin / Engineer / Tech. Our >> previous person is MIA. >> > >> > >> > >> > >> > Thanks Everyone! >> > >> > >> > >> > -Chris >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > 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 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130213/7be6bcf0/attachment-0001.html From jsun at junsun.net Thu Feb 14 00:32:14 2013 From: jsun at junsun.net (Jun Sun) Date: Wed, 13 Feb 2013 13:32:14 -0800 Subject: [Freeswitch-users] how to end conference after initiator hangs up (dialout style) In-Reply-To: References: <5116D31F.8030807@junsun.net> <511B3A74.1010903@junsun.net> Message-ID: <511C065E.5040101@junsun.net> On 2/13/2013 9:36 AM, Michael Collins wrote: > > > On Tue, Feb 12, 2013 at 11:02 PM, Jun Sun > wrote: > > > Thanks, Michael. It works! > > However, an unintended side effect is that "originate" command > returns much earlier than the "conference" command. "Conference" > command waits until the initiator picks up phone before dialing > others (possibly in background with "bgdial"). "originate" command > would return almost immediately and start dialing others without > waiting for the initiator picking up phone. > > Is there a way to force "originate" command to delay the return > until the endpoint picking up the phone? > > originate > {ignore_early_media=true}sofia/internal/$myNumber at x.x.__x.x:5060 > &conference("conference_test++__flags{endconf}") > Fantastic! It works! Cheers. Jun From msc at freeswitch.org Thu Feb 14 00:47:45 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 13 Feb 2013 13:47:45 -0800 Subject: [Freeswitch-users] ESL Mod_Conference Observations In-Reply-To: References: Message-ID: On Wed, Feb 13, 2013 at 3:05 AM, Usama Zaidi wrote: > Hey, > > Looks like I found a solution to this one as well, and turns out ESL is > MUCH more powerful then I thought, I've updated the wiki page for filters > if anyone wants to have a peek > http://wiki.freeswitch.org/wiki/Mod_event_socket#filter > > Short answer, filter Conference-Unique-ID $ConfUUID > > Good catch. "Unique-ID" is the header for a channel's uuid. "Conference-Unique-ID" is the header for an active conference's uuid. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130213/6778f8ad/attachment.html From gassaad at emassembly.com Thu Feb 14 01:02:59 2013 From: gassaad at emassembly.com (George Assaad) Date: Wed, 13 Feb 2013 17:02:59 -0500 Subject: [Freeswitch-users] Best current ATA and Setup for Faxing In-Reply-To: References: <0B7A549A-767A-4DA6-B732-9999DC6B1B48@gmail.com> Message-ID: Sorry I meant to ask for the set up of the ht-502 On Feb 13, 2013 4:27 PM, "Chris Ferreira" wrote: > Well, this ATA died, and I don't have a backup of it's config when we were > using it for Fax testing. We went through a number of tweaks and firmware > upgrades. This is months ago. > > > I'm more looking for whats the best starting point for a new ATA. At the > time it seemed like it was the SPA112, but that was 6 months ago. Just > wondering if this is still a good starting point, or if I should be > checking out a new ATA or provider. > > > Thanks, > > -Chris > > > On Wed, Feb 13, 2013 at 2:47 PM, George Assaad wrote: > >> Can you please share with us the setting for fax, if you don't mind. >> On Feb 13, 2013 11:17 AM, "Nick Vines" wrote: >> >>> I havent used the cisco atas, but I've had good success with the grand >>> stream ht502s and ht702s. My preferred faxing carrier is also flowroute. >>> >>> Nick >>> >>> On Feb 13, 2013, at 8:02 AM, Chris Ferreira wrote: >>> >>> > Hi Folks, >>> > >>> > >>> > We had previously been working with the Cisco SPA112 for T.38 Faxing >>> through Freeswitch and out to Flowroute. At the time we didn't get it >>> working 100% perfectly. >>> > >>> > >>> > I now have one of these SPA112's that I need to replace because >>> someone fried it by plugging in a much larger power supply by accident. >>> > >>> > >>> > Is my scenario above still a good starting point for Faxing or should >>> I be looking at another ATA or provider other than Flowroute? >>> > >>> > >>> > >>> > >>> > >>> > Also, one more question, and this may be better suited to the business >>> list (let me know if I should move it there), but we are looking for a part >>> time as needed remote or local Freeswitch Admin / Engineer / Tech. Our >>> previous person is MIA. >>> > >>> > >>> > >>> > >>> > Thanks Everyone! >>> > >>> > >>> > >>> > -Chris >>> > >>> _________________________________________________________________________ >>> > Professional FreeSWITCH Consulting Services: >>> > consulting at freeswitch.org >>> > http://www.freeswitchsolutions.com >>> > >>> > >>> > >>> > >>> > Official FreeSWITCH Sites >>> > http://www.freeswitch.org >>> > http://wiki.freeswitch.org >>> > http://www.cluecon.com >>> > >>> > 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 >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130213/d0bd8863/attachment.html From david.villasmil.work at gmail.com Thu Feb 14 01:13:21 2013 From: david.villasmil.work at gmail.com (David Villasmil) Date: Wed, 13 Feb 2013 23:13:21 +0100 Subject: [Freeswitch-users] Nibblebill database update In-Reply-To: References: <5091F32B.2010302@xofap.com> <509374A4.1090203@bestnet.kharkov.ua> <50937971.4090809@xofap.com> <8724844.crljyV7uo8@sos> Message-ID: Hello, True, i didn't catch that David --- David Villasmil On Feb 13, 2013, at 21:19, Steven Ayre wrote: > [STATE: 42000 CODE 1064 ERROR: [unixODBC][MySQL][ODBC 5.1 > Driver][mysqld-5.5.24-0ubuntu0.12.04.1]You have an error in your SQL > syntax; check the manual that corresponds to your MySQL server version > for the right syntax to use near 'usage=usage-0.476000 WHERE > user='1001'' at line 1 > ] > > This error is coming from MySQL itself, which indicates it is actually using the ODBC connection. This isn't the same as David's issue. > > The error message starts at the first character that failed the SQL parsing. That's "usage". > > That happens to be on the list of MySQL reserved words which will give an error in 5.5 (which wouldn't have happened in 5.1 or earlier). > http://dev.mysql.com/doc/refman/5.5/en/reserved-words.html > > You either need to change your DB structure to avoid using the reserved words (best) or quote the column name in backticks (`). > > Try setting: > > > -Steve > > > > On 13 February 2013 19:59, David Villasmil wrote: >> Hello, >> >> I was having the same problem, the solution is to use: >> >> >> >> where "dsn_name" is obviously the dsn inside the /etc/odbc.ini file previously tested with "isql" >> >> >> good luck! >> >> David >> >> >> On Fri, Nov 2, 2012 at 11:25 AM, Sergey Okhapkin wrote: >>> What is the structure of "accounts" table? >>> >>> On Friday 02 November 2012 14:42:41 William Alianto wrote: >>> > Thanks for pointing that out. I think I missed that part when I edited >>> > the config. >>> > >>> > On 11/02/2012 02:22 PM, Evgeniy Movlyan wrote: >>> > > Seems you need to uncomment custom SQL-queries and rewrite it to >>> > > according to your needs. >>> > > >>> > > 02.11.2012 05:54, William Alianto ?????: >>> > >> Hi, >>> > >> >>> > >> This is my nibblebill.conf.xml >>> > >> >>> > >> >>> > >> >>> > >> >>> > >> >>> > >> >>> > >> >>> > >> >>> > >> >>> > >> >>> > >> >>> > >> >>> > >> >>> > >> >>> > >> >>> > >> >>> > >> >>> > >> >>> > >> >>> > >> >>> > >> >>> > >> >>> > >> >>> > >> >>> > >> >>> > >> >>> > >> >>> > >> >>> > >> >>> > >> >>> > >> >>> > >> >>> > >> >>> > >> >>> > >> >>> > >> >>> > >> >>> > >> >>> > >> >>> > >> >>> > >> On 11/01/2012 02:56 PM, Evgeniy Movlyan wrote: >>> > >>> Hello. >>> > >>> Show please your nibblebill.conf.xml. >>> > >>> >>> > >>> 01.11.2012 05:57, William Alianto ?????: >>> > >>>> Hi, >>> > >>>> >>> > >>>> I was trying to use nibblebill as billing control of my FS. I have >>> > >>>> configured the odbc and the nibblebill configuration. When I tried to >>> > >>>> call, the query looks ok. But when I hanged up the call, it seems like >>> > >>>> there is something wrong with the database connection, since it's not >>> > >>>> updating the database at all. I got error message on CLI >>> > >>>> >>> > >>>> 2012-11-01 10:51:00.559472 [ERR] switch_odbc.c:494 ERR: [UPDATE >>> > >>>> accounts >>> > >>>> SET usage=age-0.476000 WHERE user='1001'] >>> > >>>> [STATE: 42000 CODE 1064 ERROR: [unixODBC][MySQL][ODBC 5.1 >>> > >>>> Driver][mysqld-5.5.24-0ubuntu0.12.04.1]You have an error in your SQL >>> > >>>> syntax; check the manual that corresponds to your MySQL server version >>> > >>>> for the right syntax to use near 'usage=age-0.476000 WHERE >>> > >>>> user=001'' at line 1 >>> > >>>> ] >>> > >>>> 2012-11-01 10:51:00.559472 [ERR] mod_nibblebill.c:343 ERR: [UPDATE >>> > >>>> accounts SET usage=age-0.476000 WHERE user='1001'] >>> > >>>> [] >>> > >>>> 2012-11-01 10:51:00.559472 [CRIT] mod_nibblebill.c:542 Failed to >>> > >>>> log to >>> > >>>> database! >>> > >>>> 2012-11-01 10:51:00.559472 [DEBUG] mod_nibblebill.c:383 Doing lookup >>> > >>>> query >>> > >>>> [SELECT usage AS nibble_balance FROM accounts WHERE user=001'] >>> > >>>> 2012-11-01 10:51:00.559472 [ERR] mod_nibblebill.c:385 ERR: [SELECT >>> > >>>> usage >>> > >>>> AS nibble_balance FROM accounts WHERE user=001'] >>> > >>>> [STATE: 42000 CODE 1064 ERROR: [unixODBC][MySQL][ODBC 5.1 >>> > >>>> Driver][mysqld-5.5.24-0ubuntu0.12.04.1]You have an error in your SQL >>> > >>>> syntax; check the manual that corresponds to your MySQL server version >>> > >>>> for the right syntax to use near 'usage AS nibble_balance FROM >>> > >>>> accounts >>> > >>>> WHERE user=001'' at line 1 >>> > >>>> ] >>> > >>>> 2012-11-01 10:51:00.559472 [ERR] mod_nibblebill.c:386 Error running >>> > >>>> this >>> > >>>> query: [SELECT usage AS nibble_balance FROM accounts WHERE user=001'] >>> > >>>> >>> > >>>> Could anybody help me solve this issue? >>> > >>>> >>> > >>>> Regards >>> > >>>> >>> > >>>> _______________________________________________________________________ >>> > >>>> __ >>> > >>>> >>> > >>>> >>> > >>>> Professional FreeSWITCH Consulting Services: >>> > >>>> consulting at freeswitch.org >>> > >>>> http://www.freeswitchsolutions.com >>> > >>>> >>> > >>>> >>> > >>>> >>> > >>>> >>> > >>>> Official FreeSWITCH Sites >>> > >>>> http://www.freeswitch.org >>> > >>>> http://wiki.freeswitch.org >>> > >>>> http://www.cluecon.com >>> > >>>> >>> > >>>> 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-user >>> > >>>> s >>> > >>>> >>> > >>>> http://www.freeswitch.org >>> > >>> > _________________________________________________________________________ >>> > Professional FreeSWITCH Consulting Services: >>> > consulting at freeswitch.org >>> > http://www.freeswitchsolutions.com >>> > >>> > >>> > >>> > >>> > Official FreeSWITCH Sites >>> > http://www.freeswitch.org >>> > http://wiki.freeswitch.org >>> > http://www.cluecon.com >>> > >>> > 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 >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130213/d75d7d1f/attachment-0001.html From nasida at live.ru Thu Feb 14 01:23:39 2013 From: nasida at live.ru (Yuriy Nasida) Date: Thu, 14 Feb 2013 02:23:39 +0400 Subject: [Freeswitch-users] 486 Busy Here with Reason: Q.850; cause=16; text="NORMAL_CLEARING" In-Reply-To: References: , Message-ID: Steve, yes I use hangup_after_bridge=true in my dialplan. Probably you are correct and it looks like that it is really a reason of "NORMAL_CLEARING". Um... I don't want ignore the remote cause. Actually I want to use remote cause and ignore "NORMAL_CLEARING". May be I should not use hangup_after_bridge. Also may be I understand you not correctly. At present I fixed my issue by means of http://wiki.freeswitch.org/wiki/Variable_disable_q850_reason Thanks a lot for your response. From: steveayre at gmail.com Date: Wed, 13 Feb 2013 17:19:29 +0000 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] 486 Busy Here with Reason: Q.850; cause=16; text="NORMAL_CLEARING" Please post a debug-level log of the call. How are you hanging up the call? If with the 'hangup' app what parameters do you use? In particular if you bridge and it fails with 486 then hangup then the SIP code sent may be from the last bridge cause while the Q.850 cause is the one given to hangup. If that's the case try setting sip_ignore_remote_cause=true before the bridge and it'll only send the Q.850->SIP mapping (https://tools.ietf.org/html/rfc3398). Note there's no SIP code mapping for 16 though. If hanging up without answering the call use a different cause. -Steve On 13 February 2013 15:17, Yuriy Nasida wrote: Hi guys, I see that my FS sends sip message "486 Busy Here" with Reason: Q.850; cause=16;text="NORMAL_CLEARING". Why 16 and "NORMAL_CLEARING ? Probably it should be 17 "user busy". One of my carrier play a "fast busy" (number not available) and blames Q.850 field. Any advice are welcome.Thanks. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130214/6524bff4/attachment.html From jaykris at gmail.com Thu Feb 14 01:26:27 2013 From: jaykris at gmail.com (JP) Date: Wed, 13 Feb 2013 14:26:27 -0800 Subject: [Freeswitch-users] DTMF in IVR menu In-Reply-To: References: <2516165109928082112@unknownmsgid> Message-ID: I still am facing this problem, but wanted to provide more data in case somebody has a solution already... 1. The call path is from PSTN --> Asterisk --> FreeSWITCH. 2. play-and-get-digits works fine for multiple dtmf. But for one single dtmf I see delay in recognizing the dtmf on the FS side. 3. All traffic is RFC2833, no inband. 4. I tried setting "rtp_manual_rtp_bugs=ignore_dtmf_duration", but that did not help. 5. I see at least one other user had the same problem on the thread ( http://lists.freeswitch.org/pipermail/freeswitch-users/2012-August/086720.html), but I don't see a final solution if there was any. 6. I looked at a tcpdump from both the Asterisk and the FS boxes, but nothing obvious jumped at me. Any help is greatly appreciated. Thanks, JP On Mon, Feb 11, 2013 at 2:18 PM, JP wrote: > I am using Counterpath's eyebeam and their documentation says jitterbuffer > is dynamic and cannot be configured. I don't have this problem with > play-and-get-digits, so I don't think the RTT is a problem. I was wondering > would tweaking the following parameters help in any way? > > min_dtmf_duration > max_dtmf_duration > default_dtmf_duration > > Thanks, > JP > > On Mon, Feb 11, 2013 at 1:52 PM, Steven Ayre wrote: > >> Out of interest, do you know the RTT from phone to server, and the size >> of the jitterbuffers? >> >> Just wondering whether the lag is a result of the two combined. DTMF >> digits can also last a noticeable length of time (often ~400ms) and I >> believe get detected when they 'end'. >> >> -Steve >> >> >> >> >> On 11 February 2013 21:21, Guillermo Ruiz Camauer wrote: >> >>> I have ported an automated polling (surveys) system from Dialogic to >>> FreeSwitch and noticed the same thing. People are used to interrupting the >>> prompts as soon as they hear the option they like. Is there any way around >>> this delay? >>> >>> Guillermo >>> >>> Sent from my iPhone >>> >>> On 11/02/2013, at 18:17, Michael Collins wrote: >>> >>> Yes, there is a bit of a delay between the the time the digit is entered >>> and the time that it is officially detected and the playback of the IVR >>> greeting is interrupted. I've not noticed that being a problem, other than >>> people maybe wondering if their digits were not properly entered so they >>> hit it again. Is that what is happening? >>> >>> -MC >>> >>> On Fri, Feb 8, 2013 at 4:03 PM, JP wrote: >>> >>>> Hi, >>>> I am observing that when I use IVR menu, the main menu greeting does >>>> not get stopped as soon as I enter the dtmf digit. It plays for a couple of >>>> seconds longer and then stops and then the dtmf is processed. Is there any >>>> tuning parameters that I can tweak to make it stop sooner. >>>> >>>> Thanks, >>>> JP >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>>> >>>> >>> >>> >>> -- >>> Michael S Collins >>> Twitter: @mercutioviz >>> http://www.FreeSWITCH.org >>> http://www.ClueCon.com >>> http://www.OSTAG.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130213/b45fef99/attachment.html From jnvines at gmail.com Thu Feb 14 01:31:40 2013 From: jnvines at gmail.com (Nick Vines) Date: Wed, 13 Feb 2013 14:31:40 -0800 Subject: [Freeswitch-users] Best current ATA and Setup for Faxing In-Reply-To: References: <0B7A549A-767A-4DA6-B732-9999DC6B1B48@gmail.com> Message-ID: I set the following on the Ht502 when I am using that line for faxing. P codes given are for the 1st account on the device. 1) Disabling Line Echo Canceller, P824=1 2) Fax Mode T.38 Auto-Detect, P228=0 3) Fax Tone Detection Caller or Callee, P894=2 On freeswitch, I set this before I bridge the call. If your provider doesn't support T.38, this will cause a "Not Allowed" hangup, but with Flowroute it has been working flawlessly. Nick On Wed, Feb 13, 2013 at 2:02 PM, George Assaad wrote: > Sorry I meant to ask for the set up of the ht-502 > On Feb 13, 2013 4:27 PM, "Chris Ferreira" wrote: > >> Well, this ATA died, and I don't have a backup of it's config when we >> were using it for Fax testing. We went through a number of tweaks and >> firmware upgrades. This is months ago. >> >> >> I'm more looking for whats the best starting point for a new ATA. At the >> time it seemed like it was the SPA112, but that was 6 months ago. Just >> wondering if this is still a good starting point, or if I should be >> checking out a new ATA or provider. >> >> >> Thanks, >> >> -Chris >> >> >> On Wed, Feb 13, 2013 at 2:47 PM, George Assaad wrote: >> >>> Can you please share with us the setting for fax, if you don't mind. >>> On Feb 13, 2013 11:17 AM, "Nick Vines" wrote: >>> >>>> I havent used the cisco atas, but I've had good success with the grand >>>> stream ht502s and ht702s. My preferred faxing carrier is also flowroute. >>>> >>>> Nick >>>> >>>> On Feb 13, 2013, at 8:02 AM, Chris Ferreira wrote: >>>> >>>> > Hi Folks, >>>> > >>>> > >>>> > We had previously been working with the Cisco SPA112 for T.38 Faxing >>>> through Freeswitch and out to Flowroute. At the time we didn't get it >>>> working 100% perfectly. >>>> > >>>> > >>>> > I now have one of these SPA112's that I need to replace because >>>> someone fried it by plugging in a much larger power supply by accident. >>>> > >>>> > >>>> > Is my scenario above still a good starting point for Faxing or should >>>> I be looking at another ATA or provider other than Flowroute? >>>> > >>>> > >>>> > >>>> > >>>> > >>>> > Also, one more question, and this may be better suited to the >>>> business list (let me know if I should move it there), but we are looking >>>> for a part time as needed remote or local Freeswitch Admin / Engineer / >>>> Tech. Our previous person is MIA. >>>> > >>>> > >>>> > >>>> > >>>> > Thanks Everyone! >>>> > >>>> > >>>> > >>>> > -Chris >>>> > >>>> _________________________________________________________________________ >>>> > Professional FreeSWITCH Consulting Services: >>>> > consulting at freeswitch.org >>>> > http://www.freeswitchsolutions.com >>>> > >>>> > >>>> > >>>> > >>>> > Official FreeSWITCH Sites >>>> > http://www.freeswitch.org >>>> > http://wiki.freeswitch.org >>>> > http://www.cluecon.com >>>> > >>>> > 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 >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130213/38adffd4/attachment-0001.html From gassaad at emassembly.com Thu Feb 14 02:43:42 2013 From: gassaad at emassembly.com (George Assaad) Date: Wed, 13 Feb 2013 18:43:42 -0500 Subject: [Freeswitch-users] Best current ATA and Setup for Faxing In-Reply-To: References: <0B7A549A-767A-4DA6-B732-9999DC6B1B48@gmail.com> Message-ID: Thanks a million On Feb 13, 2013 6:05 PM, "Nick Vines" wrote: > I set the following on the Ht502 when I am using that line for faxing. P > codes given are for the 1st account on the device. > > 1) Disabling Line Echo Canceller, P824=1 > 2) Fax Mode T.38 Auto-Detect, P228=0 > 3) Fax Tone Detection Caller or Callee, P894=2 > > On freeswitch, I set this before I bridge the call. > > > If your provider doesn't support T.38, this will cause a "Not Allowed" > hangup, but with Flowroute it has been working flawlessly. > > Nick > > > On Wed, Feb 13, 2013 at 2:02 PM, George Assaad wrote: > >> Sorry I meant to ask for the set up of the ht-502 >> On Feb 13, 2013 4:27 PM, "Chris Ferreira" wrote: >> >>> Well, this ATA died, and I don't have a backup of it's config when we >>> were using it for Fax testing. We went through a number of tweaks and >>> firmware upgrades. This is months ago. >>> >>> >>> I'm more looking for whats the best starting point for a new ATA. At the >>> time it seemed like it was the SPA112, but that was 6 months ago. Just >>> wondering if this is still a good starting point, or if I should be >>> checking out a new ATA or provider. >>> >>> >>> Thanks, >>> >>> -Chris >>> >>> >>> On Wed, Feb 13, 2013 at 2:47 PM, George Assaad wrote: >>> >>>> Can you please share with us the setting for fax, if you don't mind. >>>> On Feb 13, 2013 11:17 AM, "Nick Vines" wrote: >>>> >>>>> I havent used the cisco atas, but I've had good success with the grand >>>>> stream ht502s and ht702s. My preferred faxing carrier is also flowroute. >>>>> >>>>> Nick >>>>> >>>>> On Feb 13, 2013, at 8:02 AM, Chris Ferreira >>>>> wrote: >>>>> >>>>> > Hi Folks, >>>>> > >>>>> > >>>>> > We had previously been working with the Cisco SPA112 for T.38 Faxing >>>>> through Freeswitch and out to Flowroute. At the time we didn't get it >>>>> working 100% perfectly. >>>>> > >>>>> > >>>>> > I now have one of these SPA112's that I need to replace because >>>>> someone fried it by plugging in a much larger power supply by accident. >>>>> > >>>>> > >>>>> > Is my scenario above still a good starting point for Faxing or >>>>> should I be looking at another ATA or provider other than Flowroute? >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > Also, one more question, and this may be better suited to the >>>>> business list (let me know if I should move it there), but we are looking >>>>> for a part time as needed remote or local Freeswitch Admin / Engineer / >>>>> Tech. Our previous person is MIA. >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > Thanks Everyone! >>>>> > >>>>> > >>>>> > >>>>> > -Chris >>>>> > >>>>> _________________________________________________________________________ >>>>> > Professional FreeSWITCH Consulting Services: >>>>> > consulting at freeswitch.org >>>>> > http://www.freeswitchsolutions.com >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > Official FreeSWITCH Sites >>>>> > http://www.freeswitch.org >>>>> > http://wiki.freeswitch.org >>>>> > http://www.cluecon.com >>>>> > >>>>> > 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 >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> 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 >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130213/aa42405c/attachment.html From msc at freeswitch.org Thu Feb 14 03:32:03 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 13 Feb 2013 16:32:03 -0800 Subject: [Freeswitch-users] ooH323 vs mod_h323 In-Reply-To: References: <2BD0DC60-D428-4323-A020-D9AD20E76553@freeswitch.org> <4C2EA249.4070704@coppice.org> <267C8398-6CA3-442E-B973-BE096D728B28@freeswitch.org> Message-ID: How intense are their patches? The reason I ask is that while a blind C&P of their patches into the recent version would most likely result in a GPL violation, looking at what the code does and mimicking the functionality with new code most certainly would not result in a violation. (GPL covers "derivation" but not "inspiration" as it were. It's the whole copyright "idea" vs. "expression" dichotomy...) Anyway, I was just curious what's involved. Possibly other interested parties would be in a position to help out. -MC On Wed, Feb 13, 2013 at 12:06 PM, Steven Ayre wrote: > Indeed comparing the COPYING files, the Asterisk one lacks the FLOSS > exception. So that version isn't usable, and neither is copying their > patches into the 0.9.4 version since they'd be GPL-only. > > So while 0.9.4 would be usable, it'd lack some useful functionality (Q.931 > causes are just the one I spotted from a 5 minute glance through the API). > > -Steve > > > > > On 13 February 2013 12:27, Steven Ayre wrote: > >> I was just looking at this library to play with the idea of writing a >> mod_ooh323c. It looks like the library may have forked away from the >> version used in Asterisk though? >> >> The latest version on the website ( >> http://www.obj-sys.com/telephony-objective.shtml) is 0.9.4, but cannot >> hangup with a Q.931 hangup cause. The CVS repository ( >> sourceforge.net/projects/ooh323c) doesn't seem up-to-date since that's >> only holding 0.9.0. >> >> The version in asterisk has been patched to add a q931cause parameter to >> ooHangCall so it can - but it seems to be based off 'v0.8.3m' and I don't >> know if that's before the FLOSS exception was added. >> >> So does anyone more familiar with the library know whether the repository >> had been moved elsewhere, or whether there's a more recent version >> somewhere else, and whether the one bundled in Asterisk would be usable? >> >> -Steve >> >> >> >> >> On 5 July 2010 15:28, Brian West wrote: >> >>> It has a FLOSS exception in the GPL license. >>> >>> /b >>> >>> On Jul 2, 2010, at 9:36 PM, Steve Underwood wrote: >>> >>> > On 07/03/2010 04:28 AM, Brian West wrote: >>> >> DOH... btw ooh323 is now license compatible with FreeSWITCH if anyone >>> wants to write one. >>> >> >>> >> /b >>> >> >>> > Where did you see that? The current ooh323c tarball still says GPL 2. >>> > >>> > Steve >>> >>> >>> _______________________________________________ >>> 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 >>> >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130213/d130c353/attachment-0001.html From steveayre at gmail.com Thu Feb 14 11:08:47 2013 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 14 Feb 2013 08:08:47 +0000 Subject: [Freeswitch-users] ooH323 vs mod_h323 In-Reply-To: References: <2BD0DC60-D428-4323-A020-D9AD20E76553@freeswitch.org> <4C2EA249.4070704@coppice.org> <267C8398-6CA3-442E-B973-BE096D728B28@freeswitch.org> Message-ID: The q931cause parameter gets added in this revision: http://svnview.digium.com/svn/asterisk?view=revision&revision=227898 Which as you can see is rather a large commit - they fix a lot of other issues too. q931cause alone means changing quite a lot of code though - 11 times in 6 different files, plus the modified functions that it appears within. It means changing ooHangCall, the parameters for the command on the stack, ooH323HangCall, storing q931cause on the call, and then setting it during the actual hangup. -Steve On 14 February 2013 00:32, Michael Collins wrote: > How intense are their patches? The reason I ask is that while a blind C&P > of their patches into the recent version would most likely result in a GPL > violation, looking at what the code does and mimicking the functionality > with new code most certainly would not result in a violation. (GPL covers > "derivation" but not "inspiration" as it were. It's the whole copyright > "idea" vs. "expression" dichotomy...) > > Anyway, I was just curious what's involved. Possibly other interested > parties would be in a position to help out. > > -MC > > > On Wed, Feb 13, 2013 at 12:06 PM, Steven Ayre wrote: > >> Indeed comparing the COPYING files, the Asterisk one lacks the FLOSS >> exception. So that version isn't usable, and neither is copying their >> patches into the 0.9.4 version since they'd be GPL-only. >> >> So while 0.9.4 would be usable, it'd lack some useful functionality >> (Q.931 causes are just the one I spotted from a 5 minute glance through the >> API). >> >> -Steve >> >> >> >> >> On 13 February 2013 12:27, Steven Ayre wrote: >> >>> I was just looking at this library to play with the idea of writing a >>> mod_ooh323c. It looks like the library may have forked away from the >>> version used in Asterisk though? >>> >>> The latest version on the website ( >>> http://www.obj-sys.com/telephony-objective.shtml) is 0.9.4, but cannot >>> hangup with a Q.931 hangup cause. The CVS repository ( >>> sourceforge.net/projects/ooh323c) doesn't seem up-to-date since that's >>> only holding 0.9.0. >>> >>> The version in asterisk has been patched to add a q931cause parameter to >>> ooHangCall so it can - but it seems to be based off 'v0.8.3m' and I don't >>> know if that's before the FLOSS exception was added. >>> >>> So does anyone more familiar with the library know whether the >>> repository had been moved elsewhere, or whether there's a more recent >>> version somewhere else, and whether the one bundled in Asterisk would be >>> usable? >>> >>> -Steve >>> >>> >>> >>> >>> On 5 July 2010 15:28, Brian West wrote: >>> >>>> It has a FLOSS exception in the GPL license. >>>> >>>> /b >>>> >>>> On Jul 2, 2010, at 9:36 PM, Steve Underwood wrote: >>>> >>>> > On 07/03/2010 04:28 AM, Brian West wrote: >>>> >> DOH... btw ooh323 is now license compatible with FreeSWITCH if >>>> anyone wants to write one. >>>> >> >>>> >> /b >>>> >> >>>> > Where did you see that? The current ooh323c tarball still says GPL 2. >>>> > >>>> > Steve >>>> >>>> >>>> _______________________________________________ >>>> 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 >>>> >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> >> 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 >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130214/0676cc87/attachment.html From julf at julf.com Thu Feb 14 11:38:39 2013 From: julf at julf.com (Johan Helsingius) Date: Thu, 14 Feb 2013 09:38:39 +0100 Subject: [Freeswitch-users] Freeswitch + freetdm dumps core on Ubuntu 12.10 Message-ID: <511CA28F.4090109@julf.com> Hi! Tried compiling freeswitch with freetdm on Ubuntu 12.10 using both v1.2.stable and Master from git. Both dump core. Here is what gdb says: Program received signal SIGSEGV, Segmentation fault. 0xb7e12596 in switch_loadable_module_process (key=key at entry=0x8120ea8 "mod_file_string", new_module=new_module at entry=0x82a4200) at src/switch_loadable_module.c:321 321 for (i = 0; ptr->extens[i]; i++) { What is the best way forward? Report it as a bug? Julf From avi at avimarcus.net Thu Feb 14 11:47:31 2013 From: avi at avimarcus.net (Avi Marcus) Date: Thu, 14 Feb 2013 10:47:31 +0200 Subject: [Freeswitch-users] ooH323 vs mod_h323 In-Reply-To: References: <2BD0DC60-D428-4323-A020-D9AD20E76553@freeswitch.org> <4C2EA249.4070704@coppice.org> <267C8398-6CA3-442E-B973-BE096D728B28@freeswitch.org> Message-ID: Maybe you can get that patcher to pull the lib with the newer license and contribute patches up stream :) -Avi On Thu, Feb 14, 2013 at 10:08 AM, Steven Ayre wrote: > The q931cause parameter gets added in this revision: > http://svnview.digium.com/svn/asterisk?view=revision&revision=227898 > > Which as you can see is rather a large commit - they fix a lot of other > issues too. q931cause alone means changing quite a lot of code though - 11 > times in 6 different files, plus the modified functions that it appears > within. It means changing ooHangCall, the parameters for the command on the > stack, ooH323HangCall, storing q931cause on the call, and then setting it > during the actual hangup. > > -Steve > > > > On 14 February 2013 00:32, Michael Collins wrote: > >> How intense are their patches? The reason I ask is that while a blind C&P >> of their patches into the recent version would most likely result in a GPL >> violation, looking at what the code does and mimicking the functionality >> with new code most certainly would not result in a violation. (GPL covers >> "derivation" but not "inspiration" as it were. It's the whole copyright >> "idea" vs. "expression" dichotomy...) >> >> Anyway, I was just curious what's involved. Possibly other interested >> parties would be in a position to help out. >> >> -MC >> >> >> On Wed, Feb 13, 2013 at 12:06 PM, Steven Ayre wrote: >> >>> Indeed comparing the COPYING files, the Asterisk one lacks the FLOSS >>> exception. So that version isn't usable, and neither is copying their >>> patches into the 0.9.4 version since they'd be GPL-only. >>> >>> So while 0.9.4 would be usable, it'd lack some useful functionality >>> (Q.931 causes are just the one I spotted from a 5 minute glance through the >>> API). >>> >>> -Steve >>> >>> >>> >>> >>> On 13 February 2013 12:27, Steven Ayre wrote: >>> >>>> I was just looking at this library to play with the idea of writing a >>>> mod_ooh323c. It looks like the library may have forked away from the >>>> version used in Asterisk though? >>>> >>>> The latest version on the website ( >>>> http://www.obj-sys.com/telephony-objective.shtml) is 0.9.4, but cannot >>>> hangup with a Q.931 hangup cause. The CVS repository ( >>>> sourceforge.net/projects/ooh323c) doesn't seem up-to-date since that's >>>> only holding 0.9.0. >>>> >>>> The version in asterisk has been patched to add a q931cause parameter >>>> to ooHangCall so it can - but it seems to be based off 'v0.8.3m' and I >>>> don't know if that's before the FLOSS exception was added. >>>> >>>> So does anyone more familiar with the library know whether the >>>> repository had been moved elsewhere, or whether there's a more recent >>>> version somewhere else, and whether the one bundled in Asterisk would be >>>> usable? >>>> >>>> -Steve >>>> >>>> >>>> >>>> >>>> On 5 July 2010 15:28, Brian West wrote: >>>> >>>>> It has a FLOSS exception in the GPL license. >>>>> >>>>> /b >>>>> >>>>> On Jul 2, 2010, at 9:36 PM, Steve Underwood wrote: >>>>> >>>>> > On 07/03/2010 04:28 AM, Brian West wrote: >>>>> >> DOH... btw ooh323 is now license compatible with FreeSWITCH if >>>>> anyone wants to write one. >>>>> >> >>>>> >> /b >>>>> >> >>>>> > Where did you see that? The current ooh323c tarball still says GPL 2. >>>>> > >>>>> > Steve >>>>> >>>>> >>>>> _______________________________________________ >>>>> 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 >>>>> >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> >>> 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 >>> >>> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130214/45b4287e/attachment-0001.html From steveayre at gmail.com Thu Feb 14 11:54:47 2013 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 14 Feb 2013 08:54:47 +0000 Subject: [Freeswitch-users] 486 Busy Here with Reason: Q.850; cause=16; text="NORMAL_CLEARING" In-Reply-To: References: Message-ID: hangup_after_bridge should use the hangup cause of the last successful bridge, but USER_BUSY may not be a 'successful' bridge. Put this after the bridge then reloadxml and see if you get any log entries. -Steve On 13 February 2013 22:23, Yuriy Nasida wrote: > Steve, yes I use hangup_after_bridge=true in my dialplan. Probably you > are correct and it looks like that it is really a reason of > "NORMAL_CLEARING". Um... I don't want ignore the remote cause. Actually I > want to use remote cause and ignore "NORMAL_CLEARING". May be I should not > use hangup_after_bridge. Also may be I understand you not correctly. > At present I fixed my issue by means of > http://wiki.freeswitch.org/wiki/Variable_disable_q850_reason > > Thanks a lot for your response. > > ------------------------------ > From: steveayre at gmail.com > Date: Wed, 13 Feb 2013 17:19:29 +0000 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] 486 Busy Here with Reason: Q.850; > cause=16; text="NORMAL_CLEARING" > > > Please post a debug-level log of the call. > > How are you hanging up the call? If with the 'hangup' app what parameters > do you use? > > In particular if you bridge and it fails with 486 then hangup then the SIP > code sent may be from the last bridge cause while the Q.850 cause is the > one given to hangup. If that's the case try setting > sip_ignore_remote_cause=true before the bridge and it'll only send the > Q.850->SIP mapping (https://tools.ietf.org/html/rfc3398). > > Note there's no SIP code mapping for 16 though. If hanging up without > answering the call use a different cause. > > -Steve > > > > On 13 February 2013 15:17, Yuriy Nasida wrote: > > Hi guys, > > I see that my FS sends sip message "486 Busy Here" with Reason: Q.850; > cause=16;text="NORMAL_CLEARING". Why 16 and "NORMAL_CLEARING ? Probably it > should be 17 "user busy". One of my carrier play a "fast busy" (number not > available) and blames Q.850 field. > > Any advice are welcome. > Thanks. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.org > http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel > Communication Server Official FreeSWITCH Sites > http://www.freeswitch.org http://wiki.freeswitch.org > http://www.cluecon.com 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130214/4e3a80b9/attachment.html From steveayre at gmail.com Thu Feb 14 11:57:59 2013 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 14 Feb 2013 08:57:59 +0000 Subject: [Freeswitch-users] Freeswitch + freetdm dumps core on Ubuntu 12.10 In-Reply-To: <511CA28F.4090109@julf.com> References: <511CA28F.4090109@julf.com> Message-ID: Do you have any modules.conf.xml with critical="true"? If so FS will intentionally segfault if one of those modules has an error loading. If not see http://wiki.freeswitch.org/wiki/Reporting_Bugs#Creating_A_Backtrace_With_gdb_.28Linux.2FUnix.29to get a full backtrace, and debug-level logs would also be helpful. -Steve On 14 February 2013 08:38, Johan Helsingius wrote: > Hi! > > Tried compiling freeswitch with freetdm on Ubuntu 12.10 using > both v1.2.stable and Master from git. > Both dump core. Here is what gdb says: > > Program received signal SIGSEGV, Segmentation fault. > 0xb7e12596 in switch_loadable_module_process (key=key at entry=0x8120ea8 > "mod_file_string", > new_module=new_module at entry=0x82a4200) at > src/switch_loadable_module.c:321 > 321 for (i = 0; ptr->extens[i]; i++) { > > What is the best way forward? Report it as a bug? > > Julf > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130214/610c048e/attachment.html From julf at julf.com Thu Feb 14 12:20:46 2013 From: julf at julf.com (Johan Helsingius) Date: Thu, 14 Feb 2013 10:20:46 +0100 Subject: [Freeswitch-users] Freeswitch + freetdm dumps core on Ubuntu 12.10 In-Reply-To: References: <511CA28F.4090109@julf.com> Message-ID: <511CAC6E.4030509@julf.com> > Do you have any modules.conf.xml with critical="true"? If so FS will > intentionally segfault if one of those modules has an error loading. No, critical is not set for any modules. > If not > see http://wiki.freeswitch.org/wiki/Reporting_Bugs#Creating_A_Backtrace_With_gdb_.28Linux.2FUnix.29 > to get a full backtrace, and debug-level logs would also be helpful. Will do - will just have to recompile Master, as I have stable compiled right now. Julf From itsusama at gmail.com Thu Feb 14 13:52:07 2013 From: itsusama at gmail.com (Usama Zaidi) Date: Thu, 14 Feb 2013 15:52:07 +0500 Subject: [Freeswitch-users] Perl Alarm Event Message-ID: Hi All, Freeswitch process exits on alarm event http://pastebin.freeswitch.org/20577 1. Save the script 2. Run it on terminal 3. Ctrl + C and open fs_cli 4. perlrun 5. PROFIT??? -- I'd love to change the world, but they wont gimme the source code to it -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130214/9fcf7fd9/attachment.html From julf at julf.com Thu Feb 14 13:54:26 2013 From: julf at julf.com (Johan Helsingius) Date: Thu, 14 Feb 2013 11:54:26 +0100 Subject: [Freeswitch-users] Freeswitch + freetdm dumps core on Ubuntu 12.10 In-Reply-To: References: <511CA28F.4090109@julf.com> Message-ID: <511CC262.2030906@julf.com> > If not > see http://wiki.freeswitch.org/wiki/Reporting_Bugs#Creating_A_Backtrace_With_gdb_.28Linux.2FUnix.29 > to get a full backtrace, and debug-level logs would also be helpful. Here is the info: http://pastebin.freeswitch.org/20580 Julf From itsusama at gmail.com Thu Feb 14 13:56:00 2013 From: itsusama at gmail.com (Usama Zaidi) Date: Thu, 14 Feb 2013 15:56:00 +0500 Subject: [Freeswitch-users] Perl Alarm Event Message-ID: Update:: If I execute using application 'System' instead of application 'Perl' in the dialplan it works finie... P.S. Here's the script again for reference... http://pastebin.freeswitch.org/20577 -- I'd love to change the world, but they wont gimme the source code to it -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130214/49fdad19/attachment-0001.html From mehroz.ashraf85 at gmail.com Thu Feb 14 14:07:48 2013 From: mehroz.ashraf85 at gmail.com (mehroz) Date: Thu, 14 Feb 2013 03:07:48 -0800 (PST) Subject: [Freeswitch-users] mod_perl installation error Message-ID: <1360840068571-7587401.post@n2.nabble.com> After installing perl , enabling in modules.conf, re compiling freeswitch. when i do: make mod_perl-install i get this: making install mod_perl Creating mod_perl.la... /usr/bin/ld: cannot find -lgdbm collect2: ld returned 1 exit status cat: .libs/mod_perl.log: No such file or directory make[3]: *** [mod_perl.la] Error 1 make[2]: *** [install] Error 1 make[1]: *** [mod_perl-install] Error 1 make: *** [mod_perl-install] Error 2 Stucked here........ any one? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-perl-installation-error-tp7587401.html Sent from the freeswitch-users mailing list archive at Nabble.com. From paul at cupis.co.uk Thu Feb 14 14:28:02 2013 From: paul at cupis.co.uk (Paul Cupis) Date: Thu, 14 Feb 2013 11:28:02 +0000 Subject: [Freeswitch-users] mod_perl installation error In-Reply-To: <1360840068571-7587401.post@n2.nabble.com> References: <1360840068571-7587401.post@n2.nabble.com> Message-ID: <20130214112802.GA12329@eagle.cupis.co.uk> On Thu, Feb 14, 2013 at 03:07:48AM -0800, mehroz wrote: > /usr/bin/ld: cannot find -lgdbm > Stucked here........ any one? You need to install the gdbm libraries on your system. On Debian, I guess this is: apt-get install libgdbm-dev libgdbm3 Regards, From vipkilla at gmail.com Thu Feb 14 15:58:21 2013 From: vipkilla at gmail.com (Vik Killa) Date: Thu, 14 Feb 2013 07:58:21 -0500 Subject: [Freeswitch-users] Nibblebill database update In-Reply-To: References: <5091F32B.2010302@xofap.com> <509374A4.1090203@bestnet.kharkov.ua> <50937971.4090809@xofap.com> <8724844.crljyV7uo8@sos> Message-ID: I'll take the blame for this one. I wrote the patch but didn't update the wiki. It slipped my mind since the patch was committed months after I wrote it. Sorry FS community. From steveayre at gmail.com Thu Feb 14 16:04:52 2013 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 14 Feb 2013 13:04:52 +0000 Subject: [Freeswitch-users] Nibblebill database update In-Reply-To: References: <5091F32B.2010302@xofap.com> <509374A4.1090203@bestnet.kharkov.ua> <50937971.4090809@xofap.com> <8724844.crljyV7uo8@sos> Message-ID: Don't beat yourself up too much, the Wiki is/was still correct for v1.2.stable as the patch hasn't moved across from master yet. :o) And William's issue is unrelated. -Steve On 14 February 2013 12:58, Vik Killa wrote: > I'll take the blame for this one. I wrote the patch but didn't update the > wiki. > It slipped my mind since the patch was committed months after I wrote it. > Sorry FS community. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130214/7307669f/attachment.html From shaheryarkh at gmail.com Thu Feb 14 17:01:33 2013 From: shaheryarkh at gmail.com (Muhammad Shahzad) Date: Thu, 14 Feb 2013 15:01:33 +0100 Subject: [Freeswitch-users] Perl Alarm Event In-Reply-To: References: Message-ID: perlrun is inline, so it won't work. Thank you. On Thu, Feb 14, 2013 at 11:56 AM, Usama Zaidi wrote: > Update:: If I execute using application 'System' instead of application > 'Perl' in the dialplan it works finie... > > P.S. > Here's the script again for reference... > http://pastebin.freeswitch.org/20577 > > -- > I'd love to change the world, but they wont gimme the source code to it > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +49 176 99 83 10 85 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130214/163467f4/attachment.html From david.villasmil.work at gmail.com Thu Feb 14 17:14:02 2013 From: david.villasmil.work at gmail.com (David Villasmil) Date: Thu, 14 Feb 2013 15:14:02 +0100 Subject: [Freeswitch-users] Nibblebill Message-ID: Hello guys, Is it possible to use different billing tables for different applications? Say i have 2 different context on which i want to use different tables for balance. Thanks! David From steveayre at gmail.com Thu Feb 14 17:14:29 2013 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 14 Feb 2013 14:14:29 +0000 Subject: [Freeswitch-users] Perl Alarm Event In-Reply-To: References: Message-ID: Actually it should be possible. At least on Linux all threads have their own PID, which means running the perl script thread will receive the alarm signal and it'll be isolated from other FreeSWITCH threads. perlrun just needs to install its own signal handlers when started to capture that (and possibly other) signals and pass them to the interpreter to be handled internally, then restore the original signal handlers when the script finishes. SIGALRM is often used for timeouts, so this could break a number of Perl modules and prevent them being used from mod_perl. Terminating the process is the default action on SIGALRM though, which is why it currently exits freeswitch (FS doesn't install a signal handler of its own, and mod_perl doesn't either). I've already advised Usama on IRC to file a Jira. -Steve On 14 February 2013 14:01, Muhammad Shahzad wrote:in > perlrun is inline, so it won't work. > > Thank you. > > > On Thu, Feb 14, 2013 at 11:56 AM, Usama Zaidi wrote: > >> Update:: If I execute using application 'System' instead of application >> 'Perl' in the dialplan it works finie... >> >> P.S. >> Here's the script again for reference... >> http://pastebin.freeswitch.org/20577 >> >> -- >> I'd love to change the world, but they wont gimme the source code to it >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +49 176 99 83 10 85 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130214/e8dbca44/attachment-0001.html From shaheryarkh at gmail.com Thu Feb 14 17:42:58 2013 From: shaheryarkh at gmail.com (Muhammad Shahzad) Date: Thu, 14 Feb 2013 15:42:58 +0100 Subject: [Freeswitch-users] Perl Alarm Event In-Reply-To: References: Message-ID: makes sense. Thank you. On Thu, Feb 14, 2013 at 3:14 PM, Steven Ayre wrote: > Actually it should be possible. At least on Linux all threads have their > own PID, which means running the perl script thread will receive the alarm > signal and it'll be isolated from other FreeSWITCH threads. perlrun just > needs to install its own signal handlers when started to capture that (and > possibly other) signals and pass them to the interpreter to be handled > internally, then restore the original signal handlers when the script > finishes. SIGALRM is often used for timeouts, so this could break a number > of Perl modules and prevent them being used from mod_perl. > > Terminating the process is the default action on SIGALRM though, which is > why it currently exits freeswitch (FS doesn't install a signal handler of > its own, and mod_perl doesn't either). > > I've already advised Usama on IRC to file a Jira. > > -Steve > > > > On 14 February 2013 14:01, Muhammad Shahzad wrote:in > > perlrun is inline, so it won't work. >> >> Thank you. >> >> >> On Thu, Feb 14, 2013 at 11:56 AM, Usama Zaidi wrote: >> >>> Update:: If I execute using application 'System' instead of application >>> 'Perl' in the dialplan it works finie... >>> >>> P.S. >>> Here's the script again for reference... >>> http://pastebin.freeswitch.org/20577 >>> >>> -- >>> I'd love to change the world, but they wont gimme the source code to it >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> >> -- >> Muhammad Shahzad >> ----------------------------------- >> CISCO Rich Media Communication Specialist (CRMCS) >> CISCO Certified Network Associate (CCNA) >> Cell: +49 176 99 83 10 85 >> MSN: shari_786pk at hotmail.com >> Email: shaheryarkh at googlemail.com >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +49 176 99 83 10 85 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130214/ae79fb0b/attachment.html From levend.sayar at karel.com.tr Thu Feb 14 18:03:51 2013 From: levend.sayar at karel.com.tr (Levend Sayar) Date: Thu, 14 Feb 2013 15:03:51 +0000 Subject: [Freeswitch-users] SRTP disabling In-Reply-To: References: Message-ID: Thanks for your kind responses Steve. Phones are same brand, you are right. And not Polycom or Linksys. They are Yealink phones. I found the code piece handling the RTP/SAVP part you mentioned. (sofia_glue_negotiate_sdp function). Although it is not wise, I will add a new channel variable and try to disable SRTP even if phone says SRTP is mandatory. I need that. -- _lvnd_ {^_^} On Wed, 2013-02-13 at 17:59 +0000, Steven Ayre wrote: If the phone is Polycom or Linksys have you read the notes on http://wiki.freeswitch.org/wiki/SRTP? -Steve On 13 February 2013 17:53, Steven Ayre > wrote: Nevertheless it's not being negotiated and this is possibly a bug in the phone. Adding a workaround in FS would decrease security because the phone is explicitly saying it'll only accept SRTP so sending plain RTP wouldn't be wise. I assume both phones are the same make? -Steve On 7 February 2013 13:54, Levend Sayar > wrote: But the very same phone calls another phone and talk with RTP, not SRTP if the peer does not accept SRTP Here is the SDP offer by the same phone v=0 o=- 20186 20186 IN IP4 192.168.173.69 s=SDP data c=IN IP4 192.168.173.69 t=0 0 m=audio 11782 RTP/SAVP 0 8 18 9 101 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:NzFmYjdiMjk1OTY2ODQwYzExZjM0ZmE2NGM0YWMw a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:M2MxMTE2OWFjOGY2ZjEwADEzZmZkNzAxNjRlMzFm a=crypto:3 F8_128_HMAC_SHA1_80 inline:NjkzZDg2Mjk0ZTkxMjg1YzdmYjFiNjRlMmFhNGFm a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:9 G722/8000 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=ptime:20 a=sendrecv And here is the SDP answer sent by the other phone v=0 o=- 20029 20029 IN IP4 192.168.173.65 s=SDP data c=IN IP4 192.168.173.65 t=0 0 m=audio 11794 RTP/SAVP 0 101 a=rtpmap:0 PCMU/8000 a=sendrecv a=ptime:20 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 -- ?yi ?al??malar. _lvnd_ {^_^} KAREL`UCAP On Thu, 2013-02-07 at 13:37 +0000, Steven Ayre wrote: m=audio 11780 RTP/SAVP 0 8 18 9 101 RTP/SAVP means SRTP is mandatory. You need to reconfigure the phone. If the phone sends RTP/AVP then that means plain RTP, and RTP/AVP with a a=crypto attribute means SRTP is optional. -Steve On 7 February 2013 13:26, Levend Sayar > wrote: Below is the SDP offer sent by the phone. v=0 o=- 20185 20185 IN IP4 192.168.173.69 s=SDP data c=IN IP4 192.168.173.69 t=0 0 m=audio 11780 RTP/SAVP 0 8 18 9 101 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:MTM2MjVhMGI1NDZjYmRjADU5NWVjNGVkNTNlYzA1 a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:YmExYmZhNQAzN2ZjNDgzYTRkNGU2ZjFiN2Q0MmE3 a=crypto:3 F8_128_HMAC_SHA1_80 inline:N2Q2NTRiYQAxZjA3MWY3ZjI1YTI5NjIyM2FjODYw a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:9 G722/8000 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=ptime:20 a=sendrecv And below is the SDP answer sent by FS v=0 o=FreeSWITCH 1360230601 1360230602 IN IP4 192.168.169.114 s=FreeSWITCH c=IN IP4 192.168.169.114 t=0 0 m=audio 12532 RTP/SAVP 9 101 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:l8v0R64H7CP0vEx9j0Ycdbob8bgMCpLDppWGy7Dy -- ?yi ?al??malar. _lvnd_ {^_^} KAREL`UCAP On Thu, 2013-02-07 at 13:09 +0000, Steven Ayre wrote: What I mean is you'll see two separate m=audio lines within the callee's SDP, one for 'RTP/AVP' and one for 'SRTP/AVP'. If there is no m=audio line for RTP/AVP the caller won't know of a port that's expecting RTP. So if the callee only sends SRTP/AVP the caller can't send RTP. Can you show us the SDP being sent by the phone? -Steve On 7 February 2013 11:01, Levend Sayar > wrote: Thanx Steven. Caller makes the offer for SDP but callee chooses whatever it wants. So caller can offer SRTP but callee can prefer not to talk encrypted. In our case I want FS to choose non secure media. Phone will offer SRTP on the conference call but FS must prefer RTP, not SRTP. ?yi ?al??malar. _lvnd_ {^_^} KAREL`UCAP On 7 ?ub 2013, at 11:13, "Steven Ayre" > wrote: It's also going to rely on the phone actually offering RTP/AVP as well as SRTP/AVP in their SDP - without that there'd be nowhere to send insecure RTP. -Steve On 6 February 2013 16:09, Levend Sayar > wrote: Thanks Daniel for the reply. I tried But did not work. Upon your reply I also tried But did not work either. I am doing something wrong ? ?yi ?al??malar. _lvnd_ {^_^} KAREL`UCAP On 6 ?ub 2013, at 18:00, "Daniel Ivanov" > wrote: Of course you can. Just set the secure_media var to false and you will be srtp-free in sip. On Feb 5, 2013 6:06 PM, "Levend Sayar" > wrote: Hi all. I am using FS as a conference server. Some of my phones are using SRTP , some of them not. Both type of phone can join a conference. FS can talk to each peer with SRTP or not depending on the phone itself. My question: Is it possible to disable SRTP on FS ? I suppose if i can disable SRTP, FS will talk without SRTP with each phone whether they are using SRTP or not. TIA -- _lvnd_ {^_^} _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 plain text document attachment (ATT00001) _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 plain text document attachment (ATT00001) _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130214/618af82d/attachment-0001.html From manuel at aguete.org Thu Feb 14 16:26:42 2013 From: manuel at aguete.org (=?ISO-8859-1?Q?Manuel_Dur=E1n_Aguete?=) Date: Thu, 14 Feb 2013 14:26:42 +0100 Subject: [Freeswitch-users] Audio dropped using unimrcp tts. Message-ID: Hello, I'm a newbie testing freeswitch tts with loquendo mrcp1 server, mrcp_profile: Dialplan code: I can hear all text with first word dropped. I've captured the rtp traffic and the audio between FS <-> Loquendo contains the word. After apply the fixed recommended in http://wiki.freeswitch.org/wiki/RTP_Issues#Dropped_Audio the issue persists. Any idea, suggestion, etc ? Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130214/da78ba7c/attachment.html From a.venugopan at mundio.com Thu Feb 14 20:27:08 2013 From: a.venugopan at mundio.com (Archana Venugopan) Date: Thu, 14 Feb 2013 17:27:08 +0000 Subject: [Freeswitch-users] Multicast in freeswitch Message-ID: <592A9CF93E12394E8472A6CC66E66BF236B788@Mail-Kilo.squay.com> Hi, I would like to get 'Orig-status' event in freeswitch so that I will know when the phone is being HANGED. But to get that event I read in wiki I need mod_event_multicast. http://wiki.freeswitch.org/wiki/Mod_event_multicast But I don't see in above wiki on how to compile and load mod_event_multicast. Can anyone please let me know how to compile and load multicast? Regards, Archana -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130214/5b2ee03f/attachment.html From mike at jerris.com Thu Feb 14 21:35:50 2013 From: mike at jerris.com (Michael Jerris) Date: Thu, 14 Feb 2013 13:35:50 -0500 Subject: [Freeswitch-users] Freeswitch + freetdm dumps core on Ubuntu 12.10 In-Reply-To: <511CC262.2030906@julf.com> References: <511CA28F.4090109@julf.com> <511CC262.2030906@julf.com> Message-ID: <90802214-3959-4F51-A6D9-A069B04BC7B4@jerris.com> frame 0 print *new_module->module_interface->file_interface On Feb 14, 2013, at 5:54 AM, Johan Helsingius wrote: >> If not >> see http://wiki.freeswitch.org/wiki/Reporting_Bugs#Creating_A_Backtrace_With_gdb_.28Linux.2FUnix.29 >> to get a full backtrace, and debug-level logs would also be helpful. > > Here is the info: > > http://pastebin.freeswitch.org/20580 > > Julf From anthony.minessale at gmail.com Thu Feb 14 21:41:27 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 14 Feb 2013 12:41:27 -0600 Subject: [Freeswitch-users] Freeswitch + freetdm dumps core on Ubuntu 12.10 In-Reply-To: <90802214-3959-4F51-A6D9-A069B04BC7B4@jerris.com> References: <511CA28F.4090109@julf.com> <511CC262.2030906@julf.com> <90802214-3959-4F51-A6D9-A069B04BC7B4@jerris.com> Message-ID: One step closer to making a mailing list bot to do what Steven just did ;) On Thu, Feb 14, 2013 at 12:35 PM, Michael Jerris wrote: > frame 0 > print *new_module->module_interface->file_interface > > On Feb 14, 2013, at 5:54 AM, Johan Helsingius wrote: > > >> If not > >> see > http://wiki.freeswitch.org/wiki/Reporting_Bugs#Creating_A_Backtrace_With_gdb_.28Linux.2FUnix.29 > >> to get a full backtrace, and debug-level logs would also be helpful. > > > > Here is the info: > > > > http://pastebin.freeswitch.org/20580 > > > > Julf > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130214/f8520357/attachment.html From mike at jerris.com Thu Feb 14 21:41:56 2013 From: mike at jerris.com (Michael Jerris) Date: Thu, 14 Feb 2013 13:41:56 -0500 Subject: [Freeswitch-users] SRTP disabling In-Reply-To: References: Message-ID: <2E8666CC-8667-4500-A62A-7EDD90D17000@jerris.com> Why not just ask yealink to fix this? They have been pretty good about making fixes for us. Mike On Feb 14, 2013, at 10:03 AM, Levend Sayar wrote: > Thanks for your kind responses Steve. > > Phones are same brand, you are right. And not Polycom or Linksys. They are Yealink phones. > > I found the code piece handling the RTP/SAVP part you mentioned. (sofia_glue_negotiate_sdp function). > Although it is not wise, I will add a new channel variable and try to disable SRTP even if phone says SRTP is mandatory. > I need that. > > > On Wed, 2013-02-13 at 17:59 +0000, Steven Ayre wrote: >> If the phone is Polycom or Linksys have you read the notes on http://wiki.freeswitch.org/wiki/SRTP? >> >> On 13 February 2013 17:53, Steven Ayre wrote: >> Nevertheless it's not being negotiated and this is possibly a bug in the phone. Adding a workaround in FS would decrease security because the phone is explicitly saying it'll only accept SRTP so sending plain RTP wouldn't be wise. >> >> I assume both phones are the same make? >> >> On 7 February 2013 13:54, Levend Sayar wrote: >> But the very same phone calls another phone and talk with RTP, not SRTP if the peer does not accept SRTP >> >> Here is the SDP offer by the same phone >> >> v=0 >> o=- 20186 20186 IN IP4 192.168.173.69 >> >> s=SDP data >> c=IN IP4 192.168.173.69 >> t=0 0 >> >> m=audio 11782 RTP/SAVP 0 8 18 9 101 >> a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:NzFmYjdiMjk1OTY2ODQwYzExZjM0ZmE2NGM0YWMw >> a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:M2MxMTE2OWFjOGY2ZjEwADEzZmZkNzAxNjRlMzFm >> a=crypto:3 F8_128_HMAC_SHA1_80 inline:NjkzZDg2Mjk0ZTkxMjg1YzdmYjFiNjRlMmFhNGFm >> >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:18 G729/8000 >> a=fmtp:18 annexb=no >> a=rtpmap:9 G722/8000 >> a=fmtp:101 0-15 >> a=rtpmap:101 telephone-event/8000 >> a=ptime:20 >> a=sendrecv >> >> >> And here is the SDP answer sent by the other phone >> >> v=0 >> o=- 20029 20029 IN IP4 192.168.173.65 >> s=SDP data >> c=IN IP4 192.168.173.65 >> t=0 0 >> m=audio 11794 RTP/SAVP 0 101 >> a=rtpmap:0 PCMU/8000 >> a=sendrecv >> a=ptime:20 >> >> a=fmtp:101 0-15 >> a=rtpmap:101 telephone-event/8000 >> >> >> On Thu, 2013-02-07 at 13:37 +0000, Steven Ayre wrote: >>> m=audio 11780 RTP/SAVP 0 8 18 9 101 >>> >>> >>> RTP/SAVP means SRTP is mandatory. You need to reconfigure the phone. >>> >>> >>> If the phone sends RTP/AVP then that means plain RTP, and RTP/AVP with a a=crypto attribute means SRTP is optional. >>> >>> On 7 February 2013 13:26, Levend Sayar wrote: >>> Below is the SDP offer sent by the phone. >>> >>> v=0 >>> o=- 20185 20185 IN IP4 192.168.173.69 >>> s=SDP data >>> c=IN IP4 192.168.173.69 >>> t=0 0 >>> m=audio 11780 RTP/SAVP 0 8 18 9 101 >>> a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:MTM2MjVhMGI1NDZjYmRjADU5NWVjNGVkNTNlYzA1 >>> a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:YmExYmZhNQAzN2ZjNDgzYTRkNGU2ZjFiN2Q0MmE3 >>> a=crypto:3 F8_128_HMAC_SHA1_80 inline:N2Q2NTRiYQAxZjA3MWY3ZjI1YTI5NjIyM2FjODYw >>> a=rtpmap:0 PCMU/8000 >>> a=rtpmap:8 PCMA/8000 >>> a=rtpmap:18 G729/8000 >>> a=fmtp:18 annexb=no >>> a=rtpmap:9 G722/8000 >>> a=fmtp:101 0-15 >>> a=rtpmap:101 telephone-event/8000 >>> a=ptime:20 >>> a=sendrecv >>> >>> >>> >>> And below is the SDP answer sent by FS >>> >>> v=0 >>> o=FreeSWITCH 1360230601 1360230602 IN IP4 192.168.169.114 >>> s=FreeSWITCH >>> c=IN IP4 192.168.169.114 >>> t=0 0 >>> m=audio 12532 RTP/SAVP 9 101 >>> a=rtpmap:9 G722/8000 >>> a=rtpmap:101 telephone-event/8000 >>> a=fmtp:101 0-16 >>> a=silenceSupp:off - - - - >>> a=ptime:20 >>> a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:l8v0R64H7CP0vEx9j0Ycdbob8bgMCpLDppWGy7Dy >>> >>> On Thu, 2013-02-07 at 13:09 +0000, Steven Ayre wrote: >>>> What I mean is you'll see two separate m=audio lines within the callee's SDP, one for 'RTP/AVP' and one for 'SRTP/AVP'. If there is no m=audio line for RTP/AVP the caller won't know of a port that's expecting RTP. So if the callee only sends SRTP/AVP the caller can't send RTP. >>>> >>>> >>>> Can you show us the SDP being sent by the phone? >>>> >>>> >>>> >>>> On 7 February 2013 11:01, Levend Sayar wrote: >>>> Thanx Steven. >>>> >>>> >>>> Caller makes the offer for SDP but callee chooses whatever it wants. So caller can offer SRTP but callee can prefer not to talk encrypted. In our case I want FS to choose non secure media. >>>> Phone will offer SRTP on the conference call but FS must prefer RTP, not SRTP. >>>> >>>> On 7 ?ub 2013, at 11:13, "Steven Ayre" wrote: >>>> >>>> >>>>> It's also going to rely on the phone actually offering RTP/AVP as well as SRTP/AVP in their SDP - without that there'd be nowhere to send insecure RTP. >>>>> >>>>> >>>>> On 6 February 2013 16:09, Levend Sayar wrote: >>>>> Thanks Daniel for the reply. >>>>> >>>>> >>>>> I tried >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> But did not work. Upon your reply I also tried >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> But did not work either. I am doing something wrong ? >>>>> >>>>> On 6 ?ub 2013, at 18:00, "Daniel Ivanov" wrote: >>>>> >>>>> >>>>> >>>>>> Of course you can. Just set the secure_media var to false and you will be srtp-free in sip. >>>>>> >>>>>> On Feb 5, 2013 6:06 PM, "Levend Sayar" wrote: >>>>>> Hi all. >>>>>> >>>>>> I am using FS as a conference server. Some of my phones are using SRTP , some of them not. Both type of phone can >>>>>> join a conference. FS can talk to each peer with SRTP or not depending on the phone itself. >>>>>> >>>>>> My question: >>>>>> >>>>>> Is it possible to disable SRTP on FS ? >>>>>> >>>>>> I suppose if i can disable SRTP, FS will talk without SRTP with each phone whether they are using SRTP or not. >>>>>> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130214/78aaf910/attachment-0001.html From sherifomran2000 at yahoo.com Thu Feb 14 21:55:25 2013 From: sherifomran2000 at yahoo.com (Sherif Omran) Date: Thu, 14 Feb 2013 10:55:25 -0800 (PST) Subject: [Freeswitch-users] find sip phone ip In-Reply-To: <2E8666CC-8667-4500-A62A-7EDD90D17000@jerris.com> Message-ID: <1360868125.3129.YahooMailClassic@web141201.mail.bf1.yahoo.com> hello guys, I need to find the ip address of a sip phone registered to a freeswitch server any body knows how? thanks in advance regards, Sherif -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130214/55f2fd82/attachment.html From levend.sayar at karel.com.tr Thu Feb 14 21:56:53 2013 From: levend.sayar at karel.com.tr (Levend Sayar) Date: Thu, 14 Feb 2013 18:56:53 +0000 Subject: [Freeswitch-users] SRTP disabling In-Reply-To: <2E8666CC-8667-4500-A62A-7EDD90D17000@jerris.com> References: <2E8666CC-8667-4500-A62A-7EDD90D17000@jerris.com> Message-ID: In normal situations, I will do so. But right now I don't have chance to wait for them to make a fix. Besides, I already made the necessary fix on FS. I changed sdp parser. It ignores a=crypto attributes and set media protocol as RTP/AVP even if coming SDP offer is RTP/SAVP. I bind this behaviour to a new proprieratary channel attribute. It works just fine according to my smoke tests. -- _lvnd_ {^_^} On Thu, 2013-02-14 at 13:41 -0500, Michael Jerris wrote: Why not just ask yealink to fix this? They have been pretty good about making fixes for us. Mike On Feb 14, 2013, at 10:03 AM, Levend Sayar > wrote: Thanks for your kind responses Steve. Phones are same brand, you are right. And not Polycom or Linksys. They are Yealink phones. I found the code piece handling the RTP/SAVP part you mentioned. (sofia_glue_negotiate_sdp function). Although it is not wise, I will add a new channel variable and try to disable SRTP even if phone says SRTP is mandatory. I need that. On Wed, 2013-02-13 at 17:59 +0000, Steven Ayre wrote: If the phone is Polycom or Linksys have you read the notes on http://wiki.freeswitch.org/wiki/SRTP? On 13 February 2013 17:53, Steven Ayre > wrote: Nevertheless it's not being negotiated and this is possibly a bug in the phone. Adding a workaround in FS would decrease security because the phone is explicitly saying it'll only accept SRTP so sending plain RTP wouldn't be wise. I assume both phones are the same make? On 7 February 2013 13:54, Levend Sayar > wrote: But the very same phone calls another phone and talk with RTP, not SRTP if the peer does not accept SRTP Here is the SDP offer by the same phone v=0 o=- 20186 20186 IN IP4 192.168.173.69 s=SDP data c=IN IP4 192.168.173.69 t=0 0 m=audio 11782 RTP/SAVP 0 8 18 9 101 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:NzFmYjdiMjk1OTY2ODQwYzExZjM0ZmE2NGM0YWMw a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:M2MxMTE2OWFjOGY2ZjEwADEzZmZkNzAxNjRlMzFm a=crypto:3 F8_128_HMAC_SHA1_80 inline:NjkzZDg2Mjk0ZTkxMjg1YzdmYjFiNjRlMmFhNGFm a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:9 G722/8000 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=ptime:20 a=sendrecv And here is the SDP answer sent by the other phone v=0 o=- 20029 20029 IN IP4 192.168.173.65 s=SDP data c=IN IP4 192.168.173.65 t=0 0 m=audio 11794 RTP/SAVP 0 101 a=rtpmap:0 PCMU/8000 a=sendrecv a=ptime:20 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 On Thu, 2013-02-07 at 13:37 +0000, Steven Ayre wrote: m=audio 11780 RTP/SAVP 0 8 18 9 101 RTP/SAVP means SRTP is mandatory. You need to reconfigure the phone. If the phone sends RTP/AVP then that means plain RTP, and RTP/AVP with a a=crypto attribute means SRTP is optional. On 7 February 2013 13:26, Levend Sayar > wrote: Below is the SDP offer sent by the phone. v=0 o=- 20185 20185 IN IP4 192.168.173.69 s=SDP data c=IN IP4 192.168.173.69 t=0 0 m=audio 11780 RTP/SAVP 0 8 18 9 101 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:MTM2MjVhMGI1NDZjYmRjADU5NWVjNGVkNTNlYzA1 a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:YmExYmZhNQAzN2ZjNDgzYTRkNGU2ZjFiN2Q0MmE3 a=crypto:3 F8_128_HMAC_SHA1_80 inline:N2Q2NTRiYQAxZjA3MWY3ZjI1YTI5NjIyM2FjODYw a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:9 G722/8000 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=ptime:20 a=sendrecv And below is the SDP answer sent by FS v=0 o=FreeSWITCH 1360230601 1360230602 IN IP4 192.168.169.114 s=FreeSWITCH c=IN IP4 192.168.169.114 t=0 0 m=audio 12532 RTP/SAVP 9 101 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:l8v0R64H7CP0vEx9j0Ycdbob8bgMCpLDppWGy7Dy On Thu, 2013-02-07 at 13:09 +0000, Steven Ayre wrote: What I mean is you'll see two separate m=audio lines within the callee's SDP, one for 'RTP/AVP' and one for 'SRTP/AVP'. If there is no m=audio line for RTP/AVP the caller won't know of a port that's expecting RTP. So if the callee only sends SRTP/AVP the caller can't send RTP. Can you show us the SDP being sent by the phone? On 7 February 2013 11:01, Levend Sayar > wrote: Thanx Steven. Caller makes the offer for SDP but callee chooses whatever it wants. So caller can offer SRTP but callee can prefer not to talk encrypted. In our case I want FS to choose non secure media. Phone will offer SRTP on the conference call but FS must prefer RTP, not SRTP. On 7 ?ub 2013, at 11:13, "Steven Ayre" > wrote: It's also going to rely on the phone actually offering RTP/AVP as well as SRTP/AVP in their SDP - without that there'd be nowhere to send insecure RTP. On 6 February 2013 16:09, Levend Sayar > wrote: Thanks Daniel for the reply. I tried But did not work. Upon your reply I also tried But did not work either. I am doing something wrong ? On 6 ?ub 2013, at 18:00, "Daniel Ivanov" > wrote: Of course you can. Just set the secure_media var to false and you will be srtp-free in sip. On Feb 5, 2013 6:06 PM, "Levend Sayar" > wrote: Hi all. I am using FS as a conference server. Some of my phones are using SRTP , some of them not. Both type of phone can join a conference. FS can talk to each peer with SRTP or not depending on the phone itself. My question: Is it possible to disable SRTP on FS ? I suppose if i can disable SRTP, FS will talk without SRTP with each phone whether they are using SRTP or not. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130214/f38420e9/attachment.html From jnvines at gmail.com Thu Feb 14 22:08:14 2013 From: jnvines at gmail.com (Nick Vines) Date: Thu, 14 Feb 2013 11:08:14 -0800 Subject: [Freeswitch-users] find sip phone ip In-Reply-To: <1360868125.3129.YahooMailClassic@web141201.mail.bf1.yahoo.com> References: <2E8666CC-8667-4500-A62A-7EDD90D17000@jerris.com> <1360868125.3129.YahooMailClassic@web141201.mail.bf1.yahoo.com> Message-ID: >From the fs command line > sofia status profile PROFILE_NAME reg That will show you the registrations active, and you should be able to see the IP in the contact field. Nick On Thu, Feb 14, 2013 at 10:55 AM, Sherif Omran wrote: > hello guys, > > I need to find the ip address of a sip phone registered to a freeswitch > server > > any body knows how? > > thanks in advance > > regards, > Sherif > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130214/6d96594b/attachment-0001.html From julf at julf.com Thu Feb 14 22:45:28 2013 From: julf at julf.com (Johan Helsingius) Date: Thu, 14 Feb 2013 20:45:28 +0100 Subject: [Freeswitch-users] Freeswitch + freetdm dumps core on Ubuntu 12.10 In-Reply-To: <90802214-3959-4F51-A6D9-A069B04BC7B4@jerris.com> References: <511CA28F.4090109@julf.com> <511CC262.2030906@julf.com> <90802214-3959-4F51-A6D9-A069B04BC7B4@jerris.com> Message-ID: <511D3ED8.5080008@julf.com> On 14/02/13 19:35, Michael Jerris wrote: > frame 0 > print *new_module->module_interface->file_interface (gdb) frame 0 #0 0xb7e11d66 in switch_loadable_module_process (key=key at entry=0x8120eb0 "mod_file_string", new_module=new_module at entry=0x8295c08) at src/switch_loadable_module.c:321 321 for (i = 0; ptr->extens[i]; i++) { (gdb) print *new_module->module_interface->file_interface $1 = {interface_name = 0xb5417fd4 "mod_file_string", file_open = 0xb5417d00 , file_close = 0xb5417980 , file_truncate = 0x0, file_read = 0xb5417c30 , file_write = 0x0, file_read_video = 0xb54178b0 , file_write_video = 0x0, file_seek = 0x0, file_set_string = 0xb541a080, file_get_string = 0x0, extens = 0x0, rwlock = 0x8295ba0, refs = 0, reflock = 0x8295bd8, parent = 0x8295ab8, next = 0x0} From mike at jerris.com Thu Feb 14 22:59:40 2013 From: mike at jerris.com (Michael Jerris) Date: Thu, 14 Feb 2013 14:59:40 -0500 Subject: [Freeswitch-users] Freeswitch + freetdm dumps core on Ubuntu 12.10 In-Reply-To: <511D3ED8.5080008@julf.com> References: <511CA28F.4090109@julf.com> <511CC262.2030906@julf.com> <90802214-3959-4F51-A6D9-A069B04BC7B4@jerris.com> <511D3ED8.5080008@julf.com> Message-ID: mod_file_string was merged in to mod_dptools a while back, are you still trying to load the old .so module for it? On Feb 14, 2013, at 2:45 PM, Johan Helsingius wrote: > On 14/02/13 19:35, Michael Jerris wrote: >> frame 0 >> print *new_module->module_interface->file_interface > > (gdb) frame 0 > #0 0xb7e11d66 in switch_loadable_module_process (key=key at entry=0x8120eb0 > "mod_file_string", > new_module=new_module at entry=0x8295c08) at src/switch_loadable_module.c:321 > 321 for (i = 0; ptr->extens[i]; i++) { > > (gdb) print *new_module->module_interface->file_interface > $1 = {interface_name = 0xb5417fd4 "mod_file_string", > file_open = 0xb5417d00 , > file_close = 0xb5417980 , file_truncate = 0x0, > file_read = 0xb5417c30 , file_write = 0x0, > file_read_video = 0xb54178b0 , file_write_video = 0x0, > file_seek = 0x0, > file_set_string = 0xb541a080, file_get_string = 0x0, extens = 0x0, rwlock = > 0x8295ba0, refs = 0, > reflock = 0x8295bd8, parent = 0x8295ab8, next = 0x0} > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 benoit.raymond at amiconcept.com Thu Feb 14 23:04:27 2013 From: benoit.raymond at amiconcept.com (Benoit Raymond) Date: Thu, 14 Feb 2013 15:04:27 -0500 Subject: [Freeswitch-users] find sip phone ip In-Reply-To: References: <2E8666CC-8667-4500-A62A-7EDD90D17000@jerris.com> <1360868125.3129.YahooMailClassic@web141201.mail.bf1.yahoo.com> Message-ID: <09bc01ce0aee$7e5818c0$7b084a40$@amiconcept.com> If your phone is behind NAT, the Call-ID (from the command Nick provided you) will also give you the internal IP which is kinda neat. Also, you can include the phone number after the ?reg? to only show registration for this particular phone: > sofia status profile PROFILE_NAME reg PHONE_NUMBER Benoit De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Nick Vines Envoy? : 14 f?vrier 2013 14:08 ? : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] find sip phone ip >From the fs command line > sofia status profile PROFILE_NAME reg That will show you the registrations active, and you should be able to see the IP in the contact field. Nick On Thu, Feb 14, 2013 at 10:55 AM, Sherif Omran wrote: hello guys, I need to find the ip address of a sip phone registered to a freeswitch server any body knows how? thanks in advance regards, Sherif _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130214/587b9fcc/attachment.html From msc at freeswitch.org Thu Feb 14 23:39:54 2013 From: msc at freeswitch.org (Michael Collins) Date: Thu, 14 Feb 2013 12:39:54 -0800 Subject: [Freeswitch-users] Audio dropped using unimrcp tts. In-Reply-To: References: Message-ID: Try adding a sleep in your dialplan prior to the call to "speak" application. Start with 1000 ms and see if it makes any difference, then tweak the sleep time from there. -MC On Thu, Feb 14, 2013 at 5:26 AM, Manuel Dur?n Aguete wrote: > Hello, > > I'm a newbie testing freeswitch tts with loquendo mrcp1 server, > > mrcp_profile: > > > > > > > > > > > > > > > > > > > > > Dialplan code: > > > > > > > > > > > I can hear all text with first word dropped. I've captured the rtp traffic > and the audio between FS <-> Loquendo contains the word. After apply the > fixed recommended in > http://wiki.freeswitch.org/wiki/RTP_Issues#Dropped_Audio the issue > persists. > > Any idea, suggestion, etc ? > > Thank you. > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130214/31366656/attachment-0001.html From admin at blindi.net Fri Feb 15 01:08:59 2013 From: admin at blindi.net (Thomas Hoellriegel) Date: Thu, 14 Feb 2013 23:08:59 +0100 (CET) Subject: [Freeswitch-users] skype client 2.x on march not working? In-Reply-To: References: Message-ID: Hi all, i have a question. i have installed skype on freeswitch. Microsoft changed on march the servers. Is this a problem of old skype version under 4.x? Can i update the 2.X version to current version? What skype versions supported fs? Thanks. --------------- Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: http://www.blindi.net/callback homepage: http://www.blindi.net blinde-misc mailingliste f?r blinde. anmeldung unter: http://www.blindi.net/mailman/listinfo/blinde-misc From msc at freeswitch.org Fri Feb 15 01:41:57 2013 From: msc at freeswitch.org (Michael Collins) Date: Thu, 14 Feb 2013 14:41:57 -0800 Subject: [Freeswitch-users] FreeSWITCH HA BoF Meetings Message-ID: Hello! Yesterday we had a nice discussion about all things HA. Eliot Gable gave us a lot of very good information about the state of mod_ha_development and all the work that he's done over the past 4+ years in the world of HA and FreeSWITCH. It's now time to let those who have a vested interest in HA to get together with Eliot and continuing the discussion. To that end we propose a FreeSWITCH HA BoF (Birds of a Feather) conference. Eliot says that he can be available after 8PM EST. I would like to propose that the meetup be Tuesday evenings at 8:00PM EST in the main FreeSWITCH public conference. If you would like to meetup then please email me off-list and CC Eliot ( egable+freeswitch at gmail.com) so that we can keep track of who is most interested in this topic. Thanks! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130214/00107af7/attachment.html From brian at freeswitch.org Fri Feb 15 03:38:20 2013 From: brian at freeswitch.org (Brian West) Date: Thu, 14 Feb 2013 18:38:20 -0600 Subject: [Freeswitch-users] http://infiltrated.net Message-ID: <28D73E30-E2D0-4382-86A3-8BEDCA02B566@freeswitch.org> Does anyone know if these lists are updated still? -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire http://freeswitchcookbook.com http://freeswitchbook.com T: +1.213.286.0410 | F: +1.213.286.0401 | M: +1.918.424.WEST iNUM: +883 5100 1286 0410 From brian at freeswitch.org Fri Feb 15 03:40:35 2013 From: brian at freeswitch.org (Brian West) Date: Thu, 14 Feb 2013 18:40:35 -0600 Subject: [Freeswitch-users] ooH323 vs mod_h323 In-Reply-To: References: <2BD0DC60-D428-4323-A020-D9AD20E76553@freeswitch.org> <4C2EA249.4070704@coppice.org> <267C8398-6CA3-442E-B973-BE096D728B28@freeswitch.org> Message-ID: If someone wanted to build ooh323 module for FreeSWITCH we do have permission to do so using the lib. /b On Feb 14, 2013, at 2:08 AM, Steven Ayre wrote: > The q931cause parameter gets added in this revision: > http://svnview.digium.com/svn/asterisk?view=revision&revision=227898 > > Which as you can see is rather a large commit - they fix a lot of other > issues too. q931cause alone means changing quite a lot of code though - 11 > times in 6 different files, plus the modified functions that it appears > within. It means changing ooHangCall, the parameters for the command on the > stack, ooH323HangCall, storing q931cause on the call, and then setting it > during the actual hangup. > > -Steve > -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire http://freeswitchcookbook.com http://freeswitchbook.com T: +1.213.286.0410 | F: +1.213.286.0401 | M: +1.918.424.WEST iNUM: +883 5100 1286 0410 From freeswitch-list at puzzled.xs4all.nl Fri Feb 15 04:52:58 2013 From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists) Date: Fri, 15 Feb 2013 02:52:58 +0100 Subject: [Freeswitch-users] http://infiltrated.net In-Reply-To: <28D73E30-E2D0-4382-86A3-8BEDCA02B566@freeswitch.org> References: <28D73E30-E2D0-4382-86A3-8BEDCA02B566@freeswitch.org> Message-ID: <511D94FA.80509@puzzled.xs4all.nl> On 02/15/2013 01:38 AM, Brian West wrote: > Does anyone know if these lists are updated still? The date on the blacklist says Thu Feb 14 19:10:03 CST 2013: http://www.infiltrated.net/blacklisted Regards, Patrick From bdfoster at endigotech.com Fri Feb 15 05:30:22 2013 From: bdfoster at endigotech.com (Brian Foster) Date: Thu, 14 Feb 2013 21:30:22 -0500 Subject: [Freeswitch-users] http://infiltrated.net In-Reply-To: <511D94FA.80509@puzzled.xs4all.nl> References: <28D73E30-E2D0-4382-86A3-8BEDCA02B566@freeswitch.org> <511D94FA.80509@puzzled.xs4all.nl> Message-ID: <48B2B026-AA57-483B-BF41-40583A7E9056@endigotech.com> Looks like it's updated hourly. Sent from my iPhone On Feb 14, 2013, at 8:52 PM, Patrick Lists wrote: > On 02/15/2013 01:38 AM, Brian West wrote: >> Does anyone know if these lists are updated still? > > The date on the blacklist says Thu Feb 14 19:10:03 CST 2013: > > http://www.infiltrated.net/blacklisted > > Regards, > Patrick > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 brian at freeswitch.org Fri Feb 15 05:39:30 2013 From: brian at freeswitch.org (Brian West) Date: Thu, 14 Feb 2013 20:39:30 -0600 Subject: [Freeswitch-users] http://infiltrated.net In-Reply-To: <511D94FA.80509@puzzled.xs4all.nl> References: <28D73E30-E2D0-4382-86A3-8BEDCA02B566@freeswitch.org> <511D94FA.80509@puzzled.xs4all.nl> Message-ID: <5F2F78A9-78A0-42EC-A6F2-76FE8A6A66CA@freeswitch.org> 0.0.0.0 .99 .161.101.234 This is at the bottom of the list so that concerns me. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire T: +1.918.420.9266 | F: +1.918.420.9267 | M: +1.918.424.WEST iNUM: +883 5100 1420 9266 UK: +44 20 3298 4900 ISN: 410*543 On Feb 14, 2013, at 7:52 PM, Patrick Lists wrote: > On 02/15/2013 01:38 AM, Brian West wrote: >> Does anyone know if these lists are updated still? > > The date on the blacklist says Thu Feb 14 19:10:03 CST 2013: > > http://www.infiltrated.net/blacklisted > > Regards, > Patrick > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 curriegrad2004 at gmail.com Fri Feb 15 07:41:18 2013 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Thu, 14 Feb 2013 20:41:18 -0800 Subject: [Freeswitch-users] http://infiltrated.net In-Reply-To: <5F2F78A9-78A0-42EC-A6F2-76FE8A6A66CA@freeswitch.org> References: <28D73E30-E2D0-4382-86A3-8BEDCA02B566@freeswitch.org> <511D94FA.80509@puzzled.xs4all.nl> <5F2F78A9-78A0-42EC-A6F2-76FE8A6A66CA@freeswitch.org> Message-ID: How does it concern you Mr. West? On Thu, Feb 14, 2013 at 6:39 PM, Brian West wrote: > 0.0.0.0 > .99 > .161.101.234 > > This is at the bottom of the list so that concerns me. > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH_Wire > T: +1.918.420.9266 | F: +1.918.420.9267 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9266 > UK: +44 20 3298 4900 > ISN: 410*543 > > > > > > On Feb 14, 2013, at 7:52 PM, Patrick Lists wrote: > >> On 02/15/2013 01:38 AM, Brian West wrote: >>> Does anyone know if these lists are updated still? >> >> The date on the blacklist says Thu Feb 14 19:10:03 CST 2013: >> >> http://www.infiltrated.net/blacklisted >> >> Regards, >> Patrick >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 william.suffill at gmail.com Fri Feb 15 08:40:25 2013 From: william.suffill at gmail.com (William Suffill) Date: Fri, 15 Feb 2013 00:40:25 -0500 Subject: [Freeswitch-users] http://infiltrated.net In-Reply-To: References: <28D73E30-E2D0-4382-86A3-8BEDCA02B566@freeswitch.org> <511D94FA.80509@puzzled.xs4all.nl> <5F2F78A9-78A0-42EC-A6F2-76FE8A6A66CA@freeswitch.org> Message-ID: At one point it was suggested to pipe said rules into iptables firewall if I recall correctly so the theory would apply that all ips in that list should be publicly routable ipv4 ips. On Thu, Feb 14, 2013 at 11:41 PM, curriegrad2004 wrote: > How does it concern you Mr. West? > > On Thu, Feb 14, 2013 at 6:39 PM, Brian West wrote: > > 0.0.0.0 > > .99 > > .161.101.234 > > > > This is at the bottom of the list so that concerns me. > > -- > > Brian West > > brian at freeswitch.org > > FreeSWITCH Solutions, LLC > > PO BOX PO BOX 2531 > > Brookfield, WI 53008-2531 > > Twitter: @FreeSWITCH_Wire > > T: +1.918.420.9266 | F: +1.918.420.9267 | M: +1.918.424.WEST > > iNUM: +883 5100 1420 9266 > > UK: +44 20 3298 4900 > > ISN: 410*543 > > > > > > > > > > > > On Feb 14, 2013, at 7:52 PM, Patrick Lists < > freeswitch-list at puzzled.xs4all.nl> wrote: > > > >> On 02/15/2013 01:38 AM, Brian West wrote: > >>> Does anyone know if these lists are updated still? > >> > >> The date on the blacklist says Thu Feb 14 19:10:03 CST 2013: > >> > >> http://www.infiltrated.net/blacklisted > >> > >> Regards, > >> Patrick > >> > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> 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 > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130215/10e55160/attachment-0001.html From bdfoster at endigotech.com Fri Feb 15 08:45:40 2013 From: bdfoster at endigotech.com (Brian Foster) Date: Fri, 15 Feb 2013 00:45:40 -0500 Subject: [Freeswitch-users] http://infiltrated.net In-Reply-To: <5F2F78A9-78A0-42EC-A6F2-76FE8A6A66CA@freeswitch.org> References: <28D73E30-E2D0-4382-86A3-8BEDCA02B566@freeswitch.org> <511D94FA.80509@puzzled.xs4all.nl> <5F2F78A9-78A0-42EC-A6F2-76FE8A6A66CA@freeswitch.org> Message-ID: <443E6852-5485-482D-AC1E-195FDE71CCF5@endigotech.com> Yea doesn't seem accurate but you could run a script a you're fetching it to validate the IP and if it isn't valid or it's a local ip then throw it out. At least it will be closer to your target. Sent from my iPhone On Feb 14, 2013, at 9:39 PM, Brian West wrote: > 0.0.0.0 > .99 > .161.101.234 > > This is at the bottom of the list so that concerns me. > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH_Wire > T: +1.918.420.9266 | F: +1.918.420.9267 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9266 > UK: +44 20 3298 4900 > ISN: 410*543 > > > > > > On Feb 14, 2013, at 7:52 PM, Patrick Lists wrote: > >> On 02/15/2013 01:38 AM, Brian West wrote: >>> Does anyone know if these lists are updated still? >> >> The date on the blacklist says Thu Feb 14 19:10:03 CST 2013: >> >> http://www.infiltrated.net/blacklisted >> >> Regards, >> Patrick >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 avi at avimarcus.net Fri Feb 15 11:49:37 2013 From: avi at avimarcus.net (Avi Marcus) Date: Fri, 15 Feb 2013 10:49:37 +0200 Subject: [Freeswitch-users] http://infiltrated.net In-Reply-To: <443E6852-5485-482D-AC1E-195FDE71CCF5@endigotech.com> References: <28D73E30-E2D0-4382-86A3-8BEDCA02B566@freeswitch.org> <511D94FA.80509@puzzled.xs4all.nl> <5F2F78A9-78A0-42EC-A6F2-76FE8A6A66CA@freeswitch.org> <443E6852-5485-482D-AC1E-195FDE71CCF5@endigotech.com> Message-ID: Does someone have such a script that... 1) removes old entries 2) validates new list 3) adds new entries to iptables ..? I presume the community might appreciate it. -Avi On Fri, Feb 15, 2013 at 7:45 AM, Brian Foster wrote: > Yea doesn't seem accurate but you could run a script a you're fetching it > to validate the IP and if it isn't valid or it's a local ip then throw it > out. At least it will be closer to your target. > > Sent from my iPhone > > On Feb 14, 2013, at 9:39 PM, Brian West wrote: > > > 0.0.0.0 > > .99 > > .161.101.234 > > > > This is at the bottom of the list so that concerns me. > > -- > > Brian West > > brian at freeswitch.org > > FreeSWITCH Solutions, LLC > > PO BOX PO BOX 2531 > > Brookfield, WI 53008-2531 > > Twitter: @FreeSWITCH_Wire > > T: +1.918.420.9266 | F: +1.918.420.9267 | M: +1.918.424.WEST > > iNUM: +883 5100 1420 9266 > > UK: +44 20 3298 4900 > > ISN: 410*543 > > > > > > > > > > > > On Feb 14, 2013, at 7:52 PM, Patrick Lists < > freeswitch-list at puzzled.xs4all.nl> wrote: > > > >> On 02/15/2013 01:38 AM, Brian West wrote: > >>> Does anyone know if these lists are updated still? > >> > >> The date on the blacklist says Thu Feb 14 19:10:03 CST 2013: > >> > >> http://www.infiltrated.net/blacklisted > >> > >> Regards, > >> Patrick > >> > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> 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 > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130215/c9b178b4/attachment.html From julf at julf.com Fri Feb 15 12:32:49 2013 From: julf at julf.com (Johan Helsingius) Date: Fri, 15 Feb 2013 10:32:49 +0100 Subject: [Freeswitch-users] Freeswitch + freetdm dumps core on Ubuntu 12.10 In-Reply-To: References: <511CA28F.4090109@julf.com> <511CC262.2030906@julf.com> <90802214-3959-4F51-A6D9-A069B04BC7B4@jerris.com> <511D3ED8.5080008@julf.com> Message-ID: <511E00C1.40602@julf.com> > mod_file_string was merged in to mod_dptools a while back, > are you still trying to load the old .so module for it? Ah! Found some old config files hanging around from a previous version! That did it! Thanks! Julf From iam at onnet.su Fri Feb 15 12:33:42 2013 From: iam at onnet.su (Kirill Sysoev) Date: Fri, 15 Feb 2013 13:33:42 +0400 Subject: [Freeswitch-users] hangup error missing remote port Message-ID: <511E00F6.7090801@onnet.su> Hi Allan! Did you ever manage to solve this issue? I saw the same behavior yesterday while was trying to accept incoming calls from CS2K FreeSWITCH Version 1.3.13b+git~20130214T200725Z~b14fd4a5a4 (git b14fd4a 2013-02-14 20:07:25Z) Best regards, Kirill ######################################################################### *Allan Piske*zyryznet at gmail.com /Sun Sep 4 08:44:25 MSD 2011/ Hi, I'm having some trouble discovering the source and solution to one problem. I setup a FS box for receiving faxes, and a already have another FS box as SBC. SPA3102 ---> opensips ---> FS_SBC --> opensips --> FS_FAX this works ok, call gets answered, i hear the fax tone, the SPA3102 changes to T38 and everyone is happy. Nortel CS2000 ------> FS_SBC -------> opensips(regitrar,rtpproxy,etc) ---> FS_FAX but his doen't. This little devil CS2000 sends every invite with m=image header along with m=audio. With aparently FS doesn't like. like that: freeswitch at internal > recv 1215 bytes from udp/[10.143.82.250]:5060 at 08:26:08.226858: ------------------------------------------------------------------------ INVITE sip:4730365302 at 10.143.82.253 :5060;transport=UDP;user=phone SIP/2.0 Record-Route: f: :5060 ;user=phone>;tag=-45026-10d9cdb-2668a317-10d9cdb t: :5060;user=phone> i: 416cf3e01041960a13c410d9cdb1d2cba42b82409c10f14bb8-0322-4481 CSeq: 1 INVITE User-agent: CS2000_NGSS/9.0 P-Asserted-Identity: ;user=phone> Allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK,UPDATE Via: SIP/2.0/UDP 10.143.82.250;branch=z9hG4bK1d78.5262ccd1.0 v: SIP/2.0/UDP PAE1CS2K:5060;maddr=10.150.65.16;branch=z9hG4bK-10d9cdb-1d2cba43-57ebb21d Max-Forwards: 69 m: k: 100rel c: application/sdp l: 420 v=0 o=PVG 1315110636830 1315110636830 IN IP4 10.152.204.202 s=- p=+1 6135555555 c=IN IP4 10.152.204.202 t=0 0 m=audio 55920 RTP/AVP 18 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=fmtp:18 annexb=no m=image 64112 udptl t38 a=T38FaxVersion:0 a=T38FaxMaxBuffer:1100 a=T38FaxMaxDatagram:612 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxUdpEC:t38UDPRedundancy But then we have a problem ... My beloved FS_SBC changes the SDP before sending it to the next hop .. as it should becase they are on diferent subnets and set to proxy_media 2011-09-04 01:26:08.231102 [DEBUG] sofia_glue.c:1759 sofia/tpa/4730365302 Patched SDP --- v=0 o=PVG 1315110636830 1315110636830 IN IP4 10.152.204.202 s=- p=+1 6135555555 c=IN IP4 10.152.204.202 t=0 0 m=audio 55920 RTP/AVP 18 8 101 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 m=image 64112 udptl t38 a=T38FaxVersion:0 a=T38FaxMaxBuffer:1100 a=T38FaxMaxDatagram:612 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxUdpEC:t38UDPRedundancy +++ v=0 o=FreeSWITCH 2718891449 2718891450 IN IP4 A.B.C.D s=FreeSWITCH p=+1 6135555555 c=IN IP4 A.B.C.D t=0 0 m=audio 19262 RTP/AVP 18 8 101 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 m=image 19262 udptl t38 a=T38FaxVersion:0 a=T38FaxMaxBuffer:1100 a=T38FaxMaxDatagram:612 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxUdpEC:t38UDPRedundancy so we have now 2 m= fields with the same port !!!!! then when it gets answered on the FS_FAX box .. we got disconnection on the FS_SBC box 2011-09-04 01:26:08.257064 [DEBUG] sofia.c:4761 Channel sofia/tpa/4730365302 entering state [completing][200] 2011-09-04 01:26:08.257064 [DEBUG] sofia.c:4772 Remote SDP: v=0 o=FreeSWITCH 1315079202 1315079203 IN IP4 189.45.192.19 s=FreeSWITCH c=IN IP4 189.45.192.19 t=0 0 m=audio 31436 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 m=image 0 udptl 19 2011-09-04 01:26:08.258409 [DEBUG] sofia.c:4761 Channel sofia/tpa/4730365302 entering state [ready][200] 2011-09-04 01:26:08.258409 [DEBUG] switch_channel.c:2830 (sofia/tpa/4730365302) Callstate Change RINGING -> ACTIVE 2011-09-04 01:26:08.258409 [DEBUG] switch_channel.c:2842 Send signal sofia/voxip/4784064435 at 10.150.65.16 :5060 [BREAK] 2011-09-04 01:26:08.258409 [NOTICE] sofia.c:5318 Channel [sofia/tpa/4730365302] has been answered 2011-09-04 01:26:08.258409 [DEBUG] sofia_glue.c:2774 Set Codec sofia/tpa/4730365302 PROXY/8000 20 ms 160 samples 0 bits 2011-09-04 01:26:08.258409 [DEBUG] sofia_glue.c:3079 PROXY AUDIO RTP [sofia/tpa/4730365302] A.B.C.D:19262->W.X.Y.Z:0 codec: 0 ms: 20 2011-09-04 01:26:08.258409 [ERR] sofia_glue.c:3512 AUDIO RTP REPORTS ERROR: [Missing remote port] 2011-09-04 01:26:08.258409 [DEBUG] switch_channel.c:2563 (sofia/tpa/4730365302) Callstate Change ACTIVE -> HANGUP 2011-09-04 01:26:08.258409 [NOTICE] sofia_glue.c:3513 Hangup sofia/tpa/4730365302 [CS_CONSUME_MEDIA] [DESTINATION_OUT_OF_ORDER] 2011-09-04 01:26:08.258409 [DEBUG] switch_channel.c:2579 Send signal sofia/tpa/4730365302 [KILL] 2011-09-04 01:26:08.258409 [DEBUG] switch_core_session.c:1116 Send signal sofia/tpa/4730365302 [BREAK] 2011-09-04 01:26:08.259605 [DEBUG] switch_ivr_originate.c:3299 Originate Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER] 2011-09-04 01:26:08.259605 [INFO] mod_dptools.c:2647 Originate Failed. Cause: DESTINATION_OUT_OF_ORDER port = 0 !!! is that maybe a fault in opensips in the middle because the both m= fields with the same port ? , or one of the FS boxes ? something is broken here, sadly :( Appreciate any help. Allan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130215/a653d8f7/attachment-0001.html From steveayre at gmail.com Fri Feb 15 12:49:58 2013 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 15 Feb 2013 09:49:58 +0000 Subject: [Freeswitch-users] ooH323 vs mod_h323 In-Reply-To: References: <2BD0DC60-D428-4323-A020-D9AD20E76553@freeswitch.org> <4C2EA249.4070704@coppice.org> <267C8398-6CA3-442E-B973-BE096D728B28@freeswitch.org> Message-ID: Hi Brian, The issue would be that Asterisk's modifications are on a version predating the FLOSS exemption and their project is GPL - so presumably their modifications are all GPL-only? -Steve On 15 February 2013 00:40, Brian West wrote: > If someone wanted to build ooh323 module for FreeSWITCH we do have > permission to do so using the lib. > > /b > > On Feb 14, 2013, at 2:08 AM, Steven Ayre wrote: > > > The q931cause parameter gets added in this revision: > > http://svnview.digium.com/svn/asterisk?view=revision&revision=227898 > > > > Which as you can see is rather a large commit - they fix a lot of other > > issues too. q931cause alone means changing quite a lot of code though - > 11 > > times in 6 different files, plus the modified functions that it appears > > within. It means changing ooHangCall, the parameters for the command on > the > > stack, ooH323HangCall, storing q931cause on the call, and then setting it > > during the actual hangup. > > > > -Steve > > > > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX PO BOX 2531 > Brookfield, WI 53008-2531 > > Twitter: @FreeSWITCH_Wire > http://freeswitchcookbook.com > http://freeswitchbook.com > > T: +1.213.286.0410 | F: +1.213.286.0401 | M: +1.918.424.WEST > iNUM: +883 5100 1286 0410 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130215/f08208d5/attachment.html From a.venugopan at mundio.com Fri Feb 15 13:49:50 2013 From: a.venugopan at mundio.com (Archana Venugopan) Date: Fri, 15 Feb 2013 10:49:50 +0000 Subject: [Freeswitch-users] Event status Message-ID: <592A9CF93E12394E8472A6CC66E66BF236B89E@Mail-Kilo.squay.com> Hi, I am trying to get event status in freeswitch. For which I have passed the below line in my lua script. But I am getting error. element = e:getHeader("Orig-status") Error: 2013-02-15 10:41:14.177168 [ERR] mod_lua.cpp:198 /usr/local/freeswitch/scripts/directory.lua:201: attempt to index global 'e' (a nil value) stack traceback: /usr/local/freeswitch/scripts/directory.lua:201: in main chunk 2013-02-15 10:41:14.177168 [ERR] mod_lua.cpp:264 LUA script parse/execute error! I have installed mod_event_multicast too. Can anyone please point me out why I am not able to fetch element value. Many thanks. Regards, Archana -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130215/9d12d006/attachment.html From gmaruzz at gmail.com Fri Feb 15 14:28:50 2013 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Fri, 15 Feb 2013 12:28:50 +0100 Subject: [Freeswitch-users] skype client 2.x on march not working? In-Reply-To: References: Message-ID: Hello Thomas, just follow the wiki page strictly and you'll be good to go. http://wiki.freeswitch.org/wiki/Skypopen Use the installer.pl and the skype client provided by installer.pl Anything that is not on the wiki page, or that is done differently from the wiki page, is unsupported and will take much of your time. -giovanni On Thu, Feb 14, 2013 at 11:08 PM, Thomas Hoellriegel wrote: > Hi all, i have a question. > i have installed skype on freeswitch. > Microsoft changed on march the servers. > Is this a problem of old skype version under 4.x? > Can i update the 2.X version to current version? > What skype versions supported fs? > Thanks. > > > --------------- > Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: > http://www.blindi.net/callback > homepage: http://www.blindi.net > blinde-misc mailingliste f?r blinde. anmeldung unter: > http://www.blindi.net/mailman/**listinfo/blinde-misc > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130215/0487e7b1/attachment.html From steveayre at gmail.com Fri Feb 15 15:00:09 2013 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 15 Feb 2013 12:00:09 +0000 Subject: [Freeswitch-users] Event status In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF236B89E@Mail-Kilo.squay.com> References: <592A9CF93E12394E8472A6CC66E66BF236B89E@Mail-Kilo.squay.com> Message-ID: <4EE72A95-E1CB-44E4-A67A-42EF45C651DE@gmail.com> That error means there is no variable 'e' Steve On 15 Feb 2013, at 10:49, Archana Venugopan wrote: > Hi, > > I am trying to get event status in freeswitch. For which I have passed the below line in my lua script. But I am getting error. > > element = e:getHeader("Orig-status") > > Error: > 2013-02-15 10:41:14.177168 [ERR] mod_lua.cpp:198 /usr/local/freeswitch/scripts/directory.lua:201: attempt to index global 'e' (a nil value) > stack traceback: > /usr/local/freeswitch/scripts/directory.lua:201: in main chunk > 2013-02-15 10:41:14.177168 [ERR] mod_lua.cpp:264 LUA script parse/execute error! > > I have installed mod_event_multicast too. Can anyone please point me out why I am not able to fetch element value. Many thanks. > > Regards, > Archana > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130215/6c28ca2f/attachment.html From brian at freeswitch.org Fri Feb 15 17:46:35 2013 From: brian at freeswitch.org (Brian West) Date: Fri, 15 Feb 2013 08:46:35 -0600 Subject: [Freeswitch-users] http://infiltrated.net In-Reply-To: References: <28D73E30-E2D0-4382-86A3-8BEDCA02B566@freeswitch.org> <511D94FA.80509@puzzled.xs4all.nl> <5F2F78A9-78A0-42EC-A6F2-76FE8A6A66CA@freeswitch.org> Message-ID: <390F8718-0D40-4F26-86CB-558115D456C6@freeswitch.org> BLOCK ALL THE NETWORKS! 0.0.0.0 shouldn't be in that list. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire T: +1.918.420.9266 | F: +1.918.420.9267 | M: +1.918.424.WEST iNUM: +883 5100 1420 9266 UK: +44 20 3298 4900 ISN: 410*543 On Feb 14, 2013, at 10:41 PM, curriegrad2004 wrote: > How does it concern you Mr. West? > > On Thu, Feb 14, 2013 at 6:39 PM, Brian West wrote: >> 0.0.0.0 >> .99 >> .161.101.234 From brian at freeswitch.org Fri Feb 15 17:47:14 2013 From: brian at freeswitch.org (Brian West) Date: Fri, 15 Feb 2013 08:47:14 -0600 Subject: [Freeswitch-users] ooH323 vs mod_h323 In-Reply-To: References: <2BD0DC60-D428-4323-A020-D9AD20E76553@freeswitch.org> <4C2EA249.4070704@coppice.org> <267C8398-6CA3-442E-B973-BE096D728B28@freeswitch.org> Message-ID: I'll see what I can do here. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire T: +1.918.420.9266 | F: +1.918.420.9267 | M: +1.918.424.WEST iNUM: +883 5100 1420 9266 UK: +44 20 3298 4900 ISN: 410*543 On Feb 15, 2013, at 3:49 AM, Steven Ayre wrote: > Hi Brian, > > The issue would be that Asterisk's modifications are on a version predating the FLOSS exemption and their project is GPL - so presumably their modifications are all GPL-only? > > -Steve > > > > On 15 February 2013 00:40, Brian West wrote: > If someone wanted to build ooh323 module for FreeSWITCH we do have permission to do so using the lib. > > /b From brian at freeswitch.org Fri Feb 15 17:48:26 2013 From: brian at freeswitch.org (Brian West) Date: Fri, 15 Feb 2013 08:48:26 -0600 Subject: [Freeswitch-users] ooH323 vs mod_h323 In-Reply-To: References: <2BD0DC60-D428-4323-A020-D9AD20E76553@freeswitch.org> <4C2EA249.4070704@coppice.org> <267C8398-6CA3-442E-B973-BE096D728B28@freeswitch.org> Message-ID: So I have the impression you or someone has an ooh323 module and hasn't contributed it back to the project yet? -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire T: +1.918.420.9266 | F: +1.918.420.9267 | M: +1.918.424.WEST iNUM: +883 5100 1420 9266 UK: +44 20 3298 4900 ISN: 410*543 On Feb 15, 2013, at 3:49 AM, Steven Ayre wrote: > Hi Brian, > > The issue would be that Asterisk's modifications are on a version predating the FLOSS exemption and their project is GPL - so presumably their modifications are all GPL-only? > > -Steve > > > > On 15 February 2013 00:40, Brian West wrote: > If someone wanted to build ooh323 module for FreeSWITCH we do have permission to do so using the lib. > > /b > From yyu0378 at gmail.com Fri Feb 15 12:25:19 2013 From: yyu0378 at gmail.com (yyu0378) Date: Fri, 15 Feb 2013 18:25:19 +0900 Subject: [Freeswitch-users] About timer X fired? Message-ID: <511DFEFF.9040903@gmail.com> hi: My fs send call from one gatway to another,but in cli has many like timer X fired.If this is my seesion-timeout is wrong.The seesion-timeout defaults is 30 minutes.Other gateways is one hour.Did i change it?Thank you !! From manuel at aguete.org Fri Feb 15 12:20:21 2013 From: manuel at aguete.org (=?ISO-8859-1?Q?Manuel_Dur=E1n_Aguete?=) Date: Fri, 15 Feb 2013 10:20:21 +0100 Subject: [Freeswitch-users] Audio dropped using unimrcp tts. In-Reply-To: References: Message-ID: Hello, I've tried with different values for sleep, but the problem persists, in http://pastebin.freeswitch.org/20594 i've posted the call log. I've enabled MOD_UNIMRCP_DEBUG_AUDIO_QUEUE in mod_unircp.cm the audio streams generated can be downloaded from https://www.dropbox.com/sh/tcgnktzzby0h2ns/v7ViiUyObB. I've changed the speak data to and only "3 4 5 6 7 8 9 0" is played in softphone. Thank you for your help. Regards. 2013/2/14 Michael Collins > Try adding a sleep in your dialplan prior to the call to "speak" > application. Start with 1000 ms and see if it makes any difference, then > tweak the sleep time from there. > > -MC > > On Thu, Feb 14, 2013 at 5:26 AM, Manuel Dur?n Aguete wrote: > >> Hello, >> >> I'm a newbie testing freeswitch tts with loquendo mrcp1 server, >> >> mrcp_profile: >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> Dialplan code: >> >> >> >> >> >> >> >> >> >> >> I can hear all text with first word dropped. I've captured the rtp >> traffic and the audio between FS <-> Loquendo contains the word. After >> apply the fixed recommended in >> http://wiki.freeswitch.org/wiki/RTP_Issues#Dropped_Audio the issue >> persists. >> >> Any idea, suggestion, etc ? >> >> Thank you. >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130215/c2e01e34/attachment.html From khorsmann at gmail.com Fri Feb 15 14:26:11 2013 From: khorsmann at gmail.com (Karsten Horsmann) Date: Fri, 15 Feb 2013 12:26:11 +0100 Subject: [Freeswitch-users] originate to loopback/answer and park a lua script from another lua script Message-ID: Hello! as described in Anthonys mail http://lists.freeswitch.org/pipermail/freeswitch-users/2013-January/091769.html i want also to originate my loop.lua to an loopback channel. No problem from fs_cli. I want to originate the loop.lua from my mod_lua start lua script. How can i do this? The freeswitch.Session() didnt work as aspected. -- Kind Regards *Karsten Horsmann* From jayesh1017 at gmail.com Fri Feb 15 18:04:40 2013 From: jayesh1017 at gmail.com (Jayesh Nambiar) Date: Fri, 15 Feb 2013 20:34:40 +0530 Subject: [Freeswitch-users] post record processing for mod_callcenter module Message-ID: Hi All, I have been using this variable record_post_process_exec_app in my dialplan for post-processing of recordings where I actually convert the WAV files into an MP3 using Lame. I earlier used to record it in mp3 directly using the mod_shout module, but it was very CPU extensive. I wish to do the same for calls that are recorded using the mod_callcenter module. In callcenter groups, I can define record_template, but I dont find a place where I can do post record processing. One way I figured to do this, was in the dialplan do this: The above works but the challenge is when you want the agent name in the recording filename; because there is no way the agent name is known while we are in the dialplan. Basically the record template in groups is defined as follows: Can we also have a record-post-parameter config parameter for callcenter module calls. Thanks for any suggestions in getting this done using a different approach !! --- Jayesh -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130215/06a820c6/attachment-0001.html From bpriddy at bryantschools.org Fri Feb 15 18:16:32 2013 From: bpriddy at bryantschools.org (Blake Priddy) Date: Fri, 15 Feb 2013 09:16:32 -0600 Subject: [Freeswitch-users] HELP Message-ID: hey guys so we just got in our Adtran 908e and I was wondering if someone could assist me on sending calls to freeswitch from this? -- *Blakelund Priddy* Network Systems Engineer Bryant Public School District Bryant, Arkansas 72022 http://www.bryantschools.org p 501-653-5038 f 501-847-5656 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130215/01fdfeba/attachment.html From steveayre at gmail.com Fri Feb 15 18:37:09 2013 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 15 Feb 2013 15:37:09 +0000 Subject: [Freeswitch-users] ooH323 vs mod_h323 In-Reply-To: References: <2BD0DC60-D428-4323-A020-D9AD20E76553@freeswitch.org> <4C2EA249.4070704@coppice.org> <267C8398-6CA3-442E-B973-BE096D728B28@freeswitch.org> Message-ID: There's a module for Asterisk but none for FS I'm aware of yet - I'm only wondering about the feasibility of writing one. -Steve On 15 February 2013 14:48, Brian West wrote: > So I have the impression you or someone has an ooh323 module and hasn't > contributed it back to the project yet? > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH_Wire > T: +1.918.420.9266 | F: +1.918.420.9267 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9266 > UK: +44 20 3298 4900 > ISN: 410*543 > > > > > > On Feb 15, 2013, at 3:49 AM, Steven Ayre wrote: > > > Hi Brian, > > > > The issue would be that Asterisk's modifications are on a version > predating the FLOSS exemption and their project is GPL - so presumably > their modifications are all GPL-only? > > > > -Steve > > > > > > > > On 15 February 2013 00:40, Brian West wrote: > > If someone wanted to build ooh323 module for FreeSWITCH we do have > permission to do so using the lib. > > > > /b > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130215/a73af2b3/attachment.html From a.venugopan at mundio.com Fri Feb 15 19:14:40 2013 From: a.venugopan at mundio.com (Archana Venugopan) Date: Fri, 15 Feb 2013 16:14:40 +0000 Subject: [Freeswitch-users] In-band and out band call issues Message-ID: <592A9CF93E12394E8472A6CC66E66BF236C16D@Mail-Kilo.squay.com> Hi, Am facing with a typical issue. In my phone settings, if i enable 'Force RFC2833 Out-of-Band DTMF' then my voicemail is not working properly( If I record any greeting or VM, only part of the message is being recorded) but offnet calls for banks or shops is working. If 'Force RFC2833 Out-of-Band DTMF' is disabled my voicemail is working properly but When I call offnet numbers like Banks or shops, then my key press is not being recognised i.e no DTMF detection. My internal .xml and external settings for DTMF is Please let me know what needs to be changed so that both voicemail and offnet numbers work properly. Regards, Archana -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130215/d64e6342/attachment.html From steveayre at gmail.com Fri Feb 15 20:21:33 2013 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 15 Feb 2013 17:21:33 +0000 Subject: [Freeswitch-users] http://infiltrated.net In-Reply-To: <390F8718-0D40-4F26-86CB-558115D456C6@freeswitch.org> References: <28D73E30-E2D0-4382-86A3-8BEDCA02B566@freeswitch.org> <511D94FA.80509@puzzled.xs4all.nl> <5F2F78A9-78A0-42EC-A6F2-76FE8A6A66CA@freeswitch.org> <390F8718-0D40-4F26-86CB-558115D456C6@freeswitch.org> Message-ID: But your box would be very secure ;o) -Steve On 15 February 2013 14:46, Brian West wrote: > BLOCK ALL THE NETWORKS! 0.0.0.0 shouldn't be in that list. > > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH_Wire > T: +1.918.420.9266 | F: +1.918.420.9267 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9266 > UK: +44 20 3298 4900 > ISN: 410*543 > > > > > > On Feb 14, 2013, at 10:41 PM, curriegrad2004 > wrote: > > > How does it concern you Mr. West? > > > > On Thu, Feb 14, 2013 at 6:39 PM, Brian West > wrote: > >> 0.0.0.0 > >> .99 > >> .161.101.234 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130215/0e528b4a/attachment.html From bdfoster at endigotech.com Fri Feb 15 20:40:08 2013 From: bdfoster at endigotech.com (Brian Foster) Date: Fri, 15 Feb 2013 12:40:08 -0500 Subject: [Freeswitch-users] http://infiltrated.net In-Reply-To: References: <28D73E30-E2D0-4382-86A3-8BEDCA02B566@freeswitch.org> <511D94FA.80509@puzzled.xs4all.nl> <5F2F78A9-78A0-42EC-A6F2-76FE8A6A66CA@freeswitch.org> <443E6852-5485-482D-AC1E-195FDE71CCF5@endigotech.com> Message-ID: <3E86D336-A7EE-488C-996B-641299A5EF06@endigotech.com> It wouldn't be hard to create a bash script to do this Sent from my iPhone On Feb 15, 2013, at 3:49 AM, Avi Marcus wrote: > Does someone have such a script that... > 1) removes old entries > 2) validates new list > 3) adds new entries to iptables > > ..? I presume the community might appreciate it. > > -Avi > > On Fri, Feb 15, 2013 at 7:45 AM, Brian Foster wrote: >> Yea doesn't seem accurate but you could run a script a you're fetching it to validate the IP and if it isn't valid or it's a local ip then throw it out. At least it will be closer to your target. >> >> Sent from my iPhone >> >> On Feb 14, 2013, at 9:39 PM, Brian West wrote: >> >> > 0.0.0.0 >> > .99 >> > .161.101.234 >> > >> > This is at the bottom of the list so that concerns me. >> > -- >> > Brian West >> > brian at freeswitch.org >> > FreeSWITCH Solutions, LLC >> > PO BOX PO BOX 2531 >> > Brookfield, WI 53008-2531 >> > Twitter: @FreeSWITCH_Wire >> > T: +1.918.420.9266 | F: +1.918.420.9267 | M: +1.918.424.WEST >> > iNUM: +883 5100 1420 9266 >> > UK: +44 20 3298 4900 >> > ISN: 410*543 >> > >> > >> > >> > >> > >> > On Feb 14, 2013, at 7:52 PM, Patrick Lists wrote: >> > >> >> On 02/15/2013 01:38 AM, Brian West wrote: >> >>> Does anyone know if these lists are updated still? >> >> >> >> The date on the blacklist says Thu Feb 14 19:10:03 CST 2013: >> >> >> >> http://www.infiltrated.net/blacklisted >> >> >> >> Regards, >> >> Patrick >> >> >> >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://wiki.freeswitch.org >> >> http://www.cluecon.com >> >> >> >> 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 >> > >> > >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > 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 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130215/18b26e7a/attachment-0001.html From krice at freeswitch.org Fri Feb 15 21:00:49 2013 From: krice at freeswitch.org (Ken Rice) Date: Fri, 15 Feb 2013 12:00:49 -0600 Subject: [Freeswitch-users] =?iso-8859-1?q?Hey_Guys=2C_Don=B9t_Forget_toda?= =?iso-8859-1?q?y_is_the_Friday_Free_For_All_on_888_this_afternoon=2E=2E?= =?iso-8859-1?q?=2E=2E?= Message-ID: Joins us for some Fun at sip:888 at conference.freeswitch.org or use one of the normal PSTN access numbers Todays Topic: Killer Asteroids! (or whatever else we feel like talking about) When? This afternoon... Targeted start time is 3P EST, but join us whenever you feel like it...lets get it started and see if we can keep it going all afternoon Who should join? EVERYONE... Get in here... -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130215/f0cc609b/attachment.html From cmrienzo at gmail.com Fri Feb 15 21:02:32 2013 From: cmrienzo at gmail.com (Christopher Rienzo) Date: Fri, 15 Feb 2013 13:02:32 -0500 Subject: [Freeswitch-users] Audio dropped using unimrcp tts. In-Reply-To: References: Message-ID: Send PCAP with the RTSP signaling and RTP audio. I'm curious when the server starts streaming the audio. On Fri, Feb 15, 2013 at 4:20 AM, Manuel Dur?n Aguete wrote: > Hello, > > I've tried with different values for sleep, but the problem persists, in > http://pastebin.freeswitch.org/20594 i've posted the call log. > I've enabled MOD_UNIMRCP_DEBUG_AUDIO_QUEUE in mod_unircp.cm the audio > streams generated can be downloaded from > https://www.dropbox.com/sh/tcgnktzzby0h2ns/v7ViiUyObB. > > I've changed the speak data to > > > > > > > > > > > > > > > and only "3 4 5 6 7 8 9 0" is played in softphone. > > Thank you for your help. > > Regards. > > > > > > 2013/2/14 Michael Collins > >> Try adding a sleep in your dialplan prior to the call to "speak" >> application. Start with 1000 ms and see if it makes any difference, then >> tweak the sleep time from there. >> >> -MC >> >> On Thu, Feb 14, 2013 at 5:26 AM, Manuel Dur?n Aguete wrote: >> >>> Hello, >>> >>> I'm a newbie testing freeswitch tts with loquendo mrcp1 server, >>> >>> mrcp_profile: >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> Dialplan code: >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> I can hear all text with first word dropped. I've captured the rtp >>> traffic and the audio between FS <-> Loquendo contains the word. After >>> apply the fixed recommended in >>> http://wiki.freeswitch.org/wiki/RTP_Issues#Dropped_Audio the issue >>> persists. >>> >>> Any idea, suggestion, etc ? >>> >>> Thank you. >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130215/70dd1940/attachment.html From jack at livecall.com Fri Feb 15 21:46:31 2013 From: jack at livecall.com (Jack) Date: Fri, 15 Feb 2013 10:46:31 -0800 Subject: [Freeswitch-users] WebRTC to SIP Message-ID: <511E8287.8040501@livecall.com> Has anyone been successful in connecting to freeswitch with this? http://kapejod.org/webrtc/sip-audio-demo.html I am able to register on freeswitch and place a call to the webpage, which rings, but no audio. This is for Canary Beta browser only right now. Jack From msc at freeswitch.org Fri Feb 15 21:49:23 2013 From: msc at freeswitch.org (Michael Collins) Date: Fri, 15 Feb 2013 10:49:23 -0800 Subject: [Freeswitch-users] HELP In-Reply-To: References: Message-ID: Is that PRI on one side and SIP on the other? -MC On Fri, Feb 15, 2013 at 7:16 AM, Blake Priddy wrote: > hey guys so we just got in our Adtran 908e and I was wondering if someone > could assist me on sending calls to freeswitch from this? > > -- > > *Blakelund Priddy* > Network Systems Engineer > Bryant Public School District > Bryant, Arkansas 72022 > http://www.bryantschools.org > p 501-653-5038 > f 501-847-5656 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130215/17ee6bd2/attachment-0001.html From bpriddy at bryantschools.org Fri Feb 15 22:09:43 2013 From: bpriddy at bryantschools.org (Blake Priddy) Date: Fri, 15 Feb 2013 13:09:43 -0600 Subject: [Freeswitch-users] HELP In-Reply-To: References: Message-ID: They are in the same network. On Fri, Feb 15, 2013 at 12:49 PM, Michael Collins wrote: > Is that PRI on one side and SIP on the other? > -MC > > On Fri, Feb 15, 2013 at 7:16 AM, Blake Priddy wrote: > >> hey guys so we just got in our Adtran 908e and I was wondering if someone >> could assist me on sending calls to freeswitch from this? >> >> -- >> >> *Blakelund Priddy* >> Network Systems Engineer >> Bryant Public School District >> Bryant, Arkansas 72022 >> http://www.bryantschools.org >> p 501-653-5038 >> f 501-847-5656 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- *Blakelund Priddy* Network Systems Engineer Bryant Public School District Bryant, Arkansas 72022 http://www.bryantschools.org p 501-653-5038 f 501-847-5656 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130215/eec0e09a/attachment.html From msc at freeswitch.org Fri Feb 15 22:14:22 2013 From: msc at freeswitch.org (Michael Collins) Date: Fri, 15 Feb 2013 11:14:22 -0800 Subject: [Freeswitch-users] originate to loopback/answer and park a lua script from another lua script In-Reply-To: References: Message-ID: Can you share with us a small Lua code sample demonstrating what you are trying to do? That might make it easier for us to help. -MC On Fri, Feb 15, 2013 at 3:26 AM, Karsten Horsmann wrote: > Hello! > > as described in Anthonys mail > > http://lists.freeswitch.org/pipermail/freeswitch-users/2013-January/091769.html > i want also to originate my loop.lua to an loopback channel. > > No problem from fs_cli. I want to originate the loop.lua from my > mod_lua start lua script. > > How can i do this? The freeswitch.Session() didnt work as aspected. > > > -- > Kind Regards > *Karsten Horsmann* > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130215/0a1a268e/attachment.html From msc at freeswitch.org Fri Feb 15 22:28:13 2013 From: msc at freeswitch.org (Michael Collins) Date: Fri, 15 Feb 2013 11:28:13 -0800 Subject: [Freeswitch-users] HELP In-Reply-To: References: Message-ID: I'm not familiar with the Adtran 908e, but a quick goog search says: The Total Access 908e IP Business Gateway provides SIP-gateway functionality, a robust IP router, firewall and VPN functionality, and support for a number of analog and digital interfaces for existing phone equipment, or for the combination of an IP PBX and an analog fax, phones, or modem. - Four T1/dual Ethernet platform, 8 analog FXS interfaces, integrated DSX-1 - Ideal for networks transitioning to VoIP and an all IP voice and data WAN So can you draw a picture of what's plugged in to what? I'm interested in seeing this work because we've had some CudaTel customer who have Adtran's from a carrier and the PRI's have had challenges, particularly with faxing. If you can get yours working like a champ then we'll all be happy. :) -Michael On Fri, Feb 15, 2013 at 11:09 AM, Blake Priddy wrote: > They are in the same network. > > > On Fri, Feb 15, 2013 at 12:49 PM, Michael Collins wrote: > >> Is that PRI on one side and SIP on the other? >> -MC >> >> On Fri, Feb 15, 2013 at 7:16 AM, Blake Priddy wrote: >> >>> hey guys so we just got in our Adtran 908e and I was wondering if >>> someone could assist me on sending calls to freeswitch from this? >>> >>> -- >>> >>> *Blakelund Priddy* >>> Network Systems Engineer >>> Bryant Public School District >>> Bryant, Arkansas 72022 >>> http://www.bryantschools.org >>> p 501-653-5038 >>> f 501-847-5656 >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > > -- > > *Blakelund Priddy* > Network Systems Engineer > Bryant Public School District > Bryant, Arkansas 72022 > http://www.bryantschools.org > p 501-653-5038 > f 501-847-5656 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130215/ea2097ae/attachment-0001.html From msc at freeswitch.org Fri Feb 15 22:34:50 2013 From: msc at freeswitch.org (Michael Collins) Date: Fri, 15 Feb 2013 11:34:50 -0800 Subject: [Freeswitch-users] Door intercom/gate controller In-Reply-To: References: Message-ID: Can you show us the dialplan for sending a call to the gate controller? Also, pastebin the console debug output for a call to the gate controller where you press digits but the controller doesn't respond. Thanks, MC On Wed, Feb 13, 2013 at 2:42 AM, Mr Nathan Downes wrote: > Hi list,**** > > ** ** > > I have a gate controller that only understands RFC 2833 but it calls > endpoints that can only provide inband DTMF, I can?t seem to get the digits > recognised by the gate controller to trigger the relay using > start_dtmf_generate, as they are coming back to the a-leg? Rather than out > the b-leg.. It works fine when I call a SPA502g as it will provide RFC > 2833.. is there some trickery I can achieve this? Or am I just doing it > wrong?? J**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130215/c9ac3f0c/attachment.html From jpyle at fidelityvoice.com Fri Feb 15 22:38:07 2013 From: jpyle at fidelityvoice.com (Jeff Pyle) Date: Fri, 15 Feb 2013 14:38:07 -0500 Subject: [Freeswitch-users] which sofia gateway received the call Message-ID: Hello, Is there a variable name available in the dialplan that will contain the name of the gateway that received the call? I see "variable_sip_gateway_name" and "variable_sip_gateway" on the Channel Variables page under Info App Variable Names. In my case both (without variable_) are empty. - Jeff -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130215/9fd980a8/attachment.html From jpyle at fidelityvoice.com Fri Feb 15 22:40:46 2013 From: jpyle at fidelityvoice.com (Jeff Pyle) Date: Fri, 15 Feb 2013 14:40:46 -0500 Subject: [Freeswitch-users] HELP In-Reply-To: References: Message-ID: Blake, We use the TA900-series with quite a bit of success. Tell me more about the PRI (DNIS digits, switch type) and I should be able to help you with a config template. - Jeff On Fri, Feb 15, 2013 at 2:28 PM, Michael Collins wrote: > I'm not familiar with the Adtran 908e, but a quick goog search says: > > The Total Access 908e IP Business Gateway provides SIP-gateway > functionality, a robust IP router, firewall and VPN functionality, and > support for a number of analog and digital interfaces for existing phone > equipment, or for the combination of an IP PBX and an analog fax, phones, > or modem. > > - Four T1/dual Ethernet platform, 8 analog FXS interfaces, integrated > DSX-1 > - Ideal for networks transitioning to VoIP and an all IP voice and > data WAN > > > So can you draw a picture of what's plugged in to what? I'm interested in > seeing this work because we've had some CudaTel customer who have Adtran's > from a carrier and the PRI's have had challenges, particularly with faxing. > If you can get yours working like a champ then we'll all be happy. :) > > -Michael > > > On Fri, Feb 15, 2013 at 11:09 AM, Blake Priddy wrote: > >> They are in the same network. >> >> >> On Fri, Feb 15, 2013 at 12:49 PM, Michael Collins wrote: >> >>> Is that PRI on one side and SIP on the other? >>> -MC >>> >>> On Fri, Feb 15, 2013 at 7:16 AM, Blake Priddy >> > wrote: >>> >>>> hey guys so we just got in our Adtran 908e and I was wondering if >>>> someone could assist me on sending calls to freeswitch from this? >>>> >>>> -- >>>> >>>> *Blakelund Priddy* >>>> Network Systems Engineer >>>> Bryant Public School District >>>> Bryant, Arkansas 72022 >>>> http://www.bryantschools.org >>>> p 501-653-5038 >>>> f 501-847-5656 >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>>> >>>> >>> >>> >>> -- >>> Michael S Collins >>> Twitter: @mercutioviz >>> http://www.FreeSWITCH.org >>> http://www.ClueCon.com >>> http://www.OSTAG.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> >> -- >> >> *Blakelund Priddy* >> Network Systems Engineer >> Bryant Public School District >> Bryant, Arkansas 72022 >> http://www.bryantschools.org >> p 501-653-5038 >> f 501-847-5656 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130215/678b47e1/attachment-0001.html From bpriddy at bryantschools.org Fri Feb 15 22:55:49 2013 From: bpriddy at bryantschools.org (Blake Priddy) Date: Fri, 15 Feb 2013 13:55:49 -0600 Subject: [Freeswitch-users] HELP In-Reply-To: References: Message-ID: ATT provides our PRI. And I want to send 5 digits to the PBX and also I have a fax server i need to send 5 digits too. On Fri, Feb 15, 2013 at 1:40 PM, Jeff Pyle wrote: > Blake, > > We use the TA900-series with quite a bit of success. Tell me more about > the PRI (DNIS digits, switch type) and I should be able to help you with a > config template. > > > - Jeff > > > > On Fri, Feb 15, 2013 at 2:28 PM, Michael Collins wrote: > >> I'm not familiar with the Adtran 908e, but a quick goog search says: >> >> The Total Access 908e IP Business Gateway provides SIP-gateway >> functionality, a robust IP router, firewall and VPN functionality, and >> support for a number of analog and digital interfaces for existing phone >> equipment, or for the combination of an IP PBX and an analog fax, phones, >> or modem. >> >> - Four T1/dual Ethernet platform, 8 analog FXS interfaces, integrated >> DSX-1 >> - Ideal for networks transitioning to VoIP and an all IP voice and >> data WAN >> >> >> So can you draw a picture of what's plugged in to what? I'm interested in >> seeing this work because we've had some CudaTel customer who have Adtran's >> from a carrier and the PRI's have had challenges, particularly with faxing. >> If you can get yours working like a champ then we'll all be happy. :) >> >> -Michael >> >> >> On Fri, Feb 15, 2013 at 11:09 AM, Blake Priddy > > wrote: >> >>> They are in the same network. >>> >>> >>> On Fri, Feb 15, 2013 at 12:49 PM, Michael Collins wrote: >>> >>>> Is that PRI on one side and SIP on the other? >>>> -MC >>>> >>>> On Fri, Feb 15, 2013 at 7:16 AM, Blake Priddy < >>>> bpriddy at bryantschools.org> wrote: >>>> >>>>> hey guys so we just got in our Adtran 908e and I was wondering if >>>>> someone could assist me on sending calls to freeswitch from this? >>>>> >>>>> -- >>>>> >>>>> *Blakelund Priddy* >>>>> Network Systems Engineer >>>>> Bryant Public School District >>>>> Bryant, Arkansas 72022 >>>>> http://www.bryantschools.org >>>>> p 501-653-5038 >>>>> f 501-847-5656 >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> 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 >>>>> >>>>> >>>> >>>> >>>> -- >>>> Michael S Collins >>>> Twitter: @mercutioviz >>>> http://www.FreeSWITCH.org >>>> http://www.ClueCon.com >>>> http://www.OSTAG.org >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>>> >>>> >>> >>> >>> -- >>> >>> *Blakelund Priddy* >>> Network Systems Engineer >>> Bryant Public School District >>> Bryant, Arkansas 72022 >>> http://www.bryantschools.org >>> p 501-653-5038 >>> f 501-847-5656 >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- *Blakelund Priddy* Network Systems Engineer Bryant Public School District Bryant, Arkansas 72022 http://www.bryantschools.org p 501-653-5038 f 501-847-5656 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130215/c163c50e/attachment.html From Gary.Chen at lightyear.net Fri Feb 15 23:07:36 2013 From: Gary.Chen at lightyear.net (Gary Chen) Date: Fri, 15 Feb 2013 15:07:36 -0500 Subject: [Freeswitch-users] Prolem with SIP OPTION ping Message-ID: <1BE259695B00604384DEBAC3027DF4092FE5B93CA1@lyepkyexc.LYEPLYEXC.local> Version: Freeswitch 1.2.5.3 with ODBC as core db. For redundancy purpose, I have two FS running behind a Kamailio SIP proxy. My Kamailio uses diapatcher to send call to FS1 first. If FS1 is not available, Kamailio will send call to FS2. Kamailio frequently send out SIP OPTION message to check each FS. If no reponse back from one of FS, it will mark it as Inactive and stop sending call to it. The problem I have is that when the odbc connection gone bad, the FS is still up and running and it still reply back "200 OK" to SIP OPTION from Kamailio. So Kamailio dispatcher will not mark it as Inactive and continue sending call to the bad FS and call will fail. Is there anyway I can make FS not reponse back to Kamailio with "200 OK" when core ODBC is down? Or shutdown the FS when core ODBC is down? Gary -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130215/e913390a/attachment-0001.html From david.villasmil.work at gmail.com Fri Feb 15 23:18:07 2013 From: david.villasmil.work at gmail.com (David Villasmil) Date: Fri, 15 Feb 2013 21:18:07 +0100 Subject: [Freeswitch-users] Nibblebill In-Reply-To: References: Message-ID: I don't think it is possible... I'm trying to do this in the dialplan: and I still see: [UPDATE sip_users SET balance=balance-0.010867 WHERE id='12345'] I should note that "sip_users" is the db_table I setup in "nibblebill.conf.xml"... Has anyone tried doing this?? I want use use nibblebill to deduct monet from 2 different tables depending on the context, so I thought I could try setting the db_table, db_column_cash and db_column_account in the contex... thanks David On Thu, Feb 14, 2013 at 3:14 PM, David Villasmil < david.villasmil.work at gmail.com> wrote: > Hello guys, > > Is it possible to use different billing tables for different applications? > > Say i have 2 different context on which i want to use different tables for > balance. > > > Thanks! > > David -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130215/92909fde/attachment.html From david.villasmil.work at gmail.com Fri Feb 15 23:20:10 2013 From: david.villasmil.work at gmail.com (David Villasmil) Date: Fri, 15 Feb 2013 21:20:10 +0100 Subject: [Freeswitch-users] About timer X fired? In-Reply-To: <511DFEFF.9040903@gmail.com> References: <511DFEFF.9040903@gmail.com> Message-ID: Not that I can help you a lot, but it'd be a good idea to paste your log?? On Fri, Feb 15, 2013 at 10:25 AM, yyu0378 wrote: > hi: > > My fs send call from one gatway to another,but in cli has many like > timer X fired.If this is my seesion-timeout is wrong.The seesion-timeout > defaults is 30 minutes.Other gateways is one hour.Did i change it?Thank > you !! > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130215/2ccf4c58/attachment.html From ash at url.net.au Sat Feb 16 00:36:09 2013 From: ash at url.net.au (Ashley Breeden) Date: Sat, 16 Feb 2013 08:36:09 +1100 Subject: [Freeswitch-users] http://infiltrated.net In-Reply-To: <3E86D336-A7EE-488C-996B-641299A5EF06@endigotech.com> References: <28D73E30-E2D0-4382-86A3-8BEDCA02B566@freeswitch.org> <511D94FA.80509@puzzled.xs4all.nl> <5F2F78A9-78A0-42EC-A6F2-76FE8A6A66CA@freeswitch.org> <443E6852-5485-482D-AC1E-195FDE71CCF5@endigotech.com> <3E86D336-A7EE-488C-996B-641299A5EF06@endigotech.com> Message-ID: <452AB046-0DBE-4B27-AACB-03FDD9F5A65A@url.net.au> You could try running the list through this script - http://zwitterion.org/software/aggregate-cidr-addresses/ I think it will still allow 0.0.0.0 but if there is no mask on the address it will add 0.0.0.0/32. Ash. On 16/02/2013, at 4:40 AM, Brian Foster wrote: > It wouldn't be hard to create a bash script to do this > > Sent from my iPhone > > On Feb 15, 2013, at 3:49 AM, Avi Marcus wrote: > >> Does someone have such a script that... >> 1) removes old entries >> 2) validates new list >> 3) adds new entries to iptables >> >> ..? I presume the community might appreciate it. >> >> -Avi >> >> On Fri, Feb 15, 2013 at 7:45 AM, Brian Foster wrote: >> Yea doesn't seem accurate but you could run a script a you're fetching it to validate the IP and if it isn't valid or it's a local ip then throw it out. At least it will be closer to your target. >> >> Sent from my iPhone >> >> On Feb 14, 2013, at 9:39 PM, Brian West wrote: >> >> > 0.0.0.0 >> > .99 >> > .161.101.234 >> > >> > This is at the bottom of the list so that concerns me. >> > -- >> > Brian West >> > brian at freeswitch.org >> > FreeSWITCH Solutions, LLC >> > PO BOX PO BOX 2531 >> > Brookfield, WI 53008-2531 >> > Twitter: @FreeSWITCH_Wire >> > T: +1.918.420.9266 | F: +1.918.420.9267 | M: +1.918.424.WEST >> > iNUM: +883 5100 1420 9266 >> > UK: +44 20 3298 4900 >> > ISN: 410*543 >> > >> > >> > >> > >> > >> > On Feb 14, 2013, at 7:52 PM, Patrick Lists wrote: >> > >> >> On 02/15/2013 01:38 AM, Brian West wrote: >> >>> Does anyone know if these lists are updated still? >> >> >> >> The date on the blacklist says Thu Feb 14 19:10:03 CST 2013: >> >> >> >> http://www.infiltrated.net/blacklisted >> >> >> >> Regards, >> >> Patrick >> >> >> >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://wiki.freeswitch.org >> >> http://www.cluecon.com >> >> >> >> 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 >> > >> > >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > 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 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130216/33556665/attachment-0001.html From cal.leeming at simplicitymedialtd.co.uk Sat Feb 16 01:17:21 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Fri, 15 Feb 2013 22:17:21 +0000 Subject: [Freeswitch-users] http://infiltrated.net In-Reply-To: <28D73E30-E2D0-4382-86A3-8BEDCA02B566@freeswitch.org> References: <28D73E30-E2D0-4382-86A3-8BEDCA02B566@freeswitch.org> Message-ID: Curious, I just ran this list on a client network (which runs a non voice related service) and found a whole bunch of known abusive users, as well as some new ones we didn't know about. Thanks for this list Brian, always good to find useful blacklists :) Cal On Fri, Feb 15, 2013 at 12:38 AM, Brian West wrote: > Does anyone know if these lists are updated still? > > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX PO BOX 2531 > Brookfield, WI 53008-2531 > > Twitter: @FreeSWITCH_Wire > http://freeswitchcookbook.com > http://freeswitchbook.com > > T: +1.213.286.0410 | F: +1.213.286.0401 | M: +1.918.424.WEST > iNUM: +883 5100 1286 0410 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130215/fd8d1f0e/attachment.html From jpyle at fidelityvoice.com Sat Feb 16 01:19:53 2013 From: jpyle at fidelityvoice.com (Jeff Pyle) Date: Fri, 15 Feb 2013 17:19:53 -0500 Subject: [Freeswitch-users] HELP In-Reply-To: References: Message-ID: <511EB489.4090000@fidelityvoice.com> Bryce, Here are the relevant voice pieces - you'll want to season to taste. I recommend AOS version R10.3.2. The config may be slightly different on A4.xx. ! interface eth 0/1 ip address 192.168.150.10 255.255.255.0 media-gateway ip primary no shutdown ! interface t1 0/3 description PRI to ATT tdm-group 1 timeslots 1-24 speed 64 no shutdown ! interface pri 1 connect t1 0/3 tdm-group 1 role user no shutdown ! isdn-group 1 connect pri 1 ! timing-source t1 0/3 ! voice dial-plan 1 operator-assisted 411 voice dial-plan 2 local NXXXXXX voice dial-plan 3 local NXXNXXXXXX voice dial-plan 6 local 1NXXNXXXXXX voice dial-plan 7 international 011XXXXX$ ! voice trunk T01 type isdn desc TRUNK TO ATT PRI resource-selection linear descending connect isdn-group 1 modem-passthrough t38 t38 redundancy high-speed 1 t38 max-buffer 262 t38 max-datagram 176 t38 error-correction redundancy rtp delay-mode adaptive ! voice trunk T02 type sip reject-external desc TRUNK TO FS sip-server primary 192.168.150.20 ! voice grouped-trunk TO_FS trunk T02 accept NXXNXXXXXX cost 10 ! voice grouped-trunk TO_PRI trunk T01 accept $ cost 10 ! ip sip grammar from host local ! This assumes 10-digit DNIS from AT&T in the "accept NXXNXXXXXX cost 10" statement in the TO_FS group. If you receive 5, you'll want to adapt it accordingly. If your fax server is the standalone variety, you may want to consider connecting it to the FXS ports of the 908e. In this scenario the 908e operates like a glorified TDM channel bank. If it's on Freeswitch, this SIP configuration should work fine. I've had no issues on R10.3.2 (or later versions of A4.x) talking to Freeswitch's rxfax application with T.38 or G711 passthru from a 908e. As you debug your configuration... "debug sip stack messages" will show you all the SIP traffic. "debug isdn l2-formatted" will show you all the ISDN layer-3 messages. Both are extremely helpful. "debug voice verbose" can be handy in more complicated routing setups. "un all" will turn off all the debug statements you've accumulated. In my opinion P-Asserted-Identity / Privacy header support is a nice addition to this basic config, along with a corresponding configuration on Freeswitch. - Jeff Blake Priddy wrote: > ATT provides our PRI. And I want to send 5 digits to the PBX and also > I have a fax server i need to send 5 digits too. > > > On Fri, Feb 15, 2013 at 1:40 PM, Jeff Pyle > wrote: > > Blake, > > We use the TA900-series with quite a bit of success. Tell me more > about the PRI (DNIS digits, switch type) and I should be able to > help you with a config template. > > > - Jeff > > > > On Fri, Feb 15, 2013 at 2:28 PM, Michael Collins > > wrote: > > I'm not familiar with the Adtran 908e, but a quick goog search > says: > > The Total Access 908e IP Business Gateway provides SIP-gateway > functionality, a robust IP router, firewall and VPN > functionality, and support for a number of analog and digital > interfaces for existing phone equipment, or for the > combination of an IP PBX and an analog fax, phones, or modem. > > * Four T1/dual Ethernet platform, 8 analog FXS interfaces, > integrated DSX-1 > * Ideal for networks transitioning to VoIP and an all IP > voice and data WAN > > > So can you draw a picture of what's plugged in to what? I'm > interested in seeing this work because we've had some CudaTel > customer who have Adtran's from a carrier and the PRI's have > had challenges, particularly with faxing. If you can get yours > working like a champ then we'll all be happy. :) > > -Michael > > > On Fri, Feb 15, 2013 at 11:09 AM, Blake Priddy > > > wrote: > > They are in the same network. > > > On Fri, Feb 15, 2013 at 12:49 PM, Michael Collins > > wrote: > > Is that PRI on one side and SIP on the other? > -MC > > On Fri, Feb 15, 2013 at 7:16 AM, Blake Priddy > > wrote: > > hey guys so we just got in our Adtran 908e and I > was wondering if someone could assist me on > sending calls to freeswitch from this? > > -- > > *Blakelund Priddy* > Network Systems Engineer > Bryant Public School District > Bryant, Arkansas 72022 > http://www.bryantschools.org > > p 501-653-5038 > f 501-847-5656 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel > Communication Server > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication > Server > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > > -- > > *Blakelund Priddy* > Network Systems Engineer > Bryant Public School District > Bryant, Arkansas 72022 > http://www.bryantschools.org > p 501-653-5038 > f 501-847-5656 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > > -- > > *Blakelund Priddy* > Network Systems Engineer > Bryant Public School District > Bryant, Arkansas 72022 > http://www.bryantschools.org > p 501-653-5038 > f 501-847-5656 > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130215/23e6c393/attachment-0001.html From dujinfang at gmail.com Sat Feb 16 02:36:47 2013 From: dujinfang at gmail.com (Seven Du) Date: Sat, 16 Feb 2013 07:36:47 +0800 Subject: [Freeswitch-users] ooH323 vs mod_h323 In-Reply-To: References: <2BD0DC60-D428-4323-A020-D9AD20E76553@freeswitch.org> <4C2EA249.4070704@coppice.org> <267C8398-6CA3-442E-B973-BE096D728B28@freeswitch.org> Message-ID: So we basically have 3 options, Opal, h323 and ooh323c, what's the difference among them? Also, I'm aware that video is not supported in the current mod_opal and mod_h323, is it possible(or how hard) to add video support? Or does ooh323c easy to support video? We'd like to contribute a little money to speed it up if it can support video and some want to work on it. -- Seven Du http://www.freeswitch.org.cn http://about.me/dujinfang http://www.dujinfang.com Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Friday, February 15, 2013 at 11:37 PM, Steven Ayre wrote: > There's a module for Asterisk but none for FS I'm aware of yet - I'm only wondering about the feasibility of writing one. > > -Steve > > > > On 15 February 2013 14:48, Brian West wrote: > > So I have the impression you or someone has an ooh323 module and hasn't contributed it back to the project yet? > > -- > > Brian West > > brian at freeswitch.org (mailto:brian at freeswitch.org) > > FreeSWITCH Solutions, LLC > > PO BOX PO BOX 2531 > > Brookfield, WI 53008-2531 > > Twitter: @FreeSWITCH_Wire > > T: +1.918.420.9266 (tel:%2B1.918.420.9266) | F: +1.918.420.9267 (tel:%2B1.918.420.9267) | M: +1.918.424.WEST > > iNUM: +883 5100 1420 9266 (tel:%2B883%205100%201420%209266) > > UK: +44 20 3298 4900 (tel:%2B44%2020%203298%204900) > > ISN: 410*543 > > > > > > > > > > > > On Feb 15, 2013, at 3:49 AM, Steven Ayre wrote: > > > > > Hi Brian, > > > > > > The issue would be that Asterisk's modifications are on a version predating the FLOSS exemption and their project is GPL - so presumably their modifications are all GPL-only? > > > > > > -Steve > > > > > > > > > > > > On 15 February 2013 00:40, Brian West wrote: > > > If someone wanted to build ooh323 module for FreeSWITCH we do have permission to do so using the lib. > > > > > > /b > > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org (mailto: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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto: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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130216/f5fa4d66/attachment.html From bdfoster at endigotech.com Sat Feb 16 03:22:09 2013 From: bdfoster at endigotech.com (Brian Foster) Date: Fri, 15 Feb 2013 19:22:09 -0500 Subject: [Freeswitch-users] http://infiltrated.net In-Reply-To: References: <28D73E30-E2D0-4382-86A3-8BEDCA02B566@freeswitch.org> Message-ID: <0669FEC2-11CA-4775-A5D1-68353DF848F0@endigotech.com> We've got a process that does this: 1) use fail2ban to jail abusive traffic 2) capture the logs, parse and put in couchdb database 3) use the database as a basis for jail, publish findings as a text file usable for a jail on a node Something like that (might not be extremely scalable, but could work) would be extremely nice for the community to participate in. Theoretically you could have some logic in there that gave each entry a score based on the source of the blacklist entry, how many times its been reported, etc. and when it hits a certain point value, publish in the text file available for download to all interested parties. Sent from my iPhone On Feb 15, 2013, at 5:17 PM, "Cal Leeming [Simplicity Media Ltd]" wrote: > Curious, I just ran this list on a client network (which runs a non voice related service) and found a whole bunch of known abusive users, as well as some new ones we didn't know about. > > Thanks for this list Brian, always good to find useful blacklists :) > > Cal > > On Fri, Feb 15, 2013 at 12:38 AM, Brian West wrote: >> Does anyone know if these lists are updated still? >> >> -- >> Brian West >> brian at freeswitch.org >> FreeSWITCH Solutions, LLC >> PO BOX PO BOX 2531 >> Brookfield, WI 53008-2531 >> >> Twitter: @FreeSWITCH_Wire >> http://freeswitchcookbook.com >> http://freeswitchbook.com >> >> T: +1.213.286.0410 | F: +1.213.286.0401 | M: +1.918.424.WEST >> iNUM: +883 5100 1286 0410 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130215/efdfd419/attachment.html From steveayre at gmail.com Sat Feb 16 03:55:12 2013 From: steveayre at gmail.com (Steven Ayre) Date: Sat, 16 Feb 2013 00:55:12 +0000 Subject: [Freeswitch-users] Prolem with SIP OPTION ping In-Reply-To: <1BE259695B00604384DEBAC3027DF4092FE5B93CA1@lyepkyexc.LYEPLYEXC.local> References: <1BE259695B00604384DEBAC3027DF4092FE5B93CA1@lyepkyexc.LYEPLYEXC.local> Message-ID: <2F3EA3E2-F727-4C0D-A986-6E3141D70963@gmail.com> You could run a lua in the background monitoring the core db as a startup-script, and if it's down 'fsctl pause'. Or externally via ESL. I believe FS will then reply 503 not 200 which you could treat as a failure on Kamalio. Steve On 15 Feb 2013, at 20:07, Gary Chen wrote: > Version: Freeswitch 1.2.5.3 with ODBC as core db. > > For redundancy purpose, I have two FS running behind a Kamailio SIP proxy. My Kamailio uses diapatcher to send call to FS1 first. If FS1 is not available, Kamailio will send call to FS2. Kamailio frequently send out SIP OPTION message to check each FS. If no reponse back from one of FS, it will mark it as Inactive and stop sending call to it. The problem I have is that when the odbc connection gone bad, the FS is still up and running and it still reply back ?200 OK? to SIP OPTION from Kamailio. So Kamailio dispatcher will not mark it as Inactive and continue sending call to the bad FS and call will fail. > > Is there anyway I can make FS not reponse back to Kamailio with ?200 OK? when core ODBC is down? > Or shutdown the FS when core ODBC is down? > > Gary > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130216/28930189/attachment-0001.html From steveayre at gmail.com Sat Feb 16 03:52:56 2013 From: steveayre at gmail.com (Steven Ayre) Date: Sat, 16 Feb 2013 00:52:56 +0000 Subject: [Freeswitch-users] ooH323 vs mod_h323 In-Reply-To: References: <2BD0DC60-D428-4323-A020-D9AD20E76553@freeswitch.org> <4C2EA249.4070704@coppice.org> <267C8398-6CA3-442E-B973-BE096D728B28@freeswitch.org> Message-ID: <77B9FF50-213F-4BF6-917E-FBB006DF5C76@gmail.com> mod_opal and mod_h323 are based on the opal and h323plus libraries. Both are forks of the open323 library (which is discontinued). h323plus aims to remain API compatible with open323, while it seems opal aims to become a more generic telecoms library with support for SIP IAX etc too. ooh323c is a completely separate library with no shared code. mod_h323 and mod_opal both seem to have stability problems. Partly that seems to come from the heavy dependance of the libraries on specific matching versions of ptlib, partly that seems to be the libraries themselves. Both Asterisk and YATE appear to have stability problems using h323plus/openh323. Anecdotally the ooh323c Asterisk module is more stable... I'd need to look into it further, but the ooh323c library implements H323 and leaves RTP/codecs to the app. h323plus/opal implement them themselves (overridable I imagine). That means adding support to ooh323c may end up being easier, in theory. But that's a guess without researching it... Steve On 15 Feb 2013, at 23:36, Seven Du wrote: > So we basically have 3 options, Opal, h323 and ooh323c, what's the difference among them? > > Also, I'm aware that video is not supported in the current mod_opal and mod_h323, is it possible(or how hard) to add video support? Or does ooh323c easy to support video? > > We'd like to contribute a little money to speed it up if it can support video and some want to work on it. > > -- > Seven Du > http://www.freeswitch.org.cn > http://about.me/dujinfang > http://www.dujinfang.com > > Sent with Sparrow > > On Friday, February 15, 2013 at 11:37 PM, Steven Ayre wrote: > >> There's a module for Asterisk but none for FS I'm aware of yet - I'm only wondering about the feasibility of writing one. >> >> -Steve >> >> >> >> On 15 February 2013 14:48, Brian West wrote: >>> So I have the impression you or someone has an ooh323 module and hasn't contributed it back to the project yet? >>> -- >>> Brian West >>> brian at freeswitch.org >>> FreeSWITCH Solutions, LLC >>> PO BOX PO BOX 2531 >>> Brookfield, WI 53008-2531 >>> Twitter: @FreeSWITCH_Wire >>> T: +1.918.420.9266 | F: +1.918.420.9267 | M: +1.918.424.WEST >>> iNUM: +883 5100 1420 9266 >>> UK: +44 20 3298 4900 >>> ISN: 410*543 >>> >>> >>> >>> >>> >>> On Feb 15, 2013, at 3:49 AM, Steven Ayre wrote: >>> >>> > Hi Brian, >>> > >>> > The issue would be that Asterisk's modifications are on a version predating the FLOSS exemption and their project is GPL - so presumably their modifications are all GPL-only? >>> > >>> > -Steve >>> > >>> > >>> > >>> > On 15 February 2013 00:40, Brian West wrote: >>> > If someone wanted to build ooh323 module for FreeSWITCH we do have permission to do so using the lib. >>> > >>> > /b >>> > >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130216/7715c6f5/attachment.html From schoch+freeswitch.org at xwin32.com Sat Feb 16 04:51:46 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Fri, 15 Feb 2013 17:51:46 -0800 Subject: [Freeswitch-users] FAX problems In-Reply-To: References: <20130119191943.ca09ab6d@mail.tritonwest.net> <5102AF93.2000201@coppice.org> Message-ID: Thanks to the helpful people on the conference call, I got HylaFAX working today. It turned out I needed to set not only the "fax_enable_t38" variable, but also the "fax_enable_t38_request" variable. Further input showed that the best solution was to change the context in the spandsp.conf.xml from "default" to "hylafax", and make a separate "hylafax" dialplan that has this: ...then the normal dialing rules. But as usual, I still have problems. :-( Now I can't get the old analog FAX machine to successfully send FAXes. I have it plugged into a cheap ATA, the OBi. It seemed to be working at one time, but that may have been a fluke. On calls through it, I have fax_enable_t38 and fax_enable_t38_request set, and I also set "execute_on_answer=t38_gateway peer" before the call is bridged to the gateway. Should I be using something else? -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130215/b24bff33/attachment.html From egable+freeswitch at gmail.com Sat Feb 16 08:58:25 2013 From: egable+freeswitch at gmail.com (Eliot Gable) Date: Sat, 16 Feb 2013 00:58:25 -0500 Subject: [Freeswitch-users] FreeSWITCH HA BoF Meetings In-Reply-To: References: Message-ID: I am usually just getting my kids in bed at 8pm, so 8:15pm is typically better timing. On Feb 15, 2013 7:26 PM, "Michael Collins" wrote: > will do. > > On Fri, Feb 15, 2013 at 1:01 PM, Giovanni Maruzzelli wrote: > >> ouch! at 08:00pm EST will be 02:00am my time. >> Not sure I'll be able to attend the live discussion. Please record it as >> always, please :) >> >> -giovanni >> >> >> On Fri, Feb 15, 2013 at 9:59 PM, Giovanni Maruzzelli wrote: >> >>> I'm very much interested, thanks Eliot for your efforts so far. >>> >>> -giovanni >>> >>> On Thu, Feb 14, 2013 at 11:41 PM, Michael Collins wrote: >>> >>>> Hello! >>>> >>>> Yesterday we had a nice discussion >>>> about all things >>>> HA. Eliot Gable gave us a lot of very good information about the state of >>>> mod_ha_development and all the work that he's done over the past 4+ years >>>> in the world of HA and FreeSWITCH. >>>> >>>> It's now time to let those who have a vested interest in HA to get >>>> together with Eliot and continuing the discussion. To that end we propose a >>>> FreeSWITCH HA BoF (Birds of a Feather) conference. Eliot says that he can >>>> be available after 8PM EST. I would like to propose that the meetup be >>>> Tuesday evenings at 8:00PM EST in the main FreeSWITCH public conference. >>>> >>>> If you would like to meetup then please email me off-list and CC Eliot ( >>>> egable+freeswitch at gmail.com) so that we can keep track of who is most >>>> interested in this topic. >>>> >>>> Thanks! >>>> >>>> -- >>>> Michael S Collins >>>> Twitter: @mercutioviz >>>> http://www.FreeSWITCH.org >>>> http://www.ClueCon.com >>>> http://www.OSTAG.org >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>>> >>>> >>> >>> >>> -- >>> Sincerely, >>> >>> Giovanni Maruzzelli >>> Cell : +39-347-2665618 >>> >> >> >> >> -- >> Sincerely, >> >> Giovanni Maruzzelli >> Cell : +39-347-2665618 >> > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130216/78ed7da0/attachment-0001.html From egable+freeswitch at gmail.com Sat Feb 16 09:03:57 2013 From: egable+freeswitch at gmail.com (Eliot Gable) Date: Sat, 16 Feb 2013 01:03:57 -0500 Subject: [Freeswitch-users] WebRTC to SIP In-Reply-To: <511E8287.8040501@livecall.com> References: <511E8287.8040501@livecall.com> Message-ID: A week and a half ago Tony was making test calls which also had no audio. I know he was working on the issue, but don't know how far he got since then. I certainly would not consider it "baked" at this point, especially given current browser support and flux. I also don't really know any more details than that. On Feb 15, 2013 1:49 PM, "Jack" wrote: > Has anyone been successful in connecting to freeswitch with this? > http://kapejod.org/webrtc/sip-audio-demo.html > > I am able to register on freeswitch and place a call to the webpage, > which rings, but no audio. > > This is for Canary Beta browser only right now. > > Jack > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130216/fd5e2907/attachment.html From krice at freeswitch.org Sat Feb 16 09:06:08 2013 From: krice at freeswitch.org (Ken Rice) Date: Sat, 16 Feb 2013 00:06:08 -0600 Subject: [Freeswitch-users] FreeSWITCH HA BoF Meetings In-Reply-To: Message-ID: Is earlier and option? Say like 6 EST or something? On 2/15/13 11:58 PM, "Eliot Gable" wrote: > I am usually just getting my kids in bed at 8pm, so 8:15pm is typically better > timing. > On Feb 15, 2013 7:26 PM, "Michael Collins" wrote: >> will do. >> >> On Fri, Feb 15, 2013 at 1:01 PM, Giovanni Maruzzelli >> wrote: >>> ouch! at 08:00pm EST will be 02:00am my time. >>> Not sure I'll be able to attend the live discussion. Please record it as >>> always, please :) >>> >>> -giovanni >>> >>> >>> On Fri, Feb 15, 2013 at 9:59 PM, Giovanni Maruzzelli >>> wrote: >>>> I'm very much interested, thanks Eliot for your efforts so far. >>>> >>>> -giovanni >>>> >>>> On Thu, Feb 14, 2013 at 11:41 PM, Michael Collins >>>> wrote: >>>>> Hello! >>>>> >>>>> Yesterday we had a nice discussion >>>>> about all things >>>>> HA. Eliot Gable gave us a lot of very good information about the state of >>>>> mod_ha_development and all the work that he's done over the past 4+ years >>>>> in the world of HA and FreeSWITCH.? >>>>> >>>>> It's now time to let those who have a vested interest in HA to get >>>>> together with Eliot and continuing the discussion. To that end we propose >>>>> a FreeSWITCH HA BoF (Birds of a Feather) conference. Eliot says that he >>>>> can be available after 8PM EST. I would like to propose that the meetup be >>>>> Tuesday evenings at 8:00PM EST in the main FreeSWITCH public conference. >>>>> >>>>> If you would like to meetup then please email me off-list and CC Eliot >>>>> (egable+freeswitch at gmail.com ) so >>>>> that we can keep track of who is most interested in this topic. >>>>> >>>>> Thanks! -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130216/f319d794/attachment.html From egable+freeswitch at gmail.com Sat Feb 16 09:11:18 2013 From: egable+freeswitch at gmail.com (Eliot Gable) Date: Sat, 16 Feb 2013 01:11:18 -0500 Subject: [Freeswitch-users] Multicast in freeswitch In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF236B788@Mail-Kilo.squay.com> References: <592A9CF93E12394E8472A6CC66E66BF236B788@Mail-Kilo.squay.com> Message-ID: You can use mod_event_socket and watch for SWITCH_CHANNEL_STATE, SWITCH_CHANNEL_CALLSTATE, SWITCH_CHANNEL_CREATE, SWITCH_CHANNEL_ORIGINATE, SWITCH_CHANNEL_BRIDGE, and any other event of interest. Then you can track the state of each call for whatever purpose you might want to use that info for. On Feb 14, 2013 12:31 PM, "Archana Venugopan" wrote: > Hi,**** > > I would like to get ?Orig-status? event in freeswitch so that I will know > when the phone is being HANGED. But to get that event I read in wiki I need > mod_event_multicast.**** > > ** ** > > http://wiki.freeswitch.org/wiki/Mod_event_multicast**** > > ** ** > > But I don?t see in above wiki on how to compile and load > mod_event_multicast. Can anyone please let me know how to compile and load > multicast?**** > > ** ** > > Regards,**** > > Archana**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130216/12e14dd0/attachment.html From msc at freeswitch.org Sat Feb 16 09:49:32 2013 From: msc at freeswitch.org (Michael Collins) Date: Fri, 15 Feb 2013 22:49:32 -0800 Subject: [Freeswitch-users] FreeSWITCH HA BoF Meetings In-Reply-To: References: Message-ID: We'll call it 8:15 but people can start showing up at 8pm if they want to chitchat like we do at the beginning of the weekly conf call. -MC On Fri, Feb 15, 2013 at 9:58 PM, Eliot Gable wrote: > I am usually just getting my kids in bed at 8pm, so 8:15pm is typically > better timing. > On Feb 15, 2013 7:26 PM, "Michael Collins" wrote: > >> will do. >> >> On Fri, Feb 15, 2013 at 1:01 PM, Giovanni Maruzzelli wrote: >> >>> ouch! at 08:00pm EST will be 02:00am my time. >>> Not sure I'll be able to attend the live discussion. Please record it as >>> always, please :) >>> >>> -giovanni >>> >>> >>> On Fri, Feb 15, 2013 at 9:59 PM, Giovanni Maruzzelli wrote: >>> >>>> I'm very much interested, thanks Eliot for your efforts so far. >>>> >>>> -giovanni >>>> >>>> On Thu, Feb 14, 2013 at 11:41 PM, Michael Collins wrote: >>>> >>>>> Hello! >>>>> >>>>> Yesterday we had a nice discussion >>>>> about all >>>>> things HA. Eliot Gable gave us a lot of very good information about the >>>>> state of mod_ha_development and all the work that he's done over the past >>>>> 4+ years in the world of HA and FreeSWITCH. >>>>> >>>>> It's now time to let those who have a vested interest in HA to get >>>>> together with Eliot and continuing the discussion. To that end we propose a >>>>> FreeSWITCH HA BoF (Birds of a Feather) conference. Eliot says that he can >>>>> be available after 8PM EST. I would like to propose that the meetup be >>>>> Tuesday evenings at 8:00PM EST in the main FreeSWITCH public conference. >>>>> >>>>> If you would like to meetup then please email me off-list and CC Eliot >>>>> (egable+freeswitch at gmail.com) so that we can keep track of who is >>>>> most interested in this topic. >>>>> >>>>> Thanks! >>>>> >>>>> -- >>>>> Michael S Collins >>>>> Twitter: @mercutioviz >>>>> http://www.FreeSWITCH.org >>>>> http://www.ClueCon.com >>>>> http://www.OSTAG.org >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> 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 >>>>> >>>>> >>>> >>>> >>>> -- >>>> Sincerely, >>>> >>>> Giovanni Maruzzelli >>>> Cell : +39-347-2665618 >>>> >>> >>> >>> >>> -- >>> Sincerely, >>> >>> Giovanni Maruzzelli >>> Cell : +39-347-2665618 >>> >> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130215/1bb00171/attachment-0001.html From brian at tarricone.org Sat Feb 16 13:04:34 2013 From: brian at tarricone.org (Brian J. Tarricone) Date: Sat, 16 Feb 2013 02:04:34 -0800 Subject: [Freeswitch-users] codec negotiation weirdness Message-ID: Hi all, I'm experiencing some strange behavior with codec negotiation. I'm using the default early negotiation, and I'm setting inbound-codec-negotiation to "scrooge" so FS will always have the final word on what codec is used. (I've also tried "greedy" with the same results. "generous" does not meet my needs, unfortunately.) I'm setting inbound-codec-prefs to the following: opus at 48000h@20i,isac at 32000h@30i,speex at 16000h@20i,speex at 8000h@20i,PCMU However, I'm seeing FS select PCMU for a client that advertises both isac/32k and PCMU. The relevant portion of the FS log is below. 2013-02-16 01:45:06.947647 [DEBUG] sofia.c:5589 Remote SDP: v=0 o=- 1772524861 1772524861 IN IP4 10.70.209.126 s=- c=IN IP4 10.70.209.126 t=0 0 m=audio 57178 RTP/AVP 103 104 0 8 106 105 13 126 c=IN IP4 10.70.209.126 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec Compare [ISAC:103:16000:30:0]/[opus:116:48000:20:32000] 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec Compare [ISAC:104:32000:30:0]/[opus:116:48000:20:32000] 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec Compare [PCMU:0:8000:20:64000]/[opus:116:48000:20:32000] 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec Compare [PCMA:8:8000:20:64000]/[opus:116:48000:20:32000] 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec Compare [CN:106:32000:20:0]/[opus:116:48000:20:32000] 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec Compare [CN:105:16000:20:0]/[opus:116:48000:20:32000] 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec Compare [CN:13:8000:20:0]/[opus:116:48000:20:32000] 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec Compare [telephone-event:126:8000:20:0]/[opus:116:48000:20:32000] 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5268 Set 2833 dtmf send/recv payload to 126 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec Compare [ISAC:103:16000:30:0]/[SPEEX:99:16000:20:42200] 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec Compare [ISAC:104:32000:30:0]/[SPEEX:99:16000:20:42200] 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec Compare [PCMU:0:8000:20:64000]/[SPEEX:99:16000:20:42200] 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec Compare [PCMA:8:8000:20:64000]/[SPEEX:99:16000:20:42200] 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec Compare [CN:106:32000:20:0]/[SPEEX:99:16000:20:42200] 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec Compare [CN:105:16000:20:0]/[SPEEX:99:16000:20:42200] 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec Compare [CN:13:8000:20:0]/[SPEEX:99:16000:20:42200] 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec Compare [telephone-event:126:8000:20:0]/[SPEEX:99:16000:20:42200] 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5268 Set 2833 dtmf send/recv payload to 126 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec Compare [ISAC:103:16000:30:0]/[SPEEX:99:8000:20:24600] 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec Compare [ISAC:104:32000:30:0]/[SPEEX:99:8000:20:24600] 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec Compare [PCMU:0:8000:20:64000]/[SPEEX:99:8000:20:24600] 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec Compare [PCMA:8:8000:20:64000]/[SPEEX:99:8000:20:24600] 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec Compare [CN:106:32000:20:0]/[SPEEX:99:8000:20:24600] 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec Compare [CN:105:16000:20:0]/[SPEEX:99:8000:20:24600] 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec Compare [CN:13:8000:20:0]/[SPEEX:99:8000:20:24600] 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec Compare [telephone-event:126:8000:20:0]/[SPEEX:99:8000:20:24600] 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5268 Set 2833 dtmf send/recv payload to 126 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec Compare [ISAC:103:16000:30:0]/[PCMU:0:8000:20:64000] 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec Compare [ISAC:104:32000:30:0]/[PCMU:0:8000:20:64000] 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5161 Bah HUMBUG! Sticking with PCMU at 8000h@20i As you can see, it first compares all offered codecs against opus/48k, next skips isac completely, then goes to speex/16k, speex/8k, and PCMU, and eventually grudgingly accepts PCMU. Interestingly, if I drop opus from the list, and leave the rest as-is, it does select isac/32k, but weirdly: 2013-02-16 01:51:10.691314 [DEBUG] sofia_glue.c:5139 Audio Codec Compare [ISAC:103:16000:30:0]/[isac:99:32000:30:160000] 2013-02-16 01:51:10.691314 [DEBUG] sofia_glue.c:5161 Bah HUMBUG! Sticking with isac at 32000h@30i It picks isac/32k after only examining the isac/16k line in the SDP, before it even gets to isac/32k. (In fact those are the only 2 lines after the SDP is printed; it makes the decision after making just one comparison.) Any ideas? Am I just misunderstanding how this works? Thanks, Brian From brian at tarricone.org Sat Feb 16 14:40:07 2013 From: brian at tarricone.org (Brian J. Tarricone) Date: Sat, 16 Feb 2013 03:40:07 -0800 Subject: [Freeswitch-users] codec negotiation weirdness In-Reply-To: References: Message-ID: Sorry about the self reply, but I figured out something new: if I don't specify the codec params in inbound-codec-prefs, and instead just specify: opus,isac,speex at 16000h@20i,speex at 8000h@20i,PCMU ... then isac does get selected when opus isn't advertised by the client: 2013-02-16 03:23:53.045715 [DEBUG] sofia_glue.c:5139 Audio Codec Compare [ISAC:103:16000:30:0]/[opus:116:48000:40:32000] 2013-02-16 03:23:53.045715 [DEBUG] sofia_glue.c:5139 Audio Codec Compare [ISAC:104:32000:30:0]/[opus:116:48000:40:32000] 2013-02-16 03:23:53.045715 [DEBUG] sofia_glue.c:5139 Audio Codec Compare [PCMU:0:8000:20:64000]/[opus:116:48000:40:32000] 2013-02-16 03:23:53.045715 [DEBUG] sofia_glue.c:5139 Audio Codec Compare [PCMA:8:8000:20:64000]/[opus:116:48000:40:32000] 2013-02-16 03:23:53.045715 [DEBUG] sofia_glue.c:5139 Audio Codec Compare [CN:106:32000:20:0]/[opus:116:48000:40:32000] 2013-02-16 03:23:53.045715 [DEBUG] sofia_glue.c:5139 Audio Codec Compare [CN:105:16000:20:0]/[opus:116:48000:40:32000] 2013-02-16 03:23:53.045715 [DEBUG] sofia_glue.c:5139 Audio Codec Compare [CN:13:8000:20:0]/[opus:116:48000:40:32000] 2013-02-16 03:23:53.045715 [DEBUG] sofia_glue.c:5139 Audio Codec Compare [telephone-event:126:8000:20:0]/[opus:116:48000:40:32000] 2013-02-16 03:23:53.045715 [DEBUG] sofia_glue.c:5268 Set 2833 dtmf send/recv payload to 126 2013-02-16 03:23:53.045715 [DEBUG] sofia_glue.c:5139 Audio Codec Compare [ISAC:103:16000:30:0]/[isac:99:32000:60:160000] 2013-02-16 03:23:53.045715 [DEBUG] sofia_glue.c:5161 Bah HUMBUG! Sticking with isac at 32000h@60i However, in that case it picks a 60ms ptime, which is not what I want. FS startup messages seem to indicate the isac version I'm using does support 30ms: 2013-02-16 03:21:49.163553 [NOTICE] switch_loadable_module.c:187 Adding Codec isac 99 isac 32000hz 60ms 160000bps 2013-02-16 03:21:49.163562 [NOTICE] switch_loadable_module.c:187 Adding Codec isac 99 isac 32000hz 30ms 160000bps 2013-02-16 03:21:49.163567 [NOTICE] switch_loadable_module.c:187 Adding Codec isac 99 isac 16000hz 60ms 53400bps 2013-02-16 03:21:49.163572 [NOTICE] switch_loadable_module.c:187 Adding Codec isac 99 isac 16000hz 30ms 53400bps Why won't it let me select 30ms? Why does FS auto-select 60ms when I don't specify a preference? Forgot to mention before: I'm using current git master, rev 9733740. On Sat, Feb 16, 2013 at 2:04 AM, Brian J. Tarricone wrote: > Hi all, > > I'm experiencing some strange behavior with codec negotiation. I'm > using the default early negotiation, and I'm setting > inbound-codec-negotiation to "scrooge" so FS will always have the > final word on what codec is used. (I've also tried "greedy" with the > same results. "generous" does not meet my needs, unfortunately.) > > I'm setting inbound-codec-prefs to the following: > > opus at 48000h@20i,isac at 32000h@30i,speex at 16000h@20i,speex at 8000h@20i,PCMU > > However, I'm seeing FS select PCMU for a client that advertises both > isac/32k and PCMU. The relevant portion of the FS log is below. > > 2013-02-16 01:45:06.947647 [DEBUG] sofia.c:5589 Remote SDP: > v=0 > o=- 1772524861 1772524861 IN IP4 10.70.209.126 > s=- > c=IN IP4 10.70.209.126 > t=0 0 > m=audio 57178 RTP/AVP 103 104 0 8 106 105 13 126 > c=IN IP4 10.70.209.126 > a=rtpmap:103 ISAC/16000 > a=rtpmap:104 ISAC/32000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:106 CN/32000 > a=rtpmap:105 CN/16000 > a=rtpmap:13 CN/8000 > a=rtpmap:126 telephone-event/8000 > > 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec > Compare [ISAC:103:16000:30:0]/[opus:116:48000:20:32000] > 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec > Compare [ISAC:104:32000:30:0]/[opus:116:48000:20:32000] > 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec > Compare [PCMU:0:8000:20:64000]/[opus:116:48000:20:32000] > 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec > Compare [PCMA:8:8000:20:64000]/[opus:116:48000:20:32000] > 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec > Compare [CN:106:32000:20:0]/[opus:116:48000:20:32000] > 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec > Compare [CN:105:16000:20:0]/[opus:116:48000:20:32000] > 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec > Compare [CN:13:8000:20:0]/[opus:116:48000:20:32000] > 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec > Compare [telephone-event:126:8000:20:0]/[opus:116:48000:20:32000] > 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5268 Set 2833 dtmf > send/recv payload to 126 > 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec > Compare [ISAC:103:16000:30:0]/[SPEEX:99:16000:20:42200] > 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec > Compare [ISAC:104:32000:30:0]/[SPEEX:99:16000:20:42200] > 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec > Compare [PCMU:0:8000:20:64000]/[SPEEX:99:16000:20:42200] > 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec > Compare [PCMA:8:8000:20:64000]/[SPEEX:99:16000:20:42200] > 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec > Compare [CN:106:32000:20:0]/[SPEEX:99:16000:20:42200] > 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec > Compare [CN:105:16000:20:0]/[SPEEX:99:16000:20:42200] > 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec > Compare [CN:13:8000:20:0]/[SPEEX:99:16000:20:42200] > 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec > Compare [telephone-event:126:8000:20:0]/[SPEEX:99:16000:20:42200] > 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5268 Set 2833 dtmf > send/recv payload to 126 > 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec > Compare [ISAC:103:16000:30:0]/[SPEEX:99:8000:20:24600] > 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec > Compare [ISAC:104:32000:30:0]/[SPEEX:99:8000:20:24600] > 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec > Compare [PCMU:0:8000:20:64000]/[SPEEX:99:8000:20:24600] > 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec > Compare [PCMA:8:8000:20:64000]/[SPEEX:99:8000:20:24600] > 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec > Compare [CN:106:32000:20:0]/[SPEEX:99:8000:20:24600] > 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec > Compare [CN:105:16000:20:0]/[SPEEX:99:8000:20:24600] > 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec > Compare [CN:13:8000:20:0]/[SPEEX:99:8000:20:24600] > 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec > Compare [telephone-event:126:8000:20:0]/[SPEEX:99:8000:20:24600] > 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5268 Set 2833 dtmf > send/recv payload to 126 > 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec > Compare [ISAC:103:16000:30:0]/[PCMU:0:8000:20:64000] > 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec > Compare [ISAC:104:32000:30:0]/[PCMU:0:8000:20:64000] > 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec > Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] > 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5161 Bah HUMBUG! > Sticking with PCMU at 8000h@20i > > As you can see, it first compares all offered codecs against opus/48k, > next skips isac completely, then goes to speex/16k, speex/8k, and > PCMU, and eventually grudgingly accepts PCMU. > > Interestingly, if I drop opus from the list, and leave the rest as-is, > it does select isac/32k, but weirdly: > > 2013-02-16 01:51:10.691314 [DEBUG] sofia_glue.c:5139 Audio Codec > Compare [ISAC:103:16000:30:0]/[isac:99:32000:30:160000] > 2013-02-16 01:51:10.691314 [DEBUG] sofia_glue.c:5161 Bah HUMBUG! > Sticking with isac at 32000h@30i > > It picks isac/32k after only examining the isac/16k line in the SDP, > before it even gets to isac/32k. (In fact those are the only 2 lines > after the SDP is printed; it makes the decision after making just one > comparison.) > > Any ideas? Am I just misunderstanding how this works? > > Thanks, > Brian From a.venugopan at mundio.com Sat Feb 16 16:05:27 2013 From: a.venugopan at mundio.com (Archana Venugopan) Date: Sat, 16 Feb 2013 13:05:27 +0000 Subject: [Freeswitch-users] Multicast in freeswitch In-Reply-To: References: <592A9CF93E12394E8472A6CC66E66BF236B788@Mail-Kilo.squay.com> Message-ID: <592A9CF93E12394E8472A6CC66E66BF23790FC@Mail-Kilo.squay.com> Hi, Thanks. I read mod_event_socket wiki, it has some format like this sendmsg call-command: hangup hangup-cause: Should I need to pass on in my script? And how am i to use that hangup message in lua script? Sorry I am not clear from wiki on these. Please help. Thanks Regards, Archana From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Eliot Gable Sent: 16 February 2013 06:11 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Multicast in freeswitch You can use mod_event_socket and watch for SWITCH_CHANNEL_STATE, SWITCH_CHANNEL_CALLSTATE, SWITCH_CHANNEL_CREATE, SWITCH_CHANNEL_ORIGINATE, SWITCH_CHANNEL_BRIDGE, and any other event of interest. Then you can track the state of each call for whatever purpose you might want to use that info for. On Feb 14, 2013 12:31 PM, "Archana Venugopan" > wrote: Hi, I would like to get 'Orig-status' event in freeswitch so that I will know when the phone is being HANGED. But to get that event I read in wiki I need mod_event_multicast. http://wiki.freeswitch.org/wiki/Mod_event_multicast But I don't see in above wiki on how to compile and load mod_event_multicast. Can anyone please let me know how to compile and load multicast? Regards, Archana _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130216/8c6bd464/attachment-0001.html From admin at blindi.net Sat Feb 16 16:34:11 2013 From: admin at blindi.net (Thomas Hoellriegel) Date: Sat, 16 Feb 2013 14:34:11 +0100 (CET) Subject: [Freeswitch-users] Problem skype calls very bad in quality In-Reply-To: References: Message-ID: Hi all, i have installed skype for linux from the wiki-page. These works fine. My problem: when i place incoming or outboundcalls the sound is very clearlly. The quality is for example a very bad gsm mobile phone line. My Fs works in a datacenter 100mbit network. I have setup a free german voicechat: dorf_telefonchat I demonstrate the problem: Then you call the chat, your heare the bad ton quality. Can your help please? thanks. --------------- Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: http://www.blindi.net/callback homepage: http://www.blindi.net blinde-misc mailingliste f?r blinde. anmeldung unter: http://www.blindi.net/mailman/listinfo/blinde-misc From egable+freeswitch at gmail.com Sat Feb 16 16:45:50 2013 From: egable+freeswitch at gmail.com (Eliot Gable) Date: Sat, 16 Feb 2013 08:45:50 -0500 Subject: [Freeswitch-users] FreeSWITCH HA BoF Meetings In-Reply-To: References: Message-ID: I have the kids from 6pm Eastern until they go to bed at 8pm. On Sat, Feb 16, 2013 at 1:06 AM, Ken Rice wrote: > Is earlier and option? Say like 6 EST or something? > > > > On 2/15/13 11:58 PM, "Eliot Gable" wrote: > > I am usually just getting my kids in bed at 8pm, so 8:15pm is typically > better timing. > On Feb 15, 2013 7:26 PM, "Michael Collins" wrote: > > will do. > > On Fri, Feb 15, 2013 at 1:01 PM, Giovanni Maruzzelli > wrote: > > ouch! at 08:00pm EST will be 02:00am my time. > Not sure I'll be able to attend the live discussion. Please record it as > always, please :) > > -giovanni > > > On Fri, Feb 15, 2013 at 9:59 PM, Giovanni Maruzzelli > wrote: > > I'm very much interested, thanks Eliot for your efforts so far. > > -giovanni > > On Thu, Feb 14, 2013 at 11:41 PM, Michael Collins > wrote: > > Hello! > > Yesterday we had a nice discussion < > http://wiki.freeswitch.org/wiki/FS_weekly_2013_02_13> about all things > HA. Eliot Gable gave us a lot of very good information about the state of > mod_ha_development and all the work that he's done over the past 4+ years > in the world of HA and FreeSWITCH. > > > It's now time to let those who have a vested interest in HA to get > together with Eliot and continuing the discussion. To that end we propose a > FreeSWITCH HA BoF (Birds of a Feather) conference. Eliot says that he can > be available after 8PM EST. I would like to propose that the meetup be > Tuesday evenings at 8:00PM EST in the main FreeSWITCH public conference. > > If you would like to meetup then please email me off-list and CC Eliot ( > egable+freeswitch at gmail.com > > ) so that we can keep track of who is most interested in this topic. > > Thanks! > > > -- > Ken > > *http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > * > irc.freenode.net #freeswitch > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Eliot Gable "We do not inherit the Earth from our ancestors: we borrow it from our children." ~David Brower "I decided the words were too conservative for me. We're not borrowing from our children, we're stealing from them--and it's not even considered to be a crime." ~David Brower "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to live; not live to eat.) ~Marcus Tullius Cicero -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130216/f49836b9/attachment.html From egable+freeswitch at gmail.com Sat Feb 16 16:59:09 2013 From: egable+freeswitch at gmail.com (Eliot Gable) Date: Sat, 16 Feb 2013 08:59:09 -0500 Subject: [Freeswitch-users] Multicast in freeswitch In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF23790FC@Mail-Kilo.squay.com> References: <592A9CF93E12394E8472A6CC66E66BF236B788@Mail-Kilo.squay.com> <592A9CF93E12394E8472A6CC66E66BF23790FC@Mail-Kilo.squay.com> Message-ID: If you are using LUA, you probably want setHangupHook to do whatever you are doing. It kind of depends on what you are doing though. Perhaps you should provide more details about what you are trying to do and how you are trying to do it. On Sat, Feb 16, 2013 at 8:05 AM, Archana Venugopan wrote: > Hi,**** > > ** ** > > Thanks. I read mod_event_socket wiki, it has some format like this**** > > ** ** > > sendmsg **** > > call-command: hangup**** > > hangup-cause: **** > > ** ** > > ** ** > > Should I need to pass on in my script? And how am i to use that hangup > message in lua script? Sorry I am not clear from wiki on these. Please > help. Thanks**** > > ** ** > > Regards,**** > > Archana**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Eliot Gable > *Sent:* 16 February 2013 06:11 > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Multicast in freeswitch**** > > ** ** > > You can use mod_event_socket and watch for SWITCH_CHANNEL_STATE, > SWITCH_CHANNEL_CALLSTATE, SWITCH_CHANNEL_CREATE, SWITCH_CHANNEL_ORIGINATE, > SWITCH_CHANNEL_BRIDGE, and any other event of interest. Then you can track > the state of each call for whatever purpose you might want to use that info > for.**** > > On Feb 14, 2013 12:31 PM, "Archana Venugopan" > wrote:**** > > Hi,**** > > I would like to get ?Orig-status? event in freeswitch so that I will know > when the phone is being HANGED. But to get that event I read in wiki I need > mod_event_multicast.**** > > **** > > http://wiki.freeswitch.org/wiki/Mod_event_multicast**** > > **** > > But I don?t see in above wiki on how to compile and load > mod_event_multicast. Can anyone please let me know how to compile and load > multicast?**** > > **** > > Regards,**** > > Archana**** > > **** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Eliot Gable "We do not inherit the Earth from our ancestors: we borrow it from our children." ~David Brower "I decided the words were too conservative for me. We're not borrowing from our children, we're stealing from them--and it's not even considered to be a crime." ~David Brower "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to live; not live to eat.) ~Marcus Tullius Cicero -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130216/2dff7e1c/attachment.html From a.venugopan at mundio.com Sat Feb 16 17:08:43 2013 From: a.venugopan at mundio.com (Archana Venugopan) Date: Sat, 16 Feb 2013 14:08:43 +0000 Subject: [Freeswitch-users] Multicast in freeswitch In-Reply-To: References: <592A9CF93E12394E8472A6CC66E66BF236B788@Mail-Kilo.squay.com> <592A9CF93E12394E8472A6CC66E66BF23790FC@Mail-Kilo.squay.com> Message-ID: <592A9CF93E12394E8472A6CC66E66BF2379123@Mail-Kilo.squay.com> Hi, I want to update my Database table, based on when the phone is ACTIVE and phone is HANGUP. I just want to get these 2 events ACTIVE and HANGUP in my lua script, so that I can put a if conditions to update my Database table. So need to know what exactly I should do to get these events in lua script. I tried this 'local Channel_State = params:getHeader("Channel-State")' but this does not give me these 2 events. Since I could not understand in detail from wiki I need suggestion on what can be done. Many thanks. Regards, Archana From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Eliot Gable Sent: 16 February 2013 13:59 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Multicast in freeswitch If you are using LUA, you probably want setHangupHook to do whatever you are doing. It kind of depends on what you are doing though. Perhaps you should provide more details about what you are trying to do and how you are trying to do it. On Sat, Feb 16, 2013 at 8:05 AM, Archana Venugopan > wrote: Hi, Thanks. I read mod_event_socket wiki, it has some format like this sendmsg call-command: hangup hangup-cause: Should I need to pass on in my script? And how am i to use that hangup message in lua script? Sorry I am not clear from wiki on these. Please help. Thanks Regards, Archana From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Eliot Gable Sent: 16 February 2013 06:11 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Multicast in freeswitch You can use mod_event_socket and watch for SWITCH_CHANNEL_STATE, SWITCH_CHANNEL_CALLSTATE, SWITCH_CHANNEL_CREATE, SWITCH_CHANNEL_ORIGINATE, SWITCH_CHANNEL_BRIDGE, and any other event of interest. Then you can track the state of each call for whatever purpose you might want to use that info for. On Feb 14, 2013 12:31 PM, "Archana Venugopan" > wrote: Hi, I would like to get 'Orig-status' event in freeswitch so that I will know when the phone is being HANGED. But to get that event I read in wiki I need mod_event_multicast. http://wiki.freeswitch.org/wiki/Mod_event_multicast But I don't see in above wiki on how to compile and load mod_event_multicast. Can anyone please let me know how to compile and load multicast? Regards, Archana _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -- Eliot Gable "We do not inherit the Earth from our ancestors: we borrow it from our children." ~David Brower "I decided the words were too conservative for me. We're not borrowing from our children, we're stealing from them--and it's not even considered to be a crime." ~David Brower "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to live; not live to eat.) ~Marcus Tullius Cicero -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130216/84454bd6/attachment-0001.html From gmaruzz at gmail.com Sat Feb 16 17:09:49 2013 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Sat, 16 Feb 2013 15:09:49 +0100 Subject: [Freeswitch-users] FreeSWITCH HA BoF Meetings In-Reply-To: References: Message-ID: We in Europe and Asia will be better served by conference calls between your 09:00 and 14:00 EST (would be 15:00 - 20:00 CEST), maybe during weekend. But we old civilizations understand things, so we'll happily listen to recordings, and then post our comments in the mailing list :) -giovanni On Sat, Feb 16, 2013 at 2:45 PM, Eliot Gable wrote: > I have the kids from 6pm Eastern until they go to bed at 8pm. > > On Sat, Feb 16, 2013 at 1:06 AM, Ken Rice wrote: > >> Is earlier and option? Say like 6 EST or something? >> >> >> >> On 2/15/13 11:58 PM, "Eliot Gable" wrote: >> >> I am usually just getting my kids in bed at 8pm, so 8:15pm is typically >> better timing. >> On Feb 15, 2013 7:26 PM, "Michael Collins" wrote: >> >> will do. >> >> On Fri, Feb 15, 2013 at 1:01 PM, Giovanni Maruzzelli >> wrote: >> >> ouch! at 08:00pm EST will be 02:00am my time. >> Not sure I'll be able to attend the live discussion. Please record it as >> always, please :) >> >> -giovanni >> >> >> On Fri, Feb 15, 2013 at 9:59 PM, Giovanni Maruzzelli >> wrote: >> >> I'm very much interested, thanks Eliot for your efforts so far. >> >> -giovanni >> >> On Thu, Feb 14, 2013 at 11:41 PM, Michael Collins >> wrote: >> >> Hello! >> >> Yesterday we had a nice discussion < >> http://wiki.freeswitch.org/wiki/FS_weekly_2013_02_13> about all things >> HA. Eliot Gable gave us a lot of very good information about the state of >> mod_ha_development and all the work that he's done over the past 4+ years >> in the world of HA and FreeSWITCH. >> >> >> It's now time to let those who have a vested interest in HA to get >> together with Eliot and continuing the discussion. To that end we propose a >> FreeSWITCH HA BoF (Birds of a Feather) conference. Eliot says that he can >> be available after 8PM EST. I would like to propose that the meetup be >> Tuesday evenings at 8:00PM EST in the main FreeSWITCH public conference. >> >> If you would like to meetup then please email me off-list and CC Eliot ( >> egable+freeswitch at gmail.com > >> ) so that we can keep track of who is most interested in this topic. >> >> Thanks! >> >> >> -- >> Ken >> >> *http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> * >> irc.freenode.net #freeswitch >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > > -- > Eliot Gable > > "We do not inherit the Earth from our ancestors: we borrow it from our > children." ~David Brower > > "I decided the words were too conservative for me. We're not borrowing > from our children, we're stealing from them--and it's not even considered > to be a crime." ~David Brower > > "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to live; > not live to eat.) ~Marcus Tullius Cicero > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130216/befe3b70/attachment.html From egable+freeswitch at gmail.com Sat Feb 16 19:51:42 2013 From: egable+freeswitch at gmail.com (Eliot Gable) Date: Sat, 16 Feb 2013 11:51:42 -0500 Subject: [Freeswitch-users] FreeSWITCH HA BoF Meetings In-Reply-To: References: Message-ID: 10:00am EST on Sundays is the only time on the weekend during the timeframe you listed where I can guarantee availability for 1.5 - 2 hrs. On Sat, Feb 16, 2013 at 9:09 AM, Giovanni Maruzzelli wrote: > We in Europe and Asia will be better served by conference calls between > your 09:00 and 14:00 EST (would be 15:00 - 20:00 CEST), maybe during > weekend. > > But we old civilizations understand things, so we'll happily listen to > recordings, and then post our comments in the mailing list :) > > -giovanni > > > On Sat, Feb 16, 2013 at 2:45 PM, Eliot Gable wrote: > >> I have the kids from 6pm Eastern until they go to bed at 8pm. >> >> On Sat, Feb 16, 2013 at 1:06 AM, Ken Rice wrote: >> >>> Is earlier and option? Say like 6 EST or something? >>> >>> >>> >>> On 2/15/13 11:58 PM, "Eliot Gable" wrote: >>> >>> I am usually just getting my kids in bed at 8pm, so 8:15pm is typically >>> better timing. >>> On Feb 15, 2013 7:26 PM, "Michael Collins" wrote: >>> >>> will do. >>> >>> On Fri, Feb 15, 2013 at 1:01 PM, Giovanni Maruzzelli >>> wrote: >>> >>> ouch! at 08:00pm EST will be 02:00am my time. >>> Not sure I'll be able to attend the live discussion. Please record it as >>> always, please :) >>> >>> -giovanni >>> >>> >>> On Fri, Feb 15, 2013 at 9:59 PM, Giovanni Maruzzelli >>> wrote: >>> >>> I'm very much interested, thanks Eliot for your efforts so far. >>> >>> -giovanni >>> >>> On Thu, Feb 14, 2013 at 11:41 PM, Michael Collins >>> wrote: >>> >>> Hello! >>> >>> Yesterday we had a nice discussion < >>> http://wiki.freeswitch.org/wiki/FS_weekly_2013_02_13> about all things >>> HA. Eliot Gable gave us a lot of very good information about the state of >>> mod_ha_development and all the work that he's done over the past 4+ years >>> in the world of HA and FreeSWITCH. >>> >>> >>> It's now time to let those who have a vested interest in HA to get >>> together with Eliot and continuing the discussion. To that end we propose a >>> FreeSWITCH HA BoF (Birds of a Feather) conference. Eliot says that he can >>> be available after 8PM EST. I would like to propose that the meetup be >>> Tuesday evenings at 8:00PM EST in the main FreeSWITCH public conference. >>> >>> If you would like to meetup then please email me off-list and CC Eliot ( >>> egable+freeswitch at gmail.com > >>> ) so that we can keep track of who is most interested in this topic. >>> >>> Thanks! >>> >>> >>> -- >>> Ken >>> >>> *http://www.FreeSWITCH.org >>> http://www.ClueCon.com >>> http://www.OSTAG.org >>> * >>> irc.freenode.net #freeswitch >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> >> -- >> Eliot Gable >> >> "We do not inherit the Earth from our ancestors: we borrow it from our >> children." ~David Brower >> >> "I decided the words were too conservative for me. We're not borrowing >> from our children, we're stealing from them--and it's not even considered >> to be a crime." ~David Brower >> >> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to live; >> not live to eat.) ~Marcus Tullius Cicero >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > > -- > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Eliot Gable "We do not inherit the Earth from our ancestors: we borrow it from our children." ~David Brower "I decided the words were too conservative for me. We're not borrowing from our children, we're stealing from them--and it's not even considered to be a crime." ~David Brower "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to live; not live to eat.) ~Marcus Tullius Cicero -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130216/c64682c2/attachment-0001.html From egable+freeswitch at gmail.com Sat Feb 16 19:56:33 2013 From: egable+freeswitch at gmail.com (Eliot Gable) Date: Sat, 16 Feb 2013 11:56:33 -0500 Subject: [Freeswitch-users] Multicast in freeswitch In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF2379123@Mail-Kilo.squay.com> References: <592A9CF93E12394E8472A6CC66E66BF236B788@Mail-Kilo.squay.com> <592A9CF93E12394E8472A6CC66E66BF23790FC@Mail-Kilo.squay.com> <592A9CF93E12394E8472A6CC66E66BF2379123@Mail-Kilo.squay.com> Message-ID: You still have not provided enough information. Is your LUA script stand-alone, or is it being launched from FreeSWITCH? How are you launching? Where are you launching it? Is it being launched from the dialplan or from an API call? Is the script just to update a field in the database or does it do more than that? On Sat, Feb 16, 2013 at 9:08 AM, Archana Venugopan wrote: > Hi,**** > > ** ** > > I want to update my Database table, based on when the phone is ACTIVE and > phone is HANGUP. I just want to get these 2 events ACTIVE and HANGUP in my > lua script, so that I can put a if conditions to update my Database table. > **** > > So need to know what exactly I should do to get these events in lua > script. I tried this ?local Channel_State = > params:getHeader("Channel-State")? but this does not give me these 2 > events. **** > > Since I could not understand in detail from wiki I need suggestion on what > can be done. **** > > ** ** > > Many thanks.**** > > ** ** > > Regards,**** > > Archana**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Eliot Gable > *Sent:* 16 February 2013 13:59 > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Multicast in freeswitch**** > > ** ** > > If you are using LUA, you probably want setHangupHook to do whatever you > are doing. It kind of depends on what you are doing though. Perhaps you > should provide more details about what you are trying to do and how you are > trying to do it.**** > > ** ** > > On Sat, Feb 16, 2013 at 8:05 AM, Archana Venugopan > wrote:**** > > Hi,**** > > **** > > Thanks. I read mod_event_socket wiki, it has some format like this**** > > **** > > sendmsg **** > > call-command: hangup**** > > hangup-cause: **** > > **** > > **** > > Should I need to pass on in my script? And how am i to use that hangup > message in lua script? Sorry I am not clear from wiki on these. Please > help. Thanks**** > > **** > > Regards,**** > > Archana**** > > **** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Eliot Gable > *Sent:* 16 February 2013 06:11 > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Multicast in freeswitch**** > > **** > > You can use mod_event_socket and watch for SWITCH_CHANNEL_STATE, > SWITCH_CHANNEL_CALLSTATE, SWITCH_CHANNEL_CREATE, SWITCH_CHANNEL_ORIGINATE, > SWITCH_CHANNEL_BRIDGE, and any other event of interest. Then you can track > the state of each call for whatever purpose you might want to use that info > for.**** > > On Feb 14, 2013 12:31 PM, "Archana Venugopan" > wrote:**** > > Hi,**** > > I would like to get ?Orig-status? event in freeswitch so that I will know > when the phone is being HANGED. But to get that event I read in wiki I need > mod_event_multicast.**** > > **** > > http://wiki.freeswitch.org/wiki/Mod_event_multicast**** > > **** > > But I don?t see in above wiki on how to compile and load > mod_event_multicast. Can anyone please let me know how to compile and load > multicast?**** > > **** > > Regards,**** > > Archana**** > > **** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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**** > > > > **** > > ** ** > > -- > Eliot Gable > > "We do not inherit the Earth from our ancestors: we borrow it from our > children." ~David Brower > > "I decided the words were too conservative for me. We're not borrowing > from our children, we're stealing from them--and it's not even considered > to be a crime." ~David Brower > > "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to live; > not live to eat.) ~Marcus Tullius Cicero **** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Eliot Gable "We do not inherit the Earth from our ancestors: we borrow it from our children." ~David Brower "I decided the words were too conservative for me. We're not borrowing from our children, we're stealing from them--and it's not even considered to be a crime." ~David Brower "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to live; not live to eat.) ~Marcus Tullius Cicero -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130216/ba928529/attachment.html From geocooper at gmail.com Sat Feb 16 21:48:41 2013 From: geocooper at gmail.com (George Cooper) Date: Sat, 16 Feb 2013 13:48:41 -0500 Subject: [Freeswitch-users] NAT settings Message-ID: <5CEABE87-BD3C-4E34-9C64-8D26E2130F89@gmail.com> Hi, I just started trying to work with freeswitch using the FS 1.06 book( although I am running FS ver 1.2.6+git.) , FS cookbook,, and wiki. I and have a question regarding NAT traversal. I have been stuck trying to get phones to register and be able to call each other using ext 1000 to 1002 for example. I couldn't get the phones to register until I read an archive regarding polycom NAT issues. The freeswitch server I'm connecting to is hosted on a remote public IP. I'm using a polycom 335, grandstream xpx 2020, and an x-lite client on a Local 192 network. The below settings seem to have fixed my registration issue. Is there a reason that I would NOT set the parameter settings in sip_profiles/internal.xml to true: and to add the variable "sip-force-contact" for each user in directory/default/1000.xml - 1019.xml ? Sorry if this has been gone over, but I'm not sure if there is something I'm missing because the wiki ( http://wiki.freeswitch.org/wiki/NAT ) says "NAT just works" and the internal.xml file says ""... Thanks for any clarity you can provide, Geo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130216/8f9b2822/attachment-0001.html From lists at kavun.ch Sun Feb 17 02:10:08 2013 From: lists at kavun.ch (Emrah) Date: Sat, 16 Feb 2013 18:10:08 -0500 Subject: [Freeswitch-users] Changing codec during calls Message-ID: <6EF8B8E9-4CC5-480C-B1C1-09CBC533470C@kavun.ch> Hi all, How do we go about renegotiating the codecs of a call that is already established? I tried uuid_media_reneg to no avail. E.g.: Upgrade someone who's entered a 16k conference from PCMU to G722. Cheers and thanks, Emrah From lloyd.aloysius at gmail.com Sun Feb 17 05:58:01 2013 From: lloyd.aloysius at gmail.com (Lloyd Aloysius) Date: Sat, 16 Feb 2013 21:58:01 -0500 Subject: [Freeswitch-users] mod_directory - Cannot locate domain Message-ID: Hi All I have a strange problem with the mod_directory. I use xml_curl for all the applications. Only mod_directory giving this trouble. EXECUTE sofia/sipinterface_1/marv at mydomain.com directory(mydomain.com mydomain.com) 2013-02-16 21:43:04.601315 [DEBUG] mod_directory.c:518 [mydomain.com] rwlock 2013-02-16 21:43:04.621318 [ERR] switch_xml.c:1650 Error[[error near line 1]: unclosed tag ] 2013-02-16 21:43:04.621318 [WARNING] mod_directory.c:545 Cannot locate domain mydomain.com -- my direcotroy.conf xml curl output
Any help is appreciated. Thanks Lloyd -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130216/5b84e333/attachment.html From anthony.minessale at gmail.com Sun Feb 17 08:37:51 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 16 Feb 2013 23:37:51 -0600 Subject: [Freeswitch-users] ooH323 vs mod_h323 In-Reply-To: <77B9FF50-213F-4BF6-917E-FBB006DF5C76@gmail.com> References: <2BD0DC60-D428-4323-A020-D9AD20E76553@freeswitch.org> <4C2EA249.4070704@coppice.org> <267C8398-6CA3-442E-B973-BE096D728B28@freeswitch.org> <77B9FF50-213F-4BF6-917E-FBB006DF5C76@gmail.com> Message-ID: Fun fact... Guess who Digium paid to hash out the skeleton for that ooh323 mod....?? ;) On Feb 15, 2013 7:01 PM, "Steven Ayre" wrote: > mod_opal and mod_h323 are based on the opal and h323plus libraries. Both > are forks of the open323 library (which is discontinued). h323plus aims to > remain API compatible with open323, while it seems opal aims to become a > more generic telecoms library with support for SIP IAX etc too. > > ooh323c is a completely separate library with no shared code. > > mod_h323 and mod_opal both seem to have stability problems. Partly that > seems to come from the heavy dependance of the libraries on specific > matching versions of ptlib, partly that seems to be the libraries > themselves. Both Asterisk and YATE appear to have stability problems using > h323plus/openh323. Anecdotally the ooh323c Asterisk module is more stable... > > I'd need to look into it further, but the ooh323c library implements H323 > and leaves RTP/codecs to the app. h323plus/opal implement them themselves > (overridable I imagine). That means adding support to ooh323c may end up > being easier, in theory. But that's a guess without researching it... > > Steve > > > > On 15 Feb 2013, at 23:36, Seven Du wrote: > > So we basically have 3 options, Opal, h323 and ooh323c, what's the > difference among them? > > Also, I'm aware that video is not supported in the current mod_opal and > mod_h323, is it possible(or how hard) to add video support? Or does > ooh323c easy to support video? > > We'd like to contribute a little money to speed it up if it can support > video and some want to work on it. > > -- > Seven Du > http://www.freeswitch.org.cn > http://about.me/dujinfang > http://www.dujinfang.com > > Sent with Sparrow > > On Friday, February 15, 2013 at 11:37 PM, Steven Ayre wrote: > > There's a module for Asterisk but none for FS I'm aware of yet - I'm only > wondering about the feasibility of writing one. > > -Steve > > > > On 15 February 2013 14:48, Brian West wrote: > > So I have the impression you or someone has an ooh323 module and hasn't > contributed it back to the project yet? > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH_Wire > T: +1.918.420.9266 | F: +1.918.420.9267 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9266 > UK: +44 20 3298 4900 > ISN: 410*543 > > > > > > On Feb 15, 2013, at 3:49 AM, Steven Ayre wrote: > > > Hi Brian, > > > > The issue would be that Asterisk's modifications are on a version > predating the FLOSS exemption and their project is GPL - so presumably > their modifications are all GPL-only? > > > > -Steve > > > > > > > > On 15 February 2013 00:40, Brian West wrote: > > If someone wanted to build ooh323 module for FreeSWITCH we do have > permission to do so using the lib. > > > > /b > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130216/b5a16813/attachment.html From sparklezou at 163.com Sun Feb 17 11:34:33 2013 From: sparklezou at 163.com (sparklezou) Date: Sun, 17 Feb 2013 16:34:33 +0800 Subject: [Freeswitch-users] About the "hardware fax detection" on Sangoma A101, support on Freeswitch or NOT Message-ID: <40abdc6c.1078.13ce74a50ad.Coremail.sparklezou@163.com> Hi All, The Sangoma A101 board support "hardware fax detection". It configured in /etc/wanpipe/wanpipe1.conf configureation file. TDMV_HW_FAX_DETECT = YES # YES: receive fax 1100hz events from hardware Is there any configuration on the dial plan, if detect the fax, goes to the predefined dial plan context. something the same in asterisk http://wiki.sangoma.com/wanpipe-linux-asterisk-fax-detect Thanks! 2013-02-17 sparklezou -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130217/c9ca94be/attachment-0001.html From joakim+freeswitch-users at gissberg.nu Sun Feb 17 10:29:52 2013 From: joakim+freeswitch-users at gissberg.nu (J G) Date: Sun, 17 Feb 2013 17:29:52 +1000 Subject: [Freeswitch-users] Problems setting up mod_gsmopen Message-ID: Hi! I am trying to figure out how to get my E173 to work with Freeswitch. I accually had it working a few months ago and don't know if I have changed anything to break it. But atleast we know it is proven to have worked and has voice unlocked. The interface registers in gsmopen very randomly, something I have read is a problem for others too. Usually it connects to the provider via USB2, and sometimes USB0, but it is very random. This is not my main problem though, it is to get it working at all. On the (random) ring interface I get this output via picocom if I call the number: RING +CLIP: "+61468440XXX",145,,,,0 That should be the "controldevice_name", right? The other two are silent, but I did get some scrambled output on one, ...once. If I get it to connect to the network, and try to call out, I get nothing on one interface, and an error if I select the other: 19.218876 [ERR] switch_core_io.c:806 gsmopen/ANY/148 has no write codec. What does that error try to explain to me? Sorry for a very messy post, but everything feels so random and I can't even reproduce for good logfiles. Make a call, sighup the logfile, and the next call 10 seconds later the output is something totaly different, it's that random! I can work on proper logfiles if it's needed. - J -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130217/27019f52/attachment.html From emamirazavi at gmail.com Sun Feb 17 11:31:46 2013 From: emamirazavi at gmail.com (Sayyed Mohammad Emami Razavi) Date: Sun, 17 Feb 2013 12:01:46 +0330 Subject: [Freeswitch-users] Auto attendant :: act on no digit entry Message-ID: I want to transfer customers to operator(ext=100) on no digit entry after autoattendant with xml IVR! can anybody help me?! Blue.box apparently can not do this task! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130217/3cfe440e/attachment.html From covici at ccs.covici.com Sun Feb 17 13:00:57 2013 From: covici at ccs.covici.com (covici at ccs.covici.com) Date: Sun, 17 Feb 2013 05:00:57 -0500 Subject: [Freeswitch-users] Auto attendant :: act on no digit entry In-Reply-To: References: Message-ID: <24015.1361095257@ccs.covici.com> The only way I have found to do this is to have something right after the execute menu in your dialplan which will transfer or do whatever you want, this will activate after the timeout expires. Sayyed Mohammad Emami Razavi wrote: > I want to transfer customers to operator(ext=100) on no digit entry after > autoattendant with xml IVR! > can anybody help me?! Blue.box apparently can not do this task! > > ---------------------------------------------------- > Alternatives: > > ---------------------------------------------------- > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici at ccs.covici.com From gmaruzz at gmail.com Sun Feb 17 13:31:52 2013 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Sun, 17 Feb 2013 11:31:52 +0100 Subject: [Freeswitch-users] Problems setting up mod_gsmopen In-Reply-To: References: Message-ID: I would tell you to restart from scratch, and follow strictly the gsmopen wiki page. You probably messed up the config. After having done that (eg: refollow the steps in wiki page and reinstall a config file) if you still have problems post here BOTH your config file, a debug output, and all info requested in the section "report bugs and problems" of the wikki page. No one can help you without those info. -giovanni On 2/17/13, J G wrote: > Hi! > > I am trying to figure out how to get my E173 to work with Freeswitch. I > accually had it working a few months ago and don't know if I have changed > anything to break it. But atleast we know it is proven to have worked and > has voice unlocked. > > The interface registers in gsmopen very randomly, something I have read is > a problem for others too. Usually it connects to the provider via USB2, and > sometimes USB0, but it is very random. This is not my main problem though, > it is to get it working at all. > > On the (random) ring interface I get this output via picocom if I call the > number: > > RING > > +CLIP: "+61468440XXX",145,,,,0 > > That should be the "controldevice_name", right? > > The other two are silent, but I did get some scrambled output on one, > ...once. > > If I get it to connect to the network, and try to call out, I get nothing > on one interface, and an error if I select the other: > > 19.218876 [ERR] switch_core_io.c:806 gsmopen/ANY/148 has no write codec. > > What does that error try to explain to me? > > Sorry for a very messy post, but everything feels so random and I can't > even reproduce for good logfiles. Make a call, sighup the logfile, and the > next call 10 seconds later the output is something totaly different, it's > that random! > > I can work on proper logfiles if it's needed. > > - J > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From gmaruzz at gmail.com Sun Feb 17 13:44:36 2013 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Sun, 17 Feb 2013 11:44:36 +0100 Subject: [Freeswitch-users] Problem skype calls very bad in quality In-Reply-To: References: Message-ID: be sure to use 16khz sounds etc. Skype is 16khz. Skype call would normally be superior in quality respect to a gsm call. If you are using conference, be sure to be at 16khz. Other than that, follow the wiki strictly, particularly about which distro, and don't use virtual machines (eg real hardware has better timing) or openvz containers. Obviously, don't use weak or overloaded machines, and not 32 bit, or (horror!) machines with a graphic interface :) On 2/16/13, Thomas Hoellriegel wrote: > Hi all, i have installed skype for linux from the wiki-page. > These works fine. > My problem: when i place incoming or outboundcalls the sound is very > clearlly. > The quality is for example a very bad gsm mobile phone line. > My Fs works in a datacenter 100mbit network. > I have setup a free german voicechat: > dorf_telefonchat > I demonstrate the problem: > Then you call the chat, your heare the bad ton quality. > > Can your help please? > thanks. > > > --------------- > Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: > http://www.blindi.net/callback > homepage: http://www.blindi.net > blinde-misc mailingliste f?r blinde. anmeldung unter: > http://www.blindi.net/mailman/listinfo/blinde-misc > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From sparklezou at 163.com Sun Feb 17 15:38:08 2013 From: sparklezou at 163.com (sparklezou) Date: Sun, 17 Feb 2013 20:38:08 +0800 Subject: [Freeswitch-users] Please suggest one "SIP Provider" Message-ID: <3852fa31.3c24.13ce8295481.Coremail.sparklezou@163.com> Hi All, Could you please share your experience of the "SIP provider"? Which one is good at the quality and price? Thanks! 2013-02-17 sparklezou -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130217/89a5d5fd/attachment.html From cal.leeming at simplicitymedialtd.co.uk Sun Feb 17 17:47:17 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Sun, 17 Feb 2013 14:47:17 +0000 Subject: [Freeswitch-users] Please suggest one "SIP Provider" In-Reply-To: <3852fa31.3c24.13ce8295481.Coremail.sparklezou@163.com> References: <3852fa31.3c24.13ce8295481.Coremail.sparklezou@163.com> Message-ID: Hello, I assume your referring to SIP trunk providers. Here is a recent thread for inbound DID providers; http://freeswitch-users.2379917.n2.nabble.com/DID-providers-any-thoughts-jan-2013-td7586851.html As for outbound traffic, it entirely depends on; * Which countries you wish to target * Where your own server is located * What service level you are trying to achieve * How much you have to spend * Are you looking or ingress (inbound) or egress (outbound) >From the perspective of the UK, here is a brief breakdown of the categories (from best to worst); * Direct with a state/national telecoms company (such as BT or Kingston) - usually come with minimum commitment levels and very high set up/monthly costs, suitable only for large existing customer bases * Through a wholesaler that uses only the above routes (such as Simwood) - they give lower traffic customers the ability to use these high quality routes without the cost.. yes they are reselling, but it's one of the rare occasions where reselling it is justified, as you will save a lot of money in several circumstances * Through a reseller of a reseller of the above (always ask the company what routes they use, and try to cut out the useless middleman) * Through a reseller of a reseller of a reseller of a reseller of a reseller that is using grey routes (seriously, this really does happen a lot) It is also worth noting that providers are known to lie about the quality and sources of their routes, so rely heavily on community reviews to figure out if a company is good. And, even if a company is reselling wholesaler routes, you must make sure you perform inter-op right from the beginning, else you might hit some fun surprises along the way (we have been stung by this before). Remember, crappy cheap SIP trunk providers are 10 a penny these days, and it really depends on what you are trying to achieve. If you want to build say an automated phone alerting system that calls people with an automated electronic voice (such as PagerDuty), then using crappy routes might be acceptable for your business case. However, if you are trying to build an enterprise product for voice, then you need to make careful choices about what sacrifices (if any) you're going to make on quality vs cost. On a last note, we have trialed several "reputable" providers over the last 6 months, and most of them ended in tears - be weary. I hope this helps. Cal On Sun, Feb 17, 2013 at 12:38 PM, sparklezou wrote: > ** > ** > Hi All, > > Could you please share your experience of the "SIP provider"? > > Which one is good at the quality and price? > > Thanks! > > 2013-02-17 > ------------------------------ > sparklezou > ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130217/3cfc4ff7/attachment-0001.html From cal.leeming at simplicitymedialtd.co.uk Sun Feb 17 17:52:58 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Sun, 17 Feb 2013 14:52:58 +0000 Subject: [Freeswitch-users] NAT settings In-Reply-To: <5CEABE87-BD3C-4E34-9C64-8D26E2130F89@gmail.com> References: <5CEABE87-BD3C-4E34-9C64-8D26E2130F89@gmail.com> Message-ID: Although I can't answer your question from direct know, here are some relevant links; NDLB-force-rport (taken from http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#NDLB-force-rport ) This will force FreeSWITCH to send SIP responses to the network port from which they were received. Use at your own risk! For more information see NAT Traversal. safe = param that does force-rport behavior only on endpoints we know are safe to do so on. This is a dirty hack to try to work with certain endpoints behind sonicwall which does not use the same port when it does nat, when the devices do not support rport, while not breaking devices that acutally use different ports that force-rport will break Also found these; http://freeswitch-users.2379917.n2.nabble.com/NDLB-force-rport-safe-td5911932.html http://lists.freeswitch.org/pipermail/freeswitch-users/2011-November/077600.html http://lists.freeswitch.org/pipermail/freeswitch-users/2012-July/085658.html The last update on this was Brian West was ( http://lists.freeswitch.org/pipermail/freeswitch-users/2008-September/034336.html) Try adding this param to your sofia profile. It will break cisco phones or any other phone that follows the sip spec. This explicitly breaks RFC to accommodate broken phones. in your sofia profile. This information is somewhat out of date and there were some changes to the way NAT works since then, so I'm not 100% sure. If anyone else can chime in to confirm (possibly even Brian West himself) that'd be good. Hope this helps Cal On Sat, Feb 16, 2013 at 6:48 PM, George Cooper wrote: > Hi, > > I just started trying to work with freeswitch using the FS 1.06 book( > although I am running FS ver 1.2.6+git.) , FS cookbook,, and wiki. > > I and have a question regarding NAT traversal. > > I have been stuck trying to get phones to register and be able to call > each other using ext 1000 to 1002 for example. I couldn't get the phones to > register until I read an archive regarding polycom NAT issues. > > The freeswitch server I'm connecting to is hosted on a remote public IP. > I'm using a polycom 335, grandstream xpx 2020, and an x-lite client on a > Local 192 network. > > The below settings seem to have fixed my registration issue. > > Is there a reason that I would NOT set the parameter settings in > sip_profiles/internal.xml to true: > > > > > > > and to add the variable "sip-force-contact" for each user in > directory/default/1000.xml - 1019.xml ? > > > > > > > > > value="NDLB-connectile-dysfunction"/> > > > > > > value="$${outbound_caller_name}"/> > value="$${outbound_caller_id}"/> > > > > > > Sorry if this has been gone over, but I'm not sure if there is something > I'm missing because the wiki ( http://wiki.freeswitch.org/wiki/NAT ) says > "NAT just works" and the internal.xml file says ""... > > Thanks for any clarity you can provide, > > Geo > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130217/d3b86916/attachment.html From egable+freeswitch at gmail.com Sun Feb 17 18:50:58 2013 From: egable+freeswitch at gmail.com (Eliot Gable) Date: Sun, 17 Feb 2013 10:50:58 -0500 Subject: [Freeswitch-users] FreeSWITCH Message Bus / Shared Key Value Store In-Reply-To: References: <20130212204251.7066d944@mail.tritonwest.net> Message-ID: ZMQ also does not work with a fork, which is needed in order to execute any system commands (like iptables or anything else which has no programming API). That pretty much eliminates ZMQ as a possibility. I did some research this weekend, and of all the possibilities I could find, the one that caught my attention the most was Spread: http://www.spread.org/ There are some drawbacks to it, namely: 1) It requires any marketing material mentioning FreeSWITCH or any project / solution utilizing FreeSWITCH to also include a prepared statement about the use of the Spread toolkit. This is a fairly major licensing issue, as all commercial solutions utilizing FreeSWITCH as a core component would then also need to mention the use of the Spread toolkit within FreeSWITCH. It is essentially a "viral clause," which would be very tedious for people to make sure they honor correctly. 2) The toolkit uses a stand-alone daemon which would then need to be monitored separately from FreeSWITCH and would be another point of failure, adding complexity to the system. Basically, if that daemon were to crash, FreeSWITCH would need to know about it and would need to either respawn it or shut down. This would bring the need for Pacemaker or something similar back into the picture for any viable HA solution. Alternatively, we could write some code similar to daemontools into FreeSWTICH which respawns the daemon if it dies, but we would have to test the impact of this respawning on the overall cluster to determine if it impacts visibility of the node at any point in time. Another alternative would be to wrap the daemon into a thread inside FreeSWITCH such that if the daemon caused a segfault or something, it would force FreeSWITCH to terminate, as well. I have not done a code review of the daemon yet to determine if this is viable alternative, but assuming they have not coded it on LSD or something, it is more than likely possible. 3) They boast about how it can handle up to 8,000 1KB messages per second. I don't consider that boast-worthy. When I worked at Broadvox a few years ago, I had a FS pair which ran around 380 calls per second (760 sessions per second). Each call generates dozens of events. That hardware was getting dated when I left Broadvox, and today's hardware along with the performance improvements done to FS since then means we could conceivably have a single node which runs over 1k calls per second firing dozens of events per call. That means a single box could completely consume the message bandwidth of the entire Spread network. Imagine trying to have 64 such boxes running. We are really in need of a solution which boasts hundreds of thousands of messages per second. Spread seems like it might be off by an order of magnitude and then some. Despite these issues, Spread still seems to come closer to our needs than any other solution I found. FYI, I also looked at the Corosync IPC system, and was not at all impressed. On paper, Spread exceeds Corosync's capabilities by a fair margin. There are some strategies for mitigating issue #3 with Spread, as well. For example, we could limit messages across the Spread network to things like heartbeats and / or other HA and synchronization related messages. Basically, think of it like a D-channel on a PRI. For sending high packet per second streams of messages, we could do standard unicast connections or even try straight up mutlicasting to all nodes on the LAN. Sending heartbeats every 10ms across the Spread network would put a 64-node cluster at 6,400 messages per second just with heartbeats. That would still leave a decent amount of message bandwidth available for other types of negotiation messages and should still allow for sub-second fail-over detection and reaction. Of course, this is all assuming we can actually get 8,000 1KB messages per second out of a 64-node cluster. There are likely lots of things that impact how many messages per second Spread can handle. A lot of it has to do with network latency and CPU power. Spread uses acknowledgements and message reordering to ensure delivery in a way that accounts for things like node membership changes during the time the message is in transit and whether the message has been received by all nodes in the cluster. Network latency is probably one of the biggest factors in how many messages per second it can handle. On a faster network link, the messages per second would be higher and on a slower network, it would be lower. Obviously, CPU processing time and scheduling is important, as well. If one system is extremely overloaded and the Spread daemon is being starved for CPU resources, that will add extra latency in processing and also reduce message throughput. Obviously, this could also impact whether the node is seen as visible. So, this is one more argument for why we would need to try to run the daemon as a thread under FreeSWITCH. Then it has the same scheduling priority as FreeSWITCH and cannot be starved for resources by FreeSWITCH itself. It also would exhibit the same amount of resource starvation FreeSWITCH experiences on the node, so would more accurately reflect the state of FreeSWITCH on the node. If anyone has any other suggestions than Spread, I would like to hear it. Also, some feedback on item #1 would be great, as I cannot really judge for everyone else how willing they are to accept such a licensing clause. On Tue, Feb 12, 2013 at 8:48 PM, Jo?o Mesquita wrote: > I have used ZeroMQ in the past for this sorts of things but it really > won't be able to detect failures really fast. It is not made for this. > Maybe we can gather the requirements for such message bus? Zmq for example > provides you with this cool interface to build messaging protocols on top > of it but it does not provide reliability when it comes to endpoint to > endpoint connection without a heartbeat implemented on the user end. Can > this be used for FS as well? Anyhow, just throwing some ideas... > > Jo?o Mesquita > FreeSWITCH? Solutions > > > On Tue, Feb 12, 2013 at 5:42 PM, Dave R. Kompel wrote: > >> ** >> I've done a few experments with using both Redis, and the evil "Microsoft >> Azure Service bus" (the server on prem based version) to extend the >> eventing system to have global PUB/SUB. This way things like registrations, >> and Limit stuff could be made global. >> >> I'm looking for a way, in my carrier switch implmentation, to implment >> both HA Failover and Scaleout clustering. >> >> --Dave >> >> ------------------------------ >> *From:* Eliot Gable [mailto:egable+freeswitch at gmail.com] >> *To:* FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org >> ] >> *Sent:* Tue, 12 Feb 2013 05:49:13 -0800 >> *Subject:* [Freeswitch-users] FreeSWITCH Message Bus / Shared Key Value >> Store >> >> >> Tony and Mike and I had a discussion last night about FreeSWITCH with >> regards to implementing some form of core message bus or shared key-value >> store. We discussed a few different options, but did not really settle on >> anything. If you are writing modules or using FreeSWITCH in a multi-node >> setting, please share what features / functionality you would like to see >> implemented in this regard, how you would use it, and why you want to see >> the specific mechanism of your choice rather than some alternative. Also, >> please consider and mention whether "cluster awareness" is something that >> factors into your use case. By this, I mean having each FS node have some >> idea about the state / status of each other node in terms of taking calls >> vs acting as a standby or slave node, etc. >> >> -- >> Eliot Gable >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Eliot Gable "We do not inherit the Earth from our ancestors: we borrow it from our children." ~David Brower "I decided the words were too conservative for me. We're not borrowing from our children, we're stealing from them--and it's not even considered to be a crime." ~David Brower "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to live; not live to eat.) ~Marcus Tullius Cicero -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130217/38d99963/attachment-0001.html From krice at freeswitch.org Sun Feb 17 19:32:44 2013 From: krice at freeswitch.org (Ken Rice) Date: Sun, 17 Feb 2013 10:32:44 -0600 Subject: [Freeswitch-users] FreeSWITCH Message Bus / Shared Key Value Store In-Reply-To: Message-ID: The Spread License is pretty much a deal killer imo... Unless they are willing to eliminate that Clause for use with FreeSWITCH... That?s as bad as the AGPL... On 2/17/13 9:50 AM, "Eliot Gable" wrote: > ZMQ also does not work with a fork, which is needed in order to execute any > system commands (like iptables or anything else which has no programming API). > That pretty much eliminates ZMQ as a possibility.? > > I did some research this weekend, and of all the possibilities I could find, > the one that caught my attention the most was Spread:?http://www.spread.org/ > > There are some drawbacks to it, namely: > > 1) It requires any marketing material mentioning FreeSWITCH or any project / > solution utilizing FreeSWITCH to also include a prepared statement about the > use of the Spread toolkit. This is a fairly major licensing issue, as all > commercial solutions utilizing FreeSWITCH as a core component would then also > need to mention the use of the Spread toolkit within FreeSWITCH. It is > essentially a "viral clause," which would be very tedious for people to make > sure they honor correctly. > > 2) The toolkit uses a stand-alone daemon which would then need to be monitored > separately from FreeSWITCH and would be another point of failure, adding > complexity to the system. Basically, if that daemon were to crash, FreeSWITCH > would need to know about it and would need to either respawn it or shut down. > This would bring the need for Pacemaker or something similar back into the > picture for any viable HA solution. Alternatively, we could write some code > similar to daemontools into FreeSWTICH which respawns the daemon if it dies, > but we would have to test the impact of this respawning on the overall cluster > to determine if it impacts visibility of the node at any point in time. > Another alternative would be to wrap the daemon into a thread inside > FreeSWITCH such that if the daemon caused a segfault or something, it would > force FreeSWITCH to terminate, as well. I have not done a code review of the > daemon yet to determine if this is viable alternative, but assuming they have > not coded it on LSD or something, it is more than likely possible.? > > 3) They boast about how it can handle up to 8,000 1KB messages per second. I > don't consider that boast-worthy. When I worked at Broadvox a few years ago, I > had a FS pair which ran around 380 calls per second (760 sessions per second). > Each call generates dozens of events. That hardware was getting dated when I > left Broadvox, and today's hardware along with the performance improvements > done to FS since then means we could conceivably have a single node which runs > over 1k calls per second firing dozens of events per call. That means a single > box could completely consume the message bandwidth of the entire Spread > network. Imagine trying to have 64 such boxes running. We are really in need > of a solution which boasts hundreds of thousands of messages per second. > Spread seems like it might be off by an order of magnitude and then some. > > Despite these issues, Spread still seems to come closer to our needs than any > other solution I found. FYI, I also looked at the Corosync IPC system, and was > not at all impressed. On paper, Spread exceeds Corosync's capabilities by a > fair margin. > > There are some strategies for mitigating issue #3 with Spread, as well. For > example, we could limit messages across the Spread network to things like > heartbeats and / or other HA and synchronization related messages. Basically, > think of it like a D-channel on a PRI. For sending high packet per second > streams of messages, we could do standard unicast connections or even try > straight up mutlicasting to all nodes on the LAN. Sending heartbeats every > 10ms across the Spread network would put a 64-node cluster at 6,400 messages > per second just with heartbeats. That would still leave a decent amount of > message bandwidth available for other types of negotiation messages and should > still allow for sub-second fail-over detection and reaction. Of course, this > is all assuming we can actually get 8,000 1KB messages per second out of a > 64-node cluster.? > > There are likely lots of things that impact how many messages per second > Spread can handle. A lot of it has to do with network latency and CPU power. > Spread uses acknowledgements and message reordering to ensure delivery in a > way that accounts for things like node membership changes during the time the > message is in transit and whether the message has been received by all nodes > in the cluster. Network latency is probably one of the biggest factors in how > many messages per second it can handle. On a faster network link, the messages > per second would be higher and on a slower network, it would be lower. > Obviously, CPU processing time and scheduling is important, as well. If one > system is extremely overloaded and the Spread daemon is being starved for CPU > resources, that will add extra latency in processing and also reduce message > throughput. Obviously, this could also impact whether the node is seen as > visible. So, this is one more argument for why we would need to try to run the > daemon as a thread under FreeSWITCH. Then it has the same scheduling priority > as FreeSWITCH and cannot be starved for resources by FreeSWITCH itself. It > also would exhibit the same amount of resource starvation FreeSWITCH > experiences on the node, so would more accurately reflect the state of > FreeSWITCH on the node.? > > If anyone has any other suggestions than Spread, I would like to hear it. > Also, some feedback on item #1 would be great, as I cannot really judge for > everyone else how willing they are to accept such a licensing clause. > > > > On Tue, Feb 12, 2013 at 8:48 PM, Jo?o Mesquita > wrote: >> I have used ZeroMQ in the past for this sorts of things but it really won't >> be able to detect failures really fast. It is not made for this. Maybe we can >> gather the requirements for such message bus? Zmq for example provides you >> with this cool interface to build messaging protocols on top of it but it >> does not provide reliability when it comes to endpoint to endpoint connection >> without a heartbeat implemented on the user end. Can this be used for FS as >> well? Anyhow, just throwing some ideas... >> >> Jo?o Mesquita >> FreeSWITCH? Solutions >> >> >> On Tue, Feb 12, 2013 at 5:42 PM, Dave R. Kompel wrote: >>> I've done a few experments with using both Redis, and the evil "Microsoft >>> Azure Service bus" (the server on prem based version) to extend the eventing >>> system to have global PUB/SUB. This way things like registrations, and Limit >>> stuff could be made global. >>> ? >>> I'm looking for a way, in my carrier switch implmentation, to implment both >>> HA Failover and Scaleout clustering. >>> ? >>> --Dave >>> >>>> >>>> From: Eliot Gable [mailto:egable+freeswitch at gmail.com >>>> ] >>>> To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] >>>> Sent: Tue, 12 Feb 2013 05:49:13 -0800 >>>> Subject: [Freeswitch-users] FreeSWITCH Message Bus / Shared Key Value Store >>>> >>>> >>>> Tony and Mike and I had a discussion last night about FreeSWITCH with >>>> regards to implementing some form of core message bus or shared key-value >>>> store. We discussed a few different options, but did not really settle on >>>> anything. If you are writing modules or using FreeSWITCH in a multi-node >>>> setting, please share what features / functionality you would like to see >>>> implemented in this regard, how you would use it, and why you want to see >>>> the specific mechanism of your choice rather than some alternative. Also, >>>> please consider and mention whether "cluster awareness" is something that >>>> factors into your use case. By this, I mean having each FS node have some >>>> idea about the state / status of each other node in terms of taking calls >>>> vs acting as a standby or slave node, etc.? -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130217/9d974850/attachment.html From julf at julf.com Sun Feb 17 20:13:08 2013 From: julf at julf.com (Johan Helsingius) Date: Sun, 17 Feb 2013 18:13:08 +0100 Subject: [Freeswitch-users] Problems with sip provider Message-ID: <51210FA4.1010200@julf.com> I am using 3 different SIP providers. 2 of them work just fine, but I have never managed to get the third one, telfort (NL) to work with Freeswitch. Here is my gateway definition: sofia status shows: external::telfort gateway sip:020NNNNNNN at sip.telefoniedienst.nl REGED but when I try to dial out on that provider, I get: 2013-02-17 18:04:19.216205 [NOTICE] sofia.c:6385 Hangup sofia/external/06MMMMMMMM [CS_EXCHANGE_MEDIA] [CALL_REJECTED] The strange thing is that everything works OK when I configure the parameters into the ADSL modem/gateway - a ZyXEL P-2601HN-F1: SIP Server Address: sip.telefoniedienst.nl Port: 5060 REGISTER Server Address: sip.telefoniedienst.nl Port: 5060 SIP Service Domain: tel.telefoniedienst.nl Port: 5060 RTP Port Range: 16384-32767 Enable Early Media Account Number: 020NNNNNNN Username: 020NNNNNNN Password: XXXXXXX Compression Type: G.711ALaw, G.729, G.722 Active G.168 Echo Cancellation so somehow I am not mapping that information properly into the Freeswitch conf... Julf From avi at avimarcus.net Sun Feb 17 20:23:23 2013 From: avi at avimarcus.net (Avi Marcus) Date: Sun, 17 Feb 2013 19:23:23 +0200 Subject: [Freeswitch-users] Problems with sip provider In-Reply-To: <51210FA4.1010200@julf.com> References: <51210FA4.1010200@julf.com> Message-ID: If they are rejecting it, I'd ask them why. Are you bridging to sofia/gateway/telfort ? Also, if you changed the password on the gateway, did you killgw the old load and restart it? reloadxml doesn't restart gateways that are already connected. -Avi On Sun, Feb 17, 2013 at 7:13 PM, Johan Helsingius wrote: > I am using 3 different SIP providers. 2 of them work just fine, but > I have never managed to get the third one, telfort (NL) to work with > Freeswitch. > > Here is my gateway definition: > > > > > > > > > > > > sofia status shows: > > external::telfort gateway sip:020NNNNNNN at sip.telefoniedienst.nl REGED > > but when I try to dial out on that provider, I get: > > 2013-02-17 18:04:19.216205 [NOTICE] sofia.c:6385 Hangup > sofia/external/06MMMMMMMM [CS_EXCHANGE_MEDIA] [CALL_REJECTED] > > The strange thing is that everything works OK when I configure > the parameters into the ADSL modem/gateway - a ZyXEL P-2601HN-F1: > > SIP Server Address: sip.telefoniedienst.nl Port: 5060 > REGISTER Server Address: sip.telefoniedienst.nl Port: 5060 > SIP Service Domain: tel.telefoniedienst.nl Port: 5060 > RTP Port Range: 16384-32767 > Enable Early Media > Account Number: 020NNNNNNN > Username: 020NNNNNNN > Password: XXXXXXX > Compression Type: G.711ALaw, G.729, G.722 > Active G.168 Echo Cancellation > > so somehow I am not mapping that information properly into > the Freeswitch conf... > > Julf > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130217/5f50f5d2/attachment-0001.html From anthony.minessale at gmail.com Sun Feb 17 20:28:25 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 17 Feb 2013 11:28:25 -0600 Subject: [Freeswitch-users] FreeSWITCH Message Bus / Shared Key Value Store In-Reply-To: References: Message-ID: Restrictive licenses ... Ain't nobody got time fo dat! On Sun, Feb 17, 2013 at 10:32 AM, Ken Rice wrote: > The Spread License is pretty much a deal killer imo... Unless they are > willing to eliminate that Clause for use with FreeSWITCH... That?s as bad > as the AGPL... > > > > On 2/17/13 9:50 AM, "Eliot Gable" wrote: > > ZMQ also does not work with a fork, which is needed in order to execute > any system commands (like iptables or anything else which has no > programming API). That pretty much eliminates ZMQ as a possibility. > > I did some research this weekend, and of all the possibilities I could > find, the one that caught my attention the most was Spread: > http://www.spread.org/ > > There are some drawbacks to it, namely: > > 1) It requires any marketing material mentioning FreeSWITCH or any project > / solution utilizing FreeSWITCH to also include a prepared statement about > the use of the Spread toolkit. This is a fairly major licensing issue, as > all commercial solutions utilizing FreeSWITCH as a core component would > then also need to mention the use of the Spread toolkit within FreeSWITCH. > It is essentially a "viral clause," which would be very tedious for people > to make sure they honor correctly. > > 2) The toolkit uses a stand-alone daemon which would then need to be > monitored separately from FreeSWITCH and would be another point of failure, > adding complexity to the system. Basically, if that daemon were to crash, > FreeSWITCH would need to know about it and would need to either respawn it > or shut down. This would bring the need for Pacemaker or something similar > back into the picture for any viable HA solution. Alternatively, we could > write some code similar to daemontools into FreeSWTICH which respawns the > daemon if it dies, but we would have to test the impact of this respawning > on the overall cluster to determine if it impacts visibility of the node at > any point in time. Another alternative would be to wrap the daemon into a > thread inside FreeSWITCH such that if the daemon caused a segfault or > something, it would force FreeSWITCH to terminate, as well. I have not done > a code review of the daemon yet to determine if this is viable alternative, > but assuming they have not coded it on LSD or something, it is more than > likely possible. > > 3) They boast about how it can handle up to 8,000 1KB messages per second. > I don't consider that boast-worthy. When I worked at Broadvox a few years > ago, I had a FS pair which ran around 380 calls per second (760 sessions > per second). Each call generates dozens of events. That hardware was > getting dated when I left Broadvox, and today's hardware along with the > performance improvements done to FS since then means we could conceivably > have a single node which runs over 1k calls per second firing dozens of > events per call. That means a single box could completely consume the > message bandwidth of the entire Spread network. Imagine trying to have 64 > such boxes running. We are really in need of a solution which boasts > hundreds of thousands of messages per second. Spread seems like it might be > off by an order of magnitude and then some. > > Despite these issues, Spread still seems to come closer to our needs than > any other solution I found. FYI, I also looked at the Corosync IPC system, > and was not at all impressed. On paper, Spread exceeds Corosync's > capabilities by a fair margin. > > There are some strategies for mitigating issue #3 with Spread, as well. > For example, we could limit messages across the Spread network to things > like heartbeats and / or other HA and synchronization related messages. > Basically, think of it like a D-channel on a PRI. For sending high packet > per second streams of messages, we could do standard unicast connections or > even try straight up mutlicasting to all nodes on the LAN. Sending > heartbeats every 10ms across the Spread network would put a 64-node cluster > at 6,400 messages per second just with heartbeats. That would still leave a > decent amount of message bandwidth available for other types of negotiation > messages and should still allow for sub-second fail-over detection and > reaction. Of course, this is all assuming we can actually get 8,000 1KB > messages per second out of a 64-node cluster. > > There are likely lots of things that impact how many messages per second > Spread can handle. A lot of it has to do with network latency and CPU > power. Spread uses acknowledgements and message reordering to ensure > delivery in a way that accounts for things like node membership changes > during the time the message is in transit and whether the message has been > received by all nodes in the cluster. Network latency is probably one of > the biggest factors in how many messages per second it can handle. On a > faster network link, the messages per second would be higher and on a > slower network, it would be lower. Obviously, CPU processing time and > scheduling is important, as well. If one system is extremely overloaded and > the Spread daemon is being starved for CPU resources, that will add extra > latency in processing and also reduce message throughput. Obviously, this > could also impact whether the node is seen as visible. So, this is one more > argument for why we would need to try to run the daemon as a thread under > FreeSWITCH. Then it has the same scheduling priority as FreeSWITCH and > cannot be starved for resources by FreeSWITCH itself. It also would exhibit > the same amount of resource starvation FreeSWITCH experiences on the node, > so would more accurately reflect the state of FreeSWITCH on the node. > > If anyone has any other suggestions than Spread, I would like to hear it. > Also, some feedback on item #1 would be great, as I cannot really judge for > everyone else how willing they are to accept such a licensing clause. > > > > On Tue, Feb 12, 2013 at 8:48 PM, Jo?o Mesquita > wrote: > > I have used ZeroMQ in the past for this sorts of things but it really > won't be able to detect failures really fast. It is not made for this. > Maybe we can gather the requirements for such message bus? Zmq for example > provides you with this cool interface to build messaging protocols on top > of it but it does not provide reliability when it comes to endpoint to > endpoint connection without a heartbeat implemented on the user end. Can > this be used for FS as well? Anyhow, just throwing some ideas... > > Jo?o Mesquita > FreeSWITCH? Solutions > > > On Tue, Feb 12, 2013 at 5:42 PM, Dave R. Kompel wrote: > > I've done a few experments with using both Redis, and the evil "Microsoft > Azure Service bus" (the server on prem based version) to extend the > eventing system to have global PUB/SUB. This way things like registrations, > and Limit stuff could be made global. > > I'm looking for a way, in my carrier switch implmentation, to implment > both HA Failover and Scaleout clustering. > > --Dave > > ------------------------------ > *From:* Eliot Gable [mailto:egable+freeswitch at gmail.com< > mailto:egable%2Bfreeswitch at gmail.com > ] > > *To:* FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org > ] > *Sent:* Tue, 12 Feb 2013 05:49:13 -0800 > *Subject:* [Freeswitch-users] FreeSWITCH Message Bus / Shared Key Value > Store > > > Tony and Mike and I had a discussion last night about FreeSWITCH with > regards to implementing some form of core message bus or shared key-value > store. We discussed a few different options, but did not really settle on > anything. If you are writing modules or using FreeSWITCH in a multi-node > setting, please share what features / functionality you would like to see > implemented in this regard, how you would use it, and why you want to see > the specific mechanism of your choice rather than some alternative. Also, > please consider and mention whether "cluster awareness" is something that > factors into your use case. By this, I mean having each FS node have some > idea about the state / status of each other node in terms of taking calls > vs acting as a standby or slave node, etc. > > > -- > Ken > *http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > *irc.freenode.net #freeswitch > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130217/50b5cd20/attachment.html From avi at avimarcus.net Sun Feb 17 20:32:31 2013 From: avi at avimarcus.net (Avi Marcus) Date: Sun, 17 Feb 2013 19:32:31 +0200 Subject: [Freeswitch-users] FreeSWITCH Message Bus / Shared Key Value Store In-Reply-To: References: <20130212204251.7066d944@mail.tritonwest.net> Message-ID: Please excuse my ignorance, but can you explain again why zmq won't work? Regarding boasting about messaging capability.. it was made to be quite lightweight - supposedly it can easily saturate a 1gigabit link with message size over 150 bytes each: http://www.zeromq.org/results:10gbe-tests If you want to use it for a queue, then it's lack of persistence and assurance that messages get through might not be helpful, but for a broadcast to whoever wants to listen it sounds great for heartbeats, events, etc. -Avi Marcus 1-718-989-9485 (USA) 1-866-202-5850 (USA & Canada Toll Free) 02-372-1570 (Israel) 020-3298-2875 (UK) On Sun, Feb 17, 2013 at 5:50 PM, Eliot Gable wrote: > ZMQ also does not work with a fork, which is needed in order to execute > any system commands (like iptables or anything else which has no > programming API). That pretty much eliminates ZMQ as a possibility. > > I did some research this weekend, and of all the possibilities I could > find, the one that caught my attention the most was Spread: > http://www.spread.org/ > > There are some drawbacks to it, namely: > > 1) It requires any marketing material mentioning FreeSWITCH or any project > / solution utilizing FreeSWITCH to also include a prepared statement about > the use of the Spread toolkit. This is a fairly major licensing issue, as > all commercial solutions utilizing FreeSWITCH as a core component would > then also need to mention the use of the Spread toolkit within FreeSWITCH.. > It is essentially a "viral clause," which would be very tedious for people > to make sure they honor correctly. > > 2) The toolkit uses a stand-alone daemon which would then need to be > monitored separately from FreeSWITCH and would be another point of failure, > adding complexity to the system. Basically, if that daemon were to crash, > FreeSWITCH would need to know about it and would need to either respawn it > or shut down. This would bring the need for Pacemaker or something similar > back into the picture for any viable HA solution. Alternatively, we could > write some code similar to daemontools into FreeSWTICH which respawns the > daemon if it dies, but we would have to test the impact of this respawning > on the overall cluster to determine if it impacts visibility of the node at > any point in time. Another alternative would be to wrap the daemon into a > thread inside FreeSWITCH such that if the daemon caused a segfault or > something, it would force FreeSWITCH to terminate, as well. I have not done > a code review of the daemon yet to determine if this is viable alternative, > but assuming they have not coded it on LSD or something, it is more than > likely possible. > > 3) They boast about how it can handle up to 8,000 1KB messages per second.. > I don't consider that boast-worthy. When I worked at Broadvox a few years > ago, I had a FS pair which ran around 380 calls per second (760 sessions > per second). Each call generates dozens of events. That hardware was > getting dated when I left Broadvox, and today's hardware along with the > performance improvements done to FS since then means we could conceivably > have a single node which runs over 1k calls per second firing dozens of > events per call. That means a single box could completely consume the > message bandwidth of the entire Spread network. Imagine trying to have 64 > such boxes running. We are really in need of a solution which boasts > hundreds of thousands of messages per second. Spread seems like it might be > off by an order of magnitude and then some. > > Despite these issues, Spread still seems to come closer to our needs than > any other solution I found. FYI, I also looked at the Corosync IPC system, > and was not at all impressed. On paper, Spread exceeds Corosync's > capabilities by a fair margin. > > There are some strategies for mitigating issue #3 with Spread, as well. > For example, we could limit messages across the Spread network to things > like heartbeats and / or other HA and synchronization related messages. > Basically, think of it like a D-channel on a PRI. For sending high packet > per second streams of messages, we could do standard unicast connections or > even try straight up mutlicasting to all nodes on the LAN. Sending > heartbeats every 10ms across the Spread network would put a 64-node cluster > at 6,400 messages per second just with heartbeats. That would still leave a > decent amount of message bandwidth available for other types of negotiation > messages and should still allow for sub-second fail-over detection and > reaction. Of course, this is all assuming we can actually get 8,000 1KB > messages per second out of a 64-node cluster. > > There are likely lots of things that impact how many messages per second > Spread can handle. A lot of it has to do with network latency and CPU > power. Spread uses acknowledgements and message reordering to ensure > delivery in a way that accounts for things like node membership changes > during the time the message is in transit and whether the message has been > received by all nodes in the cluster. Network latency is probably one of > the biggest factors in how many messages per second it can handle. On a > faster network link, the messages per second would be higher and on a > slower network, it would be lower. Obviously, CPU processing time and > scheduling is important, as well. If one system is extremely overloaded and > the Spread daemon is being starved for CPU resources, that will add extra > latency in processing and also reduce message throughput. Obviously, this > could also impact whether the node is seen as visible. So, this is one more > argument for why we would need to try to run the daemon as a thread under > FreeSWITCH. Then it has the same scheduling priority as FreeSWITCH and > cannot be starved for resources by FreeSWITCH itself. It also would exhibit > the same amount of resource starvation FreeSWITCH experiences on the node, > so would more accurately reflect the state of FreeSWITCH on the node. > > If anyone has any other suggestions than Spread, I would like to hear it. > Also, some feedback on item #1 would be great, as I cannot really judge for > everyone else how willing they are to accept such a licensing clause. > > > > On Tue, Feb 12, 2013 at 8:48 PM, Jo?o Mesquita wrote: > >> I have used ZeroMQ in the past for this sorts of things but it really >> won't be able to detect failures really fast. It is not made for this. >> Maybe we can gather the requirements for such message bus? Zmq for example >> provides you with this cool interface to build messaging protocols on top >> of it but it does not provide reliability when it comes to endpoint to >> endpoint connection without a heartbeat implemented on the user end. Can >> this be used for FS as well? Anyhow, just throwing some ideas... >> >> Jo?o Mesquita >> FreeSWITCH? Solutions >> >> >> On Tue, Feb 12, 2013 at 5:42 PM, Dave R. Kompel wrote: >> >>> ** >>> I've done a few experments with using both Redis, and the evil >>> "Microsoft Azure Service bus" (the server on prem based version) to extend >>> the eventing system to have global PUB/SUB. This way things like >>> registrations, and Limit stuff could be made global. >>> >>> I'm looking for a way, in my carrier switch implmentation, to implment >>> both HA Failover and Scaleout clustering. >>> >>> --Dave >>> >>> ------------------------------ >>> *From:* Eliot Gable [mailto:egable+freeswitch at gmail.com] >>> *To:* FreeSWITCH Users Help [mailto: >>> freeswitch-users at lists.freeswitch.org] >>> *Sent:* Tue, 12 Feb 2013 05:49:13 -0800 >>> *Subject:* [Freeswitch-users] FreeSWITCH Message Bus / Shared Key Value >>> Store >>> >>> >>> Tony and Mike and I had a discussion last night about FreeSWITCH with >>> regards to implementing some form of core message bus or shared key-value >>> store. We discussed a few different options, but did not really settle on >>> anything. If you are writing modules or using FreeSWITCH in a multi-node >>> setting, please share what features / functionality you would like to see >>> implemented in this regard, how you would use it, and why you want to see >>> the specific mechanism of your choice rather than some alternative. Also, >>> please consider and mention whether "cluster awareness" is something that >>> factors into your use case. By this, I mean having each FS node have some >>> idea about the state / status of each other node in terms of taking calls >>> vs acting as a standby or slave node, etc. >>> >>> -- >>> Eliot Gable >>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > > -- > Eliot Gable > > "We do not inherit the Earth from our ancestors: we borrow it from our > children." ~David Brower > > "I decided the words were too conservative for me. We're not borrowing > from our children, we're stealing from them--and it's not even considered > to be a crime." ~David Brower > > "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to live; > not live to eat.) ~Marcus Tullius Cicero > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130217/04c88ae8/attachment-0001.html From julf at julf.com Sun Feb 17 20:33:44 2013 From: julf at julf.com (Johan Helsingius) Date: Sun, 17 Feb 2013 18:33:44 +0100 Subject: [Freeswitch-users] Problems with sip provider In-Reply-To: References: <51210FA4.1010200@julf.com> Message-ID: <51211478.1080108@julf.com> > If they are rejecting it, I'd ask them why. I did. "What equipment are you using?" "freeswitch" "we don't support that" > Are you bridging to sofia/gateway/telfort ? Yes. > Also, if you changed the password on the gateway, did you killgw the old load > and restart it? reloadxml doesn't restart gateways that are already connected. I restarted freeswitch. Julf From anthony.minessale at gmail.com Sun Feb 17 20:51:54 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 17 Feb 2013 11:51:54 -0600 Subject: [Freeswitch-users] FreeSWITCH Message Bus / Shared Key Value Store In-Reply-To: References: <20130212204251.7066d944@mail.tritonwest.net> Message-ID: It has a strict philosophy on not using locks or mutexes and only doing com over socket. When you create an instance of it, it uses its own threads internally and that model gets messed up severely if you call fork from the process (even when you are doing it for something else like system command). Its basically an engine waiting for a host program rather than something you can integrated into a more complex application. Also the multicast has issues, there is no way to specificy host local scope for hops. Finally, their philosophy is to ignore the dilemma of sub/pub. There are no subscriptions to events, there is only send all events to everyone and let them filter locally. Its easy to believe this is adequate when doing pre-determined test scenarios but we know in reality that in a busy FS box we, cant send *all* the events to *everyone* and let them filter locally. On Sun, Feb 17, 2013 at 11:32 AM, Avi Marcus wrote: > Please excuse my ignorance, but can you explain again why zmq won't work? > > Regarding boasting about messaging capability.. it was made to be quite > lightweight - supposedly it can easily saturate a 1gigabit link with > message size over 150 bytes each: > http://www.zeromq.org/results:10gbe-tests > > If you want to use it for a queue, then it's lack of persistence and > assurance that messages get through might not be helpful, but for a > broadcast to whoever wants to listen it sounds great for heartbeats, > events, etc. > > -Avi Marcus > > 1-718-989-9485 (USA) > 1-866-202-5850 (USA & Canada Toll Free) > 02-372-1570 (Israel) > 020-3298-2875 (UK) > > > On Sun, Feb 17, 2013 at 5:50 PM, Eliot Gable wrote: > >> ZMQ also does not work with a fork, which is needed in order to execute >> any system commands (like iptables or anything else which has no >> programming API). That pretty much eliminates ZMQ as a possibility. >> >> I did some research this weekend, and of all the possibilities I could >> find, the one that caught my attention the most was Spread: >> http://www.spread.org/ >> >> There are some drawbacks to it, namely: >> >> 1) It requires any marketing material mentioning FreeSWITCH or any >> project / solution utilizing FreeSWITCH to also include a prepared >> statement about the use of the Spread toolkit. This is a fairly major >> licensing issue, as all commercial solutions utilizing FreeSWITCH as a core >> component would then also need to mention the use of the Spread toolkit >> within FreeSWITCH. It is essentially a "viral clause," which would be very >> tedious for people to make sure they honor correctly. >> >> 2) The toolkit uses a stand-alone daemon which would then need to be >> monitored separately from FreeSWITCH and would be another point of failure, >> adding complexity to the system. Basically, if that daemon were to crash, >> FreeSWITCH would need to know about it and would need to either respawn it >> or shut down. This would bring the need for Pacemaker or something similar >> back into the picture for any viable HA solution. Alternatively, we could >> write some code similar to daemontools into FreeSWTICH which respawns the >> daemon if it dies, but we would have to test the impact of this respawning >> on the overall cluster to determine if it impacts visibility of the node at >> any point in time. Another alternative would be to wrap the daemon into a >> thread inside FreeSWITCH such that if the daemon caused a segfault or >> something, it would force FreeSWITCH to terminate, as well. I have not done >> a code review of the daemon yet to determine if this is viable alternative, >> but assuming they have not coded it on LSD or something, it is more than >> likely possible. >> >> 3) They boast about how it can handle up to 8,000 1KB messages per >> second. I don't consider that boast-worthy. When I worked at Broadvox a few >> years ago, I had a FS pair which ran around 380 calls per second (760 >> sessions per second). Each call generates dozens of events. That hardware >> was getting dated when I left Broadvox, and today's hardware along with the >> performance improvements done to FS since then means we could conceivably >> have a single node which runs over 1k calls per second firing dozens of >> events per call. That means a single box could completely consume the >> message bandwidth of the entire Spread network. Imagine trying to have 64 >> such boxes running. We are really in need of a solution which boasts >> hundreds of thousands of messages per second. Spread seems like it might be >> off by an order of magnitude and then some. >> >> Despite these issues, Spread still seems to come closer to our needs than >> any other solution I found. FYI, I also looked at the Corosync IPC system, >> and was not at all impressed. On paper, Spread exceeds Corosync's >> capabilities by a fair margin. >> >> There are some strategies for mitigating issue #3 with Spread, as well. >> For example, we could limit messages across the Spread network to things >> like heartbeats and / or other HA and synchronization related messages. >> Basically, think of it like a D-channel on a PRI. For sending high packet >> per second streams of messages, we could do standard unicast connections or >> even try straight up mutlicasting to all nodes on the LAN. Sending >> heartbeats every 10ms across the Spread network would put a 64-node cluster >> at 6,400 messages per second just with heartbeats. That would still leave a >> decent amount of message bandwidth available for other types of negotiation >> messages and should still allow for sub-second fail-over detection and >> reaction. Of course, this is all assuming we can actually get 8,000 1KB >> messages per second out of a 64-node cluster. >> >> There are likely lots of things that impact how many messages per second >> Spread can handle. A lot of it has to do with network latency and CPU >> power. Spread uses acknowledgements and message reordering to ensure >> delivery in a way that accounts for things like node membership changes >> during the time the message is in transit and whether the message has been >> received by all nodes in the cluster. Network latency is probably one of >> the biggest factors in how many messages per second it can handle. On a >> faster network link, the messages per second would be higher and on a >> slower network, it would be lower. Obviously, CPU processing time and >> scheduling is important, as well. If one system is extremely overloaded and >> the Spread daemon is being starved for CPU resources, that will add extra >> latency in processing and also reduce message throughput. Obviously, this >> could also impact whether the node is seen as visible. So, this is one more >> argument for why we would need to try to run the daemon as a thread under >> FreeSWITCH. Then it has the same scheduling priority as FreeSWITCH and >> cannot be starved for resources by FreeSWITCH itself. It also would exhibit >> the same amount of resource starvation FreeSWITCH experiences on the node, >> so would more accurately reflect the state of FreeSWITCH on the node. >> >> If anyone has any other suggestions than Spread, I would like to hear it. >> Also, some feedback on item #1 would be great, as I cannot really judge for >> everyone else how willing they are to accept such a licensing clause. >> >> >> >> On Tue, Feb 12, 2013 at 8:48 PM, Jo?o Mesquita wrote: >> >>> I have used ZeroMQ in the past for this sorts of things but it really >>> won't be able to detect failures really fast. It is not made for this. >>> Maybe we can gather the requirements for such message bus? Zmq for example >>> provides you with this cool interface to build messaging protocols on top >>> of it but it does not provide reliability when it comes to endpoint to >>> endpoint connection without a heartbeat implemented on the user end. Can >>> this be used for FS as well? Anyhow, just throwing some ideas... >>> >>> Jo?o Mesquita >>> FreeSWITCH? Solutions >>> >>> >>> On Tue, Feb 12, 2013 at 5:42 PM, Dave R. Kompel wrote: >>> >>>> ** >>>> I've done a few experments with using both Redis, and the evil >>>> "Microsoft Azure Service bus" (the server on prem based version) to extend >>>> the eventing system to have global PUB/SUB. This way things like >>>> registrations, and Limit stuff could be made global. >>>> >>>> I'm looking for a way, in my carrier switch implmentation, to implment >>>> both HA Failover and Scaleout clustering. >>>> >>>> --Dave >>>> >>>> ------------------------------ >>>> *From:* Eliot Gable [mailto:egable+freeswitch at gmail.com] >>>> *To:* FreeSWITCH Users Help [mailto: >>>> freeswitch-users at lists.freeswitch.org] >>>> *Sent:* Tue, 12 Feb 2013 05:49:13 -0800 >>>> *Subject:* [Freeswitch-users] FreeSWITCH Message Bus / Shared Key >>>> Value Store >>>> >>>> >>>> Tony and Mike and I had a discussion last night about FreeSWITCH with >>>> regards to implementing some form of core message bus or shared key-value >>>> store. We discussed a few different options, but did not really settle on >>>> anything. If you are writing modules or using FreeSWITCH in a multi-node >>>> setting, please share what features / functionality you would like to see >>>> implemented in this regard, how you would use it, and why you want to see >>>> the specific mechanism of your choice rather than some alternative. Also, >>>> please consider and mention whether "cluster awareness" is something that >>>> factors into your use case. By this, I mean having each FS node have some >>>> idea about the state / status of each other node in terms of taking calls >>>> vs acting as a standby or slave node, etc. >>>> >>>> -- >>>> Eliot Gable >>>> >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> >> -- >> Eliot Gable >> >> "We do not inherit the Earth from our ancestors: we borrow it from our >> children." ~David Brower >> >> "I decided the words were too conservative for me. We're not borrowing >> from our children, we're stealing from them--and it's not even considered >> to be a crime." ~David Brower >> >> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to live; >> not live to eat.) ~Marcus Tullius Cicero >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130217/2ae3e3f5/attachment-0001.html From avi at avimarcus.net Sun Feb 17 21:10:24 2013 From: avi at avimarcus.net (Avi Marcus) Date: Sun, 17 Feb 2013 20:10:24 +0200 Subject: [Freeswitch-users] FreeSWITCH Message Bus / Shared Key Value Store In-Reply-To: References: <20130212204251.7066d944@mail.tritonwest.net> Message-ID: Regarding #3, it seems they got to it for zeromq v3. > > From ?MQ 3.x, filtering happens at the publisher side, when using a > connected protocol (tcp: *or ipc:*). Using the epgm:// protocol, > filtering happens at the subscriber side. In ?MQ/2.x, all filtering happened at the subscriber side. -Avi On Sun, Feb 17, 2013 at 7:51 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > It has a strict philosophy on not using locks or mutexes and only doing > com over socket. When you create an instance of it, it uses its own > threads internally and that model gets messed up severely if you call fork > from the process (even when you are doing it for something else like system > command). Its basically an engine waiting for a host program rather than > something you can integrated into a more complex application. Also the > multicast has issues, there is no way to specificy host local scope for > hops. Finally, their philosophy is to ignore the dilemma of sub/pub. > There are no subscriptions to events, there is only send all events to > everyone and let them filter locally. Its easy to believe this is adequate > when doing pre-determined test scenarios but we know in reality that in a > busy FS box we, cant send *all* the events to *everyone* and let them > filter locally. > > > > On Sun, Feb 17, 2013 at 11:32 AM, Avi Marcus wrote: > >> Please excuse my ignorance, but can you explain again why zmq won't work? >> >> Regarding boasting about messaging capability.. it was made to be quite >> lightweight - supposedly it can easily saturate a 1gigabit link with >> message size over 150 bytes each: >> http://www.zeromq.org/results:10gbe-tests >> >> If you want to use it for a queue, then it's lack of persistence and >> assurance that messages get through might not be helpful, but for a >> broadcast to whoever wants to listen it sounds great for heartbeats, >> events, etc. >> >> -Avi Marcus >> >> 1-718-989-9485 (USA) >> 1-866-202-5850 (USA & Canada Toll Free) >> 02-372-1570 (Israel) >> 020-3298-2875 (UK) >> >> >> On Sun, Feb 17, 2013 at 5:50 PM, Eliot Gable > > wrote: >> >>> ZMQ also does not work with a fork, which is needed in order to execute >>> any system commands (like iptables or anything else which has no >>> programming API). That pretty much eliminates ZMQ as a possibility. >>> >>> I did some research this weekend, and of all the possibilities I could >>> find, the one that caught my attention the most was Spread: >>> http://www.spread.org/ >>> >>> There are some drawbacks to it, namely: >>> >>> 1) It requires any marketing material mentioning FreeSWITCH or any >>> project / solution utilizing FreeSWITCH to also include a prepared >>> statement about the use of the Spread toolkit. This is a fairly major >>> licensing issue, as all commercial solutions utilizing FreeSWITCH as a core >>> component would then also need to mention the use of the Spread toolkit >>> within FreeSWITCH. It is essentially a "viral clause," which would be very >>> tedious for people to make sure they honor correctly. >>> >>> 2) The toolkit uses a stand-alone daemon which would then need to be >>> monitored separately from FreeSWITCH and would be another point of failure, >>> adding complexity to the system. Basically, if that daemon were to crash, >>> FreeSWITCH would need to know about it and would need to either respawn it >>> or shut down. This would bring the need for Pacemaker or something similar >>> back into the picture for any viable HA solution. Alternatively, we could >>> write some code similar to daemontools into FreeSWTICH which respawns the >>> daemon if it dies, but we would have to test the impact of this respawning >>> on the overall cluster to determine if it impacts visibility of the node at >>> any point in time. Another alternative would be to wrap the daemon into a >>> thread inside FreeSWITCH such that if the daemon caused a segfault or >>> something, it would force FreeSWITCH to terminate, as well. I have not done >>> a code review of the daemon yet to determine if this is viable alternative, >>> but assuming they have not coded it on LSD or something, it is more than >>> likely possible. >>> >>> 3) They boast about how it can handle up to 8,000 1KB messages per >>> second. I don't consider that boast-worthy. When I worked at Broadvox a few >>> years ago, I had a FS pair which ran around 380 calls per second (760 >>> sessions per second). Each call generates dozens of events. That hardware >>> was getting dated when I left Broadvox, and today's hardware along with the >>> performance improvements done to FS since then means we could conceivably >>> have a single node which runs over 1k calls per second firing dozens of >>> events per call. That means a single box could completely consume the >>> message bandwidth of the entire Spread network. Imagine trying to have 64 >>> such boxes running. We are really in need of a solution which boasts >>> hundreds of thousands of messages per second. Spread seems like it might be >>> off by an order of magnitude and then some. >>> >>> Despite these issues, Spread still seems to come closer to our needs >>> than any other solution I found. FYI, I also looked at the Corosync IPC >>> system, and was not at all impressed. On paper, Spread exceeds Corosync's >>> capabilities by a fair margin. >>> >>> There are some strategies for mitigating issue #3 with Spread, as well. >>> For example, we could limit messages across the Spread network to things >>> like heartbeats and / or other HA and synchronization related messages. >>> Basically, think of it like a D-channel on a PRI. For sending high packet >>> per second streams of messages, we could do standard unicast connections or >>> even try straight up mutlicasting to all nodes on the LAN. Sending >>> heartbeats every 10ms across the Spread network would put a 64-node cluster >>> at 6,400 messages per second just with heartbeats. That would still leave a >>> decent amount of message bandwidth available for other types of negotiation >>> messages and should still allow for sub-second fail-over detection and >>> reaction. Of course, this is all assuming we can actually get 8,000 1KB >>> messages per second out of a 64-node cluster. >>> >>> There are likely lots of things that impact how many messages per second >>> Spread can handle. A lot of it has to do with network latency and CPU >>> power. Spread uses acknowledgements and message reordering to ensure >>> delivery in a way that accounts for things like node membership changes >>> during the time the message is in transit and whether the message has been >>> received by all nodes in the cluster. Network latency is probably one of >>> the biggest factors in how many messages per second it can handle. On a >>> faster network link, the messages per second would be higher and on a >>> slower network, it would be lower. Obviously, CPU processing time and >>> scheduling is important, as well. If one system is extremely overloaded and >>> the Spread daemon is being starved for CPU resources, that will add extra >>> latency in processing and also reduce message throughput. Obviously, this >>> could also impact whether the node is seen as visible. So, this is one more >>> argument for why we would need to try to run the daemon as a thread under >>> FreeSWITCH. Then it has the same scheduling priority as FreeSWITCH and >>> cannot be starved for resources by FreeSWITCH itself. It also would exhibit >>> the same amount of resource starvation FreeSWITCH experiences on the node, >>> so would more accurately reflect the state of FreeSWITCH on the node. >>> >>> If anyone has any other suggestions than Spread, I would like to hear >>> it. Also, some feedback on item #1 would be great, as I cannot really judge >>> for everyone else how willing they are to accept such a licensing clause. >>> >>> >>> >>> On Tue, Feb 12, 2013 at 8:48 PM, Jo?o Mesquita >> > wrote: >>> >>>> I have used ZeroMQ in the past for this sorts of things but it really >>>> won't be able to detect failures really fast. It is not made for this. >>>> Maybe we can gather the requirements for such message bus? Zmq for example >>>> provides you with this cool interface to build messaging protocols on top >>>> of it but it does not provide reliability when it comes to endpoint to >>>> endpoint connection without a heartbeat implemented on the user end. Can >>>> this be used for FS as well? Anyhow, just throwing some ideas... >>>> >>>> Jo?o Mesquita >>>> FreeSWITCH? Solutions >>>> >>>> >>>> On Tue, Feb 12, 2013 at 5:42 PM, Dave R. Kompel wrote: >>>> >>>>> ** >>>>> I've done a few experments with using both Redis, and the evil >>>>> "Microsoft Azure Service bus" (the server on prem based version) to extend >>>>> the eventing system to have global PUB/SUB. This way things like >>>>> registrations, and Limit stuff could be made global. >>>>> >>>>> I'm looking for a way, in my carrier switch implmentation, to implment >>>>> both HA Failover and Scaleout clustering. >>>>> >>>>> --Dave >>>>> >>>>> ------------------------------ >>>>> *From:* Eliot Gable [mailto:egable+freeswitch at gmail.com] >>>>> *To:* FreeSWITCH Users Help [mailto: >>>>> freeswitch-users at lists.freeswitch.org] >>>>> *Sent:* Tue, 12 Feb 2013 05:49:13 -0800 >>>>> *Subject:* [Freeswitch-users] FreeSWITCH Message Bus / Shared Key >>>>> Value Store >>>>> >>>>> >>>>> Tony and Mike and I had a discussion last night about FreeSWITCH with >>>>> regards to implementing some form of core message bus or shared key-value >>>>> store. We discussed a few different options, but did not really settle on >>>>> anything. If you are writing modules or using FreeSWITCH in a multi-node >>>>> setting, please share what features / functionality you would like to see >>>>> implemented in this regard, how you would use it, and why you want to see >>>>> the specific mechanism of your choice rather than some alternative. Also, >>>>> please consider and mention whether "cluster awareness" is something that >>>>> factors into your use case. By this, I mean having each FS node have some >>>>> idea about the state / status of each other node in terms of taking calls >>>>> vs acting as a standby or slave node, etc. >>>>> >>>>> -- >>>>> Eliot Gable >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> 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 >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>>> >>>> >>> >>> >>> -- >>> Eliot Gable >>> >>> "We do not inherit the Earth from our ancestors: we borrow it from our >>> children." ~David Brower >>> >>> "I decided the words were too conservative for me. We're not borrowing >>> from our children, we're stealing from them--and it's not even considered >>> to be a crime." ~David Brower >>> >>> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to live; >>> not live to eat.) ~Marcus Tullius Cicero >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130217/b51ada3a/attachment-0001.html From vbvbrj at gmail.com Sun Feb 17 21:13:44 2013 From: vbvbrj at gmail.com (Mimiko) Date: Sun, 17 Feb 2013 20:13:44 +0200 Subject: [Freeswitch-users] Adtran In-Reply-To: References: Message-ID: <51211DD8.2000202@gmail.com> On 15.02.2013 21:40, Jeff Pyle wrote: > Blake, > > We use the TA900-series with quite a bit of success. Tell me more about > the PRI (DNIS digits, switch type) and I should be able to help you with > a config template. > > > - Jeff > > > > On Fri, Feb 15, 2013 at 2:28 PM, Michael Collins > wrote: > > I'm not familiar with the Adtran 908e, but a quick goog search says: > > The Total Access 908e IP Business Gateway provides SIP-gateway > functionality, a robust IP router, firewall and VPN functionality, > and support for a number of analog and digital interfaces for > existing phone equipment, or for the combination of an IP PBX and an > analog fax, phones, or modem. > > * Four T1/dual Ethernet platform, 8 analog FXS interfaces, > integrated DSX-1 > * Ideal for networks transitioning to VoIP and an all IP voice and > data WAN Oh, I see Adtran have some interesting VoIP gatways. How Adtran 900e series is good in connecting FS to a legay PBX? Any issue to switch to them instead of more expensive Audiocodec? -- Mimiko desu. From shaheryarkh at gmail.com Sun Feb 17 22:29:21 2013 From: shaheryarkh at gmail.com (Muhammad Shahzad) Date: Sun, 17 Feb 2013 19:29:21 +0000 Subject: [Freeswitch-users] Problems with sip provider In-Reply-To: <51211478.1080108@julf.com> References: <51210FA4.1010200@julf.com> <51211478.1080108@julf.com> Message-ID: Can you provide SIP trace of the call on FS and ZyXEL? That may provide some details of the problem. I remembered a similar problem i faced with an ITSP in Pakistan, and after tracing problem for a couple days i found that they were using User Agent filter to allow calls only from their "preferred" devices. So, we had to hack that to get it working with FS, http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#user-agent-string Thank you. On Sun, Feb 17, 2013 at 5:33 PM, Johan Helsingius wrote: > > If they are rejecting it, I'd ask them why. > > I did. > > "What equipment are you using?" > "freeswitch" > "we don't support that" > > > Are you bridging to sofia/gateway/telfort ? > > Yes. > > > Also, if you changed the password on the gateway, did you killgw the old > load > > and restart it? reloadxml doesn't restart gateways that are already > connected. > > I restarted freeswitch. > > Julf > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +49 176 99 83 10 85 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130217/9492d4a5/attachment.html From julf at julf.com Sun Feb 17 22:42:23 2013 From: julf at julf.com (Johan Helsingius) Date: Sun, 17 Feb 2013 20:42:23 +0100 Subject: [Freeswitch-users] Problems with sip provider In-Reply-To: References: <51210FA4.1010200@julf.com> <51211478.1080108@julf.com> Message-ID: <5121329F.6050200@julf.com> > Can you provide SIP trace of the call on FS and ZyXEL? That may provide some > details of the problem. On FS, no prob, a bit harder on the ZyXEL, as I don't have access to the outgoing (ADSL) side of it. > I remembered a similar problem i faced with an ITSP in Pakistan, and after > tracing problem for a couple days i found that they were using User Agent filter > to allow calls only from their "preferred" devices. So, we had to hack that to > get it working with FS, Ouch! Nasty! But in the case of telfort.nl, they do support 3rd party hardware. This is what their guide says for the settings: SIP proxy: sip.telefoniedienst.nl SIP service location: sip.telefoniedienst.nl SIP signaling port: UDP 5060 SIP register service location: tel.telefoniedienst.nl SIP register signaling port: UDP 5060 SIP domain: tel.telefoniedienst.nl URL type coding: SIP Registration interval: 6000 seconds Session expires: 1800 seconds Reserved RTP port range: UDP 16384-32767 Codecs: First negotiated G.729 secondly negotiated G.711 a-law type DTMF mode: PCM Sending INVITE Anonymous: Not permitted Echo Cancellation G.168: In use Voice Activation Detection: Not in use SIP SUBSCRIBES messages: Not permitted Early Media acknowledge: Not in Use Julf From gozdal at gmail.com Sun Feb 17 23:07:06 2013 From: gozdal at gmail.com (Marcin Gozdalik) Date: Sun, 17 Feb 2013 21:07:06 +0100 Subject: [Freeswitch-users] FreeSWITCH Message Bus / Shared Key Value Store In-Reply-To: References: <20130212204251.7066d944@mail.tritonwest.net> Message-ID: 2013/2/17 Eliot Gable : > ZMQ also does not work with a fork, which is needed in order to execute any > system commands (like iptables or anything else which has no programming > API). That pretty much eliminates ZMQ as a possibility. My understanding of ZMQ and Spread is that they solve very different problems and they operate on different levels. ZMQ is on much lower levels and does not > I did some research this weekend, and of all the possibilities I could find, > the one that caught my attention the most was Spread: http://www.spread.org/ I did use Spread in a big project, comprising of 40+ servers for cluster membership. It worked very well most of the time, but failed without proper understanding on our side. It took some config files voo-doo (tuning some timeout knobs) to get it running reliably. > If anyone has any other suggestions than Spread, I would like to hear it. > Also, some feedback on item #1 would be great, as I cannot really judge for > everyone else how willing they are to accept such a licensing clause. ZooKeeper? Something based on Paxos: http://libpaxos.sourceforge.net/ ? http://openreplica.org/ ? Never used any of them though. -- Marcin Gozdalik From gozdal at gmail.com Sun Feb 17 23:09:16 2013 From: gozdal at gmail.com (Marcin Gozdalik) Date: Sun, 17 Feb 2013 21:09:16 +0100 Subject: [Freeswitch-users] FreeSWITCH Message Bus / Shared Key Value Store In-Reply-To: References: <20130212204251.7066d944@mail.tritonwest.net> Message-ID: Sent previous mail too fast :/ 2013/2/17 Marcin Gozdalik : > 2013/2/17 Eliot Gable : > >> ZMQ also does not work with a fork, which is needed in order to execute any >> system commands (like iptables or anything else which has no programming >> API). That pretty much eliminates ZMQ as a possibility. > > My understanding of ZMQ and Spread is that they solve very different > problems and they operate on different levels. ZMQ is on much lower > levels and does not... ...guarantee anything like Spread: "Spread services range from reliable message passing to fully ordered messages with delivery guarantees, even in case of computer failures and network partitions". ZMQ is basically just a socket, but a bit easier to use. >> I did some research this weekend, and of all the possibilities I could find, >> the one that caught my attention the most was Spread: http://www.spread.org/ > > > I did use Spread in a big project, comprising of 40+ servers for > cluster membership. It worked very well most of the time, but failed > without proper understanding on our side. It took some config files > voo-doo (tuning some timeout knobs) to get it running reliably. > >> If anyone has any other suggestions than Spread, I would like to hear it. >> Also, some feedback on item #1 would be great, as I cannot really judge for >> everyone else how willing they are to accept such a licensing clause. > > ZooKeeper? > Something based on Paxos: http://libpaxos.sourceforge.net/ ? > http://openreplica.org/ ? > > Never used any of them though. -- Marcin Gozdalik From jaganthoutam at gmail.com Sun Feb 17 23:13:36 2013 From: jaganthoutam at gmail.com (Jagadish Thoutam) Date: Mon, 18 Feb 2013 01:43:36 +0530 Subject: [Freeswitch-users] Multiple Billings with Nibble Billing Message-ID: Hi All, Is there Any option to do Multiple Billing in single call. Like at the same time one call can effect with different accounts like USER ----> RESELLER3--->RESELLER2---->RESELLER3 to detect the balance. Thanks Jagadish -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130218/5bba8628/attachment.html From tomasz.szuster at gmail.com Sun Feb 17 23:18:50 2013 From: tomasz.szuster at gmail.com (Tomasz Szuster) Date: Sun, 17 Feb 2013 21:18:50 +0100 Subject: [Freeswitch-users] Multiple registration issue Message-ID: Hi, I'm trying to use limit logins to one extension using *max-registrations-per-extension = 2* I've set this in *sofia profile internal.* * * >From what I see this is not working as expected. I've logged in one extensions from 3 different machines at the same time, and successfully established connections. To test I used two computers with X-lite, and Sipdroit on my android phone. Additional I've used: *multiple-registrations = false* * * Still far from expectations I've observed this on FreeSWITCH (Version 1.2.6 git a424765 2013-01-04 15:45:59Z) running on ubuntu 12.04 x86_64 I want to achieve situation when extension can be used only by one user at one time. Bellow is my internal sofia profile: Will you be able to help me ? -- Regards Tomasz -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130217/70baeabd/attachment-0001.html From brian.wiese.freeswitch at gmail.com Sun Feb 17 23:32:08 2013 From: brian.wiese.freeswitch at gmail.com (Brian Wiese) Date: Sun, 17 Feb 2013 14:32:08 -0600 Subject: [Freeswitch-users] mod_directory - Cannot locate domain In-Reply-To: References: Message-ID: Lloyd: Just guessing... it looks like the section tag is not closed. ~Brian On Sat, Feb 16, 2013 at 8:58 PM, Lloyd Aloysius wrote: > Hi All > > I have a strange problem with the mod_directory. I use xml_curl for all the > applications. Only mod_directory giving this trouble. > > > EXECUTE sofia/sipinterface_1/marv at mydomain.com directory(mydomain.com > mydomain.com) > 2013-02-16 21:43:04.601315 [DEBUG] mod_directory.c:518 [mydomain.com] rwlock > > 2013-02-16 21:43:04.621318 [ERR] switch_xml.c:1650 Error[[error near line > 1]: unclosed tag ] > 2013-02-16 21:43:04.621318 [WARNING] mod_directory.c:545 Cannot locate > domain mydomain.com > > -- > > my direcotroy.conf xml curl output > > >
> > > > > > > > > > > > > > > > > > > > > > > Any help is appreciated. > > Thanks > Lloyd > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 steveayre at gmail.com Sun Feb 17 23:36:18 2013 From: steveayre at gmail.com (Steven Ayre) Date: Sun, 17 Feb 2013 20:36:18 +0000 Subject: [Freeswitch-users] Multiple registration issue In-Reply-To: References: Message-ID: Registrations don't get logged out, they get renewed/expire or replaced. Your 3rd machine will replace the first registration. -Steve On 17 February 2013 20:18, Tomasz Szuster wrote: > Hi, > > > I'm trying to use limit logins to one extension using > > *max-registrations-per-extension = 2* > > > I've set this in *sofia profile internal.* > * > * > From what I see this is not working as expected. > > I've logged in one extensions from 3 different machines at the same time, > and successfully established connections. > To test I used two computers with X-lite, and Sipdroit on my android phone. > > Additional I've used: > > *multiple-registrations = false* > * > * > Still far from expectations > > I've observed this on FreeSWITCH (Version 1.2.6 git a424765 2013-01-04 > 15:45:59Z) running on ubuntu 12.04 x86_64 > > I want to achieve situation when extension can be used only by one user at > one time. > > > Bellow is my internal sofia profile: > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > value="false"/> > > > > > > value="$${global_codec_prefs}"/> > value="$${global_codec_prefs}"/> > > > > > > > > > value="${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> > > value="$${domain},$${local_ip_v4}"/> > value="$${presence_privacy}"/> > > > > > > > > > > > > > > > value="true"/> > > > > > > > value="$${domain}"/> > > > > > > > > > > Will you be able to help me ? > > -- > Regards > > Tomasz > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130217/4334d123/attachment.html From cal.leeming at simplicitymedialtd.co.uk Sun Feb 17 23:55:17 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Sun, 17 Feb 2013 20:55:17 +0000 Subject: [Freeswitch-users] Storing CDR UUIDs in MySQL Message-ID: Hi all, Just spent a little while comparing the performance of heavy INSERT/SELECT performance against UUID fields in a MySQL database. Full performance breakdown and write up can be found here; http://blog.simplicitymedialtd.co.uk/?p=437 So far, the best option seems to be storing the UUID as a BINARY(16) - anyone have any thoughts on this? Thanks Cal -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130217/07e5156b/attachment.html From steveayre at gmail.com Sun Feb 17 23:55:44 2013 From: steveayre at gmail.com (Steven Ayre) Date: Sun, 17 Feb 2013 20:55:44 +0000 Subject: [Freeswitch-users] Problems with sip provider In-Reply-To: <51211478.1080108@julf.com> References: <51210FA4.1010200@julf.com> <51211478.1080108@julf.com> Message-ID: > > "What equipment are you using?" > "freeswitch" > "we don't support that" A generic SIP standards-compliant client... They may be trying to answer the wrong question. You don't need them to tell you how to configure your device to talk to them (a FS specific question), you want them to tell you why the authentication failed (specific to their end). Once we know that it should be possible to know what needs to be adjusted. -Steve On 17 February 2013 17:33, Johan Helsingius wrote: > > If they are rejecting it, I'd ask them why. > > I did. > > "What equipment are you using?" > "freeswitch" > "we don't support that" > > > Are you bridging to sofia/gateway/telfort ? > > Yes. > > > Also, if you changed the password on the gateway, did you killgw the old > load > > and restart it? reloadxml doesn't restart gateways that are already > connected. > > I restarted freeswitch. > > Julf > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130217/dc80b582/attachment-0001.html From tomasz.szuster at gmail.com Sun Feb 17 23:56:54 2013 From: tomasz.szuster at gmail.com (Tomasz Szuster) Date: Sun, 17 Feb 2013 21:56:54 +0100 Subject: [Freeswitch-users] Multiple registration issue In-Reply-To: References: Message-ID: Thx for explanation. But how this can be possible that I can call at the same time from 3 different softphones registered into one extension, even if I've set max-registrations-per-extension and multiple-gistrations ? Tom On Sun, Feb 17, 2013 at 9:36 PM, Steven Ayre wrote: > Registrations don't get logged out, they get renewed/expire or replaced. > > Your 3rd machine will replace the first registration. > > -Steve > > > > On 17 February 2013 20:18, Tomasz Szuster wrote: > >> Hi, >> >> >> I'm trying to use limit logins to one extension using >> >> *max-registrations-per-extension = 2* >> >> >> I've set this in *sofia profile internal.* >> * >> * >> From what I see this is not working as expected. >> >> I've logged in one extensions from 3 different machines at the same time, >> and successfully established connections. >> To test I used two computers with X-lite, and Sipdroit on my android >> phone. >> >> Additional I've used: >> >> *multiple-registrations = false* >> * >> * >> Still far from expectations >> >> I've observed this on FreeSWITCH (Version 1.2.6 git a424765 2013-01-04 >> 15:45:59Z) running on ubuntu 12.04 x86_64 >> >> I want to achieve situation when extension can be used only by one user >> at one time. >> >> >> Bellow is my internal sofia profile: >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> > value="false"/> >> >> >> >> >> >> > value="$${global_codec_prefs}"/> >> > value="$${global_codec_prefs}"/> >> >> >> >> >> >> >> >> >> > value="${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> >> >> > value="$${domain},$${local_ip_v4}"/> >> > value="$${presence_privacy}"/> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> > value="true"/> >> >> >> >> >> >> >> > value="$${domain}"/> >> > value="$${domain}"/> >> >> >> >> >> >> >> >> >> Will you be able to help me ? >> >> -- >> Regards >> >> Tomasz >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Pozdrawiam Tomasz -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130217/9f2aa57a/attachment.html From steveayre at gmail.com Mon Feb 18 00:07:28 2013 From: steveayre at gmail.com (Steven Ayre) Date: Sun, 17 Feb 2013 21:07:28 +0000 Subject: [Freeswitch-users] Storing CDR UUIDs in MySQL In-Reply-To: References: Message-ID: Just bear in mind that UUIDs may not always be 128bit. Although frankly that's unlikely to the be case unless inbound-use-callid-as-uuid is used so if this is for your own application then that should be fine. -Steve On 17 February 2013 20:55, Cal Leeming [Simplicity Media Ltd] < cal.leeming at simplicitymedialtd.co.uk> wrote: > Hi all, > > Just spent a little while comparing the performance of heavy INSERT/SELECT > performance against UUID fields in a MySQL database. > > Full performance breakdown and write up can be found here; > http://blog.simplicitymedialtd.co.uk/?p=437 > > So far, the best option seems to be storing the UUID as a BINARY(16) - > anyone have any thoughts on this? > > Thanks > > Cal > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130217/0678d3c4/attachment-0001.html From steveayre at gmail.com Mon Feb 18 00:11:48 2013 From: steveayre at gmail.com (Steven Ayre) Date: Sun, 17 Feb 2013 21:11:48 +0000 Subject: [Freeswitch-users] Storing CDR UUIDs in MySQL In-Reply-To: References: Message-ID: Another possibility you haven't looked at is having a table of UUIDs with a BIGINT auto_increment PK and using that as keys in other takes with a foreign key relationship (whether defined or implied). That would give you a smaller integer (hence faster select/insert/join/etc) that's still guaranteed to be unique. Though the auto_increment INSERT could then become a bottleneck (hint run a recent 5.5/5.6 MySQL version at the very least since that bottleneck while still present was very much reduced). -Steve On 17 February 2013 20:55, Cal Leeming [Simplicity Media Ltd] < cal.leeming at simplicitymedialtd.co.uk> wrote: > Hi all, > > Just spent a little while comparing the performance of heavy INSERT/SELECT > performance against UUID fields in a MySQL database. > > Full performance breakdown and write up can be found here; > http://blog.simplicitymedialtd.co.uk/?p=437 > > So far, the best option seems to be storing the UUID as a BINARY(16) - > anyone have any thoughts on this? > > Thanks > > Cal > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130217/7dca34df/attachment.html From steveayre at gmail.com Mon Feb 18 00:29:03 2013 From: steveayre at gmail.com (Steven Ayre) Date: Sun, 17 Feb 2013 21:29:03 +0000 Subject: [Freeswitch-users] Multiple registration issue In-Reply-To: References: Message-ID: Registration and call authentication are unrelated, although usually configured together. Registration is solely for you to register the IP:port where your user is available to *receive *calls. You can call into FreeSWITCH without being registered. Calls (INVITE) are authenticated separately - you'll see INVITE, 401 Unauthorised, 2nd INVITE. -Steve On 17 February 2013 20:56, Tomasz Szuster wrote: > Thx for explanation. > > But how this can be possible that I can call at the same time from 3 > different softphones registered into one extension, even if I've set > max-registrations-per-extension and multiple-gistrations ? > > Tom > > > On Sun, Feb 17, 2013 at 9:36 PM, Steven Ayre wrote: > >> Registrations don't get logged out, they get renewed/expire or replaced. >> >> Your 3rd machine will replace the first registration. >> >> -Steve >> >> >> >> On 17 February 2013 20:18, Tomasz Szuster wrote: >> >>> Hi, >>> >>> >>> I'm trying to use limit logins to one extension using >>> >>> *max-registrations-per-extension = 2* >>> >>> >>> I've set this in *sofia profile internal.* >>> * >>> * >>> From what I see this is not working as expected. >>> >>> I've logged in one extensions from 3 different machines at the same >>> time, and successfully established connections. >>> To test I used two computers with X-lite, and Sipdroit on my android >>> phone. >>> >>> Additional I've used: >>> >>> *multiple-registrations = false* >>> * >>> * >>> Still far from expectations >>> >>> I've observed this on FreeSWITCH (Version 1.2.6 git a424765 2013-01-04 >>> 15:45:59Z) running on ubuntu 12.04 x86_64 >>> >>> I want to achieve situation when extension can be used only by one user >>> at one time. >>> >>> >>> Bellow is my internal sofia profile: >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >> value="false"/> >>> >>> >>> >>> >>> >>> >> value="$${global_codec_prefs}"/> >>> >> value="$${global_codec_prefs}"/> >>> >>> >>> >>> >>> >>> >>> >>> >>> >> value="${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> >>> >>> >> value="$${domain},$${local_ip_v4}"/> >>> >> value="$${presence_privacy}"/> >>> >> value="generous"/> >>> >>> >>> >>> >> value="$${internal_tls_port}"/> >>> >>> >>> >>> >>> >>> >>> >>> >>> >> value="$${internal_auth_calls}"/> >>> >> value="true"/> >>> >>> >>> >>> >>> >>> >>> >> value="$${domain}"/> >>> >> value="$${domain}"/> >>> >>> >>> >>> >>> >>> >>> >>> >>> Will you be able to help me ? >>> >>> -- >>> Regards >>> >>> Tomasz >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > > -- > Pozdrawiam > Tomasz > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130217/447a300e/attachment-0001.html From jaybinks at gmail.com Mon Feb 18 00:30:16 2013 From: jaybinks at gmail.com (jay binks) Date: Mon, 18 Feb 2013 07:30:16 +1000 Subject: [Freeswitch-users] Please suggest one "SIP Provider" In-Reply-To: <3852fa31.3c24.13ce8295481.Coremail.sparklezou@163.com> References: <3852fa31.3c24.13ce8295481.Coremail.sparklezou@163.com> Message-ID: HAHA ok then !! www.netsip.com.au provide great service , excellent call quality at reasonable prices. what geographic region are you in ? :) Jay On 17 February 2013 22:38, sparklezou wrote: > Hi All, > > Could you please share your experience of the "SIP provider"? > > Which one is good at the quality and price? > > Thanks! > > 2013-02-17 > ________________________________ > sparklezou > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -- Sincerely Jay From steveayre at gmail.com Mon Feb 18 00:31:05 2013 From: steveayre at gmail.com (Steven Ayre) Date: Sun, 17 Feb 2013 21:31:05 +0000 Subject: [Freeswitch-users] ooH323 vs mod_h323 In-Reply-To: References: <2BD0DC60-D428-4323-A020-D9AD20E76553@freeswitch.org> <4C2EA249.4070704@coppice.org> <267C8398-6CA3-442E-B973-BE096D728B28@freeswitch.org> <77B9FF50-213F-4BF6-917E-FBB006DF5C76@gmail.com> Message-ID: Ha. Wouldn't surprise me at all. :o> On 17 February 2013 05:37, Anthony Minessale wrote: > Fun fact... > > Guess who Digium paid to hash out the skeleton for that ooh323 mod....?? > > ;) > On Feb 15, 2013 7:01 PM, "Steven Ayre" wrote: > >> mod_opal and mod_h323 are based on the opal and h323plus libraries. Both >> are forks of the open323 library (which is discontinued). h323plus aims to >> remain API compatible with open323, while it seems opal aims to become a >> more generic telecoms library with support for SIP IAX etc too. >> >> ooh323c is a completely separate library with no shared code. >> >> mod_h323 and mod_opal both seem to have stability problems. Partly that >> seems to come from the heavy dependance of the libraries on specific >> matching versions of ptlib, partly that seems to be the libraries >> themselves. Both Asterisk and YATE appear to have stability problems using >> h323plus/openh323. Anecdotally the ooh323c Asterisk module is more stable... >> >> I'd need to look into it further, but the ooh323c library implements H323 >> and leaves RTP/codecs to the app. h323plus/opal implement them themselves >> (overridable I imagine). That means adding support to ooh323c may end up >> being easier, in theory. But that's a guess without researching it... >> >> Steve >> >> >> >> On 15 Feb 2013, at 23:36, Seven Du wrote: >> >> So we basically have 3 options, Opal, h323 and ooh323c, what's the >> difference among them? >> >> Also, I'm aware that video is not supported in the current mod_opal and >> mod_h323, is it possible(or how hard) to add video support? Or does >> ooh323c easy to support video? >> >> We'd like to contribute a little money to speed it up if it can support >> video and some want to work on it. >> >> -- >> Seven Du >> http://www.freeswitch.org.cn >> http://about.me/dujinfang >> http://www.dujinfang.com >> >> Sent with Sparrow >> >> On Friday, February 15, 2013 at 11:37 PM, Steven Ayre wrote: >> >> There's a module for Asterisk but none for FS I'm aware of yet - I'm only >> wondering about the feasibility of writing one. >> >> -Steve >> >> >> >> On 15 February 2013 14:48, Brian West wrote: >> >> So I have the impression you or someone has an ooh323 module and hasn't >> contributed it back to the project yet? >> -- >> Brian West >> brian at freeswitch.org >> FreeSWITCH Solutions, LLC >> PO BOX PO BOX 2531 >> Brookfield, WI 53008-2531 >> Twitter: @FreeSWITCH_Wire >> T: +1.918.420.9266 | F: +1.918.420.9267 | M: +1.918.424.WEST >> iNUM: +883 5100 1420 9266 >> UK: +44 20 3298 4900 >> ISN: 410*543 >> >> >> >> >> >> On Feb 15, 2013, at 3:49 AM, Steven Ayre wrote: >> >> > Hi Brian, >> > >> > The issue would be that Asterisk's modifications are on a version >> predating the FLOSS exemption and their project is GPL - so presumably >> their modifications are all GPL-only? >> > >> > -Steve >> > >> > >> > >> > On 15 February 2013 00:40, Brian West wrote: >> > If someone wanted to build ooh323 module for FreeSWITCH we do have >> permission to do so using the lib. >> > >> > /b >> > >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130217/4bb73427/attachment.html From roger.castaldo at gmail.com Mon Feb 18 00:50:38 2013 From: roger.castaldo at gmail.com (Roger Castaldo) Date: Sun, 17 Feb 2013 16:50:38 -0500 Subject: [Freeswitch-users] FreeSWITCH Message Bus / Shared Key Value Store In-Reply-To: References: <20130212204251.7066d944@mail.tritonwest.net> Message-ID: This is just my two cents but could we not use snmp as a basis for something like this since it would allow monitoring eternally using ones own systems, or allow each FreeSWITCH server to monitor the other servers over snmp. At least I think it could be an option to explore anyways. On Feb 17, 2013 3:12 PM, "Marcin Gozdalik" wrote: > Sent previous mail too fast :/ > > 2013/2/17 Marcin Gozdalik : > > 2013/2/17 Eliot Gable : > > > >> ZMQ also does not work with a fork, which is needed in order to execute > any > >> system commands (like iptables or anything else which has no programming > >> API). That pretty much eliminates ZMQ as a possibility. > > > > My understanding of ZMQ and Spread is that they solve very different > > problems and they operate on different levels. ZMQ is on much lower > > levels and does not... > > ...guarantee anything like Spread: "Spread services range from > reliable message passing to fully ordered messages with delivery > guarantees, even in case of computer failures and network partitions". > ZMQ is basically just a socket, but a bit easier to use. > > >> I did some research this weekend, and of all the possibilities I could > find, > >> the one that caught my attention the most was Spread: > http://www.spread.org/ > > > > > > I did use Spread in a big project, comprising of 40+ servers for > > cluster membership. It worked very well most of the time, but failed > > without proper understanding on our side. It took some config files > > voo-doo (tuning some timeout knobs) to get it running reliably. > > > >> If anyone has any other suggestions than Spread, I would like to hear > it. > >> Also, some feedback on item #1 would be great, as I cannot really judge > for > >> everyone else how willing they are to accept such a licensing clause. > > > > ZooKeeper? > > Something based on Paxos: http://libpaxos.sourceforge.net/ ? > > http://openreplica.org/ ? > > > > Never used any of them though. > > > -- > Marcin Gozdalik > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130217/26571ea9/attachment-0001.html From emamirazavi at gmail.com Sun Feb 17 18:38:35 2013 From: emamirazavi at gmail.com (Sayyed Mohammad Emami Razavi) Date: Sun, 17 Feb 2013 19:08:35 +0330 Subject: [Freeswitch-users] Auto attendant :: act on no digit entry Message-ID: Freeswitch has not problem! It does this perfectly! but when you go to "number manager" module of Blue.box and edit your number terminates on NO ANSWER and then select External Xfer and select your number! your settings will be saved in DB but not in FS XML! I think it's a good idea if i edit BB code! What is your opinion? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130217/ae2a78b9/attachment.html From cal.leeming at simplicitymedialtd.co.uk Mon Feb 18 01:13:57 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Sun, 17 Feb 2013 22:13:57 +0000 Subject: [Freeswitch-users] Storing CDR UUIDs in MySQL In-Reply-To: References: Message-ID: Thanks for the feedback Steve, In ref to auto inc, actually, I have! :) In the code on the blog, it shows that the tables uses a BIGINT auto increment for each row, and the performance was still good. However, using this approach introduces an unnecessary JOIN and uses more index and table space - unless I've missed something? Could you elaborate more on the UUIDs that may not be 128-bit? My google-fu failed me :/ Cal On Sun, Feb 17, 2013 at 9:11 PM, Steven Ayre wrote: > Another possibility you haven't looked at is having a table of UUIDs with > a BIGINT auto_increment PK and using that as keys in other takes with a > foreign key relationship (whether defined or implied). That would give you > a smaller integer (hence faster select/insert/join/etc) that's still > guaranteed to be unique. Though the auto_increment INSERT could then become > a bottleneck (hint run a recent 5.5/5.6 MySQL version at the very least > since that bottleneck while still present was very much reduced). > > -Steve > > > > On 17 February 2013 20:55, Cal Leeming [Simplicity Media Ltd] < > cal.leeming at simplicitymedialtd.co.uk> wrote: > >> Hi all, >> >> Just spent a little while comparing the performance of heavy >> INSERT/SELECT performance against UUID fields in a MySQL database. >> >> Full performance breakdown and write up can be found here; >> http://blog.simplicitymedialtd.co.uk/?p=437 >> >> So far, the best option seems to be storing the UUID as a BINARY(16) - >> anyone have any thoughts on this? >> >> Thanks >> >> Cal >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130217/76c0330d/attachment.html From dujinfang at gmail.com Mon Feb 18 01:39:37 2013 From: dujinfang at gmail.com (Seven Du) Date: Mon, 18 Feb 2013 06:39:37 +0800 Subject: [Freeswitch-users] Storing CDR UUIDs in MySQL In-Reply-To: References: Message-ID: you can use any arbitrary string as uuid, say originate {origination_uuid=my-uuid-that-is-not-128-bit}user/1000 ?. On Monday, February 18, 2013 at 6:13 AM, Cal Leeming [Simplicity Media Ltd] wrote: > Thanks for the feedback Steve, > > In ref to auto inc, actually, I have! :) In the code on the blog, it shows that the tables uses a BIGINT auto increment for each row, and the performance was still good. However, using this approach introduces an unnecessary JOIN and uses more index and table space - unless I've missed something? > > Could you elaborate more on the UUIDs that may not be 128-bit? My google-fu failed me :/ > > Cal > > On Sun, Feb 17, 2013 at 9:11 PM, Steven Ayre wrote: > > Another possibility you haven't looked at is having a table of UUIDs with a BIGINT auto_increment PK and using that as keys in other takes with a foreign key relationship (whether defined or implied). That would give you a smaller integer (hence faster select/insert/join/etc) that's still guaranteed to be unique. Though the auto_increment INSERT could then become a bottleneck (hint run a recent 5.5/5.6 MySQL version at the very least since that bottleneck while still present was very much reduced). > > > > -Steve > > > > > > > > On 17 February 2013 20:55, Cal Leeming [Simplicity Media Ltd] wrote: > > > Hi all, > > > > > > Just spent a little while comparing the performance of heavy INSERT/SELECT performance against UUID fields in a MySQL database. > > > > > > Full performance breakdown and write up can be found here; > > > http://blog.simplicitymedialtd.co.uk/?p=437 > > > > > > So far, the best option seems to be storing the UUID as a BINARY(16) - anyone have any thoughts on this? > > > > > > Thanks > > > > > > Cal > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > > > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org (mailto: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 > > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org (mailto: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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto: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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130218/0bfbd35c/attachment.html From egable+freeswitch at gmail.com Mon Feb 18 01:51:45 2013 From: egable+freeswitch at gmail.com (Eliot Gable) Date: Sun, 17 Feb 2013 17:51:45 -0500 Subject: [Freeswitch-users] FreeSWITCH Message Bus / Shared Key Value Store In-Reply-To: References: <20130212204251.7066d944@mail.tritonwest.net> Message-ID: On Sun, Feb 17, 2013 at 4:50 PM, Roger Castaldo wrote: > This is just my two cents but could we not use snmp as a basis for > something like this since it would allow monitoring eternally using ones > own systems, or allow each FreeSWITCH server to monitor the other servers > over snmp. At least I think it could be an option to explore anyways. > SNMP is not intended as an HA solution. It is slow, and unwieldy to use. Fail over time using SNMP would be measured in minutes. We are going for sub-second fail over. SNMP is really only intended for external monitoring. SNMP also does not provide a solution for the vast majority of issues which an HA solution must cover. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130217/141e9d2d/attachment-0001.html From steveayre at gmail.com Mon Feb 18 02:02:02 2013 From: steveayre at gmail.com (Steven Ayre) Date: Sun, 17 Feb 2013 23:02:02 +0000 Subject: [Freeswitch-users] Storing CDR UUIDs in MySQL In-Reply-To: References: Message-ID: > > In ref to auto inc, actually, I have! :) In the code on the blog, it shows > that the tables uses a BIGINT auto increment for each row, and the > performance was still good. However, using this approach introduces an > unnecessary JOIN and uses more index and table space - unless I've missed > something? Ah, sorry - missed that somehow. :o) In that case it might be simpler to store the UUID in a single field. Comparing multiple fields I imagine would have a larger overhead than a single comparison. The extra table does uses more space, but you're shrinking the key size enough that you're going to save that much in the other tables. It'd shrink the corresponding indexes too, and that'll speed up index lookups (which'll affect both INSERT and SELECT) - both because the comparison is smaller and because you read more index entries in a single disk access. That may be enough of a benefit that the queries are faster despite the additional JOIN. That doesn't appear to be the case in your benchmarks, but it may be worth repeating with the UUID in a single field. You don't mention many details about your test setup - CPU, RAM, Disks, MySQL configuration etc. All of which could have a large impact. The version of MySQL / InnoDB will also have a large impact - there have been many improvements between versions. 5.6 has just become GA, although few people will be running it yet. The main optimization for Innodb would be to set the buffer pool to be as large as possible to minimize disk I/O. In an ideal world as large as your entire dataset (or at least what you actively use) Though of course that's not realistic in many cases. -Steve On 17 February 2013 22:13, Cal Leeming [Simplicity Media Ltd] < cal.leeming at simplicitymedialtd.co.uk> wrote: > Thanks for the feedback Steve, > > In ref to auto inc, actually, I have! :) In the code on the blog, it shows > that the tables uses a BIGINT auto increment for each row, and the > performance was still good. However, using this approach introduces an > unnecessary JOIN and uses more index and table space - unless I've missed > something? > > Could you elaborate more on the UUIDs that may not be 128-bit? My > google-fu failed me :/ > > Cal > > > On Sun, Feb 17, 2013 at 9:11 PM, Steven Ayre wrote: > >> Another possibility you haven't looked at is having a table of UUIDs with >> a BIGINT auto_increment PK and using that as keys in other takes with a >> foreign key relationship (whether defined or implied). That would give you >> a smaller integer (hence faster select/insert/join/etc) that's still >> guaranteed to be unique. Though the auto_increment INSERT could then become >> a bottleneck (hint run a recent 5.5/5.6 MySQL version at the very least >> since that bottleneck while still present was very much reduced). >> >> -Steve >> >> >> >> On 17 February 2013 20:55, Cal Leeming [Simplicity Media Ltd] < >> cal.leeming at simplicitymedialtd.co.uk> wrote: >> >>> Hi all, >>> >>> Just spent a little while comparing the performance of heavy >>> INSERT/SELECT performance against UUID fields in a MySQL database. >>> >>> Full performance breakdown and write up can be found here; >>> http://blog.simplicitymedialtd.co.uk/?p=437 >>> >>> So far, the best option seems to be storing the UUID as a BINARY(16) - >>> anyone have any thoughts on this? >>> >>> Thanks >>> >>> Cal >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130217/86ff68bb/attachment.html From steveayre at gmail.com Mon Feb 18 02:07:06 2013 From: steveayre at gmail.com (Steven Ayre) Date: Sun, 17 Feb 2013 23:07:06 +0000 Subject: [Freeswitch-users] Storing CDR UUIDs in MySQL In-Reply-To: References: Message-ID: It's worth mentioning that the speedup I expect is BIGINT vs CHAR. BINARY(16) is a 128bit number vs 64bit BIGINT number - similar enough I wouldn't expect much difference and without requiring the extra JOIN not surprising to me that BINARY is faster. It's just less usable since it requires conversion, but that's a trivial task to do within your application. -Steve On 17 February 2013 23:02, Steven Ayre wrote: > In ref to auto inc, actually, I have! :) In the code on the blog, it shows >> that the tables uses a BIGINT auto increment for each row, and the >> performance was still good. However, using this approach introduces an >> unnecessary JOIN and uses more index and table space - unless I've missed >> something? > > > Ah, sorry - missed that somehow. :o)req > > In that case it might be simpler to store the UUID in a single field. > Comparing multiple fields I imagine would have a larger overhead than a > single comparison. > > The extra table does uses more space, but you're shrinking the key size > enough that you're going to save that much in the other tables. It'd shrink > the corresponding indexes too, and that'll speed up index lookups (which'll > affect both INSERT and SELECT) - both because the comparison is smaller and > because you read more index entries in a single disk access. That may be > enough of a benefit that the queries are faster despite the additional JOIN. > > That doesn't appear to be the case in your benchmarks, but it may be worth > repeating with the UUID in a single field. > > > > > You don't mention many details about your test setup - CPU, RAM, Disks, > MySQL configuration etc. All of which could have a large impact. The > version of MySQL / InnoDB will also have a large impact - there have been > many improvements between versions. 5.6 has just become GA, although few > people will be running it yet. > > The main optimization for Innodb would be to set the buffer pool to be as > large as possible to minimize disk I/O. In an ideal world as large as your > entire dataset (or at least what you actively use) Though of course that's > not realistic in many cases. > > -Steve > > > > > On 17 February 2013 22:13, Cal Leeming [Simplicity Media Ltd] < > cal.leeming at simplicitymedialtd.co.uk> wrote: > >> Thanks for the feedback Steve, >> >> In ref to auto inc, actually, I have! :) In the code on the blog, it >> shows that the tables uses a BIGINT auto increment for each row, and the >> performance was still good. However, using this approach introduces an >> unnecessary JOIN and uses more index and table space - unless I've missed >> something? >> >> Could you elaborate more on the UUIDs that may not be 128-bit? My >> google-fu failed me :/ >> >> Cal >> >> >> On Sun, Feb 17, 2013 at 9:11 PM, Steven Ayre wrote: >> >>> Another possibility you haven't looked at is having a table of UUIDs >>> with a BIGINT auto_increment PK and using that as keys in other takes with >>> a foreign key relationship (whether defined or implied). That would give >>> you a smaller integer (hence faster select/insert/join/etc) that's still >>> guaranteed to be unique. Though the auto_increment INSERT could then become >>> a bottleneck (hint run a recent 5.5/5.6 MySQL version at the very least >>> since that bottleneck while still present was very much reduced). >>> >>> -Steve >>> >>> >>> >>> On 17 February 2013 20:55, Cal Leeming [Simplicity Media Ltd] < >>> cal.leeming at simplicitymedialtd.co.uk> wrote: >>> >>>> Hi all, >>>> >>>> Just spent a little while comparing the performance of heavy >>>> INSERT/SELECT performance against UUID fields in a MySQL database. >>>> >>>> Full performance breakdown and write up can be found here; >>>> http://blog.simplicitymedialtd.co.uk/?p=437 >>>> >>>> So far, the best option seems to be storing the UUID as a BINARY(16) - >>>> anyone have any thoughts on this? >>>> >>>> Thanks >>>> >>>> Cal >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130217/a2552686/attachment-0001.html From steveayre at gmail.com Mon Feb 18 02:10:39 2013 From: steveayre at gmail.com (Steven Ayre) Date: Sun, 17 Feb 2013 23:10:39 +0000 Subject: [Freeswitch-users] FreeSWITCH Message Bus / Shared Key Value Store In-Reply-To: References: <20130212204251.7066d944@mail.tritonwest.net> Message-ID: It may be worth mentioning there's a mod_snmp module for external monitoring. While that's also possible via ESL, XML RPC etc where it's better tested, SNMP at least allows integration with tools like Nagios. But as Eliot says, SNMP is slow and unwieldy and requires high timeouts. It's not something you're going to be able to use for failover. -Steve On 17 February 2013 22:51, Eliot Gable wrote: > On Sun, Feb 17, 2013 at 4:50 PM, Roger Castaldo wrote: > >> This is just my two cents but could we not use snmp as a basis for >> something like this since it would allow monitoring eternally usiNng ones >> own systems, or allow each FreeSWITCH server to monitor the other servers >> over snmp. At least I think it could be an option to explore anyways. >> > > SNMP is not intended as an HA solution. It is slow, and unwieldy to use. > Fail over time using SNMP would be measured in minutes. We are going for > sub-second fail over. SNMP is really only intended for external monitoring. > SNMP also does not provide a solution for the vast majority of issues which > an HA solution must cover. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130217/6f3339f4/attachment.html From steveayre at gmail.com Mon Feb 18 02:15:55 2013 From: steveayre at gmail.com (Steven Ayre) Date: Sun, 17 Feb 2013 23:15:55 +0000 Subject: [Freeswitch-users] Storing CDR UUIDs in MySQL In-Reply-To: References: Message-ID: +1 Also as an example the 'inbound-use-callid-as-uuid' parameter I mentioned before is a Sofia profile parameter that uses the Call-ID header value from the INVITE sent by the caller as the UUID for the channel. That could be useful for example in matching up channels against SIP traces as you only need to store 1 ID not 2. I frequently see ones for example of $UUID@$ SERVER.COM - far longer than 128bits, and using characters other than 0-9A-F so that it can't be folded into bits. In this case a CHAR column becomes most flexible and the BIGINT column gives you the speed advantage of the BINARY. -Steve On 17 February 2013 22:39, Seven Du wrote: > you can use any arbitrary string as uuid, say > > originate {origination_uuid=my-uuid-that-is-not-128-bit}user/1000 ?. > > On Monday, February 18, 2013 at 6:13 AM, Cal Leeming [Simplicity Media > Ltd] wrote: > > Thanks for the feedback Steve, > > In ref to auto inc, actually, I have! :) In the code on the blog, it shows > that the tables uses a BIGINT auto increment for each row, and the > performance was still good. However, using this approach introduces an > unnecessary JOIN and uses more index and table space - unless I've missed > something? > > Could you elaborate more on the UUIDs that may not be 128-bit? My > google-fu failed me :/ > > Cal > > On Sun, Feb 17, 2013 at 9:11 PM, Steven Ayre wrote: > > Another possibility you haven't looked at is having a table of UUIDs with > a BIGINT auto_increment PK and using that as keys in other takes with a > foreign key relationship (whether defined or implied). That would give you > a smaller integer (hence faster select/insert/join/etc) that's still > guaranteed to be unique. Though the auto_increment INSERT could then become > a bottleneck (hint run a recent 5.5/5.6 MySQL version at the very least > since that bottleneck while still present was very much reduced). > > -Steve > > > > On 17 February 2013 20:55, Cal Leeming [Simplicity Media Ltd] < > cal.leeming at simplicitymedialtd.co.uk> wrote: > > Hi all, > > Just spent a little while comparing the performance of heavy INSERT/SELECT > performance against UUID fields in a MySQL database. > > Full performance breakdown and write up can be found here; > http://blog.simplicitymedialtd.co.uk/?p=437 > > So far, the best option seems to be storing the UUID as a BINARY(16) - > anyone have any thoughts on this? > > Thanks > > Cal > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130217/89f4262e/attachment.html From cal.leeming at simplicitymedialtd.co.uk Mon Feb 18 02:40:13 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Sun, 17 Feb 2013 23:40:13 +0000 Subject: [Freeswitch-users] Storing CDR UUIDs in MySQL In-Reply-To: References: Message-ID: On Sun, Feb 17, 2013 at 11:02 PM, Steven Ayre wrote: > In ref to auto inc, actually, I have! :) In the code on the blog, it shows >> that the tables uses a BIGINT auto increment for each row, and the >> performance was still good. However, using this approach introduces an >> unnecessary JOIN and uses more index and table space - unless I've missed >> something? > > > Ah, sorry - missed that somehow. :o) I *really* suck at writing these blog articles, and they aren't always a clean/easy read. I'd planned to have some really nice graphs and styling on the page, but time was my enemy! > > In that case it might be simpler to store the UUID in a single field. > Comparing multiple fields I imagine would have a larger overhead than a > single comparison. > > The extra table does uses more space, but you're shrinking the key size > enough that you're going to save that much in the other tables. It'd shrink > the corresponding indexes too, and that'll speed up index lookups (which'll > affect both INSERT and SELECT) - both because the comparison is smaller and > because you read more index entries in a single disk access. That may be > enough of a benefit that the queries are faster despite the additional JOIN. > That doesn't appear to be the case in your benchmarks, but it may be worth > repeating with the UUID in a single field. > Initially I'd planned to put all the UUIDs in a separate table and LEFT JOIN on them (hence the auto inc in the tests) I guess it also depends on how often you're referencing the UUID multiple times, the more times it's used, the more suitable a second table with a JOIN would become for saving index/table space. I agree these tests were not as extensive as they should have been, and probably should have included various JOIN statements, as well as inserting into table tables. I also hadn't tested doing a JOIN on a BINARY either - so might be worth adding this in. > > > > > You don't mention many details about your test setup - CPU, RAM, Disks, > MySQL configuration etc. All of which could have a large impact. The > version of MySQL / InnoDB will also have a large impact - there have been > many improvements between versions. 5.6 has just become GA, although few > people will be running it yet. > I was considering posting this too, but I was more focusing on the ratio difference of the numbers, rather than the numbers themselves. In this particular test though, MySQL was Percona 5.5 and, and configured with a very low innodb space / heap size, so essentially everything was being forced to disk to give a 'worst case' scenario. It would be interesting to see the difference in these numbers using different MySQL configurations for sure > > The main optimization for Innodb would be to set the buffer pool to be as > large as possible to minimize disk I/O. In an ideal world as large as your > entire dataset (or at least what you actively use) Though of course that's > not realistic in many cases. > I did a separate post about this too lol :D http://blog.simplicitymedialtd.co.uk/?p=225 These days RAM is so cheap that we tend to just throw 128GB+ RAM into the server and scale up rather than out, the speed improvement you get from having your entire InnoDB database in memory is massive. And if your chassis won't support enough RAM, then SSDs offers a considerable improvement too. > > -Steve > > > > > On 17 February 2013 22:13, Cal Leeming [Simplicity Media Ltd] < > cal.leeming at simplicitymedialtd.co.uk> wrote: > >> Thanks for the feedback Steve, >> >> In ref to auto inc, actually, I have! :) In the code on the blog, it >> shows that the tables uses a BIGINT auto increment for each row, and the >> performance was still good. However, using this approach introduces an >> unnecessary JOIN and uses more index and table space - unless I've missed >> something? >> >> Could you elaborate more on the UUIDs that may not be 128-bit? My >> google-fu failed me :/ >> >> Cal >> >> >> On Sun, Feb 17, 2013 at 9:11 PM, Steven Ayre wrote: >> >>> Another possibility you haven't looked at is having a table of UUIDs >>> with a BIGINT auto_increment PK and using that as keys in other takes with >>> a foreign key relationship (whether defined or implied). That would give >>> you a smaller integer (hence faster select/insert/join/etc) that's still >>> guaranteed to be unique. Though the auto_increment INSERT could then become >>> a bottleneck (hint run a recent 5.5/5.6 MySQL version at the very least >>> since that bottleneck while still present was very much reduced). >>> >>> -Steve >>> >>> >>> >>> On 17 February 2013 20:55, Cal Leeming [Simplicity Media Ltd] < >>> cal.leeming at simplicitymedialtd.co.uk> wrote: >>> >>>> Hi all, >>>> >>>> Just spent a little while comparing the performance of heavy >>>> INSERT/SELECT performance against UUID fields in a MySQL database. >>>> >>>> Full performance breakdown and write up can be found here; >>>> http://blog.simplicitymedialtd.co.uk/?p=437 >>>> >>>> So far, the best option seems to be storing the UUID as a BINARY(16) - >>>> anyone have any thoughts on this? >>>> >>>> Thanks >>>> >>>> Cal >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130217/e1215c9d/attachment-0001.html From cal.leeming at simplicitymedialtd.co.uk Mon Feb 18 02:47:11 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Sun, 17 Feb 2013 23:47:11 +0000 Subject: [Freeswitch-users] Storing CDR UUIDs in MySQL In-Reply-To: References: Message-ID: Ah, yeah that changes the playing field for sure. The problem with CHAR is the INSERT speed though, and I've seen similar issues on other builds too.. after you push past a few million rows, it really starts to slow down.. If the database was in memory, it might not be such an issue though. In some ways it 'feels cleaner' using a separate table and then JOIN on a BIGINT autoinc column, as the thought of JOIN'ing and GROUP BY on a BINARY seems wrong. Another option would be to take an SHA1 hash of the UUID, then convert that from hex to bin, and put that in the BINARY field.. this way you can guarantee each UUID will be the same size, not worry about changes in data schema, and get the INSERT speed benefits of the BINARY too - thoughts? Cal On Sun, Feb 17, 2013 at 11:15 PM, Steven Ayre wrote: > +1 > > Also as an example the 'inbound-use-callid-as-uuid' parameter I mentioned > before is a Sofia profile parameter that uses the Call-ID header value from > the INVITE sent by the caller as the UUID for the channel. That could be > useful for example in matching up channels against SIP traces as you only > need to store 1 ID not 2. I frequently see ones for example of $UUID@$ > SERVER.COM - far longer than 128bits, and using characters other than > 0-9A-F so that it can't be folded into bits. > > In this case a CHAR column becomes most flexible and the BIGINT column > gives you the speed advantage of the BINARY. > > -Steve > > > > On 17 February 2013 22:39, Seven Du wrote: > >> you can use any arbitrary string as uuid, say >> >> originate {origination_uuid=my-uuid-that-is-not-128-bit}user/1000 ?. >> >> On Monday, February 18, 2013 at 6:13 AM, Cal Leeming [Simplicity Media >> Ltd] wrote: >> >> Thanks for the feedback Steve, >> >> In ref to auto inc, actually, I have! :) In the code on the blog, it >> shows that the tables uses a BIGINT auto increment for each row, and the >> performance was still good. However, using this approach introduces an >> unnecessary JOIN and uses more index and table space - unless I've missed >> something? >> >> Could you elaborate more on the UUIDs that may not be 128-bit? My >> google-fu failed me :/ >> >> Cal >> >> On Sun, Feb 17, 2013 at 9:11 PM, Steven Ayre wrote: >> >> Another possibility you haven't looked at is having a table of UUIDs with >> a BIGINT auto_increment PK and using that as keys in other takes with a >> foreign key relationship (whether defined or implied). That would give you >> a smaller integer (hence faster select/insert/join/etc) that's still >> guaranteed to be unique. Though the auto_increment INSERT could then become >> a bottleneck (hint run a recent 5.5/5.6 MySQL version at the very least >> since that bottleneck while still present was very much reduced). >> >> -Steve >> >> >> >> On 17 February 2013 20:55, Cal Leeming [Simplicity Media Ltd] < >> cal.leeming at simplicitymedialtd.co.uk> wrote: >> >> Hi all, >> >> Just spent a little while comparing the performance of heavy >> INSERT/SELECT performance against UUID fields in a MySQL database. >> >> Full performance breakdown and write up can be found here; >> http://blog.simplicitymedialtd.co.uk/?p=437 >> >> So far, the best option seems to be storing the UUID as a BINARY(16) - >> anyone have any thoughts on this? >> >> Thanks >> >> Cal >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130217/63517e88/attachment.html From cal.leeming at simplicitymedialtd.co.uk Mon Feb 18 02:48:20 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Sun, 17 Feb 2013 23:48:20 +0000 Subject: [Freeswitch-users] Storing CDR UUIDs in MySQL In-Reply-To: References: Message-ID: Thanks for clarifying this! Cal On Sun, Feb 17, 2013 at 10:39 PM, Seven Du wrote: > you can use any arbitrary string as uuid, say > > originate {origination_uuid=my-uuid-that-is-not-128-bit}user/1000 ?. > > On Monday, February 18, 2013 at 6:13 AM, Cal Leeming [Simplicity Media > Ltd] wrote: > > Thanks for the feedback Steve, > > In ref to auto inc, actually, I have! :) In the code on the blog, it shows > that the tables uses a BIGINT auto increment for each row, and the > performance was still good. However, using this approach introduces an > unnecessary JOIN and uses more index and table space - unless I've missed > something? > > Could you elaborate more on the UUIDs that may not be 128-bit? My > google-fu failed me :/ > > Cal > > On Sun, Feb 17, 2013 at 9:11 PM, Steven Ayre wrote: > > Another possibility you haven't looked at is having a table of UUIDs with > a BIGINT auto_increment PK and using that as keys in other takes with a > foreign key relationship (whether defined or implied). That would give you > a smaller integer (hence faster select/insert/join/etc) that's still > guaranteed to be unique. Though the auto_increment INSERT could then become > a bottleneck (hint run a recent 5.5/5.6 MySQL version at the very least > since that bottleneck while still present was very much reduced). > > -Steve > > > > On 17 February 2013 20:55, Cal Leeming [Simplicity Media Ltd] < > cal.leeming at simplicitymedialtd.co.uk> wrote: > > Hi all, > > Just spent a little while comparing the performance of heavy INSERT/SELECT > performance against UUID fields in a MySQL database. > > Full performance breakdown and write up can be found here; > http://blog.simplicitymedialtd.co.uk/?p=437 > > So far, the best option seems to be storing the UUID as a BINARY(16) - > anyone have any thoughts on this? > > Thanks > > Cal > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130217/7cd5a44d/attachment-0001.html From bdfoster at endigotech.com Mon Feb 18 03:00:58 2013 From: bdfoster at endigotech.com (Brian Foster) Date: Sun, 17 Feb 2013 19:00:58 -0500 Subject: [Freeswitch-users] Time to start understanding the FreeSWITCH codebase Message-ID: <9982BBC9-3C49-4646-A578-51E1A1C2781C@endigotech.com> I think I've come to a stage in my life where I'm more interested than ever in learning the internals of FreeSWITCH and start contributing code to it. I've been following the project for a few years, but I've never touched the code. It's time to change that. Is there a place on the wiki someone can point me to so I can start understanding the FS core? I'm talking fairly basic, beginner level stuff. Sent from my iPhone From steveayre at gmail.com Mon Feb 18 03:42:40 2013 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 18 Feb 2013 00:42:40 +0000 Subject: [Freeswitch-users] Time to start understanding the FreeSWITCH codebase In-Reply-To: <9982BBC9-3C49-4646-A578-51E1A1C2781C@endigotech.com> References: <9982BBC9-3C49-4646-A578-51E1A1C2781C@endigotech.com> Message-ID: <17281507-28F9-4AB5-9799-A930C21F0862@gmail.com> Use the source, Luke. Digging around in the source is one way to explore how FS works. Some is easier to understand than others. Threading is heavily used so understanding mutexes etc is useful. I'd highly recommend that you read this site though: http://docs.freeswitch.org/ That documents the core API. A lot is based on APR, plus some extra stuff like the module interfaces. Steve On 18 Feb 2013, at 00:00, Brian Foster wrote: > I think I've come to a stage in my life where I'm more interested than ever in learning the internals of FreeSWITCH and start contributing code to it. I've been following the project for a few years, but I've never touched the code. It's time to change that. > > Is there a place on the wiki someone can point me to so I can start understanding the FS core? I'm talking fairly basic, beginner level stuff. > > Sent from my iPhone > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130218/5b39607f/attachment.html From steveayre at gmail.com Mon Feb 18 03:48:16 2013 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 18 Feb 2013 00:48:16 +0000 Subject: [Freeswitch-users] Storing CDR UUIDs in MySQL In-Reply-To: References: Message-ID: <5ADA3B5B-47A9-4C6F-BC9A-12A575840006@gmail.com> You could take a SHA1, but then might get collisions. That can happen with UUIDs too, but since it includes the MAC address and time stamp is less likely and would be more predictable. The originate_uuid may get collisions depending on what you pick, and the callid puts it outside your control entirely - in those situations you might need to handle collisions anyway. Sent from my iPad On 17 Feb 2013, at 23:47, "Cal Leeming [Simplicity Media Ltd]" wrote: > Ah, yeah that changes the playing field for sure. > > The problem with CHAR is the INSERT speed though, and I've seen similar issues on other builds too.. after you push past a few million rows, it really starts to slow down.. If the database was in memory, it might not be such an issue though. > > In some ways it 'feels cleaner' using a separate table and then JOIN on a BIGINT autoinc column, as the thought of JOIN'ing and GROUP BY on a BINARY seems wrong. > > Another option would be to take an SHA1 hash of the UUID, then convert that from hex to bin, and put that in the BINARY field.. this way you can guarantee each UUID will be the same size, not worry about changes in data schema, and get the INSERT speed benefits of the BINARY too - thoughts? > > Cal > > On Sun, Feb 17, 2013 at 11:15 PM, Steven Ayre wrote: >> +1 >> >> Also as an example the 'inbound-use-callid-as-uuid' parameter I mentioned before is a Sofia profile parameter that uses the Call-ID header value from the INVITE sent by the caller as the UUID for the channel. That could be useful for example in matching up channels against SIP traces as you only need to store 1 ID not 2. I frequently see ones for example of $UUID@$SERVER.COM - far longer than 128bits, and using characters other than 0-9A-F so that it can't be folded into bits. >> >> In this case a CHAR column becomes most flexible and the BIGINT column gives you the speed advantage of the BINARY. >> >> -Steve >> >> >> >> On 17 February 2013 22:39, Seven Du wrote: >>> you can use any arbitrary string as uuid, say >>> >>> originate {origination_uuid=my-uuid-that-is-not-128-bit}user/1000 ?. >>> >>> On Monday, February 18, 2013 at 6:13 AM, Cal Leeming [Simplicity Media Ltd] wrote: >>> >>>> Thanks for the feedback Steve, >>>> >>>> In ref to auto inc, actually, I have! :) In the code on the blog, it shows that the tables uses a BIGINT auto increment for each row, and the performance was still good. However, using this approach introduces an unnecessary JOIN and uses more index and table space - unless I've missed something? >>>> >>>> Could you elaborate more on the UUIDs that may not be 128-bit? My google-fu failed me :/ >>>> >>>> Cal >>>> >>>> On Sun, Feb 17, 2013 at 9:11 PM, Steven Ayre wrote: >>>>> Another possibility you haven't looked at is having a table of UUIDs with a BIGINT auto_increment PK and using that as keys in other takes with a foreign key relationship (whether defined or implied). That would give you a smaller integer (hence faster select/insert/join/etc) that's still guaranteed to be unique. Though the auto_increment INSERT could then become a bottleneck (hint run a recent 5.5/5.6 MySQL version at the very least since that bottleneck while still present was very much reduced). >>>>> >>>>> -Steve >>>>> >>>>> >>>>> >>>>> On 17 February 2013 20:55, Cal Leeming [Simplicity Media Ltd] wrote: >>>>>> Hi all, >>>>>> >>>>>> Just spent a little while comparing the performance of heavy INSERT/SELECT performance against UUID fields in a MySQL database. >>>>>> >>>>>> Full performance breakdown and write up can be found here; >>>>>> http://blog.simplicitymedialtd.co.uk/?p=437 >>>>>> >>>>>> So far, the best option seems to be storing the UUID as a BINARY(16) - anyone have any thoughts on this? >>>>>> >>>>>> Thanks >>>>>> >>>>>> Cal >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> 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 >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> 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 >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130218/cef21faa/attachment-0001.html From shaheryarkh at gmail.com Mon Feb 18 04:35:23 2013 From: shaheryarkh at gmail.com (Muhammad Shahzad) Date: Mon, 18 Feb 2013 02:35:23 +0100 Subject: [Freeswitch-users] Auto attendant :: act on no digit entry In-Reply-To: References: Message-ID: Not sure what your are trying to do or say. But it seems you have some trouble with bluebox from 2600hz. So you should go to its support forum / mailing list and report your bug there. Thank you. On Sun, Feb 17, 2013 at 4:38 PM, Sayyed Mohammad Emami Razavi < emamirazavi at gmail.com> wrote: > Freeswitch has not problem! It does this perfectly! but when you go to > "number manager" module of Blue.box and > edit your number terminates on NO ANSWER and then select External Xfer and > select your number! your settings will be saved in DB but > not in FS XML! I think it's a good idea if i edit BB code! What is your > opinion? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +49 176 99 83 10 85 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130218/f2acde11/attachment.html From shaheryarkh at gmail.com Mon Feb 18 04:55:22 2013 From: shaheryarkh at gmail.com (Muhammad Shahzad) Date: Mon, 18 Feb 2013 02:55:22 +0100 Subject: [Freeswitch-users] Change Voicemail IVR Menu Message-ID: Hi, I am trying to change menu order of FS voicemail module. I am using voicemail_ivr with xml_curl, when a voicemail user checks voicemail, i check if user has his own greeting set, then play normal voicemail menu, otherwise instead of playing message count and related options, i simply want to play voicemail preferences menu, in fact only voicemail record greeting menu. I have tried many different possibilities including many changes to voicemail_ivr.xml and voicemail_ivr.conf.xml but so far none worked. FS Wiki on voicemail_ivr module seems outdated. Till now i am only able to play standard voicemail menu through voicemail_ivr module. Do I have to modify native code of these modules (mod_voicemail and mod_voicemail_ivr) to achieve the goal? Any suggestions / guidelines? P.S. Note that sample files (voicemail_ivr.conf.xml and voicemail_ivr.xml) do not work as is, i had to modify many typos and tag attributes to get it working same as standard mod_voicemail works. I will update wiki once this problem is solved. Thank you. -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +49 176 99 83 10 85 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130218/9d04db6e/attachment.html From krice at freeswitch.org Mon Feb 18 04:45:47 2013 From: krice at freeswitch.org (Ken Rice) Date: Sun, 17 Feb 2013 19:45:47 -0600 Subject: [Freeswitch-users] Storing CDR UUIDs in MySQL In-Reply-To: <5ADA3B5B-47A9-4C6F-BC9A-12A575840006@gmail.com> Message-ID: Why don?t you do something like look at something other than mysql like PostgreSQL, we insert millions of CDRs per day there and don?t have performance issues due to Table Partitioning, this allows us to cram well over 100million new rows/month into a table, partitioned daily along something like a call_date field (derived by dropping the time from a full time/date stamp). This also allows us to do really speedy reports without having to deal with what happens if you do something like select count(*) from cdr where call_date = ?2013-02-15?; this would be a monster of a query on a table with say 90 days of CDRs at 4million CDRs/day (360million CDRs total) but by using the table partitions with a date constraint, the query planer eliminates partitions that do not match the date constraint the speeds along the query. The caveat to this is if you don?t give it some sort of date constraint you end up looking across many partitions for the data... On 2/17/13 6:48 PM, "Steven Ayre" wrote: > You could take a SHA1, but then might get collisions. That can happen with > UUIDs too, but since it includes the MAC address and time stamp is less likely > and would be more predictable. > > The originate_uuid may get collisions depending on what you pick, and the > callid puts it outside your control entirely - in those situations you might > need to handle collisions anyway. > > Sent from my iPad > > > > On 17 Feb 2013, at 23:47, "Cal Leeming [Simplicity Media Ltd]" > wrote: > >> Ah, yeah that changes the playing field for sure. >> >> The problem with CHAR is the INSERT speed though, and I've seen similar >> issues on other builds too.. after you push past a few million rows, it >> really starts to slow down.. If the database was in memory, it might not be >> such an issue though. >> >> In some ways it 'feels cleaner' using a separate table and then JOIN on a >> BIGINT autoinc column, as the thought of JOIN'ing and GROUP BY on a BINARY >> seems wrong. >> >> Another option would be to take an SHA1 hash of the UUID, then convert that >> from hex to bin, and put that in the BINARY field.. this way you can >> guarantee each UUID will be the same size, not worry about changes in data >> schema, and get the INSERT speed benefits of the BINARY too - thoughts? >> >> Cal >> >> On Sun, Feb 17, 2013 at 11:15 PM, Steven Ayre wrote: >>> +1 >>> >>> Also as an example the 'inbound-use-callid-as-uuid' parameter I mentioned >>> before is a Sofia profile parameter that uses the Call-ID header value from >>> the INVITE sent by the caller as the UUID for the channel. That could be >>> useful for example in matching up channels against SIP traces as you only >>> need to store 1 ID not 2. I frequently see ones for example of >>> $UUID@$SERVER.COM - far longer than 128bits, and using >>> characters other than 0-9A-F so that it can't be folded into bits. >>> >>> In this case a CHAR column becomes most flexible and the BIGINT column gives >>> you the speed advantage of the BINARY. >>> >>> -Steve >>> >>> >>> >>> On 17 February 2013 22:39, Seven Du wrote: >>>> >>>> you can use any arbitrary string as uuid, say >>>> >>>> originate {origination_uuid=my-uuid-that-is-not-128-bit}user/1000 ?. >>>> >>>> >>>> >>>> On Monday, February 18, 2013 at 6:13 AM, Cal Leeming [Simplicity Media Ltd] >>>> wrote: >>>> >>>>> >>>>> Thanks for the feedback Steve, >>>>> >>>>> In ref to auto inc, actually, I have! :) In the code on the blog, it shows >>>>> that the tables uses a BIGINT auto increment for each row, and the >>>>> performance was still good. However, using this approach introduces an >>>>> unnecessary JOIN and uses more index and table space - unless I've missed >>>>> something? >>>>> >>>>> Could you elaborate more on the UUIDs that may not be 128-bit? My >>>>> google-fu failed me :/ >>>>> >>>>> Cal >>>>> >>>>> On Sun, Feb 17, 2013 at 9:11 PM, Steven Ayre wrote: >>>>>> Another possibility you haven't looked at is having a table of UUIDs with >>>>>> a BIGINT auto_increment PK and using that as keys in other takes with a >>>>>> foreign key relationship (whether defined or implied). That would give >>>>>> you a smaller integer (hence faster select/insert/join/etc) that's still >>>>>> guaranteed to be unique. Though the auto_increment INSERT could then >>>>>> become a bottleneck (hint run a recent 5.5/5.6 MySQL version at the very >>>>>> least since that bottleneck while still present was very much reduced). >>>>>> >>>>>> -Steve >>>>>> >>>>>> >>>>>> >>>>>> On 17 February 2013 20:55, Cal Leeming [Simplicity Media Ltd] >>>>>> wrote: >>>>>>> Hi all, >>>>>>> >>>>>>> Just spent a little while comparing the performance of heavy >>>>>>> INSERT/SELECT performance against UUID fields in a MySQL database. >>>>>>> >>>>>>> Full performance breakdown and write up can be found here; >>>>>>> http://blog.simplicitymedialtd.co.uk/?p=437 >>>>>>> >>>>>>> So far, the best option seems to be storing the UUID as a BINARY(16) - >>>>>>> anyone have any thoughts on this? >>>>>>> >>>>>>> Thanks >>>>>>> >>>>>>> Cal >>>>>>> >>>>>>> ________________________________________________________________________>>>>>>> _ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://wiki.freeswitch.org >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> 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 >>>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> 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 >>>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> 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 >>>>> >>>> >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130217/4d39a3d1/attachment-0001.html From jaybinks at gmail.com Mon Feb 18 05:08:48 2013 From: jaybinks at gmail.com (jay binks) Date: Mon, 18 Feb 2013 12:08:48 +1000 Subject: [Freeswitch-users] Storing CDR UUIDs in MySQL In-Reply-To: References: <5ADA3B5B-47A9-4C6F-BC9A-12A575840006@gmail.com> Message-ID: your very right Ken ... . however Mysql has partitions with InnoDB backend. I do the exact same thing and it helps a heap ! On 18 February 2013 11:45, Ken Rice wrote: > Why don?t you do something like look at something other than mysql like > PostgreSQL, we insert millions of CDRs per day there and don?t have > performance issues due to Table Partitioning, this allows us to cram well > over 100million new rows/month into a table, partitioned daily along > something like a call_date field (derived by dropping the time from a full > time/date stamp). This also allows us to do really speedy reports without > having to deal with what happens if you do something like select count(*) > from cdr where call_date = ?2013-02-15?; this would be a monster of a query > on a table with say 90 days of CDRs at 4million CDRs/day (360million CDRs > total) but by using the table partitions with a date constraint, the query > planer eliminates partitions that do not match the date constraint the > speeds along the query. > > The caveat to this is if you don?t give it some sort of date constraint you > end up looking across many partitions for the data... > > > On 2/17/13 6:48 PM, "Steven Ayre" wrote: > > You could take a SHA1, but then might get collisions. That can happen with > UUIDs too, but since it includes the MAC address and time stamp is less > likely and would be more predictable. > > The originate_uuid may get collisions depending on what you pick, and the > callid puts it outside your control entirely - in those situations you might > need to handle collisions anyway. > > Sent from my iPad > > > > On 17 Feb 2013, at 23:47, "Cal Leeming [Simplicity Media Ltd]" > wrote: > > Ah, yeah that changes the playing field for sure. > > The problem with CHAR is the INSERT speed though, and I've seen similar > issues on other builds too.. after you push past a few million rows, it > really starts to slow down.. If the database was in memory, it might not be > such an issue though. > > In some ways it 'feels cleaner' using a separate table and then JOIN on a > BIGINT autoinc column, as the thought of JOIN'ing and GROUP BY on a BINARY > seems wrong. > > Another option would be to take an SHA1 hash of the UUID, then convert that > from hex to bin, and put that in the BINARY field.. this way you can > guarantee each UUID will be the same size, not worry about changes in data > schema, and get the INSERT speed benefits of the BINARY too - thoughts? > > Cal > > On Sun, Feb 17, 2013 at 11:15 PM, Steven Ayre wrote: > > +1 > > Also as an example the 'inbound-use-callid-as-uuid' parameter I mentioned > before is a Sofia profile parameter that uses the Call-ID header value from > the INVITE sent by the caller as the UUID for the channel. That could be > useful for example in matching up channels against SIP traces as you only > need to store 1 ID not 2. I frequently see ones for example of > $UUID@$SERVER.COM - far longer than 128bits, and using > characters other than 0-9A-F so that it can't be folded into bits. > > > In this case a CHAR column becomes most flexible and the BIGINT column gives > you the speed advantage of the BINARY. > > -Steve > > > > On 17 February 2013 22:39, Seven Du wrote: > > > you can use any arbitrary string as uuid, say > > originate {origination_uuid=my-uuid-that-is-not-128-bit}user/1000 ?. > > > > On Monday, February 18, 2013 at 6:13 AM, Cal Leeming [Simplicity Media Ltd] > wrote: > > > Thanks for the feedback Steve, > > In ref to auto inc, actually, I have! :) In the code on the blog, it shows > that the tables uses a BIGINT auto increment for each row, and the > performance was still good. However, using this approach introduces an > unnecessary JOIN and uses more index and table space - unless I've missed > something? > > Could you elaborate more on the UUIDs that may not be 128-bit? My google-fu > failed me :/ > > Cal > > On Sun, Feb 17, 2013 at 9:11 PM, Steven Ayre wrote: > > Another possibility you haven't looked at is having a table of UUIDs with a > BIGINT auto_increment PK and using that as keys in other takes with a > foreign key relationship (whether defined or implied). That would give you a > smaller integer (hence faster select/insert/join/etc) that's still > guaranteed to be unique. Though the auto_increment INSERT could then become > a bottleneck (hint run a recent 5.5/5.6 MySQL version at the very least > since that bottleneck while still present was very much reduced). > > -Steve > > > > On 17 February 2013 20:55, Cal Leeming [Simplicity Media Ltd] > wrote: > > Hi all, > > Just spent a little while comparing the performance of heavy INSERT/SELECT > performance against UUID fields in a MySQL database. > > Full performance breakdown and write up can be found here; > http://blog.simplicitymedialtd.co.uk/?p=437 > > So far, the best option seems to be storing the UUID as a BINARY(16) - > anyone have any thoughts on this? > > Thanks > > Cal > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > ________________________________ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > -- > Ken > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > irc.freenode.net #freeswitch > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -- Sincerely Jay From krice at freeswitch.org Mon Feb 18 05:13:58 2013 From: krice at freeswitch.org (Ken Rice) Date: Sun, 17 Feb 2013 20:13:58 -0600 Subject: [Freeswitch-users] Storing CDR UUIDs in MySQL In-Reply-To: Message-ID: Cool Jay! Last time I looked at mysql that wasn't available... (and yes it was quite a long time ago) On 2/17/13 8:08 PM, "jay binks" wrote: > your very right Ken ... . > however Mysql has partitions with InnoDB backend. > > I do the exact same thing and it helps a heap ! > > On 18 February 2013 11:45, Ken Rice wrote: >> Why don?t you do something like look at something other than mysql like >> PostgreSQL, we insert millions of CDRs per day there and don?t have >> performance issues due to Table Partitioning, this allows us to cram well >> over 100million new rows/month into a table, partitioned daily along >> something like a call_date field (derived by dropping the time from a full >> time/date stamp). This also allows us to do really speedy reports without >> having to deal with what happens if you do something like select count(*) >> from cdr where call_date = ?2013-02-15?; this would be a monster of a query >> on a table with say 90 days of CDRs at 4million CDRs/day (360million CDRs >> total) but by using the table partitions with a date constraint, the query >> planer eliminates partitions that do not match the date constraint the >> speeds along the query. >> >> The caveat to this is if you don?t give it some sort of date constraint you >> end up looking across many partitions for the data... >> >> >> On 2/17/13 6:48 PM, "Steven Ayre" wrote: >> >> You could take a SHA1, but then might get collisions. That can happen with >> UUIDs too, but since it includes the MAC address and time stamp is less >> likely and would be more predictable. >> >> The originate_uuid may get collisions depending on what you pick, and the >> callid puts it outside your control entirely - in those situations you might >> need to handle collisions anyway. >> >> Sent from my iPad >> >> >> >> On 17 Feb 2013, at 23:47, "Cal Leeming [Simplicity Media Ltd]" >> wrote: >> >> Ah, yeah that changes the playing field for sure. >> >> The problem with CHAR is the INSERT speed though, and I've seen similar >> issues on other builds too.. after you push past a few million rows, it >> really starts to slow down.. If the database was in memory, it might not be >> such an issue though. >> >> In some ways it 'feels cleaner' using a separate table and then JOIN on a >> BIGINT autoinc column, as the thought of JOIN'ing and GROUP BY on a BINARY >> seems wrong. >> >> Another option would be to take an SHA1 hash of the UUID, then convert that >> from hex to bin, and put that in the BINARY field.. this way you can >> guarantee each UUID will be the same size, not worry about changes in data >> schema, and get the INSERT speed benefits of the BINARY too - thoughts? >> >> Cal >> >> On Sun, Feb 17, 2013 at 11:15 PM, Steven Ayre wrote: >> >> +1 >> >> Also as an example the 'inbound-use-callid-as-uuid' parameter I mentioned >> before is a Sofia profile parameter that uses the Call-ID header value from >> the INVITE sent by the caller as the UUID for the channel. That could be >> useful for example in matching up channels against SIP traces as you only >> need to store 1 ID not 2. I frequently see ones for example of >> $UUID@$SERVER.COM - far longer than 128bits, and using >> characters other than 0-9A-F so that it can't be folded into bits. >> >> >> In this case a CHAR column becomes most flexible and the BIGINT column gives >> you the speed advantage of the BINARY. >> >> -Steve >> >> >> >> On 17 February 2013 22:39, Seven Du wrote: >> >> >> you can use any arbitrary string as uuid, say >> >> originate {origination_uuid=my-uuid-that-is-not-128-bit}user/1000 ?. >> >> >> >> On Monday, February 18, 2013 at 6:13 AM, Cal Leeming [Simplicity Media Ltd] >> wrote: >> >> >> Thanks for the feedback Steve, >> >> In ref to auto inc, actually, I have! :) In the code on the blog, it shows >> that the tables uses a BIGINT auto increment for each row, and the >> performance was still good. However, using this approach introduces an >> unnecessary JOIN and uses more index and table space - unless I've missed >> something? >> >> Could you elaborate more on the UUIDs that may not be 128-bit? My google-fu >> failed me :/ >> >> Cal >> >> On Sun, Feb 17, 2013 at 9:11 PM, Steven Ayre wrote: >> >> Another possibility you haven't looked at is having a table of UUIDs with a >> BIGINT auto_increment PK and using that as keys in other takes with a >> foreign key relationship (whether defined or implied). That would give you a >> smaller integer (hence faster select/insert/join/etc) that's still >> guaranteed to be unique. Though the auto_increment INSERT could then become >> a bottleneck (hint run a recent 5.5/5.6 MySQL version at the very least >> since that bottleneck while still present was very much reduced). >> >> -Steve >> >> >> >> On 17 February 2013 20:55, Cal Leeming [Simplicity Media Ltd] >> wrote: >> >> Hi all, >> >> Just spent a little while comparing the performance of heavy INSERT/SELECT >> performance against UUID fields in a MySQL database. >> >> Full performance breakdown and write up can be found here; >> http://blog.simplicitymedialtd.co.uk/?p=437 >> >> So far, the best option seems to be storing the UUID as a BINARY(16) - >> anyone have any thoughts on this? >> >> Thanks >> >> Cal >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> >> >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> >> ________________________________ >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> >> -- >> Ken >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> irc.freenode.net #freeswitch >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> > > > > -- > Sincerely > > Jay > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch From sparklezou at 163.com Mon Feb 18 06:00:24 2013 From: sparklezou at 163.com (sparklezou) Date: Mon, 18 Feb 2013 11:00:24 +0800 Subject: [Freeswitch-users] Please suggest one "SIP Provider" In-Reply-To: References: Message-ID: <3de797a2.23ab.13ceb3ec564.Coremail.sparklezou@163.com> Hi Cal, Thanks very much! 2013-02-18 sparklezou ????Cal Leeming [Simplicity Media Ltd] ?????2013-02-17 22:47 ???Re: [Freeswitch-users] Please suggest one "SIP Provider" ????"FreeSWITCH Users Help" ??? Hello, I assume your referring to SIP trunk providers. Here is a recent thread for inbound DID providers; http://freeswitch-users.2379917.n2.nabble.com/DID-providers-any-thoughts-jan-2013-td7586851.html As for outbound traffic, it entirely depends on; * Which countries you wish to target * Where your own server is located * What service level you are trying to achieve * How much you have to spend * Are you looking or ingress (inbound) or egress (outbound) From the perspective of the UK, here is a brief breakdown of the categories (from best to worst); * Direct with a state/national telecoms company (such as BT or Kingston) - usually come with minimum commitment levels and very high set up/monthly costs, suitable only for large existing customer bases * Through a wholesaler that uses only the above routes (such as Simwood) - they give lower traffic customers the ability to use these high quality routes without the cost.. yes they are reselling, but it's one of the rare occasions where reselling it is justified, as you will save a lot of money in several circumstances * Through a reseller of a reseller of the above (always ask the company what routes they use, and try to cut out the useless middleman) * Through a reseller of a reseller of a reseller of a reseller of a reseller that is using grey routes (seriously, this really does happen a lot) It is also worth noting that providers are known to lie about the quality and sources of their routes, so rely heavily on community reviews to figure out if a company is good. And, even if a company is reselling wholesaler routes, you must make sure you perform inter-op right from the beginning, else you might hit some fun surprises along the way (we have been stung by this before). Remember, crappy cheap SIP trunk providers are 10 a penny these days, and it really depends on what you are trying to achieve. If you want to build say an automated phone alerting system that calls people with an automated electronic voice (such as PagerDuty), then using crappy routes might be acceptable for your business case. However, if you are trying to build an enterprise product for voice, then you need to make careful choices about what sacrifices (if any) you're going to make on quality vs cost. On a last note, we have trialed several "reputable" providers over the last 6 months, and most of them ended in tears - be weary. I hope this helps. Cal On Sun, Feb 17, 2013 at 12:38 PM, sparklezou wrote: Hi All, Could you please share your experience of the "SIP provider"? Which one is good at the quality and price? Thanks! 2013-02-17 sparklezou _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130218/f91bdc62/attachment-0001.html From jmesquita at freeswitch.org Mon Feb 18 06:59:18 2013 From: jmesquita at freeswitch.org (=?utf-8?Q?Jo=C3=A3o_Mesquita?=) Date: Mon, 18 Feb 2013 00:59:18 -0300 Subject: [Freeswitch-users] FreeSWITCH Message Bus / Shared Key Value Store In-Reply-To: References: <20130212204251.7066d944@mail.tritonwest.net> Message-ID: <72946562-6211-4B4A-BF4D-E433470855A4@freeswitch.org> Tony, ZMQ is fixing the Pub/Sub part filtering on version 3 which I think is out now. Sent from my iPhone On Feb 17, 2013, at 2:51 PM, Anthony Minessale wrote: > It has a strict philosophy on not using locks or mutexes and only doing com over socket. When you create an instance of it, it uses its own threads internally and that model gets messed up severely if you call fork from the process (even when you are doing it for something else like system command). Its basically an engine waiting for a host program rather than something you can integrated into a more complex application. Also the multicast has issues, there is no way to specificy host local scope for hops. Finally, their philosophy is to ignore the dilemma of sub/pub. There are no subscriptions to events, there is only send all events to everyone and let them filter locally. Its easy to believe this is adequate when doing pre-determined test scenarios but we know in reality that in a busy FS box we, cant send *all* the events to *everyone* and let them filter locally. > > > > On Sun, Feb 17, 2013 at 11:32 AM, Avi Marcus wrote: >> Please excuse my ignorance, but can you explain again why zmq won't work? >> >> Regarding boasting about messaging capability.. it was made to be quite lightweight - supposedly it can easily saturate a 1gigabit link with message size over 150 bytes each: http://www.zeromq.org/results:10gbe-tests >> >> If you want to use it for a queue, then it's lack of persistence and assurance that messages get through might not be helpful, but for a broadcast to whoever wants to listen it sounds great for heartbeats, events, etc. >> >> -Avi Marcus >> >> 1-718-989-9485 (USA) >> 1-866-202-5850 (USA & Canada Toll Free) >> 02-372-1570 (Israel) >> 020-3298-2875 (UK) >> >> >> On Sun, Feb 17, 2013 at 5:50 PM, Eliot Gable wrote: >>> ZMQ also does not work with a fork, which is needed in order to execute any system commands (like iptables or anything else which has no programming API). That pretty much eliminates ZMQ as a possibility. >>> >>> I did some research this weekend, and of all the possibilities I could find, the one that caught my attention the most was Spread: http://www.spread.org/ >>> >>> There are some drawbacks to it, namely: >>> >>> 1) It requires any marketing material mentioning FreeSWITCH or any project / solution utilizing FreeSWITCH to also include a prepared statement about the use of the Spread toolkit. This is a fairly major licensing issue, as all commercial solutions utilizing FreeSWITCH as a core component would then also need to mention the use of the Spread toolkit within FreeSWITCH. It is essentially a "viral clause," which would be very tedious for people to make sure they honor correctly. >>> >>> 2) The toolkit uses a stand-alone daemon which would then need to be monitored separately from FreeSWITCH and would be another point of failure, adding complexity to the system. Basically, if that daemon were to crash, FreeSWITCH would need to know about it and would need to either respawn it or shut down. This would bring the need for Pacemaker or something similar back into the picture for any viable HA solution. Alternatively, we could write some code similar to daemontools into FreeSWTICH which respawns the daemon if it dies, but we would have to test the impact of this respawning on the overall cluster to determine if it impacts visibility of the node at any point in time. Another alternative would be to wrap the daemon into a thread inside FreeSWITCH such that if the daemon caused a segfault or something, it would force FreeSWITCH to terminate, as well. I have not done a code review of the daemon yet to determine if this is viable alternative, but assuming they have not coded it on LSD or something, it is more than likely possible. >>> >>> 3) They boast about how it can handle up to 8,000 1KB messages per second. I don't consider that boast-worthy. When I worked at Broadvox a few years ago, I had a FS pair which ran around 380 calls per second (760 sessions per second). Each call generates dozens of events. That hardware was getting dated when I left Broadvox, and today's hardware along with the performance improvements done to FS since then means we could conceivably have a single node which runs over 1k calls per second firing dozens of events per call. That means a single box could completely consume the message bandwidth of the entire Spread network. Imagine trying to have 64 such boxes running. We are really in need of a solution which boasts hundreds of thousands of messages per second. Spread seems like it might be off by an order of magnitude and then some. >>> >>> Despite these issues, Spread still seems to come closer to our needs than any other solution I found. FYI, I also looked at the Corosync IPC system, and was not at all impressed. On paper, Spread exceeds Corosync's capabilities by a fair margin. >>> >>> There are some strategies for mitigating issue #3 with Spread, as well. For example, we could limit messages across the Spread network to things like heartbeats and / or other HA and synchronization related messages. Basically, think of it like a D-channel on a PRI. For sending high packet per second streams of messages, we could do standard unicast connections or even try straight up mutlicasting to all nodes on the LAN. Sending heartbeats every 10ms across the Spread network would put a 64-node cluster at 6,400 messages per second just with heartbeats. That would still leave a decent amount of message bandwidth available for other types of negotiation messages and should still allow for sub-second fail-over detection and reaction. Of course, this is all assuming we can actually get 8,000 1KB messages per second out of a 64-node cluster. >>> >>> There are likely lots of things that impact how many messages per second Spread can handle. A lot of it has to do with network latency and CPU power. Spread uses acknowledgements and message reordering to ensure delivery in a way that accounts for things like node membership changes during the time the message is in transit and whether the message has been received by all nodes in the cluster. Network latency is probably one of the biggest factors in how many messages per second it can handle. On a faster network link, the messages per second would be higher and on a slower network, it would be lower. Obviously, CPU processing time and scheduling is important, as well. If one system is extremely overloaded and the Spread daemon is being starved for CPU resources, that will add extra latency in processing and also reduce message throughput. Obviously, this could also impact whether the node is seen as visible. So, this is one more argument for why we would need to try to run the daemon as a thread under FreeSWITCH. Then it has the same scheduling priority as FreeSWITCH and cannot be starved for resources by FreeSWITCH itself. It also would exhibit the same amount of resource starvation FreeSWITCH experiences on the node, so would more accurately reflect the state of FreeSWITCH on the node. >>> >>> If anyone has any other suggestions than Spread, I would like to hear it. Also, some feedback on item #1 would be great, as I cannot really judge for everyone else how willing they are to accept such a licensing clause. >>> >>> >>> >>> On Tue, Feb 12, 2013 at 8:48 PM, Jo?o Mesquita wrote: >>>> I have used ZeroMQ in the past for this sorts of things but it really won't be able to detect failures really fast. It is not made for this. Maybe we can gather the requirements for such message bus? Zmq for example provides you with this cool interface to build messaging protocols on top of it but it does not provide reliability when it comes to endpoint to endpoint connection without a heartbeat implemented on the user end. Can this be used for FS as well? Anyhow, just throwing some ideas... >>>> >>>> Jo?o Mesquita >>>> FreeSWITCH? Solutions >>>> >>>> >>>> On Tue, Feb 12, 2013 at 5:42 PM, Dave R. Kompel wrote: >>>>> I've done a few experments with using both Redis, and the evil "Microsoft Azure Service bus" (the server on prem based version) to extend the eventing system to have global PUB/SUB. This way things like registrations, and Limit stuff could be made global. >>>>> >>>>> I'm looking for a way, in my carrier switch implmentation, to implment both HA Failover and Scaleout clustering. >>>>> >>>>> --Dave >>>>> >>>>> From: Eliot Gable [mailto:egable+freeswitch at gmail.com] >>>>> To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] >>>>> Sent: Tue, 12 Feb 2013 05:49:13 -0800 >>>>> Subject: [Freeswitch-users] FreeSWITCH Message Bus / Shared Key Value Store >>>>> >>>>> >>>>> Tony and Mike and I had a discussion last night about FreeSWITCH with regards to implementing some form of core message bus or shared key-value store. We discussed a few different options, but did not really settle on anything. If you are writing modules or using FreeSWITCH in a multi-node setting, please share what features / functionality you would like to see implemented in this regard, how you would use it, and why you want to see the specific mechanism of your choice rather than some alternative. Also, please consider and mention whether "cluster awareness" is something that factors into your use case. By this, I mean having each FS node have some idea about the state / status of each other node in terms of taking calls vs acting as a standby or slave node, etc. >>>>> >>>>> -- >>>>> Eliot Gable >>>>> >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> 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 >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>>> >>> >>> >>> >>> -- >>> Eliot Gable >>> >>> "We do not inherit the Earth from our ancestors: we borrow it from our children." ~David Brower >>> >>> "I decided the words were too conservative for me. We're not borrowing from our children, we're stealing from them--and it's not even considered to be a crime." ~David Brower >>> >>> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to live; not live to eat.) ~Marcus Tullius Cicero >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130218/e6c97fcd/attachment-0001.html From joakim+freeswitch-users at gissberg.nu Sun Feb 17 11:57:22 2013 From: joakim+freeswitch-users at gissberg.nu (J G) Date: Sun, 17 Feb 2013 18:57:22 +1000 Subject: [Freeswitch-users] (no subject) Message-ID: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130217/a1fa47e8/attachment.html From msc at freeswitch.org Mon Feb 18 07:44:34 2013 From: msc at freeswitch.org (Michael Collins) Date: Sun, 17 Feb 2013 20:44:34 -0800 Subject: [Freeswitch-users] mod_directory - Cannot locate domain In-Reply-To: References: Message-ID: You forgot to close the
tag... -MC On Sat, Feb 16, 2013 at 6:58 PM, Lloyd Aloysius wrote: > Hi All > > I have a strange problem with the mod_directory. I use xml_curl for all > the applications. Only mod_directory giving this trouble. > > > EXECUTE sofia/sipinterface_1/marv at mydomain.com directory(mydomain.com > mydomain.com) > 2013-02-16 21:43:04.601315 [DEBUG] mod_directory.c:518 [mydomain.com] > rwlock > > 2013-02-16 21:43:04.621318 [ERR] switch_xml.c:1650 Error[[error near line > 1]: unclosed tag ] > 2013-02-16 21:43:04.621318 [WARNING] mod_directory.c:545 Cannot locate > domain mydomain.com > > -- > > my direcotroy.conf xml curl output > > >
> > > > > > > > > > > > > > > > > > > > > > > Any help is appreciated. > > Thanks > Lloyd > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130217/d985c3fb/attachment.html From msc at freeswitch.org Mon Feb 18 07:51:40 2013 From: msc at freeswitch.org (Michael Collins) Date: Sun, 17 Feb 2013 20:51:40 -0800 Subject: [Freeswitch-users] Please suggest one "SIP Provider" In-Reply-To: <3de797a2.23ab.13ceb3ec564.Coremail.sparklezou@163.com> References: <3de797a2.23ab.13ceb3ec564.Coremail.sparklezou@163.com> Message-ID: Sparklezou, If you need a reputable SIP provider in the US then I can highly recommend Flowroute. They have great service and they support both FreeSWITCH and ClueCon. :) -MC On Sun, Feb 17, 2013 at 7:00 PM, sparklezou wrote: > ** > Hi Cal, > > Thanks very much! > > 2013-02-18 > ------------------------------ > sparklezou > ------------------------------ > *????*Cal Leeming [Simplicity Media Ltd] > *?????*2013-02-17 22:47 > *???*Re: [Freeswitch-users] Please suggest one "SIP Provider" > *????*"FreeSWITCH Users Help" > *???* > > Hello, > > I assume your referring to SIP trunk providers. > > Here is a recent thread for inbound DID providers; > > http://freeswitch-users.2379917.n2.nabble.com/DID-providers-any-thoughts-jan-2013-td7586851.html > > As for outbound traffic, it entirely depends on; > > * Which countries you wish to target > * Where your own server is located > * What service level you are trying to achieve > * How much you have to spend > * Are you looking or ingress (inbound) or egress (outbound) > > From the perspective of the UK, here is a brief breakdown of the > categories (from best to worst); > > * Direct with a state/national telecoms company (such as BT or Kingston) - > usually come with minimum commitment levels and very high set up/monthly > costs, suitable only for large existing customer bases > * Through a wholesaler that uses only the above routes (such as Simwood) - > they give lower traffic customers the ability to use these high quality > routes without the cost.. yes they are reselling, but it's one of the rare > occasions where reselling it is justified, as you will save a lot of money > in several circumstances > * Through a reseller of a reseller of the above (always ask the company > what routes they use, and try to cut out the useless middleman) > * Through a reseller of a reseller of a reseller of a reseller of a > reseller that is using grey routes (seriously, this really does happen a > lot) > > It is also worth noting that providers are known to lie about the quality > and sources of their routes, so rely heavily on community reviews to figure > out if a company is good. And, even if a company is reselling wholesaler > routes, you must make sure you perform inter-op right from the beginning, > else you might hit some fun surprises along the way (we have been stung by > this before). > > Remember, crappy cheap SIP trunk providers are 10 a penny these days, and > it really depends on what you are trying to achieve. If you want to build > say an automated phone alerting system that calls people with an automated > electronic voice (such as PagerDuty), then using crappy routes might be > acceptable for your business case. However, if you are trying to build an > enterprise product for voice, then you need to make careful choices about > what sacrifices (if any) you're going to make on quality vs cost. > > On a last note, we have trialed several "reputable" providers over the > last 6 months, and most of them ended in tears - be weary. > > I hope this helps. > > Cal > > On Sun, Feb 17, 2013 at 12:38 PM, sparklezou wrote: > >> ** >> ** >> Hi All, >> >> Could you please share your experience of the "SIP provider"? >> >> Which one is good at the quality and price? >> >> Thanks! >> >> 2013-02-17 >> ------------------------------ >> sparklezou >> ** >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130217/aaeb6e1a/attachment-0001.html From msc at freeswitch.org Mon Feb 18 07:54:13 2013 From: msc at freeswitch.org (Michael Collins) Date: Sun, 17 Feb 2013 20:54:13 -0800 Subject: [Freeswitch-users] Auto attendant :: act on no digit entry In-Reply-To: References: Message-ID: I would ask on the #2600hz IRC channel or their mailing list. You're much more likely to find someone with the specific knowledge to answer this question. There are some 2600hz folks on this list but most of us are not familiar with blue.box so it would be to your advantage to try in the 2600hz community resources. -MC On Sun, Feb 17, 2013 at 7:38 AM, Sayyed Mohammad Emami Razavi < emamirazavi at gmail.com> wrote: > Freeswitch has not problem! It does this perfectly! but when you go to > "number manager" module of Blue.box and > edit your number terminates on NO ANSWER and then select External Xfer and > select your number! your settings will be saved in DB but > not in FS XML! I think it's a good idea if i edit BB code! What is your > opinion? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130217/b470c44c/attachment.html From krice at freeswitch.org Mon Feb 18 07:59:05 2013 From: krice at freeswitch.org (Ken Rice) Date: Sun, 17 Feb 2013 22:59:05 -0600 Subject: [Freeswitch-users] Its that Time of the year again... Mr Brian West is one year older.... Message-ID: Hey Guys, Every one send shout outs to BKW aka Brian West... Lets all send him a pint via Paypal by sending him a Fiver... And don?t forget to tag is personal gift so paypal doesn?t screw him over on the percentage! I?ve sent mine already! -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130217/f3ba47b7/attachment.html From msc at freeswitch.org Mon Feb 18 08:01:20 2013 From: msc at freeswitch.org (Michael Collins) Date: Sun, 17 Feb 2013 21:01:20 -0800 Subject: [Freeswitch-users] FreeSWITCH Message Bus / Shared Key Value Store In-Reply-To: <72946562-6211-4B4A-BF4D-E433470855A4@freeswitch.org> References: <20130212204251.7066d944@mail.tritonwest.net> <72946562-6211-4B4A-BF4D-E433470855A4@freeswitch.org> Message-ID: On Sun, Feb 17, 2013 at 7:59 PM, Jo?o Mesquita wrote: > Tony, ZMQ is fixing the Pub/Sub part filtering on version 3 which I think > is out now. > For those who want to read up on version 3.x of zmq: http://www.zeromq.org/topics:planning#toc1 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130217/2870f314/attachment.html From sparklezou at 163.com Mon Feb 18 08:11:22 2013 From: sparklezou at 163.com (sparklezou) Date: Mon, 18 Feb 2013 13:11:22 +0800 Subject: [Freeswitch-users] Please suggest one "SIP Provider" In-Reply-To: References: Message-ID: <76a5e417.33a9.13cebb6ace0.Coremail.sparklezou@163.com> Hi MC, Thanks! I will regist and try it. I also found the configuration guid http://wiki.freeswitch.org/wiki/Provider_Configuration:_FlowRoute one more question, how many call could be set up at the same time by one account? Thanks! 2013-02-18 sparklezou ????Michael Collins ?????2013-02-18 12:51 ???Re: [Freeswitch-users] Please suggest one "SIP Provider" ????"FreeSWITCH Users Help" ??? Sparklezou, If you need a reputable SIP provider in the US then I can highly recommend Flowroute. They have great service and they support both FreeSWITCH and ClueCon. :) -MC On Sun, Feb 17, 2013 at 7:00 PM, sparklezou wrote: Hi Cal, Thanks very much! 2013-02-18 sparklezou ????Cal Leeming [Simplicity Media Ltd] ?????2013-02-17 22:47 ???Re: [Freeswitch-users] Please suggest one "SIP Provider" ????"FreeSWITCH Users Help" ??? Hello, I assume your referring to SIP trunk providers. Here is a recent thread for inbound DID providers; http://freeswitch-users.2379917.n2.nabble.com/DID-providers-any-thoughts-jan-2013-td7586851.html As for outbound traffic, it entirely depends on; * Which countries you wish to target * Where your own server is located * What service level you are trying to achieve * How much you have to spend * Are you looking or ingress (inbound) or egress (outbound) From the perspective of the UK, here is a brief breakdown of the categories (from best to worst); * Direct with a state/national telecoms company (such as BT or Kingston) - usually come with minimum commitment levels and very high set up/monthly costs, suitable only for large existing customer bases * Through a wholesaler that uses only the above routes (such as Simwood) - they give lower traffic customers the ability to use these high quality routes without the cost.. yes they are reselling, but it's one of the rare occasions where reselling it is justified, as you will save a lot of money in several circumstances * Through a reseller of a reseller of the above (always ask the company what routes they use, and try to cut out the useless middleman) * Through a reseller of a reseller of a reseller of a reseller of a reseller that is using grey routes (seriously, this really does happen a lot) It is also worth noting that providers are known to lie about the quality and sources of their routes, so rely heavily on community reviews to figure out if a company is good. And, even if a company is reselling wholesaler routes, you must make sure you perform inter-op right from the beginning, else you might hit some fun surprises along the way (we have been stung by this before). Remember, crappy cheap SIP trunk providers are 10 a penny these days, and it really depends on what you are trying to achieve. If you want to build say an automated phone alerting system that calls people with an automated electronic voice (such as PagerDuty), then using crappy routes might be acceptable for your business case. However, if you are trying to build an enterprise product for voice, then you need to make careful choices about what sacrifices (if any) you're going to make on quality vs cost. On a last note, we have trialed several "reputable" providers over the last 6 months, and most of them ended in tears - be weary. I hope this helps. Cal On Sun, Feb 17, 2013 at 12:38 PM, sparklezou wrote: Hi All, Could you please share your experience of the "SIP provider"? Which one is good at the quality and price? Thanks! 2013-02-17 sparklezou _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130218/7ace32c4/attachment-0001.html From msc at freeswitch.org Mon Feb 18 08:11:03 2013 From: msc at freeswitch.org (Michael Collins) Date: Sun, 17 Feb 2013 21:11:03 -0800 Subject: [Freeswitch-users] Time to start understanding the FreeSWITCH codebase In-Reply-To: <17281507-28F9-4AB5-9799-A930C21F0862@gmail.com> References: <9982BBC9-3C49-4646-A578-51E1A1C2781C@endigotech.com> <17281507-28F9-4AB5-9799-A930C21F0862@gmail.com> Message-ID: >From personal experience I can say that it makes sense to start at the outer levels and work your way in. A gentle way to get into what the source code does is to look at the source code for API commands or dialplan applications with which you are very familiar or that have very simple functions. I would start with mod_commands.c and look at simple commands like "version" and "banner". Once you get the hang of those then try digging a little deeper, like with the "show" API. That API is a great example of how the FS devs handle argument parsing and use the occasional "really large" case block. If you would rather start with the dialplan stuff then check out mod_dptools.c. Look at really simple apps like "log". You'll see lots of references to the "session" which you MUST learn about if you want to do anything with a live call. One thing you'll see is that the functions in the "outer" layers will make calls to "core" functions. As homework I recommend you look in mod_dptools.c for the definition of the "playback" app. Hint: to get the whole picture you'll need to also look in switch_ivr_play_say.c. ;) Have fun and don't be afraid to ask questions here or in the IRC channel. -MC On Sun, Feb 17, 2013 at 4:42 PM, Steven Ayre wrote: > Use the source, Luke. > > Digging around in the source is one way to explore how FS works. Some is > easier to understand than others. Threading is heavily used so > understanding mutexes etc is useful. > > I'd highly recommend that you read this site though: > http://docs.freeswitch.org/ > > That documents the core API. A lot is based on APR, plus some extra stuff > like the module interfaces. > > Steve > > > > > On 18 Feb 2013, at 00:00, Brian Foster wrote: > > I think I've come to a stage in my life where I'm more interested than > ever in learning the internals of FreeSWITCH and start contributing code to > it. I've been following the project for a few years, but I've never touched > the code. It's time to change that. > > Is there a place on the wiki someone can point me to so I can start > understanding the FS core? I'm talking fairly basic, beginner level stuff. > > Sent from my iPhone > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130217/71ce701f/attachment.html From tomasz.szuster at gmail.com Mon Feb 18 08:18:05 2013 From: tomasz.szuster at gmail.com (Tomasz Szuster) Date: Mon, 18 Feb 2013 06:18:05 +0100 Subject: [Freeswitch-users] Multiple registration issue In-Reply-To: References: Message-ID: Thank you Stephen. Using your information I was able to better understand freeswitch. I've successfully setup limitation using http://wiki.freeswitch.org/wiki/Limit#max Case closed :) Tom On 17 Feb 2013 22:37, "Steven Ayre" wrote: > Registration and call authentication are unrelated, although usually > configured together. > > Registration is solely for you to register the IP:port where your user is > available to *receive *calls. > > You can call into FreeSWITCH without being registered. Calls (INVITE) are > authenticated separately - you'll see INVITE, 401 Unauthorised, 2nd INVITE. > > -Steve > > > > > On 17 February 2013 20:56, Tomasz Szuster wrote: > >> Thx for explanation. >> >> But how this can be possible that I can call at the same time from 3 >> different softphones registered into one extension, even if I've set >> max-registrations-per-extension and multiple-gistrations ? >> >> Tom >> >> >> On Sun, Feb 17, 2013 at 9:36 PM, Steven Ayre wrote: >> >>> Registrations don't get logged out, they get renewed/expire or replaced. >>> >>> Your 3rd machine will replace the first registration. >>> >>> -Steve >>> >>> >>> >>> On 17 February 2013 20:18, Tomasz Szuster wrote: >>> >>>> Hi, >>>> >>>> >>>> I'm trying to use limit logins to one extension using >>>> >>>> *max-registrations-per-extension = 2* >>>> >>>> >>>> I've set this in *sofia profile internal.* >>>> * >>>> * >>>> From what I see this is not working as expected. >>>> >>>> I've logged in one extensions from 3 different machines at the same >>>> time, and successfully established connections. >>>> To test I used two computers with X-lite, and Sipdroit on my android >>>> phone. >>>> >>>> Additional I've used: >>>> >>>> *multiple-registrations = false* >>>> * >>>> * >>>> Still far from expectations >>>> >>>> I've observed this on FreeSWITCH (Version 1.2.6 git a424765 2013-01-04 >>>> 15:45:59Z) running on ubuntu 12.04 x86_64 >>>> >>>> I want to achieve situation when extension can be used only by one user >>>> at one time. >>>> >>>> >>>> Bellow is my internal sofia profile: >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>> value="false"/> >>>> >>>> >>>> >>>> >>>> >>>> >>> value="$${global_codec_prefs}"/> >>>> >>> value="$${global_codec_prefs}"/> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>> value="${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> >>>> >>>> >>> value="$${domain},$${local_ip_v4}"/> >>>> >>> value="$${presence_privacy}"/> >>>> >>> value="generous"/> >>>> >>>> >>>> >>>> >>> value="$${internal_tls_port}"/> >>>> >>> value="$${internal_ssl_dir}"/> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>> value="$${internal_auth_calls}"/> >>>> >>> value="true"/> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>> value="$${domain}"/> >>>> >>> value="$${domain}"/> >>>> >>>> >>> value="2"/> >>>> >>>> >>>> >>>> >>>> >>>> >>>> Will you be able to help me ? >>>> >>>> -- >>>> Regards >>>> >>>> Tomasz >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> >> -- >> Pozdrawiam >> Tomasz >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130218/0c431b93/attachment-0001.html From jaybinks at gmail.com Mon Feb 18 08:35:46 2013 From: jaybinks at gmail.com (jay binks) Date: Mon, 18 Feb 2013 15:35:46 +1000 Subject: [Freeswitch-users] Time to start understanding the FreeSWITCH codebase In-Reply-To: References: <9982BBC9-3C49-4646-A578-51E1A1C2781C@endigotech.com> <17281507-28F9-4AB5-9799-A930C21F0862@gmail.com> Message-ID: Start to learn the modules you use.... especially the smaller ones. Im my systems, I used Enum quite a bit so I started digging into mod_enum years ago followed by mod_limit ( deprecated now ? ) and mod_sofia etc ... these were both enough to get me understanding the FS Side of things, I then started digging into ESL and FS core. I dont proclaim to be a guru at FS internals, but Ive had my fair share of patches accepted ( and rejected :P ) Jay On 18 February 2013 15:11, Michael Collins wrote: > >From personal experience I can say that it makes sense to start at the > outer levels and work your way in. A gentle way to get into what the source > code does is to look at the source code for API commands or dialplan > applications with which you are very familiar or that have very simple > functions. > > I would start with mod_commands.c and look at simple commands like "version" > and "banner". Once you get the hang of those then try digging a little > deeper, like with the "show" API. That API is a great example of how the FS > devs handle argument parsing and use the occasional "really large" case > block. > > If you would rather start with the dialplan stuff then check out > mod_dptools.c. Look at really simple apps like "log". You'll see lots of > references to the "session" which you MUST learn about if you want to do > anything with a live call. One thing you'll see is that the functions in the > "outer" layers will make calls to "core" functions. As homework I recommend > you look in mod_dptools.c for the definition of the "playback" app. Hint: to > get the whole picture you'll need to also look in switch_ivr_play_say.c. ;) > > Have fun and don't be afraid to ask questions here or in the IRC channel. > > -MC > > > On Sun, Feb 17, 2013 at 4:42 PM, Steven Ayre wrote: >> >> Use the source, Luke. >> >> Digging around in the source is one way to explore how FS works. Some is >> easier to understand than others. Threading is heavily used so understanding >> mutexes etc is useful. >> >> I'd highly recommend that you read this site though: >> http://docs.freeswitch.org/ >> >> That documents the core API. A lot is based on APR, plus some extra stuff >> like the module interfaces. >> >> Steve >> >> >> >> >> On 18 Feb 2013, at 00:00, Brian Foster wrote: >> >> I think I've come to a stage in my life where I'm more interested than >> ever in learning the internals of FreeSWITCH and start contributing code to >> it. I've been following the project for a few years, but I've never touched >> the code. It's time to change that. >> >> Is there a place on the wiki someone can point me to so I can start >> understanding the FS core? I'm talking fairly basic, beginner level stuff. >> >> Sent from my iPhone >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -- Sincerely Jay From a.afzali2003 at gmail.com Mon Feb 18 09:21:25 2013 From: a.afzali2003 at gmail.com (afshin afzali) Date: Mon, 18 Feb 2013 09:51:25 +0330 Subject: [Freeswitch-users] Using mod_spandsp for data exchange Message-ID: Hi, I was wondering if there is a way to use mod_spandsp for generic data exchange (i.e. file transfer). As I find out mod_spandsp can create soft modems on the system level. If I be able to bridge an inbound call to one of the modems, an external process can use AT commands to establish a connection and exchange data !!! Am I right ? Appreciate all comments, Afshin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130218/0c4b5c5b/attachment.html From julf at julf.com Mon Feb 18 09:48:28 2013 From: julf at julf.com (Johan Helsingius) Date: Mon, 18 Feb 2013 07:48:28 +0100 Subject: [Freeswitch-users] Problems with sip provider In-Reply-To: References: <51210FA4.1010200@julf.com> <51211478.1080108@julf.com> Message-ID: <5121CEBC.2020707@julf.com> > You don't need them to tell you how to configure your device to talk to them (a > FS specific question), you want them to tell you why the authentication failed > (specific to their end). Once we know that it should be possible to know what > needs to be adjusted. Indeed. But the problem is that I can't get past their (useless) first level helpdesk to talk to the people who might actually have access to their logs. Julf From julf at julf.com Mon Feb 18 09:54:33 2013 From: julf at julf.com (Johan Helsingius) Date: Mon, 18 Feb 2013 07:54:33 +0100 Subject: [Freeswitch-users] Problems with sip provider In-Reply-To: <7FA18191BBD319419C1E34EA005B43C5058E3D34B8@GUREXMB02.ASIAN.AD.ARICENT.COM> References: <51210FA4.1010200@julf.com> <7FA18191BBD319419C1E34EA005B43C5058E3D34B8@GUREXMB02.ASIAN.AD.ARICENT.COM> Message-ID: <5121D029.3070907@julf.com> > Could you please share the experience for the another 2 SIP providers? Both based in Netherlands - I use VoIP Planet (http://voipplanet.nl/) for international, and Callvoip (http://callvoip.nl/) for national/ local. Julf From avi at avimarcus.net Mon Feb 18 10:24:31 2013 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 18 Feb 2013 09:24:31 +0200 Subject: [Freeswitch-users] Change Voicemail IVR Menu In-Reply-To: References: Message-ID: Are you referring to the typos fixed in http://fisheye.freeswitch.org/changelog/freeswitch.git?cs=67111b07705372bb9be77accbe0aef3459206c0d ? If there are others, please file a jira with a patch to have it fixed. -Avi On Mon, Feb 18, 2013 at 3:55 AM, Muhammad Shahzad wrote: > Hi, > > I am trying to change menu order of FS voicemail module. I am using > voicemail_ivr with xml_curl, when a voicemail user checks voicemail, i > check if user has his own greeting set, then play normal voicemail menu, > otherwise instead of playing message count and related options, i simply > want to play voicemail preferences menu, in fact only voicemail record > greeting menu. I have tried many different possibilities including many > changes to voicemail_ivr.xml and voicemail_ivr.conf.xml but so far none > worked. FS Wiki on voicemail_ivr module seems outdated. Till now i am only > able to play standard voicemail menu through voicemail_ivr module. > > Do I have to modify native code of these modules (mod_voicemail and > mod_voicemail_ivr) to achieve the goal? Any suggestions / guidelines? > > P.S. Note that sample files (voicemail_ivr.conf.xml and voicemail_ivr.xml) > do not work as is, i had to modify many typos and tag attributes to get it > working same as standard mod_voicemail works. I will update wiki once this > problem is solved. > > Thank you. > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +49 176 99 83 10 85 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130218/1daea443/attachment.html From ckmonkey158 at yahoo.com Mon Feb 18 11:51:12 2013 From: ckmonkey158 at yahoo.com (Chris Monkey) Date: Mon, 18 Feb 2013 00:51:12 -0800 (PST) Subject: [Freeswitch-users] (no subject) Message-ID: <1361177472.11065.YahooMailNeo@web125601.mail.ne1.yahoo.com>         http://www.bernd-kettner.de/hmrtrj/hd20um/7d3jr26f7gdfsrh56e4v8azwp5&fvo0bsewy053lf7omu1kma6y=hu12nh46jv59wtv44q1uw1ommtbcdq            Chris Monkey -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130218/f2730f90/attachment.html From david.villasmil.work at gmail.com Mon Feb 18 11:57:50 2013 From: david.villasmil.work at gmail.com (David Villasmil) Date: Mon, 18 Feb 2013 09:57:50 +0100 Subject: [Freeswitch-users] Multiple Billings with Nibble Billing In-Reply-To: References: Message-ID: <9671ACC8-1039-43BB-A4C3-7281B9807E81@gmail.com> Hello, I'm going out on a limb here, but I don't think so. I've been working with nibblebill lately and it seems to me that the charging possibilities are pretty basic. You can achieve what you want with triggers, i guess. David --- David Villasmil On Feb 17, 2013, at 21:13, Jagadish Thoutam wrote: > Hi All, > > Is there Any option to do Multiple Billing in single call. > > Like at the same time one call can effect with different accounts like USER ----> RESELLER3--->RESELLER2---->RESELLER3 to detect the balance. > > > > Thanks > Jagadish > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 david.villasmil.work at gmail.com Mon Feb 18 12:00:35 2013 From: david.villasmil.work at gmail.com (David Villasmil) Date: Mon, 18 Feb 2013 10:00:35 +0100 Subject: [Freeswitch-users] Multiple Billings with Nibble Billing In-Reply-To: <9671ACC8-1039-43BB-A4C3-7281B9807E81@gmail.com> References: <9671ACC8-1039-43BB-A4C3-7281B9807E81@gmail.com> Message-ID: <5E4FF050-0342-4FBB-AA97-71E042226EC2@gmail.com> Btw, I've been trying to charge on a different table depending on the context, doesn't seem possible either. Once you set the db_table in the module conf, you're stuck with with it... (Or so it looks to me) David --- David Villasmil On Feb 18, 2013, at 9:57, David Villasmil wrote: > Hello, > > I'm going out on a limb here, but I don't think so. I've been working with nibblebill lately and it seems to me that the charging possibilities are pretty basic. > > You can achieve what you want with triggers, i guess. > > David > > --- > > David Villasmil > > > On Feb 17, 2013, at 21:13, Jagadish Thoutam wrote: > >> Hi All, >> >> Is there Any option to do Multiple Billing in single call. >> >> Like at the same time one call can effect with different accounts like USER ----> RESELLER3--->RESELLER2---->RESELLER3 to detect the balance. >> >> >> >> Thanks >> Jagadish >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 david.villasmil.work at gmail.com Mon Feb 18 12:01:36 2013 From: david.villasmil.work at gmail.com (David Villasmil) Date: Mon, 18 Feb 2013 10:01:36 +0100 Subject: [Freeswitch-users] Multiple Billings with Nibble Billing In-Reply-To: <9671ACC8-1039-43BB-A4C3-7281B9807E81@gmail.com> References: <9671ACC8-1039-43BB-A4C3-7281B9807E81@gmail.com> Message-ID: You could also use custom_sal for what you want. --- David Villasmil On Feb 18, 2013, at 9:57, David Villasmil wrote: > Hello, > > I'm going out on a limb here, but I don't think so. I've been working with nibblebill lately and it seems to me that the charging possibilities are pretty basic. > > You can achieve what you want with triggers, i guess. > > David > > --- > > David Villasmil > > > On Feb 17, 2013, at 21:13, Jagadish Thoutam wrote: > >> Hi All, >> >> Is there Any option to do Multiple Billing in single call. >> >> Like at the same time one call can effect with different accounts like USER ----> RESELLER3--->RESELLER2---->RESELLER3 to detect the balance. >> >> >> >> Thanks >> Jagadish >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 david.villasmil.work at gmail.com Mon Feb 18 12:05:36 2013 From: david.villasmil.work at gmail.com (David Villasmil) Date: Mon, 18 Feb 2013 10:05:36 +0100 Subject: [Freeswitch-users] Multiple Billings with Nibble Billing In-Reply-To: <9671ACC8-1039-43BB-A4C3-7281B9807E81@gmail.com> References: <9671ACC8-1039-43BB-A4C3-7281B9807E81@gmail.com> Message-ID: And now that i think about it, maybe i can also use custom_sql for what i need, if custom_sql can use channel variables... --- David Villasmil On Feb 18, 2013, at 9:57, David Villasmil wrote: > Hello, > > I'm going out on a limb here, but I don't think so. I've been working with nibblebill lately and it seems to me that the charging possibilities are pretty basic. > > You can achieve what you want with triggers, i guess. > > David > > --- > > David Villasmil > > > On Feb 17, 2013, at 21:13, Jagadish Thoutam wrote: > >> Hi All, >> >> Is there Any option to do Multiple Billing in single call. >> >> Like at the same time one call can effect with different accounts like USER ----> RESELLER3--->RESELLER2---->RESELLER3 to detect the balance. >> >> >> >> Thanks >> Jagadish >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 a.venugopan at mundio.com Mon Feb 18 12:17:11 2013 From: a.venugopan at mundio.com (Archana Venugopan) Date: Mon, 18 Feb 2013 09:17:11 +0000 Subject: [Freeswitch-users] Multicast in freeswitch In-Reply-To: References: <592A9CF93E12394E8472A6CC66E66BF236B788@Mail-Kilo.squay.com> <592A9CF93E12394E8472A6CC66E66BF23790FC@Mail-Kilo.squay.com> <592A9CF93E12394E8472A6CC66E66BF2379123@Mail-Kilo.squay.com> Message-ID: <592A9CF93E12394E8472A6CC66E66BF23A9EB1@Mail-Kilo.squay.com> Hi, LUA script is written by us. Ya its being lauched from dialplan. The script establishes call to mobile and our deskphone. So far it dint update the DB, it just checks the DB and establishes call. Regards, Archana From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Eliot Gable Sent: 16 February 2013 16:57 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Multicast in freeswitch You still have not provided enough information. Is your LUA script stand-alone, or is it being launched from FreeSWITCH? How are you launching? Where are you launching it? Is it being launched from the dialplan or from an API call? Is the script just to update a field in the database or does it do more than that? On Sat, Feb 16, 2013 at 9:08 AM, Archana Venugopan > wrote: Hi, I want to update my Database table, based on when the phone is ACTIVE and phone is HANGUP. I just want to get these 2 events ACTIVE and HANGUP in my lua script, so that I can put a if conditions to update my Database table. So need to know what exactly I should do to get these events in lua script. I tried this 'local Channel_State = params:getHeader("Channel-State")' but this does not give me these 2 events. Since I could not understand in detail from wiki I need suggestion on what can be done. Many thanks. Regards, Archana From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Eliot Gable Sent: 16 February 2013 13:59 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Multicast in freeswitch If you are using LUA, you probably want setHangupHook to do whatever you are doing. It kind of depends on what you are doing though. Perhaps you should provide more details about what you are trying to do and how you are trying to do it. On Sat, Feb 16, 2013 at 8:05 AM, Archana Venugopan > wrote: Hi, Thanks. I read mod_event_socket wiki, it has some format like this sendmsg call-command: hangup hangup-cause: Should I need to pass on in my script? And how am i to use that hangup message in lua script? Sorry I am not clear from wiki on these. Please help. Thanks Regards, Archana From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Eliot Gable Sent: 16 February 2013 06:11 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Multicast in freeswitch You can use mod_event_socket and watch for SWITCH_CHANNEL_STATE, SWITCH_CHANNEL_CALLSTATE, SWITCH_CHANNEL_CREATE, SWITCH_CHANNEL_ORIGINATE, SWITCH_CHANNEL_BRIDGE, and any other event of interest. Then you can track the state of each call for whatever purpose you might want to use that info for. On Feb 14, 2013 12:31 PM, "Archana Venugopan" > wrote: Hi, I would like to get 'Orig-status' event in freeswitch so that I will know when the phone is being HANGED. But to get that event I read in wiki I need mod_event_multicast. http://wiki.freeswitch.org/wiki/Mod_event_multicast But I don't see in above wiki on how to compile and load mod_event_multicast. Can anyone please let me know how to compile and load multicast? Regards, Archana _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -- Eliot Gable "We do not inherit the Earth from our ancestors: we borrow it from our children." ~David Brower "I decided the words were too conservative for me. We're not borrowing from our children, we're stealing from them--and it's not even considered to be a crime." ~David Brower "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to live; not live to eat.) ~Marcus Tullius Cicero _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -- Eliot Gable "We do not inherit the Earth from our ancestors: we borrow it from our children." ~David Brower "I decided the words were too conservative for me. We're not borrowing from our children, we're stealing from them--and it's not even considered to be a crime." ~David Brower "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to live; not live to eat.) ~Marcus Tullius Cicero -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130218/367383e7/attachment-0001.html From steveayre at gmail.com Mon Feb 18 12:55:39 2013 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 18 Feb 2013 09:55:39 +0000 Subject: [Freeswitch-users] (no subject) In-Reply-To: References: Message-ID: Yes On 17 February 2013 08:57, J G wrote: > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130218/5d460f0d/attachment.html From a.venugopan at mundio.com Mon Feb 18 13:24:51 2013 From: a.venugopan at mundio.com (Archana Venugopan) Date: Mon, 18 Feb 2013 10:24:51 +0000 Subject: [Freeswitch-users] FW: In-band and out band call issues Message-ID: <592A9CF93E12394E8472A6CC66E66BF23ABF7E@Mail-Kilo.squay.com> Can anyone please help me on this peculiar issue. Regards, Archana From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Archana Venugopan Sent: 15 February 2013 16:15 To: FreeSWITCH Users Help Subject: [Freeswitch-users] In-band and out band call issues Hi, Am facing with a typical issue. In my phone settings, if i enable 'Force RFC2833 Out-of-Band DTMF' then my voicemail is not working properly( If I record any greeting or VM, only part of the message is being recorded) but offnet calls for banks or shops is working. If 'Force RFC2833 Out-of-Band DTMF' is disabled my voicemail is working properly but When I call offnet numbers like Banks or shops, then my key press is not being recognised i.e no DTMF detection. My internal .xml and external settings for DTMF is Please let me know what needs to be changed so that both voicemail and offnet numbers work properly. Regards, Archana -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130218/fa99205b/attachment.html -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: ATT00001.txt Url: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130218/fa99205b/attachment.txt From shaheryarkh at gmail.com Mon Feb 18 13:57:21 2013 From: shaheryarkh at gmail.com (Muhammad Shahzad) Date: Mon, 18 Feb 2013 11:57:21 +0100 Subject: [Freeswitch-users] (no subject) In-Reply-To: References: Message-ID: I am getting empty emails like this, while other people can see content and respond to them. What am i missing? Do i need to change my mailing list settings to view these messages. On 18 Feb 2013 10:59, "Steven Ayre" wrote: > > Yes > > On 17 February 2013 08:57, J G wrote: >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130218/b5759c29/attachment.html From jaganthoutam at gmail.com Mon Feb 18 13:59:09 2013 From: jaganthoutam at gmail.com (Jagadish Thoutam) Date: Mon, 18 Feb 2013 16:29:09 +0530 Subject: [Freeswitch-users] Multiple Billings with Nibble Billing In-Reply-To: References: <9671ACC8-1039-43BB-A4C3-7281B9807E81@gmail.com> Message-ID: I think this can happen if i can add some more custom code in Nibble Bill source. On 18 February 2013 14:35, David Villasmil wrote: > And now that i think about it, maybe i can also use custom_sql for what i > need, if custom_sql can use channel variables... > > --- > > David Villasmil > > > On Feb 18, 2013, at 9:57, David Villasmil > wrote: > > > Hello, > > > > I'm going out on a limb here, but I don't think so. I've been working > with nibblebill lately and it seems to me that the charging possibilities > are pretty basic. > > > > You can achieve what you want with triggers, i guess. > > > > David > > > > --- > > > > David Villasmil > > > > > > On Feb 17, 2013, at 21:13, Jagadish Thoutam > wrote: > > > >> Hi All, > >> > >> Is there Any option to do Multiple Billing in single call. > >> > >> Like at the same time one call can effect with different accounts like > USER ----> RESELLER3--->RESELLER2---->RESELLER3 to detect the balance. > >> > >> > >> > >> Thanks > >> Jagadish > >> > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130218/abd0d4b1/attachment.html From vitaliy.davudov at vts24.ru Mon Feb 18 14:00:47 2013 From: vitaliy.davudov at vts24.ru (=?UTF-8?B?0JLQuNGC0LDQu9C40Lkg0JTQsNCy0YPQtNC+0LI=?=) Date: Mon, 18 Feb 2013 15:00:47 +0400 Subject: [Freeswitch-users] Memory growth Message-ID: <512209DF.6050806@vts24.ru> Hi, list! I've installed FreeSWITCH Version 1.2.3+git~20120920T220849Z~f718a5e8e6 (1.2.3; git at commit f718a5e8e6 on Thu, 20 Sep 2012 22:08:49 Z) I noticed that FreeSwtich uses a lot of memory of my system, and percent of used memory is gradually growing regardless of the number of the simultaneous calls: # uname -a Linux freeswitch1 2.6.18-308.13.1.el5 #1 SMP Tue Aug 21 17:10:18 EDT 2012 x86_64 x86_64 x86_64 GNU/Linux # top top - 14:21:39 up 140 days, 29 min, 2 users, load average: 0.11, 0.14, 0.10 Tasks: 99 total, 1 running, 96 sleeping, 2 stopped, 0 zombie Cpu(s): 1.2%us, 0.8%sy, 0.0%ni, 96.9%id, 0.1%wa, 0.2%hi, 0.7%si, 0.0%st Mem: 3781680k total, 3748360k used, 33320k free, 120708k buffers Swap: 2096472k total, 17224k used, 2079248k free, 1838356k cached PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND 9584 root -2 -10 1792m 1.3g 8252 S 2.7 36.8 935:07.85 freeswitch 3095 mysql 15 0 329m 28m 4588 S 0.7 0.8 713:07.03 mysqld At this moment FS handles no more 15 simultaneous calls, and has about 117 registered users. Is there a tool to monitor, which of processes or modules in FS is using memory and how much? -- Best regards, Vitaly. From steveayre at gmail.com Mon Feb 18 14:16:00 2013 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 18 Feb 2013 11:16:00 +0000 Subject: [Freeswitch-users] Memory growth In-Reply-To: <512209DF.6050806@vts24.ru> References: <512209DF.6050806@vts24.ru> Message-ID: Are you able to reproduce it on a newer version? Try current head of the master branch. If you can, then try running FS through valgrind which may reveal the source of any memory leaks. -Steve On 18 February 2013 11:00, ??????? ??????? wrote: > Hi, list! > I've installed FreeSWITCH Version 1.2.3+git~20120920T220849Z~f718a5e8e6 > (1.2.3; git at commit f718a5e8e6 on Thu, 20 Sep 2012 22:08:49 Z) > > I noticed that FreeSwtich uses a lot of memory of my system, and percent > of used memory is gradually growing regardless of the number of the > simultaneous calls: > > # uname -a > Linux freeswitch1 2.6.18-308.13.1.el5 #1 SMP Tue Aug 21 17:10:18 EDT > 2012 x86_64 x86_64 x86_64 GNU/Linux > > # top > top - 14:21:39 up 140 days, 29 min, 2 users, load average: 0.11, 0.14, > 0.10 > Tasks: 99 total, 1 running, 96 sleeping, 2 stopped, 0 zombie > Cpu(s): 1.2%us, 0.8%sy, 0.0%ni, 96.9%id, 0.1%wa, 0.2%hi, 0.7%si, > 0.0%st > Mem: 3781680k total, 3748360k used, 33320k free, 120708k buffers > Swap: 2096472k total, 17224k used, 2079248k free, 1838356k cached > > PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND > 9584 root -2 -10 1792m 1.3g 8252 S 2.7 36.8 935:07.85 freeswitch > 3095 mysql 15 0 329m 28m 4588 S 0.7 0.8 713:07.03 mysqld > > At this moment FS handles no more 15 simultaneous calls, and has about > 117 registered users. > Is there a tool to monitor, which of processes or modules in FS is using > memory and how much? > > -- > Best regards, > Vitaly. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130218/425cd9ee/attachment.html From steveayre at gmail.com Mon Feb 18 14:17:50 2013 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 18 Feb 2013 11:17:50 +0000 Subject: [Freeswitch-users] (no subject) In-Reply-To: References: Message-ID: No it was an empty mail, probably either a test message or an accidental send. On 18 February 2013 10:57, Muhammad Shahzad wrote: > I am getting empty emails like this, while other people can see content > and respond to them. What am i missing? Do i need to change my mailing list > settings to view these messages. > > On 18 Feb 2013 10:59, "Steven Ayre" wrote: > > > > Yes > > > > On 17 February 2013 08:57, J G > wrote: > >> > >> > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> 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 > >> > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130218/884b5f3b/attachment.html From g.d.monnezza at tiscali.it Mon Feb 18 14:35:16 2013 From: g.d.monnezza at tiscali.it (g) Date: Mon, 18 Feb 2013 12:35:16 +0100 Subject: [Freeswitch-users] Blind transfer dropped calls Message-ID: <1436977.bWcj1llolB@virtex> Hi everyone. I'm not absolute begineer with freeswitch, but pretty new after long experience with asterisk. I'm facing now with an installation (1.3.13b+git~20130213) of FS with an issue I can't solve (the same issue was present in 1.0head previous installation), so I invoke your help The scenario is simple: Grandstream (updated firmware) Ip phones connected to FS, gateway to SIP VoIP provider via trunk. Everything works great. Internal phones can talk each oters, calls can came in and go out to PSTN without problem, but ... When a phone call came and an internal phone picks it up, it is impossible to blind tranfert that call to onother internal phone. I mean that the phone i try to transfert the call rings, then it picks up the call and the call is dropped. I suspect is a codec problem, but I can't figure why. All the phones are configured the same way. Here below the log af the process, from external call incoming until the drop. I can see two suspicious steps, where a "no read codec" error and an "incopatible_destination" message appears. I think I made a trivial error, but I really can't understand where. Can please anyone point me in the right direction to solve it? Many thanks g ____ call log taken from the cli - some data is ### masked ### ___ 2013-01-25 17:15:04.770166 [DEBUG] switch_core_session.c:874 Send signal sofia/sipinterface_1/201 at 192.168.35.80 [BREAK] 2013-01-25 17:15:04.770166 [DEBUG] switch_core_session.c:874 Send signal sofia/sipinterface_1/201 at 192.168.35.80 [BREAK] 2013-01-25 17:15:04.770166 [DEBUG] sofia.c:5370 Channel sofia/sipinterface_1/201 at 192.168.35.80 entering state [received][100] 2013-01-25 17:15:04.770166 [DEBUG] sofia.c:5381 Remote SDP: v=0 o=201 8000 8004 IN IP4 192.168.35.114 s=SIP Call c=IN IP4 192.168.35.114 t=0 0 m=audio 5086 RTP/AVP 18 0 8 2 97 9 3 101 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=rtpmap:9 G722/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 a=sendonly a=ptime:20 2013-01-25 17:15:04.770166 [DEBUG] sofia.c:5810 Passing SDP to other leg. v=0 o=201 8000 8004 IN IP4 192.168.35.114 s=SIP Call c=IN IP4 192.168.35.114 t=0 0 m=audio 5086 RTP/AVP 18 0 8 2 97 9 3 101 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=rtpmap:9 G722/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 a=sendonly a=ptime:20 2013-01-25 17:15:04.770166 [DEBUG] switch_channel.c:1554 (sofia/sipinterface_1/201 at 192.168.35.80) Callstate Change ACTIVE -> HELD 2013-01-25 17:15:04.770166 [DEBUG] switch_core_session.c:790 Send signal sofia/sipinterface_3/347******** ### masked ### [BREAK] 2013-01-25 17:15:04.770166 [DEBUG] mod_sofia.c:1812 sofia/sipinterface_3/347******** ### masked ### Sending media re-direct: v=0 o=201 8000 8004 IN IP4 192.168.35.114 s=SIP Call c=IN IP4 192.168.35.114 t=0 0 m=audio 5086 RTP/AVP 18 0 8 2 97 9 3 101 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=rtpmap:9 G722/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 a=sendonly a=ptime:20 2013-01-25 17:15:04.770166 [DEBUG] switch_core_session.c:728 Send signal sofia/sipinterface_3/347******** ### masked ### [BREAK] 2013-01-25 17:15:04.770166 [DEBUG] switch_core_session.c:874 Send signal sofia/sipinterface_3/347******** ### masked ### [BREAK] 2013-01-25 17:15:04.770166 [DEBUG] sofia.c:5370 Channel sofia/sipinterface_3/347******** ### masked ### entering state [calling][0] 2013-01-25 17:15:04.810163 [DEBUG] switch_core_session.c:874 Send signal sofia/sipinterface_3/347******** ### masked ### [BREAK] 2013-01-25 17:15:04.810163 [DEBUG] switch_core_session.c:874 Send signal sofia/sipinterface_3/347******** ### masked ### [BREAK] 2013-01-25 17:15:04.830180 [DEBUG] sofia.c:4953 Passing 200 OK to other leg 2013-01-25 17:15:04.830180 [DEBUG] switch_core_session.c:790 Send signal sofia/sipinterface_1/201 at 192.168.35.80 [BREAK] 2013-01-25 17:15:04.830180 [DEBUG] sofia.c:5370 Channel sofia/sipinterface_3/347******** ### masked ### entering state [completing] [200] 2013-01-25 17:15:04.830180 [DEBUG] sofia.c:5381 Remote SDP: v=0 o=root 1558328831 1558328833 IN IP4 xxx.xxx.xxx.xxx s=VOIP ### provider masked ### c=IN IP4 xxx.xxx.xxx.xxx #### masked ### t=0 0 m=audio 46518 RTP/AVP 18 3 97 8 0 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=recvonly a=ptime:20 2013-01-25 17:15:04.830180 [DEBUG] mod_sofia.c:2363 Responding with 200 [OK] 2013-01-25 17:15:04.830180 [DEBUG] switch_core_session.c:728 Send signal sofia/sipinterface_1/201 at 192.168.35.80 [BREAK] 2013-01-25 17:15:04.830180 [DEBUG] switch_core_session.c:874 Send signal sofia/sipinterface_3/347******** ### masked ### [BREAK] 2013-01-25 17:15:04.830180 [DEBUG] switch_core_session.c:874 Send signal sofia/sipinterface_3/347******** ### masked ### [BREAK] 2013-01-25 17:15:04.830180 [DEBUG] sofia.c:5370 Channel sofia/sipinterface_3/347******** ### masked ### entering state [ready][200] 2013-01-25 17:15:04.830180 [DEBUG] switch_core_session.c:790 Send signal sofia/sipinterface_1/201 at 192.168.35.80 [BREAK] 2013-01-25 17:15:04.830180 [DEBUG] switch_core_session.c:874 Send signal sofia/sipinterface_1/201 at 192.168.35.80 [BREAK] 2013-01-25 17:15:04.830180 [DEBUG] sofia.c:5370 Channel sofia/sipinterface_1/201 at 192.168.35.80 entering state [completed][200] 2013-01-25 17:15:04.910172 [DEBUG] switch_core_session.c:874 Send signal sofia/sipinterface_1/201 at 192.168.35.80 [BREAK] 2013-01-25 17:15:04.910172 [DEBUG] switch_core_session.c:874 Send signal sofia/sipinterface_1/201 at 192.168.35.80 [BREAK] 2013-01-25 17:15:04.910172 [DEBUG] switch_core_session.c:874 Send signal sofia/sipinterface_1/201 at 192.168.35.80 [BREAK] 2013-01-25 17:15:04.910172 [DEBUG] sofia.c:5370 Channel sofia/sipinterface_1/201 at 192.168.35.80 entering state [ready][200] 2013-01-25 17:15:09.270174 [DEBUG] switch_core_session.c:874 Send signal sofia/sipinterface_1/201 at 192.168.35.80 [BREAK] 2013-01-25 17:15:09.270174 [DEBUG] sofia.c:6356 Process REFER to [203 at 192.168.35.80] 2013-01-25 17:15:09.270174 [DEBUG] switch_channel.c:2842 (sofia/sipinterface_1/201 at 192.168.35.80) Callstate Change HELD -> HANGUP 2013-01-25 17:15:09.270174 [NOTICE] switch_ivr.c:1696 Hangup sofia/sipinterface_1/201 at 192.168.35.80 [CS_HIBERNATE] [BLIND_TRANSFER] 2013-01-25 17:15:09.270174 [DEBUG] switch_channel.c:2865 Send signal sofia/sipinterface_1/201 at 192.168.35.80 [KILL] 2013-01-25 17:15:09.270174 [DEBUG] switch_ivr_bridge.c:929 sofia/sipinterface_1/201 at 192.168.35.80 skip receive message [UNBRIDGE] (channel is hungup already) 2013-01-25 17:15:09.270174 [DEBUG] switch_core_session.c:1179 Send signal sofia/sipinterface_1/201 at 192.168.35.80 [BREAK] 2013-01-25 17:15:09.270174 [DEBUG] switch_ivr.c:1705 (sofia/sipinterface_3/347******** ### masked ###) State Change CS_HIBERNATE -> CS_ROUTING 2013-01-25 17:15:09.270174 [DEBUG] switch_core_session.c:728 Send signal sofia/sipinterface_3/347******** ### masked ### [BREAK] 2013-01-25 17:15:09.270174 [DEBUG] switch_core_session.c:1179 Send signal sofia/sipinterface_3/347******** ### masked ### [BREAK] 2013-01-25 17:15:09.270174 [DEBUG] switch_core_session.c:728 Send signal sofia/sipinterface_3/347******** ### masked ### [BREAK] 2013-01-25 17:15:09.270174 [NOTICE] switch_ivr.c:1711 Transfer sofia/sipinterface_3/347******** ### masked ### to XML[203 at context_1] 2013-01-25 17:15:09.270174 [DEBUG] switch_core_state_machine.c:362 (sofia/sipinterface_1/201 at 192.168.35.80) Running State Change CS_HANGUP 2013-01-25 17:15:09.270174 [DEBUG] switch_core_state_machine.c:362 (sofia/sipinterface_3/347******** ### masked ###) Running State Change CS_ROUTING 2013-01-25 17:15:09.270174 [DEBUG] switch_channel.c:1880 (sofia/sipinterface_3/347******** ### masked ###) Callstate Change ACTIVE -> RINGING 2013-01-25 17:15:09.270174 [DEBUG] switch_core_state_machine.c:410 (sofia/sipinterface_3/347******** ### masked ###) State ROUTING 2013-01-25 17:15:09.270174 [DEBUG] mod_sofia.c:148 sofia/sipinterface_3/347******** ### masked ### SOFIA ROUTING 2013-01-25 17:15:09.270174 [DEBUG] switch_core_state_machine.c:104 sofia/sipinterface_3/347******** ### masked ### Standard ROUTING 2013-01-25 17:15:09.270174 [DEBUG] switch_core_state_machine.c:602 (sofia/sipinterface_1/201 at 192.168.35.80) State HANGUP 2013-01-25 17:15:09.270174 [DEBUG] mod_sofia.c:465 Channel sofia/sipinterface_1/201 at 192.168.35.80 hanging up, cause: BLIND_TRANSFER 2013-01-25 17:15:09.270174 [DEBUG] mod_sofia.c:509 Sending BYE to sofia/sipinterface_1/201 at 192.168.35.80 2013-01-25 17:15:09.270174 [DEBUG] switch_core_state_machine.c:47 sofia/sipinterface_1/201 at 192.168.35.80 Standard HANGUP, cause: BLIND_TRANSFER 2013-01-25 17:15:09.270174 [DEBUG] switch_core_state_machine.c:602 (sofia/sipinterface_1/201 at 192.168.35.80) State HANGUP going to sleep 2013-01-25 17:15:09.270174 [DEBUG] switch_core_state_machine.c:393 (sofia/sipinterface_1/201 at 192.168.35.80) State Change CS_HANGUP -> CS_REPORTING 2013-01-25 17:15:09.270174 [DEBUG] switch_core_session.c:1179 Send signal sofia/sipinterface_1/201 at 192.168.35.80 [BREAK] 2013-01-25 17:15:09.270174 [DEBUG] switch_core_state_machine.c:362 (sofia/sipinterface_1/201 at 192.168.35.80) Running State Change CS_REPORTING 2013-01-25 17:15:09.270174 [DEBUG] switch_core_state_machine.c:662 (sofia/sipinterface_1/201 at 192.168.35.80) State REPORTING 2013-01-25 17:15:09.270174 [DEBUG] switch_nat.c:510 mapped public port 29868 protocol UDP to localport 29868 2013-01-25 17:15:09.290165 [DEBUG] switch_nat.c:510 mapped public port 29869 protocol UDP to localport 29869 2013-01-25 17:15:09.290165 [DEBUG] switch_core_session.c:728 Send signal sofia/sipinterface_3/347******** ### masked ### [BREAK] 2013-01-25 17:15:09.290165 [DEBUG] switch_core_session.c:874 Send signal sofia/sipinterface_3/347******** ### masked ### [BREAK] 2013-01-25 17:15:09.330130 [DEBUG] switch_core_session.c:874 Send signal sofia/sipinterface_3/347******** ### masked ### [BREAK] 2013-01-25 17:15:09.330130 [DEBUG] switch_core_session.c:874 Send signal sofia/sipinterface_3/347******** ### masked ### [BREAK] 2013-01-25 17:15:09.330130 [DEBUG] switch_core_session.c:874 Send signal sofia/sipinterface_3/347******** ### masked ### [BREAK] 2013-01-25 17:15:09.730204 [DEBUG] switch_core_state_machine.c:79 sofia/sipinterface_1/201 at 192.168.35.80 Standard REPORTING, cause: BLIND_TRANSFER 2013-01-25 17:15:09.730204 [DEBUG] switch_core_state_machine.c:662 (sofia/sipinterface_1/201 at 192.168.35.80) State REPORTING going to sleep 2013-01-25 17:15:09.730204 [DEBUG] switch_core_state_machine.c:387 (sofia/sipinterface_1/201 at 192.168.35.80) State Change CS_REPORTING -> CS_DESTROY 2013-01-25 17:15:09.730204 [DEBUG] switch_core_session.c:1179 Send signal sofia/sipinterface_1/201 at 192.168.35.80 [BREAK] 2013-01-25 17:15:09.730204 [DEBUG] switch_core_session.c:1379 Session 77 (sofia/sipinterface_1/201 at 192.168.35.80) Locked, Waiting on external entities 2013-01-25 17:15:09.730204 [NOTICE] switch_core_session.c:1397 Session 77 (sofia/sipinterface_1/201 at 192.168.35.80) Ended 2013-01-25 17:15:09.730204 [NOTICE] switch_core_session.c:1399 Close Channel sofia/sipinterface_1/201 at 192.168.35.80 [CS_DESTROY] 2013-01-25 17:15:09.730204 [DEBUG] switch_core_state_machine.c:491 (sofia/sipinterface_1/201 at 192.168.35.80) Callstate Change HANGUP -> DOWN 2013-01-25 17:15:09.730204 [DEBUG] switch_core_state_machine.c:494 (sofia/sipinterface_1/201 at 192.168.35.80) Running State Change CS_DESTROY 2013-01-25 17:15:09.730204 [DEBUG] switch_core_state_machine.c:504 (sofia/sipinterface_1/201 at 192.168.35.80) State DESTROY 2013-01-25 17:15:09.730204 [DEBUG] mod_sofia.c:370 sofia/sipinterface_1/201 at 192.168.35.80 SOFIA DESTROY 2013-01-25 17:15:09.730204 [DEBUG] switch_core_state_machine.c:86 sofia/sipinterface_1/201 at 192.168.35.80 Standard DESTROY 2013-01-25 17:15:09.730204 [DEBUG] switch_core_state_machine.c:504 (sofia/sipinterface_1/201 at 192.168.35.80) State DESTROY going to sleep 2013-01-25 17:15:41.650174 [ERR] switch_core_io.c:131 sofia/sipinterface_3/347******** ### masked ### has no read codec. 2013-01-25 17:15:41.650174 [DEBUG] switch_channel.c:2842 (sofia/sipinterface_3/347******** ### masked ###) Callstate Change RINGING -> HANGUP 2013-01-25 17:15:41.650174 [NOTICE] switch_core_io.c:132 Hangup sofia/sipinterface_3/347******** ### masked ### [CS_ROUTING] [INCOMPATIBLE_DESTINATION] 2013-01-25 17:15:41.650174 [DEBUG] switch_channel.c:2865 Send signal sofia/sipinterface_3/347******** ### masked ### [KILL] 2013-01-25 17:15:41.650174 [DEBUG] switch_core_session.c:1179 Send signal sofia/sipinterface_3/347******** ### masked ### [BREAK] 2013-01-25 17:15:41.650174 [INFO] mod_dialplan_xml.c:481 Processing Gianbattista Gallus <0707966538>->203 in context context_1 Dialplan: sofia/sipinterface_3/347******** ### masked ### parsing [context_1->conditioning_callerid] continue=true Dialplan: sofia/sipinterface_3/347******** ### masked ### Regex (FAIL) [conditioning_callerid] ${internal_caller_id_number}() =~ /^.+$/ break=on-false Dialplan: sofia/sipinterface_3/347******** ### masked ### parsing [context_1->postroute_global] continue=true Dialplan: sofia/sipinterface_3/347******** ### masked ### Absolute Condition [postroute_global] Dialplan: sofia/sipinterface_3/347******** ### masked ### Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) Dialplan: sofia/sipinterface_3/347******** ### masked ### Action hash(insert/${domain_name}- last_dial/${caller_id_number}/${destination_number}) Dialplan: sofia/sipinterface_3/347******** ### masked ### Action hash(insert/${domain_name}-last_dial/global/${uuid}) Dialplan: sofia/sipinterface_3/347******** ### masked ### Action set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) Dialplan: sofia/sipinterface_3/347******** ### masked ### parsing [context_1->main_trunk_1_pattern_5] continue=true Dialplan: sofia/sipinterface_3/347******** ### masked ### Regex (FAIL) [main_trunk_1_pattern_5] destination_number(203) =~ /^0(\d+)$/ break=on-false Dialplan: sofia/sipinterface_3/347******** ### masked ### parsing [context_1->main_number_2] continue=true Dialplan: sofia/sipinterface_3/347******** ### masked ### Regex (FAIL) [main_number_2] destination_number(203) =~ /^999$/ break=on-false Dialplan: sofia/sipinterface_3/347******** ### masked ### parsing [context_1->main_number_4] continue=true Dialplan: sofia/sipinterface_3/347******** ### masked ### Regex (FAIL) [main_number_4] destination_number(203) =~ /^201$/ break=on-false Dialplan: sofia/sipinterface_3/347******** ### masked ### parsing [context_1->main_number_5] continue=true Dialplan: sofia/sipinterface_3/347******** ### masked ### Regex (FAIL) [main_number_5] destination_number(203) =~ /^202$/ break=on-false Dialplan: sofia/sipinterface_3/347******** ### masked ### parsing [context_1->main_number_6] continue=true Dialplan: sofia/sipinterface_3/347******** ### masked ### Regex (PASS) [main_number_6] destination_number(203) =~ /^203$/ break=on-false Dialplan: sofia/sipinterface_3/347******** ### masked ### Action set(hangup_after_bridge=true) Dialplan: sofia/sipinterface_3/347******** ### masked ### Action set(continue_on_fail=true) Dialplan: sofia/sipinterface_3/347******** ### masked ### Action set(vm_auto_play=false) Dialplan: sofia/sipinterface_3/347******** ### masked ### Action set(call_timeout=20) Dialplan: sofia/sipinterface_3/347******** ### masked ### Action set(ringback=${us-ring}) Dialplan: sofia/sipinterface_3/347******** ### masked ### Action set(transfer_ringback=${us-ring}) Dialplan: sofia/sipinterface_3/347******** ### masked ### Action export(sip_callee_id_name=Vanna Cuccu) Dialplan: sofia/sipinterface_3/347******** ### masked ### Action export(sip_callee_id_number=203) Dialplan: sofia/sipinterface_3/347******** ### masked ### Action bridge(user/203 at 192.168.35.80) Dialplan: sofia/sipinterface_3/347******** ### masked ### Action answer() Dialplan: sofia/sipinterface_3/347******** ### masked ### Action voicemail(default voicemail_1 203) Dialplan: sofia/sipinterface_3/347******** ### masked ### Action hangup() Dialplan: sofia/sipinterface_3/347******** ### masked ### parsing [context_1->main_number_7] continue=true Dialplan: sofia/sipinterface_3/347******** ### masked ### Regex (FAIL) [main_number_7] destination_number(203) =~ /^204$/ break=on-false Dialplan: sofia/sipinterface_3/347******** ### masked ### parsing [context_1->main_number_8] continue=true Dialplan: sofia/sipinterface_3/347******** ### masked ### Regex (FAIL) [main_number_8] destination_number(203) =~ /^200$/ break=on-false Dialplan: sofia/sipinterface_3/347******** ### masked ### parsing [context_1- >catchall] continue=false Dialplan: sofia/sipinterface_3/347******** ### masked ### Regex (PASS) [catchall] destination_number(203) =~ /^(\d+)$/ break=on-false Dialplan: sofia/sipinterface_3/347******** ### masked ### Action playback(misc/invalid_extension.wav) 2013-01-25 17:15:41.670347 [INFO] switch_channel.c:2704 sofia/sipinterface_3/347******** ### masked ### Flipping CID from "Gianbattista Gallus" <0707966538> to "Outbound Call" <347******** ### masked ###> 2013-01-25 17:15:41.670347 [DEBUG] switch_core_state_machine.c:410 (sofia/sipinterface_3/347******** ### masked ###) State ROUTING going to sleep 2013-01-25 17:15:41.670347 [DEBUG] switch_core_state_machine.c:362 (sofia/sipinterface_3/347******** ### masked ###) Running State Change CS_HANGUP 2013-01-25 17:15:41.670347 [DEBUG] switch_core_state_machine.c:602 (sofia/sipinterface_3/347******** ### masked ###) State HANGUP 2013-01-25 17:15:41.670347 [DEBUG] mod_sofia.c:465 Channel sofia/sipinterface_3/347******** ### masked ### hanging up, cause: INCOMPATIBLE_DESTINATION 2013-01-25 17:15:41.670347 [DEBUG] mod_sofia.c:509 Sending BYE to sofia/sipinterface_3/347******** ### masked ### 2013-01-25 17:15:41.670347 [DEBUG] switch_core_state_machine.c:47 sofia/sipinterface_3/347******** ### masked ### Standard HANGUP, cause: INCOMPATIBLE_DESTINATION 2013-01-25 17:15:41.670347 [DEBUG] switch_core_state_machine.c:602 (sofia/sipinterface_3/347******** ### masked ###) State HANGUP going to sleep 2013-01-25 17:15:41.670347 [DEBUG] switch_core_state_machine.c:393 (sofia/sipinterface_3/347******** ### masked ###) State Change CS_HANGUP -> CS_REPORTING 2013-01-25 17:15:41.670347 [DEBUG] switch_core_session.c:1179 Send signal sofia/sipinterface_3/347******** ### masked ### [BREAK] 2013-01-25 17:15:41.670347 [DEBUG] switch_core_state_machine.c:362 (sofia/sipinterface_3/347******** ### masked ###) Running State Change CS_REPORTING 2013-01-25 17:15:41.670347 [DEBUG] switch_core_state_machine.c:662 (sofia/sipinterface_3/347******** ### masked ###) State REPORTING 2013-01-25 17:15:42.090203 [DEBUG] switch_core_state_machine.c:79 sofia/sipinterface_3/347******** ### masked ### Standard REPORTING, cause: INCOMPATIBLE_DESTINATION 2013-01-25 17:15:42.090203 [DEBUG] switch_core_state_machine.c:662 (sofia/sipinterface_3/347******** ### masked ###) State REPORTING going to sleep 2013-01-25 17:15:42.090203 [DEBUG] switch_core_state_machine.c:387 (sofia/sipinterface_3/347******** ### masked ###) State Change CS_REPORTING -> CS_DESTROY 2013-01-25 17:15:42.090203 [DEBUG] switch_core_session.c:1179 Send signal sofia/sipinterface_3/347******** ### masked ### [BREAK] 2013-01-25 17:15:42.090203 [DEBUG] switch_core_session.c:1379 Session 78 (sofia/sipinterface_3/347******** ### masked ###) Locked, Waiting on external entities 2013-01-25 17:15:42.090203 [NOTICE] switch_core_session.c:1397 Session 78 (sofia/sipinterface_3/347******** ### masked ###) Ended 2013-01-25 17:15:42.090203 [NOTICE] switch_core_session.c:1399 Close Channel sofia/sipinterface_3/347******** ### masked ### [CS_DESTROY] 2013-01-25 17:15:42.090203 [DEBUG] switch_core_state_machine.c:491 (sofia/sipinterface_3/347******** ### masked ###) Callstate Change HANGUP -> DOWN 2013-01-25 17:15:42.090203 [DEBUG] switch_core_state_machine.c:494 (sofia/sipinterface_3/347******** ### masked ###) Running State Change CS_DESTROY 2013-01-25 17:15:42.090203 [DEBUG] switch_core_state_machine.c:504 (sofia/sipinterface_3/347******** ### masked ###) State DESTROY 2013-01-25 17:15:42.090203 [DEBUG] mod_sofia.c:370 sofia/sipinterface_3/347******** ### masked ### SOFIA DESTROY 2013-01-25 17:15:42.110145 [DEBUG] switch_nat.c:570 unmapped public port 29868 protocol UDP to localport 29868 2013-01-25 17:15:42.130212 [DEBUG] switch_nat.c:570 unmapped public port 29869 protocol UDP to localport 29869 2013-01-25 17:15:42.130212 [DEBUG] switch_core_state_machine.c:86 sofia/sipinterface_3/347******** ### masked ### Standard DESTROY 2013-01-25 17:15:42.130212 [DEBUG] switch_core_state_machine.c:504 (sofia/sipinterface_3/347******** ### masked ###) State DESTROY going to sleep From sos at sokhapkin.dyndns.org Mon Feb 18 14:47:41 2013 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Mon, 18 Feb 2013 06:47:41 -0500 Subject: [Freeswitch-users] Using mod_spandsp for data exchange In-Reply-To: References: Message-ID: <4390459.SrhBBHpkWF@sos> It 's better to exchange data over pure IP rather than over VoIP :-) On Monday 18 February 2013 09:51:25 afshin afzali wrote: > Hi, > > I was wondering if there is a way to use mod_spandsp for generic data > exchange (i.e. file transfer). As I find out mod_spandsp can create soft > modems on the system level. If I be able to bridge an inbound call to one > of the modems, an external process can use AT commands to establish a > connection and exchange data !!! Am I right ? > > Appreciate all comments, > Afshin From david.villasmil.work at gmail.com Mon Feb 18 14:49:40 2013 From: david.villasmil.work at gmail.com (David Villasmil) Date: Mon, 18 Feb 2013 12:49:40 +0100 Subject: [Freeswitch-users] Multiple Billings with Nibble Billing In-Reply-To: References: <9671ACC8-1039-43BB-A4C3-7281B9807E81@gmail.com> Message-ID: look at custom_sql, the solution might be there On Mon, Feb 18, 2013 at 11:59 AM, Jagadish Thoutam wrote: > I think this can happen if i can add some more custom code in Nibble Bill > source. > > > > On 18 February 2013 14:35, David Villasmil > wrote: > >> And now that i think about it, maybe i can also use custom_sql for what i >> need, if custom_sql can use channel variables... >> >> --- >> >> David Villasmil >> >> >> On Feb 18, 2013, at 9:57, David Villasmil >> wrote: >> >> > Hello, >> > >> > I'm going out on a limb here, but I don't think so. I've been working >> with nibblebill lately and it seems to me that the charging possibilities >> are pretty basic. >> > >> > You can achieve what you want with triggers, i guess. >> > >> > David >> > >> > --- >> > >> > David Villasmil >> > >> > >> > On Feb 17, 2013, at 21:13, Jagadish Thoutam >> wrote: >> > >> >> Hi All, >> >> >> >> Is there Any option to do Multiple Billing in single call. >> >> >> >> Like at the same time one call can effect with different accounts like >> USER ----> RESELLER3--->RESELLER2---->RESELLER3 to detect the balance. >> >> >> >> >> >> >> >> Thanks >> >> Jagadish >> >> >> >> >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://wiki.freeswitch.org >> >> http://www.cluecon.com >> >> >> >> 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 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130218/2916149a/attachment.html From a.afzali2003 at gmail.com Mon Feb 18 15:05:19 2013 From: a.afzali2003 at gmail.com (afshin afzali) Date: Mon, 18 Feb 2013 15:35:19 +0330 Subject: [Freeswitch-users] Using mod_spandsp for data exchange In-Reply-To: <4390459.SrhBBHpkWF@sos> References: <4390459.SrhBBHpkWF@sos> Message-ID: Sergey, kidding me , I know :) but I need to exchange data over dial-up connections. tnx On Mon, Feb 18, 2013 at 3:17 PM, Sergey Okhapkin wrote: > It 's better to exchange data over pure IP rather than over VoIP :-) > > On Monday 18 February 2013 09:51:25 afshin afzali wrote: > > Hi, > > > > I was wondering if there is a way to use mod_spandsp for generic data > > exchange (i.e. file transfer). As I find out mod_spandsp can create soft > > modems on the system level. If I be able to bridge an inbound call to one > > of the modems, an external process can use AT commands to establish a > > connection and exchange data !!! Am I right ? > > > > Appreciate all comments, > > Afshin > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130218/acfc0008/attachment.html From david.villasmil.work at gmail.com Mon Feb 18 16:07:58 2013 From: david.villasmil.work at gmail.com (David Villasmil) Date: Mon, 18 Feb 2013 14:07:58 +0100 Subject: [Freeswitch-users] nibblebill custom_sql_save Message-ID: Hello guys, I have the following config for nibbebill: I see this on the log, which is obviously failing: [UPDATE cards_table SET balance=balance- WHERE pin=12345;] is ${nibble_increment} not the correct variable to use? It looks like it is empty! Thanks! David -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130218/d68750fe/attachment.html From cal.leeming at simplicitymedialtd.co.uk Mon Feb 18 16:31:34 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Mon, 18 Feb 2013 13:31:34 +0000 Subject: [Freeswitch-users] Storing CDR UUIDs in MySQL In-Reply-To: References: <5ADA3B5B-47A9-4C6F-BC9A-12A575840006@gmail.com> Message-ID: Hey Ken, Thanks for the feedback. Having used MySQL/Percona for most of my days it's quite a culture change to switch over. Percona are doing amazing things with MySQL, or as my colleague says, "dragging it kicking and screaming into the 21st century". At the same time, MariaDB is really starting to gain momentum, I had a chance to meet Monty (creator of MySQL) at a conference and listen to his talk on MariaDB vs MySQL.. he openly states that no one should be continuing to use MySQL now. MariaDB looks amazing, but I just can't convince myself to put an alpha release into production, as much as I love Monty! PostgreSQL is one of the things on my todo list, it comes heavily recommended by the Django community, and I have suffered first hand at some of the problems in MySQL that apparently wouldn't have happened in PostgreSQL. http://markmail.org/thread/oy3siq3wikhdjld5 https://code.djangoproject.com/ticket/18557 https://groups.google.com/forum/#!topic/django-developers/wvpysb-fXtc/discussion The other problem is that none of our engineers are trained in PostgreSQL (including myself), and if something catastrophic were to happen then we wouldn't be able to deal with it in-house, and this scares me. So to summarize, for us at least - I think Percona MySQL still has a lot of fight left in it, MariaDB will hopefully take over one day, and PostgreSQL is on the list of "nice to haves". Cal On Mon, Feb 18, 2013 at 1:45 AM, Ken Rice wrote: > Why don?t you do something like look at something other than mysql like > PostgreSQL, we insert millions of CDRs per day there and don?t have > performance issues due to Table Partitioning, this allows us to cram well > over 100million new rows/month into a table, partitioned daily along > something like a call_date field (derived by dropping the time from a full > time/date stamp). This also allows us to do really speedy reports without > having to deal with what happens if you do something like select count(*) > from cdr where call_date = ?2013-02-15?; this would be a monster of a query > on a table with say 90 days of CDRs at 4million CDRs/day (360million CDRs > total) but by using the table partitions with a date constraint, the query > planer eliminates partitions that do not match the date constraint the > speeds along the query. > > The caveat to this is if you don?t give it some sort of date constraint > you end up looking across many partitions for the data... > > > On 2/17/13 6:48 PM, "Steven Ayre" wrote: > > You could take a SHA1, but then might get collisions. That can happen with > UUIDs too, but since it includes the MAC address and time stamp is less > likely and would be more predictable. > > The originate_uuid may get collisions depending on what you pick, and the > callid puts it outside your control entirely - in those situations you > might need to handle collisions anyway. > > Sent from my iPad > > > > On 17 Feb 2013, at 23:47, "Cal Leeming [Simplicity Media Ltd]" < > cal.leeming at simplicitymedialtd.co.uk> wrote: > > Ah, yeah that changes the playing field for sure. > > The problem with CHAR is the INSERT speed though, and I've seen similar > issues on other builds too.. after you push past a few million rows, it > really starts to slow down.. If the database was in memory, it might not be > such an issue though. > > In some ways it 'feels cleaner' using a separate table and then JOIN on a > BIGINT autoinc column, as the thought of JOIN'ing and GROUP BY on a BINARY > seems wrong. > > Another option would be to take an SHA1 hash of the UUID, then convert > that from hex to bin, and put that in the BINARY field.. this way you can > guarantee each UUID will be the same size, not worry about changes in data > schema, and get the INSERT speed benefits of the BINARY too - thoughts? > > Cal > > On Sun, Feb 17, 2013 at 11:15 PM, Steven Ayre wrote: > > +1 > > Also as an example the 'inbound-use-callid-as-uuid' parameter I mentioned > before is a Sofia profile parameter that uses the Call-ID header value from > the INVITE sent by the caller as the UUID for the channel. That could be > useful for example in matching up channels against SIP traces as you only > need to store 1 ID not 2. I frequently see ones for example of > $UUID@$SERVER.COM - far longer than 128bits, and > using characters other than 0-9A-F so that it can't be folded into bits. > > > In this case a CHAR column becomes most flexible and the BIGINT column > gives you the speed advantage of the BINARY. > > -Steve > > > > On 17 February 2013 22:39, Seven Du wrote: > > > you can use any arbitrary string as uuid, say > > originate {origination_uuid=my-uuid-that-is-not-128-bit}user/1000 ?. > > > > On Monday, February 18, 2013 at 6:13 AM, Cal Leeming [Simplicity Media > Ltd] wrote: > > > Thanks for the feedback Steve, > > In ref to auto inc, actually, I have! :) In the code on the blog, it shows > that the tables uses a BIGINT auto increment for each row, and the > performance was still good. However, using this approach introduces an > unnecessary JOIN and uses more index and table space - unless I've missed > something? > > Could you elaborate more on the UUIDs that may not be 128-bit? My > google-fu failed me :/ > > Cal > > On Sun, Feb 17, 2013 at 9:11 PM, Steven Ayre wrote: > > Another possibility you haven't looked at is having a table of UUIDs with > a BIGINT auto_increment PK and using that as keys in other takes with a > foreign key relationship (whether defined or implied). That would give you > a smaller integer (hence faster select/insert/join/etc) that's still > guaranteed to be unique. Though the auto_increment INSERT could then become > a bottleneck (hint run a recent 5.5/5.6 MySQL version at the very least > since that bottleneck while still present was very much reduced). > > -Steve > > > > On 17 February 2013 20:55, Cal Leeming [Simplicity Media Ltd] < > cal.leeming at simplicitymedialtd.co.uk> wrote: > > Hi all, > > Just spent a little while comparing the performance of heavy INSERT/SELECT > performance against UUID fields in a MySQL database. > > Full performance breakdown and write up can be found here; > http://blog.simplicitymedialtd.co.uk/?p=437 > > So far, the best option seems to be storing the UUID as a BINARY(16) - > anyone have any thoughts on this? > > Thanks > > Cal > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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://lists.freeswitch.org/mailman/options/freeswitch-users> > http://www.freeswitch.org > > > ------------------------------ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > -- > Ken > *http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > *irc.freenode.net #freeswitch > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130218/28ad1bea/attachment-0001.html From cal.leeming at simplicitymedialtd.co.uk Mon Feb 18 16:36:15 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Mon, 18 Feb 2013 13:36:15 +0000 Subject: [Freeswitch-users] Friday Free For All is Go In-Reply-To: <7B264347-DB13-42CA-9BEB-3C3B8344A483@freeswitch.org> References: <795B4931-BD8C-4E6E-8254-73C4BC9257E3@endigotech.com> <7B264347-DB13-42CA-9BEB-3C3B8344A483@freeswitch.org> Message-ID: Anyone else thinking conference call reports on a BBS?! Slightly before my time, but it'd be kick ass awesome to see. Cal On Sat, Jan 19, 2013 at 3:56 AM, Brian West wrote: > Like OMG, thats so retro! > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH_Wire > T: +1.918.420.9266 | F: +1.918.420.9267 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9266 > UK: +44 20 3298 4900 > ISN: 410*543 > > > > > > On Jan 18, 2013, at 6:51 PM, Brian Foster wrote: > > Dial up as in modem lol > > Sent from my iPhone > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130218/1cc6f200/attachment.html From egable+freeswitch at gmail.com Mon Feb 18 17:36:31 2013 From: egable+freeswitch at gmail.com (Eliot Gable) Date: Mon, 18 Feb 2013 09:36:31 -0500 Subject: [Freeswitch-users] Multicast in freeswitch In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF23A9EB1@Mail-Kilo.squay.com> References: <592A9CF93E12394E8472A6CC66E66BF236B788@Mail-Kilo.squay.com> <592A9CF93E12394E8472A6CC66E66BF23790FC@Mail-Kilo.squay.com> <592A9CF93E12394E8472A6CC66E66BF2379123@Mail-Kilo.squay.com> <592A9CF93E12394E8472A6CC66E66BF23A9EB1@Mail-Kilo.squay.com> Message-ID: On Mon, Feb 18, 2013 at 4:17 AM, Archana Venugopan wrote: > Hi,**** > > LUA script is written by us. Ya its being lauched from dialplan. The > script establishes call to mobile and our deskphone. So far it dint update > the DB, it just checks the DB and establishes call. **** > > > This is probably what you want: http://wiki.freeswitch.org/wiki/Mod_lua#session:setHangupHook As for updating the database, if you have a PostgreSQL database, you can enable the core pgsql support and use native pgsql calls from lua to access the database. Also, keep in mind that FS keeps a channels table in the database and if you just want to read the state of sessions up in FS, if you use core ODBC or core PGSQL support, you can just read the channels table for the current call state information. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130218/ffdf7a19/attachment.html From julf at julf.com Mon Feb 18 18:43:51 2013 From: julf at julf.com (Johan Helsingius) Date: Mon, 18 Feb 2013 16:43:51 +0100 Subject: [Freeswitch-users] making group of phones ring from the command line Message-ID: <51224C37.4090902@julf.com> Another really stupid one... I want to have an alarm process make a group of phones ring. There doesn't need to be a real connection, as nobody will pick up the connection, but as an alternative it could play a standard recording if answered. I have the group defined as a group in the directory but also as a context and number leading to a group call in the dialplan. So what I need is a command to fs_cli that makes a call to the group or dialplan extension. Haven't managed to figure out how to do it... Julf From steveayre at gmail.com Mon Feb 18 19:18:21 2013 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 18 Feb 2013 16:18:21 +0000 Subject: [Freeswitch-users] making group of phones ring from the command line In-Reply-To: <51224C37.4090902@julf.com> References: <51224C37.4090902@julf.com> Message-ID: Use the 'originate' API over ESL (fs_cli connects via ESL). You can give it two parameters: 1) The dialstring to call (the dialstring for the group of phones) 2) Where to bridge the call to - &playback(filename) would play a recording For the dialstring you can use a forked dial. With a defined call group use the group_call API with the +A flag. See: http://wiki.freeswitch.org/wiki/Mod_commands#originate http://wiki.freeswitch.org/wiki/Mod_commands#group_call -Steve On 18 February 2013 15:43, Johan Helsingius wrote: > Another really stupid one... > > I want to have an alarm process make a group of phones ring. > There doesn't need to be a real connection, as nobody will > pick up the connection, but as an alternative it could play > a standard recording if answered. > > I have the group defined as a group in the directory but > also as a context and number leading to a group call in > the dialplan. > > So what I need is a command to fs_cli that makes a call to > the group or dialplan extension. Haven't managed to figure > out how to do it... > > Julf > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130218/52259447/attachment-0001.html From mail at dmatthews.org Mon Feb 18 19:26:12 2013 From: mail at dmatthews.org (David Matthews) Date: Mon, 18 Feb 2013 16:26:12 +0000 Subject: [Freeswitch-users] Reading incoming skype chat messges Message-ID: <20130218162612.GB13448@dmatthews.org> I have a freeswitch install with the skypopen module, mostly working without problems. I do not see incoming chat messages on port 8021 though and that's puzzling for a variety of reasons. I've done events plain MESSAGE and I see outgoing skype messages and also both incoming and outgoing sip messages. Most puzzling is that I'm sure at one point I *did* see these incoming skypes. I am new to freeswitch and have changed configs a fair bit, I know what I've changed although not necessarily when. The only change in event_socket.conf.xml is to the password. I do see messages coming in on the cli for instance __________________________________________________________________________ 2013-02-18 15:01:47.148356 [DEBUG] skypopen_protocol.c:207 [33378fa|14c4f66] [DEBUG_SKYPE 207 ][skype101 ][IDLE,IDLE] READING: |||CHAT #wendy.hahn2/$davcmat;cc92a695ea7a38c2 ACTIVITY_TIMESTAMP 1361199703||| 2013-02-18 15:01:47.148356 [DEBUG] skypopen_protocol.c:207 [33378fa|14c4f66] [DEBUG_SKYPE 207 ][skype101 ][IDLE,IDLE] READING: |||CHATMESSAGE 966 STATUS READ||| _________________________________________________________________________ and if I do events plain ALL I think these are the events (below) that correspond to that cli log, but I nowhere see the actual content of the message. If this is me borking something as I guess it must be, any idea where that might be? _________________________________________________________________________ Event-Name: MESSAGE_QUERY Core-UUID: 1c08551c-7831-11e2-a903-cd8324092615 FreeSWITCH-Hostname: hermes.default.davcmat.uk0.bigv.io FreeSWITCH-Switchname: hermes.default.davcmat.uk0.bigv.io FreeSWITCH-IPv4: 213.138.100.41 FreeSWITCH-IPv6: 2001%3A41c8%3A51%3A29%3Afeff%3Aff%3Afe00%3Aa4c Event-Date-Local: 2013-02-18%2015%3A01%3A50 Event-Date-GMT: Mon,%2018%20Feb%202013%2015%3A01%3A50%20GMT Event-Date-Timestamp: 1361199710428367 Event-Calling-File: sofia_reg.c Event-Calling-Function: sofia_reg_handle_register Event-Calling-Line-Number: 1746 Event-Sequence: 1006992 Message-Account: sip%3A666%40hermes.dmatthews.org VM-Sofia-Profile: internal VM-Call-ID: PSF7PQoA808mC3FiUBuCRuUUUc.3q7Pl Content-Length: 720 Content-Type: text/event-plain Event-Name: MESSAGE_WAITING Core-UUID: 1c08551c-7831-11e2-a903-cd8324092615 Content-Length: 883 Content-Type: text/event-plain FreeSWITCH-Hostname: hermes.default.davcmat.uk0.bigv.io FreeSWITCH-Switchname: hermes.default.davcmat.uk0.bigv.io FreeSWITCH-IPv4: 213.138.100.41 FreeSWITCH-IPv6: 2001%3A41c8%3A51%3A29%3Afeff%3Aff%3Afe00%3Aa4c Event-Date-Local: 2013-02-18%2015%3A01%3A50 Event-Date-GMT: Mon,%2018%20Feb%202013%2015%3A01%3A50%20GMT Event-Date-Timestamp: 1361199710488338 Event-Calling-File: mod_voicemail.c Event-Calling-Function: actual_message_query_handler Event-Calling-Line-Number: 3857 Event-Sequence: 1006994 MWI-Messages-Waiting: no MWI-Message-Account: sip%3A666%40hermes.dmatthews.org Sofia-Profile: internal Call-ID: PSF7PQoA808mC3FiUBuCRuUUUc.3q7Pl -- David Matthews mail at dmatthews.org From ec.amazumdar at tatapowersed.com Mon Feb 18 09:59:10 2013 From: ec.amazumdar at tatapowersed.com (Anindo Mazumdar) Date: Mon, 18 Feb 2013 12:29:10 +0530 Subject: [Freeswitch-users] Integration of FreeSWITCH and Open IMS Message-ID: <201302180659.r1I6xAkX030223@blr.tatapowersed.com> Hello all, I am working on FreeSWITCH and Open IMS server. Just wanted to know whether their is a way by which I can integrate both. I have both of them installed as a standalone application on my machine. -- With Regards, Anindo Mazumdar From imran.moinuddin at nexdegree.com Mon Feb 18 10:57:12 2013 From: imran.moinuddin at nexdegree.com (Imran Moinuddin) Date: Mon, 18 Feb 2013 12:57:12 +0500 Subject: [Freeswitch-users] Audio Issue: Digitized background beep when speaking Message-ID: Hi everyone, Recently we've been experiencing some unusual audio issues on our SIP gateway (running Freeswitch) where people on a conference are sometimes hearing periodic / digitized 'beeps' or 'clicks' during the call (mainly when someone's speaking). You can listen to a recorded clip from the call at: https://www.dropbox.com/s/aya3j8xkxtz0wx6/AudioSampleBeep.mp3 The caller was coming in via a landline through one of our carriers. Our FS server wasn't under any particular strain for resources at the time nor does it seem too widespread (it happens pretty non-deterministic from what I can tell) but thought I'd share in case anyone has heard anything similar in the past and if there are insights worth sharing on this. Any feedback would be awesome, Imran -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130218/8d1a2d4a/attachment-0001.html From manuel at aguete.org Mon Feb 18 14:09:52 2013 From: manuel at aguete.org (=?ISO-8859-1?Q?Manuel_Dur=E1n_Aguete?=) Date: Mon, 18 Feb 2013 12:09:52 +0100 Subject: [Freeswitch-users] Audio dropped using unimrcp tts. In-Reply-To: References: Message-ID: Hello, I've update the dropbox folder with the pcap and mod_unimrcp_rx/tx files, added the log file too. https://www.dropbox.com/sh/tcgnktzzby0h2ns/v7ViiUyObB I've tested with branch 1.2 and with master, the pcap is from master code. Thank you for your help. Regards. 2013/2/15 Christopher Rienzo > Send PCAP with the RTSP signaling and RTP audio. I'm curious when the > server starts streaming the audio. > > > > On Fri, Feb 15, 2013 at 4:20 AM, Manuel Dur?n Aguete wrote: > >> Hello, >> >> I've tried with different values for sleep, but the problem persists, in >> http://pastebin.freeswitch.org/20594 i've posted the call log. >> I've enabled MOD_UNIMRCP_DEBUG_AUDIO_QUEUE in mod_unircp.cm the audio >> streams generated can be downloaded from >> https://www.dropbox.com/sh/tcgnktzzby0h2ns/v7ViiUyObB. >> >> I've changed the speak data to >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> and only "3 4 5 6 7 8 9 0" is played in softphone. >> >> Thank you for your help. >> >> Regards. >> >> >> >> >> >> 2013/2/14 Michael Collins >> >>> Try adding a sleep in your dialplan prior to the call to "speak" >>> application. Start with 1000 ms and see if it makes any difference, then >>> tweak the sleep time from there. >>> >>> -MC >>> >>> On Thu, Feb 14, 2013 at 5:26 AM, Manuel Dur?n Aguete >> > wrote: >>> >>>> Hello, >>>> >>>> I'm a newbie testing freeswitch tts with loquendo mrcp1 server, >>>> >>>> mrcp_profile: >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> Dialplan code: >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> I can hear all text with first word dropped. I've captured the rtp >>>> traffic and the audio between FS <-> Loquendo contains the word. After >>>> apply the fixed recommended in >>>> http://wiki.freeswitch.org/wiki/RTP_Issues#Dropped_Audio the issue >>>> persists. >>>> >>>> Any idea, suggestion, etc ? >>>> >>>> Thank you. >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>>> >>>> >>> >>> >>> -- >>> Michael S Collins >>> Twitter: @mercutioviz >>> http://www.FreeSWITCH.org >>> http://www.ClueCon.com >>> http://www.OSTAG.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130218/c13f8879/attachment-0001.html From kaothics at hotmail.com Mon Feb 18 19:19:35 2013 From: kaothics at hotmail.com (PolyMorphiK) Date: Mon, 18 Feb 2013 08:19:35 -0800 (PST) Subject: [Freeswitch-users] troubles with Mod_gsmopen and Huawei E153 Message-ID: <1361204375768-7587585.post@n2.nabble.com> Hello, Actually i've compiled the last Freeswitch build and mod_gsmopen on win7 with Visual Studio Ultimate without big troubles. ( i only had to replace some lib files into the right folders in order to compile correctly mod_gsmopen ) i followed each steps described on Wiki. i already did 8 months ago nd it worked fine with the dongle Huawei E153 but today there is nothing to do it seems the port isnt open from freeswitch log... Before to wonder if the dongle is correctly installed , unlocked or if the voice capability is activate i must say i did it correctly and tried with an Indian Dashboard with Call plugin to call out nd receive calls nd it works like a charm but still using mod_gsmopen there is nothing to do i even tried different port in case it was interverted or something like that but nothing. I hope someone can help me to fix that. here's the simple config : here's the log : 013-02-18 16:00:28.478618 [DEBUG] mod_gsmopen.cpp:1201 rev 1.3.13b[00000000|37 ][DEBUG_GSMOPEN 1201 ][none ][-1,-1,-1] Windows CODEPAGE Input =850 2013-02-18 16:00:28.478618 [DEBUG] mod_gsmopen.cpp:1203 rev 1.3.13b[00000000|37 ][DEBUG_GSMOPEN 1203 ][none ][-1,-1,-1] Windows CODEPAGE Input =65001 2013-02-18 16:00:28.478618 [DEBUG] mod_gsmopen.cpp:1204 rev 1.3.13b[00000000|37 ][DEBUG_GSMOPEN 1204 ][none ][-1,-1,-1] Windows CODEPAGE Output =850 2013-02-18 16:00:28.478618 [DEBUG] mod_gsmopen.cpp:1206 rev 1.3.13b[00000000|37 ][DEBUG_GSMOPEN 1206 ][none ][-1,-1,-1] Windows CODEPAGE Output =65001 2013-02-18 16:00:28.478618 [NOTICE] mod_gsmopen.cpp:1209 rev 1.3.13b[00000000|37 ][NOTICA 1209 ][none ][-1,-1,-1] GSMOPEN Charset Output Test 0 ????? 2013-02-18 16:00:28.478618 [NOTICE] mod_gsmopen.cpp:1210 rev 1.3.13b[00000000|37 ][NOTICA 1210 ][none ][-1,-1,-1] GSMOPEN Charset Output Test 1 ???^? 2013-02-18 16:00:28.478618 [NOTICE] mod_gsmopen.cpp:1211 rev 1.3.13b[00000000|37 ][NOTICA 1211 ][none ][-1,-1,-1] GSMOPEN Charset Output Test 2 ??????? 2013-02-18 16:00:28.478618 [NOTICE] mod_gsmopen.cpp:1212 rev 1.3.13b[00000000|37 ][NOTICA 1212 ][none ][-1,-1,-1] GSMOPEN Charset Output Test 3 ?????? 2013-02-18 16:00:28.478618 [NOTICE] mod_gsmopen.cpp:1213 rev 1.3.13b[00000000|37 ][NOTICA 1213 ][none ][-1,-1,-1] GSMOPEN Charset Output Test 4 ??? 2013-02-18 16:00:28.478618 [NOTICE] mod_gsmopen.cpp:1214 rev 1.3.13b[00000000|37 ][NOTICA 1214 ][none ][-1,-1,-1] GSMOPEN Charset Output Test 5 ?? 2013-02-18 16:00:28.478618 [NOTICE] mod_gsmopen.cpp:1215 rev 1.3.13b[00000000|37 ][NOTICA 1215 ][none ][-1,-1,-1] GSMOPEN Charset Output Test 6 ?? 2013-02-18 16:00:28.478618 [DEBUG] mod_gsmopen.cpp:1228 rev 1.3.13b[00000000|37 ][DEBUG_GSMOPEN 1228 ][none ][-1,-1,-1] Default globals.dialplan=XML 2013-02-18 16:00:28.478618 [DEBUG] mod_gsmopen.cpp:1230 rev 1.3.13b[00000000|37 ][DEBUG_GSMOPEN 1230 ][none ][-1,-1,-1] Default globals.destination=5000 2013-02-18 16:00:28.478618 [DEBUG] mod_gsmopen.cpp:1232 rev 1.3.13b[00000000|37 ][DEBUG_GSMOPEN 1232 ][none ][-1,-1,-1] Default globals.context=default 2013-02-18 16:00:28.478618 [DEBUG] mod_gsmopen.cpp:1240 rev 1.3.13b[00000000|37 ][DEBUG_GSMOPEN 1240 ][none ][-1,-1,-1] globals.debug=0 2013-02-18 16:00:28.478618 [DEBUG] mod_gsmopen.cpp:1242 rev 1.3.13b[00000000|37 ][DEBUG_GSMOPEN 1242 ][none ][-1,-1,-1] globals.debug=8 2013-02-18 16:00:28.478618 [DEBUG] mod_gsmopen.cpp:1248 rev 1.3.13b[00000000|37 ][DEBUG_GSMOPEN 1248 ][none ][-1,-1,-1] globals.dialplan=XML 2013-02-18 16:00:28.478618 [DEBUG] mod_gsmopen.cpp:1254 rev 1.3.13b[00000000|37 ][DEBUG_GSMOPEN 1254 ][none ][-1,-1,-1] globals.context=default 2013-02-18 16:00:28.478618 [DEBUG] mod_gsmopen.cpp:1245 rev 1.3.13b[00000000|37 ][DEBUG_GSMOPEN 1245 ][none ][-1,-1,-1] globals.hold_music= 2013-02-18 16:00:28.478618 [DEBUG] mod_gsmopen.cpp:1251 rev 1.3.13b[00000000|37 ][DEBUG_GSMOPEN 1251 ][none ][-1,-1,-1] globals.destination=1000 2013-02-18 16:00:28.478618 [WARNING] mod_gsmopen.cpp:1667 rev 1.3.13b[00000000|37 ][WARNINGA 1667 ][gsm1 ][-1, 0, 0] STARTING interface_id=1 2013-02-18 16:00:28.478618 [DEBUG] mod_gsmopen.cpp:1668 rev 1.3.13b[00000000|37 ][DEBUG_GSMOPEN 1668 ][gsm1 ][-1, 0, 0] id=1 2013-02-18 16:00:28.478618 [DEBUG] mod_gsmopen.cpp:1669 rev 1.3.13b[00000000|37 ][DEBUG_GSMOPEN 1669 ][gsm1 ][-1, 0, 0] name=gsm1 2013-02-18 16:00:28.478618 [DEBUG] mod_gsmopen.cpp:1670 rev 1.3.13b[00000000|37 ][DEBUG_GSMOPEN 1670 ][gsm1 ][-1, 0, 0] hold-music= 2013-02-18 16:00:28.478618 [DEBUG] mod_gsmopen.cpp:1671 rev 1.3.13b[00000000|37 ][DEBUG_GSMOPEN 1671 ][gsm1 ][-1, 0, 0] context=default 2013-02-18 16:00:28.478618 [DEBUG] mod_gsmopen.cpp:1672 rev 1.3.13b[00000000|37 ][DEBUG_GSMOPEN 1672 ][gsm1 ][-1, 0, 0] dialplan=XML 2013-02-18 16:00:28.478618 [DEBUG] mod_gsmopen.cpp:1673 rev 1.3.13b[00000000|37 ][DEBUG_GSMOPEN 1673 ][gsm1 ][-1, 0, 0] destination=1000 2013-02-18 16:00:28.478618 [DEBUG] mod_gsmopen.cpp:1674 rev 1.3.13b[00000000|37 ][DEBUG_GSMOPEN 1674 ][gsm1 ][-1, 0, 0] controldevice_name=COM11 2013-02-18 16:00:28.478618 [DEBUG] mod_gsmopen.cpp:1675 rev 1.3.13b[00000000|37 ][DEBUG_GSMOPEN 1675 ][gsm1 ][-1, 0, 0] controldevice_audio_name=COM10 2013-02-18 16:00:28.478618 [DEBUG] mod_gsmopen.cpp:1677 rev 1.3.13b[00000000|37 ][DEBUG_GSMOPEN 1677 ][gsm1 ][-1, 0, 0] gsmopen_serial_sync_period=300 2013-02-18 16:00:28.478618 [ERR] gsmopen_protocol.cpp:122 rev 1.3.13b[00000000|37 ][ERRORA 122 ][gsm1 ][-1, 0, 0] port COM11, NOT open 2013-02-18 16:00:28.478618 [ERR] mod_gsmopen.cpp:1684 rev 1.3.13b[00000000|37 ][ERRORA 1684 ][gsm1 ][-1, 0, 0] gsmopen_serial_init failed 2013-02-18 16:00:28.478618 [ERR] mod_gsmopen.cpp:1685 rev 1.3.13b[00000000|37 ][ERRORA 1685 ][gsm1 ][-1, 0, 0] STARTING interface_id=1 FAILED 2013-02-18 16:00:28.478618 [ERR] mod_gsmopen.cpp:2942 rev 1.3.13b[00000000|37 ][ERRORA 2942 ][gsm1 ][-1, 0, 0] ALARM on interface gsm1: 2013-02-18 16:00:28.478618 [NOTICE] switch_loadable_module.c:146 Adding Endpoint 'gsmopen' 2013-02-18 16:00:28.478618 [NOTICE] switch_loadable_module.c:298 Adding API Function 'gsm' 2013-02-18 16:00:28.478618 [NOTICE] switch_loadable_module.c:298 Adding API Function 'gsmopen' 2013-02-18 16:00:28.478618 [NOTICE] switch_loadable_module.c:298 Adding API Function 'gsmopen_boost_audio' 2013-02-18 16:00:28.478618 [NOTICE] switch_loadable_module.c:298 Adding API Function 'gsmopen_dump' 2013-02-18 16:00:28.478618 [NOTICE] switch_loadable_module.c:298 Adding API Function 'gsmopen_sendsms' 2013-02-18 16:00:28.478618 [NOTICE] switch_loadable_module.c:405 Adding Chat interface 'sms' Thank You Best Regards -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/troubles-with-Mod-gsmopen-and-Huawei-E153-tp7587585.html Sent from the freeswitch-users mailing list archive at Nabble.com. From julf at julf.com Mon Feb 18 19:39:06 2013 From: julf at julf.com (Johan Helsingius) Date: Mon, 18 Feb 2013 17:39:06 +0100 Subject: [Freeswitch-users] making group of phones ring from the command line In-Reply-To: References: <51224C37.4090902@julf.com> Message-ID: <5122592A.5060604@julf.com> > You can give it two parameters: > 1) The dialstring to call (the dialstring for the group of phones) > 2) Where to bridge the call to - &playback(filename) would play a recording Thanks, I guess my problem is partially that I can't figure out the dialstring to reach either the group directly, or the extension in the dialplan. Julf From steveayre at gmail.com Mon Feb 18 19:46:49 2013 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 18 Feb 2013 16:46:49 +0000 Subject: [Freeswitch-users] making group of phones ring from the command line In-Reply-To: <5122592A.5060604@julf.com> References: <51224C37.4090902@julf.com> <5122592A.5060604@julf.com> Message-ID: Try something along the lines of originate ${group_call(2000 at domain.com+A)} &playback(alarm.wav) The group_call call returns/expands to the dialstring for the group for you. The second parameter can either be the dialplan extension name, or &application(args). -Steve On 18 February 2013 16:39, Johan Helsingius wrote: > > You can give it two parameters: > > 1) The dialstring to call (the dialstring for the group of phones) > > 2) Where to bridge the call to - &playback(filename) would play a > recording > > Thanks, > > I guess my problem is partially that I can't figure out the dialstring > to reach either the group directly, or the extension in the dialplan. > > Julf > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130218/2adf42b5/attachment.html From julf at julf.com Mon Feb 18 20:06:59 2013 From: julf at julf.com (Johan Helsingius) Date: Mon, 18 Feb 2013 18:06:59 +0100 Subject: [Freeswitch-users] making group of phones ring from the command line In-Reply-To: References: <51224C37.4090902@julf.com> <5122592A.5060604@julf.com> Message-ID: <51225FB3.9060304@julf.com> > Try something along the lines of > > originate ${group_call(2000 at domain.com +A)} > &playback(alarm.wav) Unfortunately all permutations of the group call just give me ERR CHAN_NOT_IMPLEMENTED from fs_cli. Julf From a.venugopan at mundio.com Mon Feb 18 20:10:32 2013 From: a.venugopan at mundio.com (Archana Venugopan) Date: Mon, 18 Feb 2013 17:10:32 +0000 Subject: [Freeswitch-users] Multicast in freeswitch In-Reply-To: References: <592A9CF93E12394E8472A6CC66E66BF236B788@Mail-Kilo.squay.com> <592A9CF93E12394E8472A6CC66E66BF23790FC@Mail-Kilo.squay.com> <592A9CF93E12394E8472A6CC66E66BF2379123@Mail-Kilo.squay.com> <592A9CF93E12394E8472A6CC66E66BF23A9EB1@Mail-Kilo.squay.com> Message-ID: <592A9CF93E12394E8472A6CC66E66BF23AC0DC@Mail-Kilo.squay.com> Hi, We use MSSQL and in our channels table currently its not update channe-state status. Though in my logs I could see the below messages 2013-02-18 17:04:02.767161 [DEBUG] sofia.c:6282 Channel sofia/internal/100 at fsfailover.uk01.com entering state [terminated][487] 2013-02-18 17:04:02.767161 [DEBUG] switch_channel.c:2950 (sofia/internal/100 at fsfailover.uk01.com) Callstate Change EARLY -> HANGUP Not sure where these states are being saved. I need to invoke my lua script during HANGUP case currently lua script works till call gets established, after that I don't see its hitting any lua scripts written by us. Please suggest what can we do now? Many thanks. Regards, Archana From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Eliot Gable Sent: 18 February 2013 14:37 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Multicast in freeswitch On Mon, Feb 18, 2013 at 4:17 AM, Archana Venugopan > wrote: Hi, LUA script is written by us. Ya its being lauched from dialplan. The script establishes call to mobile and our deskphone. So far it dint update the DB, it just checks the DB and establishes call. This is probably what you want: http://wiki.freeswitch.org/wiki/Mod_lua#session:setHangupHook As for updating the database, if you have a PostgreSQL database, you can enable the core pgsql support and use native pgsql calls from lua to access the database. Also, keep in mind that FS keeps a channels table in the database and if you just want to read the state of sessions up in FS, if you use core ODBC or core PGSQL support, you can just read the channels table for the current call state information. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130218/ba1022e1/attachment-0001.html From steveayre at gmail.com Mon Feb 18 20:24:07 2013 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 18 Feb 2013 17:24:07 +0000 Subject: [Freeswitch-users] making group of phones ring from the command line In-Reply-To: <51225FB3.9060304@julf.com> References: <51224C37.4090902@julf.com> <5122592A.5060604@julf.com> <51225FB3.9060304@julf.com> Message-ID: I'm assuming you didn't include the mailto: markup that one of our email clients helpfully inserted... (probably my gmail). First check group_call is actually giving you a valid dialstring: group_call group at domain.com+A I'm thinking the problem is the ${} didn't get expanded. Try running it via the expand API (http://wiki.freeswitch.org/wiki/Mod_commands#expand): expand originate ${group_call(group at domain.com+A)} &playback(alarm.wav) -Steve On 18 February 2013 17:06, Johan Helsingius wrote: > > > Try something along the lines of > > > > originate ${group_call(2000 at domain.com +A)} > > &playback(alarm.wav) > > Unfortunately all permutations of the group call just give me > ERR CHAN_NOT_IMPLEMENTED from fs_cli. > > Julf > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 lndspereira-fs at yahoo.com Mon Feb 18 20:26:49 2013 From: lndspereira-fs at yahoo.com (lndspereira-fs at yahoo.com) Date: Mon, 18 Feb 2013 09:26:49 -0800 (PST) Subject: [Freeswitch-users] Creating a conference for an incoming call via socket interface Message-ID: <1361208409.82019.YahooMailNeo@web125802.mail.ne1.yahoo.com> Hey all. I configured the FS to answer any incoming call and I have an application that keeps listen to Freeswitch events. I would like know if there is a way to create a conference and add an incoming call to this conference using the FS socket interface. Thanks in advance for your help. Leo ? Leonardo Nogueira de S? Pereira Tel.: +55 19 3307-5589 Cel.: +55 19 9122-5943 Skype: leonardo_pereira_77 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130218/6b4f2bb0/attachment.html From julf at julf.com Mon Feb 18 20:43:23 2013 From: julf at julf.com (Johan Helsingius) Date: Mon, 18 Feb 2013 18:43:23 +0100 Subject: [Freeswitch-users] making group of phones ring from the command line In-Reply-To: References: <51224C37.4090902@julf.com> <5122592A.5060604@julf.com> <51225FB3.9060304@julf.com> Message-ID: <5122683B.8000209@julf.com> > expand originate ${group_call(group at domain.com+A)} &playback(alarm.wav) Ah! Yes, that did it! Thanks! Now I just need to figure out how to change the caller id on the group call to enable me to use a distinct ring tone... Julf From shahzad.bhatti at g-r-v.com Mon Feb 18 20:46:47 2013 From: shahzad.bhatti at g-r-v.com (Shahzad Bhatti) Date: Mon, 18 Feb 2013 22:46:47 +0500 Subject: [Freeswitch-users] Channel is not clear properly! Message-ID: Hi EveryBody! i am using Freeswitch Version 1.3.13b+git~20130218T142410Z~3d0c245f80 (git 3d0c245 2013-02-18 14:24:10Z) and using detect_speech using pocketsphinx and got an issue that my channels are not properly cleared as i noticed that the Port is Closed as i stop detect_speech in hanguphook . but when i show channels i got result as *freeswitch at internal> show channels* * uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,sent_callee_name,sent_callee_num * *ac2c751a-7a1b-11e2-b6c5-e5fd6d8cf5cf,outbound,2013-02-19 03:36:44,1361227004,sofia/internal/1000,CS_SOFT_EXECUTE,,321,,1000%192.168.0.100,detect_speech,pocketsphinx my_grammar my_grammar,,default,PCMU,8000,64000,PCMU,8000,64000,,localhost.localdomain, 1000 at 192.168.0.100,,ACTIVE,Outbound Call,1000%192.168.0.100,,ac2c751a-7a1b-11e2-b6c5-e5fd6d8cf5cf,,* i try to use hupall and uuid_kill both *freeswitch at internal> hupall* *+OK hangup all channels with cause MANAGER_REQUEST* * * *freeswitch at internal> uuid_kill ac2c751a-7a1b-11e2-b6c5-e5fd6d8cf5cf* *-ERR No such channel!* but after that i show channels again then output is *freeswitch at internal> show channels* * uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,sent_callee_name,sent_callee_num * *ac2c751a-7a1b-11e2-b6c5-e5fd6d8cf5cf,outbound,2013-02-19 03:36:44,1361227004,sofia/internal/1000,CS_SOFT_EXECUTE,,321,,1000%192.168.0.100,detect_speech,pocketsphinx my_grammar my_grammar,,default,PCMU,8000,64000,PCMU,8000,64000,,localhost.localdomain, 1000 at 192.168.0.100,,ACTIVE,Outbound Call,1000%192.168.0.100,,ac2c751a-7a1b-11e2-b6c5-e5fd6d8cf5cf,,* * * *1 total.* any positive response is highly appreciated Regards Shahzad Bhatti -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130218/6dae5527/attachment.html From brian at freeswitch.org Mon Feb 18 20:50:48 2013 From: brian at freeswitch.org (Brian West) Date: Mon, 18 Feb 2013 11:50:48 -0600 Subject: [Freeswitch-users] Friday Free For All is Go In-Reply-To: References: <795B4931-BD8C-4E6E-8254-73C4BC9257E3@endigotech.com> <7B264347-DB13-42CA-9BEB-3C3B8344A483@freeswitch.org> Message-ID: <2F429537-01AE-4E21-A051-E6283E93DA6F@freeswitch.org> So I take it you never surfed the interwebs via email? I recall downloading my QWK packet from a BBS so I could access internet email. :P I'm not that old either. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire T: +1.918.420.9266 | F: +1.918.420.9267 | M: +1.918.424.WEST iNUM: +883 5100 1420 9266 ISN: 410*543 On Feb 18, 2013, at 7:36 AM, Cal Leeming [Simplicity Media Ltd] wrote: > Anyone else thinking conference call reports on a BBS?! > > Slightly before my time, but it'd be kick ass awesome to see. > > Cal From msc at freeswitch.org Mon Feb 18 20:58:26 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 18 Feb 2013 09:58:26 -0800 Subject: [Freeswitch-users] Creating a conference for an incoming call via socket interface In-Reply-To: <1361208409.82019.YahooMailNeo@web125802.mail.ne1.yahoo.com> References: <1361208409.82019.YahooMailNeo@web125802.mail.ne1.yahoo.com> Message-ID: Yes there is! If you are listening for a particular event that tells you that you want to send a particular call to a conference then you just need to capture that call's uuid. From there you can either uuid_transfer that call to an extension in your XML dialplan that handles the conference. (examples of these are in the example configs.) Let us know how it goes. -MC On Mon, Feb 18, 2013 at 9:26 AM, lndspereira-fs at yahoo.com < lndspereira-fs at yahoo.com> wrote: > Hey all. > I configured the FS to answer any incoming call and I have an application > that keeps listen to Freeswitch events. > I would like know if there is a way to create a conference and add an > incoming call to this conference using the FS socket interface. > > Thanks in advance for your help. > > Leo > > > > Leonardo Nogueira de S? Pereira > Tel.: +55 19 3307-5589 > Cel.: +55 19 9122-5943 > Skype: leonardo_pereira_77 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130218/2b6b2fdb/attachment.html From bdfoster at endigotech.com Mon Feb 18 20:58:46 2013 From: bdfoster at endigotech.com (Brian Foster) Date: Mon, 18 Feb 2013 12:58:46 -0500 Subject: [Freeswitch-users] Friday Free For All is Go In-Reply-To: <2F429537-01AE-4E21-A051-E6283E93DA6F@freeswitch.org> References: <795B4931-BD8C-4E6E-8254-73C4BC9257E3@endigotech.com> <7B264347-DB13-42CA-9BEB-3C3B8344A483@freeswitch.org> <2F429537-01AE-4E21-A051-E6283E93DA6F@freeswitch.org> Message-ID: > :P I'm not that old either. I hate to break it to ya... Screw it, someone else can do it lol > > > > > > On Feb 18, 2013, at 7:36 AM, Cal Leeming [Simplicity Media Ltd] wrote: > >> Anyone else thinking conference call reports on a BBS?! >> >> Slightly before my time, but it'd be kick ass awesome to see. >> >> Cal > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 cal.leeming at simplicitymedialtd.co.uk Mon Feb 18 21:03:04 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Mon, 18 Feb 2013 18:03:04 +0000 Subject: [Freeswitch-users] Friday Free For All is Go In-Reply-To: <2F429537-01AE-4E21-A051-E6283E93DA6F@freeswitch.org> References: <795B4931-BD8C-4E6E-8254-73C4BC9257E3@endigotech.com> <7B264347-DB13-42CA-9BEB-3C3B8344A483@freeswitch.org> <2F429537-01AE-4E21-A051-E6283E93DA6F@freeswitch.org> Message-ID: Oh man, I actually had to google QWK - I feel ashamed! I came in around '99, fortunate to have a 56k modem, rose through the ranks of being a 'packet kiddie' on Dalnet (lol!) building Cisco scanners in mIRC script - those were the days haha Cal On Mon, Feb 18, 2013 at 5:50 PM, Brian West wrote: > So I take it you never surfed the interwebs via email? I recall > downloading my QWK packet from a BBS so I could access internet email. :P > I'm not that old either. > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH_Wire > T: +1.918.420.9266 | F: +1.918.420.9267 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9266 > ISN: 410*543 > > > > > > On Feb 18, 2013, at 7:36 AM, Cal Leeming [Simplicity Media Ltd] < > cal.leeming at simplicitymedialtd.co.uk> wrote: > > > Anyone else thinking conference call reports on a BBS?! > > > > Slightly before my time, but it'd be kick ass awesome to see. > > > > Cal > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130218/6a25b6ec/attachment.html From mario_fs at mgtech.com Mon Feb 18 21:05:47 2013 From: mario_fs at mgtech.com (Mario G) Date: Mon, 18 Feb 2013 10:05:47 -0800 Subject: [Freeswitch-users] [Solved] adding In-Reply-To in sip header not working In-Reply-To: <1769A24D-BC79-4CFD-AA88-0A1E81E346BA@mgtech.com> References: <0F4F8E19-73FC-460F-A5AB-B7A2CF8CD78B@mgtech.com> <1769A24D-BC79-4CFD-AA88-0A1E81E346BA@mgtech.com> Message-ID: <3B6503C1-BA0A-46BD-B48F-DEA3F2AD5FBB@mgtech.com> Finally! Sharing with anyone else wanting to pass the calling party ID from an external SIP account into FS then back to the standard PSTN on some ITSPs (Callcentric in this case), this is what works: Anytime before the bridge: (notice it was close to Richards idea but uses _h_ instead of _rh_) In my bridge below remove the effective caller id number like this: On Feb 4, 2013, at 12:44 PM, Mario G wrote: > Yes, in and out are the same provider, although in/out may be different accounts, could that be the ticket? > Mario G > > On Feb 4, 2013, at 10:08 AM, Richard Brady wrote: > >> That Call-ID looks fine. Just to be clear, you are sending this back to the same provider it came from right? >> >> You'll need to post a trace (anonymised) in order for me to help further. >> >> Richard >> >> On 29 January 2013 19:31, Mario G wrote: >> Thanks, I tried both suggestions but no love. When I used sip_h_In-Reply-To=${sip_call_id} used as below, the trace showed all normal but the cell phone does not ring at all. When I removed everything the cell rang but the original number was not passed. BTW, the sip_call_id was translated to (#s altered): >> sip_h_In-Reply-To=3912345-9123456295-612341 at msw1.telengy.net, could that be an issue with ATT not liking it? >> >> Mario G >> >> >> On Jan 27, 2013, at 3:02 PM, Richard Brady wrote: >> >>> Ok, nifty. They are letting you present a number you do not own as Caller ID on an outbound call if that outbound call is a forwarded leg of an inbound call. >>> >>> They do this by looking the In-Reply-To header of the INVITE for the forwarded leg, which should contain the Call-ID of the orignal leg. >>> >>> So you need to copy the Call-ID in order to authorize the Caller ID. >>> >>> A couple things: >>> >>> 1. From the docs: effective_caller_id_name Sets the effective callerid name. This is automatically exported to the B-leg; however, it is not valid in an origination string. In other words, set this before calling bridge, otherwise use origination_caller_id_name >>> >>> 2. You shouldn't care about 1 above as it should be copied across from the A leg by default and you are not modifying it, so remove effective_caller_id_name and don't bother with origination_caller_id_name either. >>> >>> 3. You should use sip_h_ not sip_rh_ because you want the header in the new INVITE going out. >>> >>> Perhaps try: >>> >>> >>> >>> Hope this helps. >>> >>> Richard >>> >>> >>> On 21 January 2013 19:54, Mario G wrote: >>> Thanks, apparently I had it wrong, the doc below states that the PBX must support it incoming, they pointed me to using effective_caller_id which I added to the bridge but it still does not work. Would love to fix this since the cell phones currently have no idea who is calling. >>> Mario G >>> >>> >>> >>> Please note that this feature is ONLY AVAILABLE for customers using a SIP PBX that either supports (or allows the configuring of) the "in-reply-to" header (defined by RFC 3261) for incoming calls which are forwarded to an outbound trunk. In these instances Callcentric will "Pass-Through" the CallerID from the original call which was received to the outbound bridged/forwarded call. >>> >>> On Jan 19, 2013, at 5:13 PM, Richard Brady wrote: >>> >>>> On 20 January 2013 00:06, Mario G wrote: >>>> I never did this so I must be missing something, I tried both below but the bridge then fails. Can anyone shed some light on what I am doing wrong. My ITSP now supports in-reply-to so I can pass the caller ID to a forwarded call from FS. >>>> >>>> In-Reply-To should contain a Call-ID not a caller ID. They are very different. >>>> >>>> The following would make a bit more sense, but still not a lot: >>>> >>>> >>>> >>>> Using In-Reply-To in a response doesn't seem right to me. I would expect it to appear in an INVITE. For for example, you get a missed call and you call the person back, then the INVITE for the callback would have a new Call-ID but the original Call-ID in the In-Reply-To header. That said, I have no idea what your ITSPs intended use for the header is. >>>> >>>> Richard -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130218/7df29b82/attachment.html From steveayre at gmail.com Mon Feb 18 21:06:16 2013 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 18 Feb 2013 18:06:16 +0000 Subject: [Freeswitch-users] making group of phones ring from the command line In-Reply-To: <5122683B.8000209@julf.com> References: <51224C37.4090902@julf.com> <5122592A.5060604@julf.com> <51225FB3.9060304@julf.com> <5122683B.8000209@julf.com> Message-ID: Have a look over the originate documentation again - there's optional additional parameters for CallerID. Whether that actually gets presented to the phone is going to depend on the route you're using to dial the phones though. For example if you're going via a SIP-PSTN provider then they'll be restricting the Caller IDs you can send (to stop spoofing numbers that aren't your own). -Steve On 18 February 2013 17:43, Johan Helsingius wrote: >> expand originate ${group_call(group at domain.com+A)} &playback(alarm.wav) > > Ah! Yes, that did it! Thanks! > > Now I just need to figure out how to change the caller id on the group call > to enable me to use a distinct ring tone... > > Julf > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 msc at freeswitch.org Mon Feb 18 21:09:27 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 18 Feb 2013 10:09:27 -0800 Subject: [Freeswitch-users] making group of phones ring from the command line In-Reply-To: <5122683B.8000209@julf.com> References: <51224C37.4090902@julf.com> <5122592A.5060604@julf.com> <51225FB3.9060304@julf.com> <5122683B.8000209@julf.com> Message-ID: expand originate {origination_caller_id_number=1234}${group_call( group at domain.com+A)} &playback(alarm.wav) -MC On Mon, Feb 18, 2013 at 9:43 AM, Johan Helsingius wrote: > > expand originate ${group_call(group at domain.com+A)} &playback(alarm.wav) > > Ah! Yes, that did it! Thanks! > > Now I just need to figure out how to change the caller id on the group call > to enable me to use a distinct ring tone... > > Julf > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130218/389edaba/attachment.html From mario_fs at mgtech.com Mon Feb 18 21:19:41 2013 From: mario_fs at mgtech.com (Mario G) Date: Mon, 18 Feb 2013 10:19:41 -0800 Subject: [Freeswitch-users] Memory growth In-Reply-To: References: <512209DF.6050806@vts24.ru> Message-ID: FYI/in case it's related: I had a substantial memory leak on the Oct 25 though Jan 29 Head versions and 1.2.3. After Feb 11 head, now a much smaller leak I can't pinpoint, see http://jira.freeswitch.org/browse/FS-5095#comment-39379. Updating to HEAD will probably fix it, I hope someone else sees the slow leak since I am getting nowhere real fast on it..... Mario G On Feb 18, 2013, at 3:16 AM, Steven Ayre wrote: > Are you able to reproduce it on a newer version? Try current head of the master branch. > > If you can, then try running FS through valgrind which may reveal the source of any memory leaks. > > -Steve > > > > On 18 February 2013 11:00, ??????? ??????? wrote: > Hi, list! > I've installed FreeSWITCH Version 1.2.3+git~20120920T220849Z~f718a5e8e6 > (1.2.3; git at commit f718a5e8e6 on Thu, 20 Sep 2012 22:08:49 Z) > > I noticed that FreeSwtich uses a lot of memory of my system, and percent > of used memory is gradually growing regardless of the number of the > simultaneous calls: > > # uname -a > Linux freeswitch1 2.6.18-308.13.1.el5 #1 SMP Tue Aug 21 17:10:18 EDT > 2012 x86_64 x86_64 x86_64 GNU/Linux > > # top > top - 14:21:39 up 140 days, 29 min, 2 users, load average: 0.11, 0.14, > 0.10 > Tasks: 99 total, 1 running, 96 sleeping, 2 stopped, 0 zombie > Cpu(s): 1.2%us, 0.8%sy, 0.0%ni, 96.9%id, 0.1%wa, 0.2%hi, 0.7%si, > 0.0%st > Mem: 3781680k total, 3748360k used, 33320k free, 120708k buffers > Swap: 2096472k total, 17224k used, 2079248k free, 1838356k cached > > PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND > 9584 root -2 -10 1792m 1.3g 8252 S 2.7 36.8 935:07.85 freeswitch > 3095 mysql 15 0 329m 28m 4588 S 0.7 0.8 713:07.03 mysqld > > At this moment FS handles no more 15 simultaneous calls, and has about > 117 registered users. > Is there a tool to monitor, which of processes or modules in FS is using > memory and how much? > > -- > Best regards, > Vitaly. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130218/8a93623f/attachment.html From msc at freeswitch.org Mon Feb 18 21:19:49 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 18 Feb 2013 10:19:49 -0800 Subject: [Freeswitch-users] Audio Issue: Digitized background beep when speaking In-Reply-To: References: Message-ID: It would help if you could isolate whether or not the audio is coming from a particular call leg. I would use pcapsipdump for this since it creates separate files for each SIP call on the system. Also, if you are using FreeTDM then you'll also need to use the ftdm trace command to capture the audio from the TDM card. Once you know where the rogue audio is originating then you can start looking for a solution. -MC On Sun, Feb 17, 2013 at 11:57 PM, Imran Moinuddin < imran.moinuddin at nexdegree.com> wrote: > Hi everyone, > > Recently we've been experiencing some unusual audio issues on our SIP > gateway (running Freeswitch) where people on a conference are sometimes > hearing periodic / digitized 'beeps' or 'clicks' during the call (mainly > when someone's speaking). You can listen to a recorded clip from the call > at: > > https://www.dropbox.com/s/aya3j8xkxtz0wx6/AudioSampleBeep.mp3 > > The caller was coming in via a landline through one of our carriers. > > Our FS server wasn't under any particular strain for resources at the time > nor does it seem too widespread (it happens pretty non-deterministic from > what I can tell) but thought I'd share in case anyone has heard anything > similar in the past and if there are insights worth sharing on this. > > Any feedback would be awesome, > > Imran > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130218/dd99c814/attachment.html From msc at freeswitch.org Mon Feb 18 21:29:29 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 18 Feb 2013 10:29:29 -0800 Subject: [Freeswitch-users] Blind transfer dropped calls In-Reply-To: <1436977.bWcj1llolB@virtex> References: <1436977.bWcj1llolB@virtex> Message-ID: I suspect that you'll need to turn off G.729 and any other codecs you don't really need. It looks like PCMU and PCMA are at the end of the list of codecs offered by the phone. If all the phones are configured the same way and if you are doing calls to/from the PSTN then there's really no reason to have all those other codecs enabled. I would have each phone set to use only PCMA (since you're in Europe) and make sure that in conf/vars.xml you have PCMA enabled in the global_codec_prefs. Those two things together will probably clear up your troubles. -MC On Mon, Feb 18, 2013 at 3:35 AM, g wrote: > Hi everyone. I'm not absolute begineer with freeswitch, but pretty new > after > long experience with asterisk. > I'm facing now with an installation (1.3.13b+git~20130213) of FS with an > issue > I can't solve (the same issue was present in 1.0head previous > installation), > so I invoke your help > The scenario is simple: Grandstream (updated firmware) Ip phones connected > to > FS, gateway to SIP VoIP provider via trunk. > Everything works great. Internal phones can talk each oters, calls can > came in > and go out to PSTN without problem, but ... > When a phone call came and an internal phone picks it up, it is impossible > to > blind tranfert that call to onother internal phone. > I mean that the phone i try to transfert the call rings, then it picks up > the > call and the call is dropped. > I suspect is a codec problem, but I can't figure why. All the phones are > configured the same way. > Here below the log af the process, from external call incoming until the > drop. > I can see two suspicious steps, where a "no read codec" error and an > "incopatible_destination" message appears. > I think I made a trivial error, but I really can't understand where. > Can please anyone point me in the right direction to solve it? > Many thanks > g > > ____ call log taken from the cli - some data is ### masked ### ___ > > 2013-01-25 17:15:04.770166 [DEBUG] switch_core_session.c:874 Send signal > sofia/sipinterface_1/201 at 192.168.35.80 [BREAK] > 2013-01-25 17:15:04.770166 [DEBUG] switch_core_session.c:874 Send signal > sofia/sipinterface_1/201 at 192.168.35.80 [BREAK] > 2013-01-25 17:15:04.770166 [DEBUG] sofia.c:5370 Channel > sofia/sipinterface_1/201 at 192.168.35.80 entering state [received][100] > 2013-01-25 17:15:04.770166 [DEBUG] sofia.c:5381 Remote SDP: > v=0 > o=201 8000 8004 IN IP4 192.168.35.114 > s=SIP Call > c=IN IP4 192.168.35.114 > t=0 0 > m=audio 5086 RTP/AVP 18 0 8 2 97 9 3 101 > a=rtpmap:18 G729/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:2 G726-32/8000 > a=rtpmap:97 iLBC/8000 > a=fmtp:97 mode=20 > a=rtpmap:9 G722/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-11 > a=sendonly > a=ptime:20 > > 2013-01-25 17:15:04.770166 [DEBUG] sofia.c:5810 Passing SDP to other leg. > v=0 > o=201 8000 8004 IN IP4 192.168.35.114 > s=SIP Call > c=IN IP4 192.168.35.114 > t=0 0 > m=audio 5086 RTP/AVP 18 0 8 2 97 9 3 101 > a=rtpmap:18 G729/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:2 G726-32/8000 > a=rtpmap:97 iLBC/8000 > a=fmtp:97 mode=20 > a=rtpmap:9 G722/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-11 > a=sendonly > a=ptime:20 > > 2013-01-25 17:15:04.770166 [DEBUG] switch_channel.c:1554 > (sofia/sipinterface_1/201 at 192.168.35.80) Callstate Change ACTIVE -> HELD > 2013-01-25 17:15:04.770166 [DEBUG] switch_core_session.c:790 Send signal > sofia/sipinterface_3/347******** ### masked ### [BREAK] > 2013-01-25 17:15:04.770166 [DEBUG] mod_sofia.c:1812 > sofia/sipinterface_3/347******** ### masked ### Sending media re-direct: > v=0 > o=201 8000 8004 IN IP4 192.168.35.114 > s=SIP Call > c=IN IP4 192.168.35.114 > t=0 0 > m=audio 5086 RTP/AVP 18 0 8 2 97 9 3 101 > a=rtpmap:18 G729/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:2 G726-32/8000 > a=rtpmap:97 iLBC/8000 > a=fmtp:97 mode=20 > a=rtpmap:9 G722/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-11 > a=sendonly > a=ptime:20 > > 2013-01-25 17:15:04.770166 [DEBUG] switch_core_session.c:728 Send signal > sofia/sipinterface_3/347******** ### masked ### [BREAK] > 2013-01-25 17:15:04.770166 [DEBUG] switch_core_session.c:874 Send signal > sofia/sipinterface_3/347******** ### masked ### [BREAK] > 2013-01-25 17:15:04.770166 [DEBUG] sofia.c:5370 Channel > sofia/sipinterface_3/347******** ### masked ### entering state [calling][0] > 2013-01-25 17:15:04.810163 [DEBUG] switch_core_session.c:874 Send signal > sofia/sipinterface_3/347******** ### masked ### [BREAK] > 2013-01-25 17:15:04.810163 [DEBUG] switch_core_session.c:874 Send signal > sofia/sipinterface_3/347******** ### masked ### [BREAK] > 2013-01-25 17:15:04.830180 [DEBUG] sofia.c:4953 Passing 200 OK to other leg > 2013-01-25 17:15:04.830180 [DEBUG] switch_core_session.c:790 Send signal > sofia/sipinterface_1/201 at 192.168.35.80 [BREAK] > 2013-01-25 17:15:04.830180 [DEBUG] sofia.c:5370 Channel > sofia/sipinterface_3/347******** ### masked ### entering state [completing] > [200] > 2013-01-25 17:15:04.830180 [DEBUG] sofia.c:5381 Remote SDP: > v=0 > o=root 1558328831 1558328833 IN IP4 xxx.xxx.xxx.xxx > s=VOIP ### provider masked ### > c=IN IP4 xxx.xxx.xxx.xxx #### masked ### > t=0 0 > m=audio 46518 RTP/AVP 18 3 97 8 0 101 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:3 GSM/8000 > a=rtpmap:97 iLBC/8000 > a=fmtp:97 mode=30 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=recvonly > a=ptime:20 > > 2013-01-25 17:15:04.830180 [DEBUG] mod_sofia.c:2363 Responding with 200 > [OK] > 2013-01-25 17:15:04.830180 [DEBUG] switch_core_session.c:728 Send signal > sofia/sipinterface_1/201 at 192.168.35.80 [BREAK] > 2013-01-25 17:15:04.830180 [DEBUG] switch_core_session.c:874 Send signal > sofia/sipinterface_3/347******** ### masked ### [BREAK] > 2013-01-25 17:15:04.830180 [DEBUG] switch_core_session.c:874 Send signal > sofia/sipinterface_3/347******** ### masked ### [BREAK] > 2013-01-25 17:15:04.830180 [DEBUG] sofia.c:5370 Channel > sofia/sipinterface_3/347******** ### masked ### entering state [ready][200] > 2013-01-25 17:15:04.830180 [DEBUG] switch_core_session.c:790 Send signal > sofia/sipinterface_1/201 at 192.168.35.80 [BREAK] > 2013-01-25 17:15:04.830180 [DEBUG] switch_core_session.c:874 Send signal > sofia/sipinterface_1/201 at 192.168.35.80 [BREAK] > 2013-01-25 17:15:04.830180 [DEBUG] sofia.c:5370 Channel > sofia/sipinterface_1/201 at 192.168.35.80 entering state [completed][200] > 2013-01-25 17:15:04.910172 [DEBUG] switch_core_session.c:874 Send signal > sofia/sipinterface_1/201 at 192.168.35.80 [BREAK] > 2013-01-25 17:15:04.910172 [DEBUG] switch_core_session.c:874 Send signal > sofia/sipinterface_1/201 at 192.168.35.80 [BREAK] > 2013-01-25 17:15:04.910172 [DEBUG] switch_core_session.c:874 Send signal > sofia/sipinterface_1/201 at 192.168.35.80 [BREAK] > 2013-01-25 17:15:04.910172 [DEBUG] sofia.c:5370 Channel > sofia/sipinterface_1/201 at 192.168.35.80 entering state [ready][200] > 2013-01-25 17:15:09.270174 [DEBUG] switch_core_session.c:874 Send signal > sofia/sipinterface_1/201 at 192.168.35.80 [BREAK] > 2013-01-25 17:15:09.270174 [DEBUG] sofia.c:6356 Process REFER to > [203 at 192.168.35.80] > 2013-01-25 17:15:09.270174 [DEBUG] switch_channel.c:2842 > (sofia/sipinterface_1/201 at 192.168.35.80) Callstate Change HELD -> HANGUP > 2013-01-25 17:15:09.270174 [NOTICE] switch_ivr.c:1696 Hangup > sofia/sipinterface_1/201 at 192.168.35.80 [CS_HIBERNATE] [BLIND_TRANSFER] > 2013-01-25 17:15:09.270174 [DEBUG] switch_channel.c:2865 Send signal > sofia/sipinterface_1/201 at 192.168.35.80 [KILL] > 2013-01-25 17:15:09.270174 [DEBUG] switch_ivr_bridge.c:929 > sofia/sipinterface_1/201 at 192.168.35.80 skip receive message [UNBRIDGE] > (channel is hungup already) > 2013-01-25 17:15:09.270174 [DEBUG] switch_core_session.c:1179 Send > signal sofia/sipinterface_1/201 at 192.168.35.80 [BREAK] > 2013-01-25 17:15:09.270174 [DEBUG] switch_ivr.c:1705 > (sofia/sipinterface_3/347******** ### masked ###) State Change > CS_HIBERNATE -> > CS_ROUTING > 2013-01-25 17:15:09.270174 [DEBUG] switch_core_session.c:728 Send signal > sofia/sipinterface_3/347******** ### masked ### [BREAK] > 2013-01-25 17:15:09.270174 [DEBUG] switch_core_session.c:1179 Send > signal sofia/sipinterface_3/347******** ### masked ### [BREAK] > 2013-01-25 17:15:09.270174 [DEBUG] switch_core_session.c:728 Send signal > sofia/sipinterface_3/347******** ### masked ### [BREAK] > 2013-01-25 17:15:09.270174 [NOTICE] switch_ivr.c:1711 Transfer > sofia/sipinterface_3/347******** ### masked ### to XML[203 at context_1] > 2013-01-25 17:15:09.270174 [DEBUG] switch_core_state_machine.c:362 > (sofia/sipinterface_1/201 at 192.168.35.80) Running State Change CS_HANGUP > 2013-01-25 17:15:09.270174 [DEBUG] switch_core_state_machine.c:362 > (sofia/sipinterface_3/347******** ### masked ###) Running State Change > CS_ROUTING > 2013-01-25 17:15:09.270174 [DEBUG] switch_channel.c:1880 > (sofia/sipinterface_3/347******** ### masked ###) Callstate Change ACTIVE > -> > RINGING > 2013-01-25 17:15:09.270174 [DEBUG] switch_core_state_machine.c:410 > (sofia/sipinterface_3/347******** ### masked ###) State ROUTING > 2013-01-25 17:15:09.270174 [DEBUG] mod_sofia.c:148 > sofia/sipinterface_3/347******** ### masked ### SOFIA ROUTING > 2013-01-25 17:15:09.270174 [DEBUG] switch_core_state_machine.c:104 > sofia/sipinterface_3/347******** ### masked ### Standard ROUTING > 2013-01-25 17:15:09.270174 [DEBUG] switch_core_state_machine.c:602 > (sofia/sipinterface_1/201 at 192.168.35.80) State HANGUP > 2013-01-25 17:15:09.270174 [DEBUG] mod_sofia.c:465 Channel > sofia/sipinterface_1/201 at 192.168.35.80 hanging up, cause: BLIND_TRANSFER > 2013-01-25 17:15:09.270174 [DEBUG] mod_sofia.c:509 Sending BYE to > sofia/sipinterface_1/201 at 192.168.35.80 > 2013-01-25 17:15:09.270174 [DEBUG] switch_core_state_machine.c:47 > sofia/sipinterface_1/201 at 192.168.35.80 Standard HANGUP, cause: > BLIND_TRANSFER > 2013-01-25 17:15:09.270174 [DEBUG] switch_core_state_machine.c:602 > (sofia/sipinterface_1/201 at 192.168.35.80) State HANGUP going to sleep > 2013-01-25 17:15:09.270174 [DEBUG] switch_core_state_machine.c:393 > (sofia/sipinterface_1/201 at 192.168.35.80) State Change CS_HANGUP -> > CS_REPORTING > 2013-01-25 17:15:09.270174 [DEBUG] switch_core_session.c:1179 Send > signal sofia/sipinterface_1/201 at 192.168.35.80 [BREAK] > 2013-01-25 17:15:09.270174 [DEBUG] switch_core_state_machine.c:362 > (sofia/sipinterface_1/201 at 192.168.35.80) Running State Change CS_REPORTING > 2013-01-25 17:15:09.270174 [DEBUG] switch_core_state_machine.c:662 > (sofia/sipinterface_1/201 at 192.168.35.80) State REPORTING > 2013-01-25 17:15:09.270174 [DEBUG] switch_nat.c:510 mapped public port > 29868 protocol UDP to localport 29868 > 2013-01-25 17:15:09.290165 [DEBUG] switch_nat.c:510 mapped public port > 29869 protocol UDP to localport 29869 > 2013-01-25 17:15:09.290165 [DEBUG] switch_core_session.c:728 Send signal > sofia/sipinterface_3/347******** ### masked ### [BREAK] > 2013-01-25 17:15:09.290165 [DEBUG] switch_core_session.c:874 Send signal > sofia/sipinterface_3/347******** ### masked ### [BREAK] > 2013-01-25 17:15:09.330130 [DEBUG] switch_core_session.c:874 Send signal > sofia/sipinterface_3/347******** ### masked ### [BREAK] > 2013-01-25 17:15:09.330130 [DEBUG] switch_core_session.c:874 Send signal > sofia/sipinterface_3/347******** ### masked ### [BREAK] > 2013-01-25 17:15:09.330130 [DEBUG] switch_core_session.c:874 Send signal > sofia/sipinterface_3/347******** ### masked ### [BREAK] > 2013-01-25 17:15:09.730204 [DEBUG] switch_core_state_machine.c:79 > sofia/sipinterface_1/201 at 192.168.35.80 Standard REPORTING, cause: > BLIND_TRANSFER > 2013-01-25 17:15:09.730204 [DEBUG] switch_core_state_machine.c:662 > (sofia/sipinterface_1/201 at 192.168.35.80) State REPORTING going to sleep > 2013-01-25 17:15:09.730204 [DEBUG] switch_core_state_machine.c:387 > (sofia/sipinterface_1/201 at 192.168.35.80) State Change CS_REPORTING -> > CS_DESTROY > 2013-01-25 17:15:09.730204 [DEBUG] switch_core_session.c:1179 Send > signal sofia/sipinterface_1/201 at 192.168.35.80 [BREAK] > 2013-01-25 17:15:09.730204 [DEBUG] switch_core_session.c:1379 Session 77 > (sofia/sipinterface_1/201 at 192.168.35.80) Locked, Waiting on external > entities > 2013-01-25 17:15:09.730204 [NOTICE] switch_core_session.c:1397 Session > 77 (sofia/sipinterface_1/201 at 192.168.35.80) Ended > 2013-01-25 17:15:09.730204 [NOTICE] switch_core_session.c:1399 Close > Channel sofia/sipinterface_1/201 at 192.168.35.80 [CS_DESTROY] > 2013-01-25 17:15:09.730204 [DEBUG] switch_core_state_machine.c:491 > (sofia/sipinterface_1/201 at 192.168.35.80) Callstate Change HANGUP -> DOWN > 2013-01-25 17:15:09.730204 [DEBUG] switch_core_state_machine.c:494 > (sofia/sipinterface_1/201 at 192.168.35.80) Running State Change CS_DESTROY > 2013-01-25 17:15:09.730204 [DEBUG] switch_core_state_machine.c:504 > (sofia/sipinterface_1/201 at 192.168.35.80) State DESTROY > 2013-01-25 17:15:09.730204 [DEBUG] mod_sofia.c:370 > sofia/sipinterface_1/201 at 192.168.35.80 SOFIA DESTROY > 2013-01-25 17:15:09.730204 [DEBUG] switch_core_state_machine.c:86 > sofia/sipinterface_1/201 at 192.168.35.80 Standard DESTROY > 2013-01-25 17:15:09.730204 [DEBUG] switch_core_state_machine.c:504 > (sofia/sipinterface_1/201 at 192.168.35.80) State DESTROY going to sleep > 2013-01-25 17:15:41.650174 [ERR] switch_core_io.c:131 > sofia/sipinterface_3/347******** ### masked ### has no read codec. > 2013-01-25 17:15:41.650174 [DEBUG] switch_channel.c:2842 > (sofia/sipinterface_3/347******** ### masked ###) Callstate Change RINGING > -> > HANGUP > 2013-01-25 17:15:41.650174 [NOTICE] switch_core_io.c:132 Hangup > sofia/sipinterface_3/347******** ### masked ### [CS_ROUTING] > [INCOMPATIBLE_DESTINATION] > 2013-01-25 17:15:41.650174 [DEBUG] switch_channel.c:2865 Send signal > sofia/sipinterface_3/347******** ### masked ### [KILL] > 2013-01-25 17:15:41.650174 [DEBUG] switch_core_session.c:1179 Send > signal sofia/sipinterface_3/347******** ### masked ### [BREAK] > 2013-01-25 17:15:41.650174 [INFO] mod_dialplan_xml.c:481 Processing > Gianbattista Gallus <0707966538>->203 in context context_1 > Dialplan: sofia/sipinterface_3/347******** ### masked ### parsing > [context_1->conditioning_callerid] continue=true > Dialplan: sofia/sipinterface_3/347******** ### masked ### Regex (FAIL) > [conditioning_callerid] ${internal_caller_id_number}() =~ /^.+$/ > break=on-false > Dialplan: sofia/sipinterface_3/347******** ### masked ### parsing > [context_1->postroute_global] continue=true > Dialplan: sofia/sipinterface_3/347******** ### masked ### Absolute > Condition > [postroute_global] > Dialplan: sofia/sipinterface_3/347******** ### masked ### Action > hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) > Dialplan: sofia/sipinterface_3/347******** ### masked ### Action > hash(insert/${domain_name}- > last_dial/${caller_id_number}/${destination_number}) > > Dialplan: sofia/sipinterface_3/347******** ### masked ### Action > hash(insert/${domain_name}-last_dial/global/${uuid}) > Dialplan: sofia/sipinterface_3/347******** ### masked ### Action > set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) > Dialplan: sofia/sipinterface_3/347******** ### masked ### parsing > [context_1->main_trunk_1_pattern_5] continue=true > Dialplan: sofia/sipinterface_3/347******** ### masked ### Regex (FAIL) > [main_trunk_1_pattern_5] destination_number(203) =~ /^0(\d+)$/ > break=on-false > Dialplan: sofia/sipinterface_3/347******** ### masked ### parsing > [context_1->main_number_2] continue=true > Dialplan: sofia/sipinterface_3/347******** ### masked ### Regex (FAIL) > [main_number_2] > destination_number(203) =~ /^999$/ break=on-false > Dialplan: sofia/sipinterface_3/347******** ### masked ### parsing > [context_1->main_number_4] continue=true > Dialplan: sofia/sipinterface_3/347******** ### masked ### Regex (FAIL) > [main_number_4] > destination_number(203) =~ /^201$/ break=on-false > Dialplan: sofia/sipinterface_3/347******** ### masked ### parsing > [context_1->main_number_5] continue=true > Dialplan: sofia/sipinterface_3/347******** ### masked ### Regex (FAIL) > [main_number_5] > destination_number(203) =~ /^202$/ break=on-false > Dialplan: sofia/sipinterface_3/347******** ### masked ### parsing > [context_1->main_number_6] continue=true > Dialplan: sofia/sipinterface_3/347******** ### masked ### Regex (PASS) > [main_number_6] > destination_number(203) =~ /^203$/ break=on-false > Dialplan: sofia/sipinterface_3/347******** ### masked ### Action > set(hangup_after_bridge=true) > Dialplan: sofia/sipinterface_3/347******** ### masked ### Action > set(continue_on_fail=true) > Dialplan: sofia/sipinterface_3/347******** ### masked ### Action > set(vm_auto_play=false) > Dialplan: sofia/sipinterface_3/347******** ### masked ### Action > set(call_timeout=20) > Dialplan: sofia/sipinterface_3/347******** ### masked ### Action > set(ringback=${us-ring}) > Dialplan: sofia/sipinterface_3/347******** ### masked ### Action > set(transfer_ringback=${us-ring}) > Dialplan: sofia/sipinterface_3/347******** ### masked ### Action > export(sip_callee_id_name=Vanna Cuccu) > Dialplan: sofia/sipinterface_3/347******** ### masked ### Action > export(sip_callee_id_number=203) > Dialplan: sofia/sipinterface_3/347******** ### masked ### Action > bridge(user/203 at 192.168.35.80) > Dialplan: sofia/sipinterface_3/347******** ### masked ### Action answer() > Dialplan: sofia/sipinterface_3/347******** ### masked ### Action > voicemail(default > voicemail_1 203) > Dialplan: sofia/sipinterface_3/347******** ### masked ### Action hangup() > Dialplan: sofia/sipinterface_3/347******** ### masked ### parsing > [context_1->main_number_7] continue=true > Dialplan: sofia/sipinterface_3/347******** ### masked ### Regex (FAIL) > [main_number_7] > destination_number(203) =~ /^204$/ break=on-false > Dialplan: sofia/sipinterface_3/347******** ### masked ### parsing > [context_1->main_number_8] continue=true > Dialplan: sofia/sipinterface_3/347******** ### masked ### Regex (FAIL) > [main_number_8] > destination_number(203) =~ /^200$/ break=on-false > Dialplan: sofia/sipinterface_3/347******** ### masked ### parsing > [context_1- > >catchall] > continue=false > Dialplan: sofia/sipinterface_3/347******** ### masked ### Regex (PASS) > [catchall] > destination_number(203) =~ /^(\d+)$/ break=on-false > Dialplan: sofia/sipinterface_3/347******** ### masked ### Action > playback(misc/invalid_extension.wav) > 2013-01-25 17:15:41.670347 [INFO] switch_channel.c:2704 > sofia/sipinterface_3/347******** ### masked ### Flipping CID from > "Gianbattista > Gallus" > <0707966538> to "Outbound Call" <347******** ### masked ###> > 2013-01-25 17:15:41.670347 [DEBUG] switch_core_state_machine.c:410 > (sofia/sipinterface_3/347******** ### masked ###) State ROUTING going to > sleep > 2013-01-25 17:15:41.670347 [DEBUG] switch_core_state_machine.c:362 > (sofia/sipinterface_3/347******** ### masked ###) Running State Change > CS_HANGUP > 2013-01-25 17:15:41.670347 [DEBUG] switch_core_state_machine.c:602 > (sofia/sipinterface_3/347******** ### masked ###) State HANGUP > 2013-01-25 17:15:41.670347 [DEBUG] mod_sofia.c:465 Channel > sofia/sipinterface_3/347******** ### masked ### hanging up, cause: > INCOMPATIBLE_DESTINATION > 2013-01-25 17:15:41.670347 [DEBUG] mod_sofia.c:509 Sending BYE to > sofia/sipinterface_3/347******** ### masked ### > 2013-01-25 17:15:41.670347 [DEBUG] switch_core_state_machine.c:47 > sofia/sipinterface_3/347******** ### masked ### Standard HANGUP, cause: > INCOMPATIBLE_DESTINATION > 2013-01-25 17:15:41.670347 [DEBUG] switch_core_state_machine.c:602 > (sofia/sipinterface_3/347******** ### masked ###) State HANGUP going to > sleep > 2013-01-25 17:15:41.670347 [DEBUG] switch_core_state_machine.c:393 > (sofia/sipinterface_3/347******** ### masked ###) State Change CS_HANGUP -> > CS_REPORTING > 2013-01-25 17:15:41.670347 [DEBUG] switch_core_session.c:1179 Send > signal sofia/sipinterface_3/347******** ### masked ### [BREAK] > 2013-01-25 17:15:41.670347 [DEBUG] switch_core_state_machine.c:362 > (sofia/sipinterface_3/347******** ### masked ###) Running State Change > CS_REPORTING > 2013-01-25 17:15:41.670347 [DEBUG] switch_core_state_machine.c:662 > (sofia/sipinterface_3/347******** ### masked ###) State REPORTING > 2013-01-25 17:15:42.090203 [DEBUG] switch_core_state_machine.c:79 > sofia/sipinterface_3/347******** ### masked ### Standard REPORTING, cause: > INCOMPATIBLE_DESTINATION > 2013-01-25 17:15:42.090203 [DEBUG] switch_core_state_machine.c:662 > (sofia/sipinterface_3/347******** ### masked ###) State REPORTING going to > sleep > 2013-01-25 17:15:42.090203 [DEBUG] switch_core_state_machine.c:387 > (sofia/sipinterface_3/347******** ### masked ###) State Change > CS_REPORTING -> > CS_DESTROY > 2013-01-25 17:15:42.090203 [DEBUG] switch_core_session.c:1179 Send > signal sofia/sipinterface_3/347******** ### masked ### [BREAK] > 2013-01-25 17:15:42.090203 [DEBUG] switch_core_session.c:1379 Session 78 > (sofia/sipinterface_3/347******** ### masked ###) Locked, Waiting on > external > entities > 2013-01-25 17:15:42.090203 [NOTICE] switch_core_session.c:1397 Session > 78 (sofia/sipinterface_3/347******** ### masked ###) Ended > 2013-01-25 17:15:42.090203 [NOTICE] switch_core_session.c:1399 Close > Channel sofia/sipinterface_3/347******** ### masked ### [CS_DESTROY] > 2013-01-25 17:15:42.090203 [DEBUG] switch_core_state_machine.c:491 > (sofia/sipinterface_3/347******** ### masked ###) Callstate Change HANGUP > -> > DOWN > 2013-01-25 17:15:42.090203 [DEBUG] switch_core_state_machine.c:494 > (sofia/sipinterface_3/347******** ### masked ###) Running State Change > CS_DESTROY > 2013-01-25 17:15:42.090203 [DEBUG] switch_core_state_machine.c:504 > (sofia/sipinterface_3/347******** ### masked ###) State DESTROY > 2013-01-25 17:15:42.090203 [DEBUG] mod_sofia.c:370 > sofia/sipinterface_3/347******** ### masked ### SOFIA DESTROY > 2013-01-25 17:15:42.110145 [DEBUG] switch_nat.c:570 unmapped public port > 29868 protocol UDP to localport 29868 > 2013-01-25 17:15:42.130212 [DEBUG] switch_nat.c:570 unmapped public port > 29869 protocol UDP to localport 29869 > 2013-01-25 17:15:42.130212 [DEBUG] switch_core_state_machine.c:86 > sofia/sipinterface_3/347******** ### masked ### Standard DESTROY > 2013-01-25 17:15:42.130212 [DEBUG] switch_core_state_machine.c:504 > (sofia/sipinterface_3/347******** ### masked ###) State DESTROY going to > sleep > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130218/3054ac3c/attachment-0001.html From julf at julf.com Mon Feb 18 21:30:08 2013 From: julf at julf.com (Johan Helsingius) Date: Mon, 18 Feb 2013 19:30:08 +0100 Subject: [Freeswitch-users] making group of phones ring from the command line In-Reply-To: References: <51224C37.4090902@julf.com> <5122592A.5060604@julf.com> <51225FB3.9060304@julf.com> <5122683B.8000209@julf.com> Message-ID: <51227330.6060301@julf.com> > Have a look over the originate documentation again - there's optional > additional parameters for CallerID. Thanks again! > Whether that actually gets presented to the phone is going to depend > on the route you're using to dial the phones though. For example if > you're going via a SIP-PSTN provider then they'll be restricting the > Caller IDs you can send (to stop spoofing numbers that aren't your > own). In this case no prob, as all the phones I want to have ringing are locally registered to my fs. No point in having the phone ring in the US when somebody is at our door in Amsterdam... :) Julf From msc at freeswitch.org Mon Feb 18 21:31:19 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 18 Feb 2013 10:31:19 -0800 Subject: [Freeswitch-users] FW: In-band and out band call issues In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF23ABF7E@Mail-Kilo.squay.com> References: <592A9CF93E12394E8472A6CC66E66BF23ABF7E@Mail-Kilo.squay.com> Message-ID: We'll need to see SIP traces and console debugs of calls where these issues are occurring. -MC On Mon, Feb 18, 2013 at 2:24 AM, Archana Venugopan wrote: > Can anyone please help me on this peculiar issue. **** > > ** ** > > Regards,**** > > Archana**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Archana > Venugopan > *Sent:* 15 February 2013 16:15 > *To:* FreeSWITCH Users Help > *Subject:* [Freeswitch-users] In-band and out band call issues**** > > ** ** > > Hi,**** > > ** ** > > Am facing with a typical issue. In my phone settings, if i enable ?Force > RFC2833 Out-of-Band DTMF? then my voicemail is not working properly( If I > record any greeting or VM, only part of the message is being recorded) but > offnet calls for banks or shops is working. If ?Force RFC2833 Out-of-Band > DTMF? is disabled my voicemail is working properly but When I call offnet > numbers like Banks or shops, then my key press is not being recognised i.e > no DTMF detection.**** > > ** ** > > My internal .xml and external settings for DTMF is **** > > ** ** > > **** > > **** > > **** > > ** ** > > Please let me know what needs to be changed so that both voicemail and > offnet numbers work properly.**** > > ** ** > > Regards,**** > > Archana**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130218/7ccac9d7/attachment.html From julf at julf.com Mon Feb 18 21:34:31 2013 From: julf at julf.com (Johan Helsingius) Date: Mon, 18 Feb 2013 19:34:31 +0100 Subject: [Freeswitch-users] making group of phones ring from the command line In-Reply-To: References: <51224C37.4090902@julf.com> <5122592A.5060604@julf.com> <51225FB3.9060304@julf.com> <5122683B.8000209@julf.com> Message-ID: <51227437.9020806@julf.com> > expand originate > {origination_caller_id_number=1234}${group_call(group at domain.com > +A)} &playback(alarm.wav) Great! Thanks! Julf From msc at freeswitch.org Mon Feb 18 21:34:23 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 18 Feb 2013 10:34:23 -0800 Subject: [Freeswitch-users] Problems with sip provider In-Reply-To: <5121CEBC.2020707@julf.com> References: <51210FA4.1010200@julf.com> <51211478.1080108@julf.com> <5121CEBC.2020707@julf.com> Message-ID: Ask them for a list of devices that they support. Once you have that we can probably help you figure out how to fool them into thinking that they are talking to one of those "supported" devices. In cases where the provider is so totally foolish like this you have two choices: * Use their foolishness against them and make them think your FreeSWITCH server is really something else * Switch providers I know that's probably not what you want to hear but it is indeed the truth. -MC On Sun, Feb 17, 2013 at 10:48 PM, Johan Helsingius wrote: > > You don't need them to tell you how to configure your device to talk to > them (a > > FS specific question), you want them to tell you why the authentication > failed > > (specific to their end). Once we know that it should be possible to know > what > > needs to be adjusted. > > Indeed. But the problem is that I can't get past their (useless) first > level helpdesk to talk to the people who might actually have access to > their logs. > > Julf > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130218/963d6dbc/attachment.html From msc at freeswitch.org Mon Feb 18 21:38:03 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 18 Feb 2013 10:38:03 -0800 Subject: [Freeswitch-users] Changing codec during calls In-Reply-To: <6EF8B8E9-4CC5-480C-B1C1-09CBC533470C@kavun.ch> References: <6EF8B8E9-4CC5-480C-B1C1-09CBC533470C@kavun.ch> Message-ID: What happened with the uuid_media_reneg command? -MC On Sat, Feb 16, 2013 at 3:10 PM, Emrah wrote: > Hi all, > > How do we go about renegotiating the codecs of a call that is already > established? > I tried uuid_media_reneg to no avail. > E.g.: Upgrade someone who's entered a 16k conference from PCMU to G722. > > Cheers and thanks, > Emrah > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130218/82986a7a/attachment.html From msc at freeswitch.org Mon Feb 18 21:44:27 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 18 Feb 2013 10:44:27 -0800 Subject: [Freeswitch-users] NAT settings In-Reply-To: References: <5CEABE87-BD3C-4E34-9C64-8D26E2130F89@gmail.com> Message-ID: NDLB-force-rport is explicitly for Polycom phones where the FreeSWITCH server is on a public IP address and the phone is behind NAT. (i.e. your scenario.) Polycoms have a known bug/limitation in that they don't support rport. (Polycom refuses to acknowledge this as an issue and has been ignoring please from the public for nearly 10 years. I doubt it will ever change.) You can set that value to "true" and it will force rport for every phone that attempts to register. Alternatively you can set it to "safe" and it will only force the rport for known broken phones (read: Polycoms) and it won't mess with other phones. So yes, in FreeSWITCH the NAT "just works" - the issue is that the devices FreeSWITCH needs to talk to do not "just work" and in fact there are many cases where those phones "just suck" at NAT. All these knobs and switches inside FreeSWITCH are meant to help these clueless devices "not get left behind." -MC On Sun, Feb 17, 2013 at 6:52 AM, Cal Leeming [Simplicity Media Ltd] < cal.leeming at simplicitymedialtd.co.uk> wrote: > Although I can't answer your question from direct know, here are some > relevant links; > > NDLB-force-rport (taken from > http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#NDLB-force-rport) > This will force FreeSWITCH to send SIP responses to the network port from > which they were received. Use at your own risk! For more information see > NAT Traversal. > > safe = param that does force-rport behavior only on endpoints we know are > safe to do so on. This is a dirty hack to try to work with certain > endpoints behind sonicwall which does not use the same port when it does > nat, when the devices do not support rport, while not breaking devices that > acutally use different ports that force-rport will break > > Also found these; > > http://freeswitch-users.2379917.n2.nabble.com/NDLB-force-rport-safe-td5911932.html > > http://lists.freeswitch.org/pipermail/freeswitch-users/2011-November/077600.html > > http://lists.freeswitch.org/pipermail/freeswitch-users/2012-July/085658.html > > The last update on this was Brian West was ( > http://lists.freeswitch.org/pipermail/freeswitch-users/2008-September/034336.html) > > Try adding this param to your sofia profile. It will break cisco > phones or any other phone that follows the sip spec. This explicitly > breaks RFC to accommodate broken phones. > in your sofia profile. > > This information is somewhat out of date and there were some changes to > the way NAT works since then, so I'm not 100% sure. > > If anyone else can chime in to confirm (possibly even Brian West himself) > that'd be good. > > Hope this helps > > Cal > > On Sat, Feb 16, 2013 at 6:48 PM, George Cooper wrote: > >> Hi, >> >> I just started trying to work with freeswitch using the FS 1.06 book( >> although I am running FS ver 1.2.6+git.) , FS cookbook,, and wiki. >> >> I and have a question regarding NAT traversal. >> >> I have been stuck trying to get phones to register and be able to call >> each other using ext 1000 to 1002 for example. I couldn't get the phones to >> register until I read an archive regarding polycom NAT issues. >> >> The freeswitch server I'm connecting to is hosted on a remote public IP. >> I'm using a polycom 335, grandstream xpx 2020, and an x-lite client on a >> Local 192 network. >> >> The below settings seem to have fixed my registration issue. >> >> Is there a reason that I would NOT set the parameter settings in >> sip_profiles/internal.xml to true: >> >> >> >> >> >> >> and to add the variable "sip-force-contact" for each user in >> directory/default/1000.xml - 1019.xml ? >> >> >> >> >> >> >> >> >> > value="NDLB-connectile-dysfunction"/> >> >> >> >> >> >> > value="$${outbound_caller_name}"/> >> > value="$${outbound_caller_id}"/> >> >> >> >> >> >> Sorry if this has been gone over, but I'm not sure if there is something >> I'm missing because the wiki ( http://wiki.freeswitch.org/wiki/NAT ) says >> "NAT just works" and the internal.xml file says ""... >> >> Thanks for any clarity you can provide, >> >> Geo >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130218/8c98aff9/attachment-0001.html From lndspereira-fs at yahoo.com Mon Feb 18 21:53:54 2013 From: lndspereira-fs at yahoo.com (lndspereira-fs at yahoo.com) Date: Mon, 18 Feb 2013 10:53:54 -0800 (PST) Subject: [Freeswitch-users] Creating a conference for an incoming call via socket interface In-Reply-To: References: <1361208409.82019.YahooMailNeo@web125802.mail.ne1.yahoo.com> Message-ID: <1361213634.36660.YahooMailNeo@web125805.mail.ne1.yahoo.com> Ok, it should work, but in this case, I would need to have the "conference resources" pre-configured in Freeswitch. It does not solve my problem :( I would like to be able to create a conference resource from the application on the fly. I'm trying to take the same approach that I use with outbound calling conferences: - bgapi originate 4001234321 &conference(confName at default) - bgapi conference confName dial 4001234321 I'm looking for something like: - bgapi conference confName at default create UUID (if this is the first conference member) - bgapi conference confName at default add UUID (if conference already exists) Is there any API/tricky to help me on that? ? Leonardo Nogueira de S? Pereira Tel.: +55 19 3307-5589 Cel.: +55 19 9122-5943 Skype: leonardo_pereira_77 ________________________________ From: Michael Collins To: FreeSWITCH Users Help Sent: Monday, February 18, 2013 2:58 PM Subject: Re: [Freeswitch-users] Creating a conference for an incoming call via socket interface Yes there is! If you are listening for a particular event that tells you that you want to send a particular call to a conference then you just need to capture that call's uuid. From there you can either uuid_transfer that call to an extension in your XML dialplan that handles the conference. (examples of these are in the example configs.) Let us know how it goes. -MC On Mon, Feb 18, 2013 at 9:26 AM, lndspereira-fs at yahoo.com wrote: Hey all. >I configured the FS to answer any incoming call and I have an application that keeps listen to Freeswitch events. >I would like know if there is a way to create a conference and add an incoming call to this conference using the FS socket interface. > >Thanks in advance for your help. > >Leo > > > > >? >Leonardo Nogueira de S? Pereira >Tel.: +55 19 3307-5589 >Cel.: +55 19 9122-5943 >Skype: leonardo_pereira_77 > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130218/f0626586/attachment.html From julian.pawlowski at gmail.com Mon Feb 18 22:01:52 2013 From: julian.pawlowski at gmail.com (Julian Pawlowski) Date: Mon, 18 Feb 2013 20:01:52 +0100 Subject: [Freeswitch-users] German voice prompts Message-ID: Hi, a few days ago I started to create german voice prompts via Text-To-Speech using MARY TTS (http://mary.dfki.de/). As there is still a lot to do I just wrote a mailing to the german mailing list of GemeinschaftPBX where I would like to suggest german readers of this mailing list may want to have a look to: https://groups.google.com/forum/?fromgroups=#!topic/gs5-users/ZyNJdAFzYEg If somebody here is willing to contribute to the project, please feel free to send a pull-request to the Github repo: https://github.com/jpawlowski/freeswitch-sounds-de Cheers, Julian -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130218/89181a48/attachment.html From msc at freeswitch.org Mon Feb 18 22:08:56 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 18 Feb 2013 11:08:56 -0800 Subject: [Freeswitch-users] Creating a conference for an incoming call via socket interface In-Reply-To: <1361213634.36660.YahooMailNeo@web125805.mail.ne1.yahoo.com> References: <1361208409.82019.YahooMailNeo@web125802.mail.ne1.yahoo.com> <1361213634.36660.YahooMailNeo@web125805.mail.ne1.yahoo.com> Message-ID: You only have to have a conference profile preconfigured. Other than that you can do basically what you have outlined below, except with uuid_transfer: bgapi uuid_transfer conference:confName at default inline Try it out and remember that the standard disclaimerapplies, so tinker a bit before reporting back that it doesn't work. ;) -MC On Mon, Feb 18, 2013 at 10:53 AM, lndspereira-fs at yahoo.com < lndspereira-fs at yahoo.com> wrote: > Ok, it should work, but in this case, I would need to have the "conference > resources" pre-configured in Freeswitch. It does not solve my problem :( > > I would like to be able to create a conference resource from the > application on the fly. > I'm trying to take the same approach that I use with outbound calling > conferences: > - bgapi originate 4001234321 &conference(confName at default) > - bgapi conference confName dial 4001234321 > > I'm looking for something like: > - bgapi conference confName at default create UUID (if this is the first > conference member) > - bgapi conference confName at default add UUID (if conference already > exists) > > Is there any API/tricky to help me on that? > > > Leonardo Nogueira de S? Pereira > Tel.: +55 19 3307-5589 > Cel.: +55 19 9122-5943 > Skype: leonardo_pereira_77 > > ------------------------------ > *From:* Michael Collins > *To:* FreeSWITCH Users Help > *Sent:* Monday, February 18, 2013 2:58 PM > *Subject:* Re: [Freeswitch-users] Creating a conference for an incoming > call via socket interface > > Yes there is! If you are listening for a particular event that tells you > that you want to send a particular call to a conference then you just need > to capture that call's uuid. From there you can either uuid_transfer that > call to an extension in your XML dialplan that handles the conference. > (examples of these are in the example configs.) > > Let us know how it goes. > > -MC > > On Mon, Feb 18, 2013 at 9:26 AM, lndspereira-fs at yahoo.com < > lndspereira-fs at yahoo.com> wrote: > > Hey all. > I configured the FS to answer any incoming call and I have an application > that keeps listen to Freeswitch events. > I would like know if there is a way to create a conference and add an > incoming call to this conference using the FS socket interface. > > Thanks in advance for your help. > > Leo > > > > Leonardo Nogueira de S? Pereira > Tel.: +55 19 3307-5589 > Cel.: +55 19 9122-5943 > Skype: leonardo_pereira_77 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130218/7955ebf6/attachment-0001.html From krice at freeswitch.org Mon Feb 18 22:22:29 2013 From: krice at freeswitch.org (Ken Rice) Date: Mon, 18 Feb 2013 13:22:29 -0600 Subject: [Freeswitch-users] Memory growth In-Reply-To: Message-ID: We?re trying to figure out where this last bit of leaking is coming from... Looks like it might be related to Registrations some how... Once we squash that there will be some more info on the 1.2 branch coming On 2/18/13 12:19 PM, "Mario G" wrote: > FYI/in case it's related: I had a substantial memory leak on the Oct 25 though > Jan 29 Head versions and 1.2.3. After Feb 11 head, now a much smaller leak I > can't pinpoint, see http://jira.freeswitch.org/browse/FS-5095#comment-39379. > Updating to HEAD will probably fix it, I hope someone else sees the slow leak > since I am getting nowhere real fast on it..... > Mario G > > On Feb 18, 2013, at 3:16 AM, Steven Ayre wrote: > >> Are you able to reproduce it on a newer version? Try current head of the >> master branch. >> >> If you can, then try running FS through valgrind which may reveal the source >> of any memory leaks. >> >> -Steve >> >> >> >> On 18 February 2013 11:00, ??????? ??????? wrote: >>> Hi, list! >>> I've installed FreeSWITCH Version 1.2.3+git~20120920T220849Z~f718a5e8e6 >>> (1.2.3; git at commit f718a5e8e6 on Thu, 20 Sep 2012 22:08:49 Z) >>> >>> I noticed that FreeSwtich uses a lot of memory of my system, and percent >>> of used memory is gradually growing regardless of the number of the >>> simultaneous calls: >>> >>> # uname -a >>> Linux freeswitch1 2.6.18-308.13.1.el5 #1 SMP Tue Aug 21 17:10:18 EDT >>> 2012 x86_64 x86_64 x86_64 GNU/Linux >>> >>> # top >>> top - 14:21:39 up 140 days, 29 min, 2 users, load average: 0.11, 0.14, >>> 0.10 >>> Tasks: 99 total, 1 running, 96 sleeping, 2 stopped, 0 zombie >>> Cpu(s): 1.2%us, 0.8%sy, 0.0%ni, 96.9%id, 0.1%wa, 0.2%hi, 0.7%si, >>> 0.0%st >>> Mem: 3781680k total, 3748360k used, 33320k free, 120708k buffers >>> Swap: 2096472k total, 17224k used, 2079248k free, 1838356k cached >>> >>> PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND >>> 9584 root -2 -10 1792m 1.3g 8252 S 2.7 36.8 935:07.85 freeswitch >>> 3095 mysql 15 0 329m 28m 4588 S 0.7 0.8 713:07.03 mysqld >>> >>> At this moment FS handles no more 15 simultaneous calls, and has about >>> 117 registered users. >>> Is there a tool to monitor, which of processes or modules in FS is using >>> memory and how much? >>> >>> -- >>> Best regards, >>> Vitaly. >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130218/edc1aad8/attachment.html From cal.leeming at simplicitymedialtd.co.uk Mon Feb 18 22:24:02 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Mon, 18 Feb 2013 19:24:02 +0000 Subject: [Freeswitch-users] Problems with sip provider In-Reply-To: References: <51210FA4.1010200@julf.com> <51211478.1080108@julf.com> <5121CEBC.2020707@julf.com> Message-ID: +1. We suffered a similar problem with our first round of providers, it took us nearly 4 months to get them to fix it, and even after they fixed it, we found other problems. It's totally your choice if you want to continue trying to get it fixed, but most of the time, it's not worth the time/effort. My two cents at least. Cal On Mon, Feb 18, 2013 at 6:34 PM, Michael Collins wrote: > Ask them for a list of devices that they support. Once you have that we > can probably help you figure out how to fool them into thinking that they > are talking to one of those "supported" devices. In cases where the > provider is so totally foolish like this you have two choices: > * Use their foolishness against them and make them think your FreeSWITCH > server is really something else > * Switch providers > > I know that's probably not what you want to hear but it is indeed the > truth. > > -MC > > > On Sun, Feb 17, 2013 at 10:48 PM, Johan Helsingius wrote: > >> > You don't need them to tell you how to configure your device to talk to >> them (a >> > FS specific question), you want them to tell you why the authentication >> failed >> > (specific to their end). Once we know that it should be possible to >> know what >> > needs to be adjusted. >> >> Indeed. But the problem is that I can't get past their (useless) first >> level helpdesk to talk to the people who might actually have access to >> their logs. >> >> Julf >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130218/8e0c7b3d/attachment.html From marketing at cluecon.com Mon Feb 18 22:38:53 2013 From: marketing at cluecon.com (Michael Collins) Date: Mon, 18 Feb 2013 11:38:53 -0800 Subject: [Freeswitch-users] FreeSWITCH Weekly News and Notes Message-ID: Greetings all! Last week was rather interesting. Initially we had planned on doing a CudaTel <> demonstration on the weekly conference call. However, interest in Eliot Gable's mod_ha_clusterand the accompanying conversationwas particularly intense. That being the case, on last week's conference call we spent most of the time talking about HA in general and how we could build a FreeSWITCH HA system. We also invited everyone who is interested in the subject to call in to the public FreeSWITCH conference at 8PM EST on Tuesday evening (Feb 19) for the first HA conference call. (Eliot won't join until about 8:15PM.) If you have a vested interest in HA for FreeSWITCH then please join the conference call. For this week's conference callwe will ask one of the participants on the Tuesday night call to give us a brief overview of the HA discussion. After that I will be doing a demonstration of the CudaTel to show off what the FreeSWITCH team has been working so hard to develop these past few years. We hope you enjoy it! For the DIY crowd you may enjoy this site that Ken Rice set up. It's dedicated to doing cool things with FreeSWITCH and the Raspberry Pi . Check it out! In ClueCon 2013 news we are getting things all set. In the coming days we will be making announcements about new sponsors, registration and hotel information, and a call for speakers for this year's event. Don't forget to plan for a full 3 days, August 6th - 8th. Also, we tend to do a lot of fun stuff on the Monday before ClueCon, so it's a good idea to arrive on or before Monday August 7. Hope you have a great week! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130218/926a8ac5/attachment-0001.html From talk2ram at gmail.com Mon Feb 18 22:39:39 2013 From: talk2ram at gmail.com (ram) Date: Tue, 19 Feb 2013 01:09:39 +0530 Subject: [Freeswitch-users] mod_flite compile error Message-ID: Hi iam having bit old Centos on hosted Platform 2.6.18-348.1.1.el5PAE #1 SMP Tue Jan 22 17:06:35 EST 2013 i686 i686 i386 GNU/Linux iam trying to install Freeswitch GIT i get following error any advice making in main ... ../build/i386-linux-gnu/lib/libflite_cmu_us_rms.a(cmu_us_rms_cg.o):(. data.rel.ro+0x38): undefined reference to `cmu_us_rms_f0_carts' ../build/i386-linux-gnu/lib/libflite_cmu_us_slt.a(cmu_us_slt_cg.o):(. data.rel.ro+0x4c): undefined reference to `cmu_us_slt_spamf0_phrase_cart' collect2: ld returned 1 exit status make[8]: *** [../bin/flite] Error 1 make[7]: *** [build/i386-linux-gnu/obj//.make_build_dirs] Error 2 make[7]: Leaving directory `/usr/src/freeswitch/libs/flite-1.5.4-current' make[6]: *** [/usr/src/freeswitch/libs/flite-1.5.4-current/build/libs/libflite_cmu_us_awb.a] Error 2 make[6]: Leaving directory `/usr/src/freeswitch/src/mod/asr_tts/mod_flite' make[5]: *** [all] Error 1 make[5]: Leaving directory `/usr/src/freeswitch/src/mod/asr_tts/mod_flite' make[4]: *** [mod_flite-all] Error 1 make[4]: Leaving directory `/usr/src/freeswitch/src/mod' make[3]: *** [all-recursive] Error 1 make[3]: Leaving directory `/usr/src/freeswitch/src' make[2]: *** [all-recursive] Error 1 make[2]: Leaving directory `/usr/src/freeswitch' make[1]: *** [all] Error 2 make[1]: Leaving directory `/usr/src/freeswitch' make: *** [current] Error 2 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130219/6a00511c/attachment.html From msc at freeswitch.org Mon Feb 18 22:41:05 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 18 Feb 2013 11:41:05 -0800 Subject: [Freeswitch-users] German voice prompts In-Reply-To: References: Message-ID: Thanks for taking the lead on this. Once you get a set of sound files ready for production please let me know. I will get them house in our freeswitch-sounds git repo so that we can keep track of them and properly roll tarballs to put out on files.freeswitch.org. -MC On Mon, Feb 18, 2013 at 11:01 AM, Julian Pawlowski < julian.pawlowski at gmail.com> wrote: > Hi, > > a few days ago I started to create german voice prompts via Text-To-Speech > using MARY TTS (http://mary.dfki.de/). > > As there is still a lot to do I just wrote a mailing to the german mailing > list of GemeinschaftPBX where I would like to suggest german readers of > this mailing list may want to have a look to: > https://groups.google.com/forum/?fromgroups=#!topic/gs5-users/ZyNJdAFzYEg > > If somebody here is willing to contribute to the project, please feel free > to send a pull-request to the Github repo: > https://github.com/jpawlowski/freeswitch-sounds-de > > > Cheers, > Julian > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130218/e736adac/attachment.html From lndspereira-fs at yahoo.com Mon Feb 18 22:42:33 2013 From: lndspereira-fs at yahoo.com (lndspereira-fs at yahoo.com) Date: Mon, 18 Feb 2013 11:42:33 -0800 (PST) Subject: [Freeswitch-users] Creating a conference for an incoming call via socket interface In-Reply-To: References: <1361208409.82019.YahooMailNeo@web125802.mail.ne1.yahoo.com> <1361213634.36660.YahooMailNeo@web125805.mail.ne1.yahoo.com> Message-ID: <1361216553.49002.YahooMailNeo@web125803.mail.ne1.yahoo.com> Michael, thank you very much. It worked like a charm. Regards, Leo ? Leonardo Nogueira de S? Pereira Tel.: +55 19 3307-5589 Cel.: +55 19 9122-5943 Skype: leonardo_pereira_77 ________________________________ From: Michael Collins To: FreeSWITCH Users Help Sent: Monday, February 18, 2013 4:08 PM Subject: Re: [Freeswitch-users] Creating a conference for an incoming call via socket interface You only have to have a conference profile preconfigured. Other than that you can do basically what you have outlined below, except with uuid_transfer: bgapi uuid_transfer conference:confName at default inline Try it out and remember that the standard disclaimer applies, so tinker a bit before reporting back that it doesn't work. ;) -MC On Mon, Feb 18, 2013 at 10:53 AM, lndspereira-fs at yahoo.com wrote: Ok, it should work, but in this case, I would need to have the "conference resources" pre-configured in Freeswitch. It does not solve my problem :( > > > >I would like to be able to create a conference resource from the application on the fly. > >I'm trying to take the same approach that I use with outbound calling conferences: >- bgapi originate 4001234321 &conference(confName at default) >- bgapi conference confName dial 4001234321 > > > >I'm looking for something like: >- bgapi conference confName at default create UUID (if this is the first conference member) > >- bgapi conference confName at default add UUID (if conference already exists) >Is there any API/tricky to help me on that? > > > >? >Leonardo Nogueira de S? Pereira >Tel.: +55 19 3307-5589 >Cel.: +55 19 9122-5943 >Skype: leonardo_pereira_77 > > > > >________________________________ > From: Michael Collins >To: FreeSWITCH Users Help >Sent: Monday, February 18, 2013 2:58 PM >Subject: Re: [Freeswitch-users] Creating a conference for an incoming call via socket interface > > >Yes there is! If you are listening for a particular event that tells you that you want to send a particular call to a conference then you just need to capture that call's uuid. From there you can either uuid_transfer that call to an extension in your XML dialplan that handles the conference. (examples of these are in the example configs.) > >Let us know how it goes. > >-MC > > >On Mon, Feb 18, 2013 at 9:26 AM, lndspereira-fs at yahoo.com wrote: > >Hey all. >>I configured the FS to answer any incoming call and I have an application that keeps listen to Freeswitch events. >>I would like know if there is a way to create a conference and add an incoming call to this conference using the FS socket interface. >> >>Thanks in advance for your help. >> >>Leo >> >> >> >> >>? >>Leonardo Nogueira de S? Pereira >>Tel.: +55 19 3307-5589 >>Cel.: +55 19 9122-5943 >>Skype: leonardo_pereira_77 >> >>_________________________________________________________________________ >>Professional FreeSWITCH Consulting Services: >>consulting at freeswitch.org >>http://www.freeswitchsolutions.com >> >> >> >> >>Official FreeSWITCH Sites >>http://www.freeswitch.org >>http://wiki.freeswitch.org >>http://www.cluecon.com >> >>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 >> >> > > >-- >Michael S Collins >Twitter: @mercutioviz >http://www.FreeSWITCH.org >http://www.ClueCon.com >http://www.OSTAG.org > > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >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 > > > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130218/510f0d55/attachment-0001.html From freeswitch-list at puzzled.xs4all.nl Mon Feb 18 23:34:19 2013 From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists) Date: Mon, 18 Feb 2013 21:34:19 +0100 Subject: [Freeswitch-users] FreeSWITCH Weekly News and Notes In-Reply-To: References: Message-ID: <5122904B.5090209@puzzled.xs4all.nl> On 02/18/2013 08:38 PM, Michael Collins wrote: [snip] > For the DIY crowd you may enjoy this site that > Ken Rice set up. It's dedicated to doing cool things with FreeSWITCH and > the Raspberry Pi . Check it out! Cool project. Noticed in the LED screen: Dial A Minnestrone Time for SpellCheckerPI? :-) Regards, Patrick From msc at freeswitch.org Mon Feb 18 23:43:31 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 18 Feb 2013 12:43:31 -0800 Subject: [Freeswitch-users] mod_flite compile error In-Reply-To: References: Message-ID: Is this a fresh git checkout? If not, try doing: git clean -fdx then do bootstrap, configure and make install again -MC On Mon, Feb 18, 2013 at 11:39 AM, ram wrote: > > Hi > > iam having bit old Centos on hosted Platform > 2.6.18-348.1.1.el5PAE #1 SMP Tue Jan 22 17:06:35 EST 2013 i686 i686 i386 > GNU/Linux > > iam trying to install Freeswitch GIT > > i get following error > > any advice > > > making in main ... > ../build/i386-linux-gnu/lib/libflite_cmu_us_rms.a(cmu_us_rms_cg.o):(. > data.rel.ro+0x38): undefined reference to `cmu_us_rms_f0_carts' > ../build/i386-linux-gnu/lib/libflite_cmu_us_slt.a(cmu_us_slt_cg.o):(. > data.rel.ro+0x4c): undefined reference to `cmu_us_slt_spamf0_phrase_cart' > collect2: ld returned 1 exit status > make[8]: *** [../bin/flite] Error 1 > make[7]: *** [build/i386-linux-gnu/obj//.make_build_dirs] Error 2 > make[7]: Leaving directory `/usr/src/freeswitch/libs/flite-1.5.4-current' > make[6]: *** > [/usr/src/freeswitch/libs/flite-1.5.4-current/build/libs/libflite_cmu_us_awb.a] > Error 2 > make[6]: Leaving directory `/usr/src/freeswitch/src/mod/asr_tts/mod_flite' > make[5]: *** [all] Error 1 > make[5]: Leaving directory `/usr/src/freeswitch/src/mod/asr_tts/mod_flite' > make[4]: *** [mod_flite-all] Error 1 > make[4]: Leaving directory `/usr/src/freeswitch/src/mod' > make[3]: *** [all-recursive] Error 1 > make[3]: Leaving directory `/usr/src/freeswitch/src' > make[2]: *** [all-recursive] Error 1 > make[2]: Leaving directory `/usr/src/freeswitch' > make[1]: *** [all] Error 2 > make[1]: Leaving directory `/usr/src/freeswitch' > make: *** [current] Error 2 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130218/d5d55b0e/attachment.html From steveayre at gmail.com Mon Feb 18 23:58:14 2013 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 18 Feb 2013 20:58:14 +0000 Subject: [Freeswitch-users] FreeSWITCH Weekly News and Notes In-Reply-To: <5122904B.5090209@puzzled.xs4all.nl> References: <5122904B.5090209@puzzled.xs4all.nl> Message-ID: That's his Italian cousin. On 18 February 2013 20:34, Patrick Lists wrote: > On 02/18/2013 08:38 PM, Michael Collins wrote: > [snip] > >> For the DIY crowd you may enjoy this site that >> Ken Rice set up. It's dedicated to doing cool things with FreeSWITCH and >> the Raspberry Pi . Check it out! > > Cool project. Noticed in the LED screen: > > Dial > A Minnestrone > > Time for SpellCheckerPI? :-) > > Regards, > Patrick > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 krice at freeswitch.org Tue Feb 19 00:01:00 2013 From: krice at freeswitch.org (Ken Rice) Date: Mon, 18 Feb 2013 15:01:00 -0600 Subject: [Freeswitch-users] FreeSWITCH Weekly News and Notes In-Reply-To: <5122904B.5090209@puzzled.xs4all.nl> Message-ID: On 2/18/13 2:34 PM, "Patrick Lists" wrote: > On 02/18/2013 08:38 PM, Michael Collins wrote: > [snip] > >> For the DIY crowd you may enjoy this site that >> Ken Rice set up. It's dedicated to doing cool things with FreeSWITCH and >> the Raspberry Pi . Check it out! > > Cool project. Noticed in the LED screen: > > Dial > A Minnestrone > > Time for SpellCheckerPI? :-) Actually that was purposely done... Seems some people think that's how you pronounce tony's last name lol.... Now if UPS will just deliver my DTMF pad to go with it lol -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org http://www.switchpi.org irc.freenode.net #freeswitch From mel0torme at gmail.com Tue Feb 19 00:30:11 2013 From: mel0torme at gmail.com (Tom C) Date: Mon, 18 Feb 2013 13:30:11 -0800 Subject: [Freeswitch-users] mod_flite compile error In-Reply-To: References: Message-ID: How much memory is available on your hosted platform? Building mod-flite requires about 450 megabytes of memory. If there is insufficient memory available, you'll receive errors like those. On a system with limited RAM, creating a swap file will allow you to build this module successfully. On Feb 18, 2013 12:46 PM, "Michael Collins" wrote: > Is this a fresh git checkout? If not, try doing: > > git clean -fdx then do bootstrap, configure and make install again > > -MC > > On Mon, Feb 18, 2013 at 11:39 AM, ram wrote: > >> >> Hi >> >> iam having bit old Centos on hosted Platform >> 2.6.18-348.1.1.el5PAE #1 SMP Tue Jan 22 17:06:35 EST 2013 i686 i686 i386 >> GNU/Linux >> >> iam trying to install Freeswitch GIT >> >> i get following error >> >> any advice >> >> >> making in main ... >> ../build/i386-linux-gnu/lib/libflite_cmu_us_rms.a(cmu_us_rms_cg.o):(. >> data.rel.ro+0x38): undefined reference to `cmu_us_rms_f0_carts' >> ../build/i386-linux-gnu/lib/libflite_cmu_us_slt.a(cmu_us_slt_cg.o):(. >> data.rel.ro+0x4c): undefined reference to `cmu_us_slt_spamf0_phrase_cart' >> collect2: ld returned 1 exit status >> make[8]: *** [../bin/flite] Error 1 >> make[7]: *** [build/i386-linux-gnu/obj//.make_build_dirs] Error 2 >> make[7]: Leaving directory `/usr/src/freeswitch/libs/flite-1.5.4-current' >> make[6]: *** >> [/usr/src/freeswitch/libs/flite-1.5.4-current/build/libs/libflite_cmu_us_awb.a] >> Error 2 >> make[6]: Leaving directory `/usr/src/freeswitch/src/mod/asr_tts/mod_flite' >> make[5]: *** [all] Error 1 >> make[5]: Leaving directory `/usr/src/freeswitch/src/mod/asr_tts/mod_flite' >> make[4]: *** [mod_flite-all] Error 1 >> make[4]: Leaving directory `/usr/src/freeswitch/src/mod' >> make[3]: *** [all-recursive] Error 1 >> make[3]: Leaving directory `/usr/src/freeswitch/src' >> make[2]: *** [all-recursive] Error 1 >> make[2]: Leaving directory `/usr/src/freeswitch' >> make[1]: *** [all] Error 2 >> make[1]: Leaving directory `/usr/src/freeswitch' >> make: *** [current] Error 2 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130218/dda33bcf/attachment.html From kaothics at hotmail.com Tue Feb 19 00:34:54 2013 From: kaothics at hotmail.com (PolyMorphiK) Date: Mon, 18 Feb 2013 13:34:54 -0800 (PST) Subject: [Freeswitch-users] troubles with Mod_gsmopen and Huawei E153 In-Reply-To: <1361204375768-7587585.post@n2.nabble.com> References: <1361204375768-7587585.post@n2.nabble.com> Message-ID: <1361223294004-7587628.post@n2.nabble.com> Maybe it comes from the CTB (Communication toolbox library) libctb-0.16 ... ?? Any idea ? Anyone ? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/troubles-with-Mod-gsmopen-and-Huawei-E153-tp7587585p7587628.html Sent from the freeswitch-users mailing list archive at Nabble.com. From gmaruzz at gmail.com Tue Feb 19 01:19:44 2013 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Mon, 18 Feb 2013 23:19:44 +0100 Subject: [Freeswitch-users] troubles with Mod_gsmopen and Huawei E153 In-Reply-To: <1361223294004-7587628.post@n2.nabble.com> References: <1361204375768-7587585.post@n2.nabble.com> <1361223294004-7587628.post@n2.nabble.com> Message-ID: you need to find the right COM ports and put then in the config file... maybe by trial and error... or looking in device manager (or how it's called nowadays). Also, the application for unlocking may be useful for finding the ports ( http://www.dc-unlocker.com/ ) -giovanni On Mon, Feb 18, 2013 at 10:34 PM, PolyMorphiK wrote: > Maybe it comes from the CTB (Communication toolbox library) libctb-0.16 ... > ?? > > Any idea ? Anyone ? > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/troubles-with-Mod-gsmopen-and-Huawei-E153-tp7587585p7587628.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130218/796963f4/attachment.html From kaothics at hotmail.com Tue Feb 19 01:48:31 2013 From: kaothics at hotmail.com (PolyMorphiK) Date: Mon, 18 Feb 2013 14:48:31 -0800 (PST) Subject: [Freeswitch-users] troubles with Mod_gsmopen and Huawei E153 In-Reply-To: References: <1361204375768-7587585.post@n2.nabble.com> <1361223294004-7587628.post@n2.nabble.com> Message-ID: <1361227711276-7587631.post@n2.nabble.com> Thanks for your advices GIovanni but i already tried. I checked on the device manager , uninstalled the device , installed again the device , reloaded mod_gsmopen , used also the command gsm reload and same result unfortunatly , i tried all the port listed on device manager and inverted them to be sure nothing was wrong there. i also tried an other E153 device and on the Dashboard Mobil partner the port listed is the same than on the device manager and of course the dashboard isnt open when i use freeswitch to make sure the port com is available. i also checked on the source code the setup of the port com baud rate 115 200 "8N1" and i've made myself the same setup to make sure it will be compatible. I guess you worked on the Development of Gsm_open Giovanni, do you have any other idea ? Thanks -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/troubles-with-Mod-gsmopen-and-Huawei-E153-tp7587585p7587631.html Sent from the freeswitch-users mailing list archive at Nabble.com. From itamar at ispbrasil.com.br Tue Feb 19 01:53:12 2013 From: itamar at ispbrasil.com.br (Itamar Reis Peixoto) Date: Mon, 18 Feb 2013 19:53:12 -0300 Subject: [Freeswitch-users] troubles with Mod_gsmopen and Huawei E153 In-Reply-To: References: <1361204375768-7587585.post@n2.nabble.com> <1361223294004-7587628.post@n2.nabble.com> Message-ID: On Mon, Feb 18, 2013 at 7:19 PM, Giovanni Maruzzelli wrote: > you need to find the right COM ports and put then in the config file... > maybe by trial and error... or looking in device manager (or how it's called > nowadays). Also, the application for unlocking may be useful for finding > the ports ( http://www.dc-unlocker.com/ ) mod_chan-dongle can auto-detect the port based on IMEI / IMSI -- ------------ Itamar Reis Peixoto From gmaruzz at celliax.org Tue Feb 19 01:58:59 2013 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Mon, 18 Feb 2013 23:58:59 +0100 Subject: [Freeswitch-users] troubles with Mod_gsmopen and Huawei E153 In-Reply-To: References: <1361204375768-7587585.post@n2.nabble.com> <1361223294004-7587628.post@n2.nabble.com> Message-ID: On Mon, Feb 18, 2013 at 11:53 PM, Itamar Reis Peixoto < itamar at ispbrasil.com.br> wrote: > > > mod_chan-dongle can auto-detect the port based on IMEI / IMSI > > on windows 8 ? Cool !!! -giovanni -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130218/1ae53887/attachment.html From kaothics at hotmail.com Tue Feb 19 02:03:16 2013 From: kaothics at hotmail.com (PolyMorphiK) Date: Mon, 18 Feb 2013 15:03:16 -0800 (PST) Subject: [Freeswitch-users] troubles with Mod_gsmopen and Huawei E153 In-Reply-To: References: <1361204375768-7587585.post@n2.nabble.com> <1361223294004-7587628.post@n2.nabble.com> Message-ID: <1361228596913-7587633.post@n2.nabble.com> i've just tried DC-Unlocker like suggest and it detects pretty well the Dongle and it shows that the Dongle is unlocked and with the voice enabled. i tried the port detected and still the same :( Driving me nuts sometimes lol Cordialy -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/troubles-with-Mod-gsmopen-and-Huawei-E153-tp7587585p7587633.html Sent from the freeswitch-users mailing list archive at Nabble.com. From dvl36.ripe.nick at gmail.com Tue Feb 19 02:33:59 2013 From: dvl36.ripe.nick at gmail.com (Dmitry Lysenko) Date: Tue, 19 Feb 2013 01:33:59 +0200 Subject: [Freeswitch-users] troubles with Mod_gsmopen and Huawei E153 In-Reply-To: <1361228596913-7587633.post@n2.nabble.com> References: <1361204375768-7587585.post@n2.nabble.com> <1361223294004-7587628.post@n2.nabble.com> <1361228596913-7587633.post@n2.nabble.com> Message-ID: Using of mod_gsmopen(and freeswitch too) on desktops systems is not good idea, IMO. Finding of "right" com-ports on Windows isn't so simple. Best Wishes, Dmitry. 2013/2/19 PolyMorphiK > i've just tried DC-Unlocker like suggest and it detects pretty well the > Dongle and it shows that the Dongle is unlocked and with the voice enabled. > i tried the port detected and still the same :( > Driving me nuts sometimes lol > > > Cordialy > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/troubles-with-Mod-gsmopen-and-Huawei-E153-tp7587585p7587633.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130219/57d24d3d/attachment.html From schoch+freeswitch.org at xwin32.com Tue Feb 19 04:17:27 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Mon, 18 Feb 2013 17:17:27 -0800 Subject: [Freeswitch-users] Pre-loading Voicemail prompts and names Message-ID: We're switching from an old phone system to Freeswitch, so I converted everyone's voicemail greetings and recorded names using sox, and dropped the WAV files to greeting_1.wav and recorded_name.wav in their respective voicemail directories. The voicemail module didn't automatically recognize these (even on restart), but I was able to set everyone's greeting by connecting to their voice mailbox from their extension and selecting greeting 1. I don't know how to do this to make it take their recorded name. I want the directory module to play their name instead of spelling it. How to I tell Freeswitch that there are recorded_name.wav files sitting there ready to be used? -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130218/40fe3966/attachment.html From kaothics at hotmail.com Tue Feb 19 06:17:41 2013 From: kaothics at hotmail.com (PolyMorphiK) Date: Mon, 18 Feb 2013 19:17:41 -0800 (PST) Subject: [Freeswitch-users] troubles with Mod_gsmopen and Huawei E153 In-Reply-To: References: <1361204375768-7587585.post@n2.nabble.com> <1361223294004-7587628.post@n2.nabble.com> <1361228596913-7587633.post@n2.nabble.com> Message-ID: <1361243861053-7587637.post@n2.nabble.com> Well, i somewhat agree with you dvl36 but if you travel a lot and want to try some scripts really fast it's not too bad to be on windows systems. And if there is a windows development that's for a reason don't you think ? ... Beside that it's not constructive to think that way so instead i'm looking for a provisory solution using Oktell SIP-GSM ... http://www.oktellpbx.com/gateways/ if someone already worked with, he is welcome to help ;) have fun ! Cordialy -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/troubles-with-Mod-gsmopen-and-Huawei-E153-tp7587585p7587637.html Sent from the freeswitch-users mailing list archive at Nabble.com. From bdfoster at endigotech.com Tue Feb 19 06:37:14 2013 From: bdfoster at endigotech.com (Brian Foster) Date: Mon, 18 Feb 2013 22:37:14 -0500 Subject: [Freeswitch-users] FreeSWITCH Weekly News and Notes In-Reply-To: References: Message-ID: <84AE3463-2CD4-4165-B29B-3C15137025E8@endigotech.com> Specifically Callie....Lol Nice project. Is the purpose of the project to basically turn a Pi into a "softphone"? Sent from my iPhone On Feb 18, 2013, at 4:01 PM, Ken Rice wrote: > On 2/18/13 2:34 PM, "Patrick Lists" > wrote: > >> On 02/18/2013 08:38 PM, Michael Collins wrote: >> [snip] >> >>> For the DIY crowd you may enjoy this site that >>> Ken Rice set up. It's dedicated to doing cool things with FreeSWITCH and >>> the Raspberry Pi . Check it out! >> >> Cool project. Noticed in the LED screen: >> >> Dial >> A Minnestrone >> >> Time for SpellCheckerPI? :-) > > Actually that was purposely done... Seems some people think that's how you > pronounce tony's last name lol.... > > Now if UPS will just deliver my DTMF pad to go with it lol > > > -- > Ken > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > http://www.switchpi.org > irc.freenode.net #freeswitch > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 talk2ram at gmail.com Tue Feb 19 08:12:18 2013 From: talk2ram at gmail.com (ram) Date: Tue, 19 Feb 2013 10:42:18 +0530 Subject: [Freeswitch-users] mod_flite compile error In-Reply-To: References: Message-ID: Hi yes, the command fixes the issue.. On Tue, Feb 19, 2013 at 2:13 AM, Michael Collins wrote: > Is this a fresh git checkout? If not, try doing: > > git clean -fdx then do bootstrap, configure and make install again > > -MC > > On Mon, Feb 18, 2013 at 11:39 AM, ram wrote: > >> >> Hi >> >> iam having bit old Centos on hosted Platform >> 2.6.18-348.1.1.el5PAE #1 SMP Tue Jan 22 17:06:35 EST 2013 i686 i686 i386 >> GNU/Linux >> >> iam trying to install Freeswitch GIT >> >> i get following error >> >> any advice >> >> >> making in main ... >> ../build/i386-linux-gnu/lib/libflite_cmu_us_rms.a(cmu_us_rms_cg.o):(. >> data.rel.ro+0x38): undefined reference to `cmu_us_rms_f0_carts' >> ../build/i386-linux-gnu/lib/libflite_cmu_us_slt.a(cmu_us_slt_cg.o):(. >> data.rel.ro+0x4c): undefined reference to `cmu_us_slt_spamf0_phrase_cart' >> collect2: ld returned 1 exit status >> make[8]: *** [../bin/flite] Error 1 >> make[7]: *** [build/i386-linux-gnu/obj//.make_build_dirs] Error 2 >> make[7]: Leaving directory `/usr/src/freeswitch/libs/flite-1.5.4-current' >> make[6]: *** >> [/usr/src/freeswitch/libs/flite-1.5.4-current/build/libs/libflite_cmu_us_awb.a] >> Error 2 >> make[6]: Leaving directory `/usr/src/freeswitch/src/mod/asr_tts/mod_flite' >> make[5]: *** [all] Error 1 >> make[5]: Leaving directory `/usr/src/freeswitch/src/mod/asr_tts/mod_flite' >> make[4]: *** [mod_flite-all] Error 1 >> make[4]: Leaving directory `/usr/src/freeswitch/src/mod' >> make[3]: *** [all-recursive] Error 1 >> make[3]: Leaving directory `/usr/src/freeswitch/src' >> make[2]: *** [all-recursive] Error 1 >> make[2]: Leaving directory `/usr/src/freeswitch' >> make[1]: *** [all] Error 2 >> make[1]: Leaving directory `/usr/src/freeswitch' >> make: *** [current] Error 2 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130219/c397771d/attachment.html From shayne.alone at gmail.com Tue Feb 19 08:47:37 2013 From: shayne.alone at gmail.com (shayne.alone at gmail.com) Date: Tue, 19 Feb 2013 09:17:37 +0330 Subject: [Freeswitch-users] outbound caller ID Message-ID: Hi all; how can I change the green "FREESWITCH" string on outbound calls? 09:07:57.171420 IP 2.22.222.20.5080 > 2.22.222.10.5060: SIP, length: 1056 E..<.9.. at .@..-.E...d.....(V.INVITE sip:12123089 at 2.22.222.10 SIP/2.0 Via: SIP/2.0/UDP 2.22.222.20:5080;rport;branch=z9hG4bKDZQUXBgeDHZpg Max-Forwards: 69 From: "18009889888" ;tag=DK6NFKH46tFam To: Call-ID: 5b2d87f7-f4f9-1230-e4b8-00505681534d CSeq: 40290074 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.2.6+git~20130104T154559Z~a4247651ca Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 203 X-FS-Support: update_display,send_info Remote-Party-ID: "18009889888" ;party=calling;screen=yes;privacy=off -- Regards, Ali R. Taleghani -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130219/3bac8140/attachment.html From lists at kavun.ch Tue Feb 19 08:54:52 2013 From: lists at kavun.ch (Emrah) Date: Tue, 19 Feb 2013 00:54:52 -0500 Subject: [Freeswitch-users] Changing codec during calls In-Reply-To: References: <6EF8B8E9-4CC5-480C-B1C1-09CBC533470C@kavun.ch> Message-ID: <35AF9E79-771D-440A-BEBB-A856D831161C@kavun.ch> Hey, In short, nothing seems to happen. I tried several phones. I also tried enabling codec renegotiation on hold/unhold thinking that the change in SDP with uuid_media_reneg would do it. This is what happens: freeswitch at internal> show channels uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,sent_callee_name,sent_callee_num 49f428e5-1d72-42d8-b0e1-5b3f29f9a4ca,inbound,2013-02-19 00:44:35,1361252675,sofia/internal/10000 at sip.domain.net,CS_EXECUTE,Emrah - Macbook Pro,10000,100.100.100.100,conf-3111,conference,3111-sip.domain.net at ek-custom-16k,XML,ek-conference,G722,16000,64000,G722,16000,64000,,cornflakes.domain.net,10000 at sip.domain.net,,ACTIVE,,,,,, 1 total. freeswitch at internal> uuid_media_reneg 49f428e5-1d72-42d8-b0e1-5b3f29f9a4ca PCMU +OK Success 2013-02-19 00:45:25.172659 [DEBUG] sofia_glue.c:2647 Local SDP: v=0 o=FreeSWITCH 1361234787 1361234790 IN IP4 1.2.3.4 s=FreeSWITCH c=IN IP4 1.2.3.4 t=0 0 m=audio 17888 RTP/AVP 9 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2013-02-19 00:45:25.172659 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/10000 at sip.domain.net [BREAK] 2013-02-19 00:45:25.182652 [DEBUG] sofia.c:5574 Channel sofia/internal/10000 at sip.domain.net entering state [calling][0] 2013-02-19 00:45:25.292661 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/10000 at sip.domain.net [BREAK] 2013-02-19 00:45:25.292661 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/10000 at sip.domain.net [BREAK] 2013-02-19 00:45:25.302683 [INFO] sofia.c:931 sofia/internal/10000 at sip.domain.net Update Callee ID to "10000" <10000> 2013-02-19 00:45:25.302683 [DEBUG] sofia.c:5574 Channel sofia/internal/10000 at sip.domain.net entering state [completing][200] 2013-02-19 00:45:25.302683 [DEBUG] sofia.c:5585 Remote SDP: v=0 o=- 3570241475 3570241476 IN IP4 10.0.0.131 s=pjmedia c=IN IP4 10.0.0.131 t=0 0 a=X-nat:0 m=audio 4010 RTP/AVP 9 101 a=rtcp:4011 IN IP4 10.0.0.131 a=rtpmap:9 G722/8000 a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 2013-02-19 00:45:25.302683 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/10000 at sip.domain.net [BREAK] 2013-02-19 00:45:25.302683 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/10000 at sip.domain.net [BREAK] 2013-02-19 00:45:25.322657 [DEBUG] sofia.c:5574 Channel sofia/internal/10000 at sip.domain.net entering state [ready][200] 2013-02-19 00:45:25.322657 [DEBUG] sofia.c:5582 Duplicate SDP v=0 o=- 3570241475 3570241476 IN IP4 10.0.0.131 s=pjmedia c=IN IP4 10.0.0.131 t=0 0 a=X-nat:0 m=audio 4010 RTP/AVP 9 101 a=rtcp:4011 IN IP4 10.0.0.131 a=rtpmap:9 G722/8000 a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 2013-02-19 00:45:25.322657 [DEBUG] sofia_glue.c:5137 Audio Codec Compare [G722:9:8000:20:64000]/[G722:9:8000:20:64000] 2013-02-19 00:45:25.322657 [DEBUG] sofia_glue.c:3022 Already using G722 2013-02-19 00:45:25.322657 [DEBUG] sofia_glue.c:5266 Set 2833 dtmf send/recv payload to 101 Thanks for any help, Emrah On Feb 18, 2013, at 1:38 PM, Michael Collins wrote: > What happened with the uuid_media_reneg command? > -MC > > On Sat, Feb 16, 2013 at 3:10 PM, Emrah wrote: > Hi all, > > How do we go about renegotiating the codecs of a call that is already established? > I tried uuid_media_reneg to no avail. > E.g.: Upgrade someone who's entered a 16k conference from PCMU to G722. > > Cheers and thanks, > Emrah > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 krice at freeswitch.org Tue Feb 19 08:56:36 2013 From: krice at freeswitch.org (Ken Rice) Date: Mon, 18 Feb 2013 23:56:36 -0600 Subject: [Freeswitch-users] outbound caller ID In-Reply-To: Message-ID: See http://wiki.freeswitch.org/wiki/Clarification:gateways#Caller_ID On 2/18/13 11:47 PM, "shayne.alone at gmail.com" wrote: > Hi all; > ?how can I change the green "FREESWITCH" string on outbound calls? > > > > > 09:07:57.171420 IP 2.22.222.20.5080 > 2.22.222.10.5060: SIP, length: 1056 > E..<.9.. at .@..-.E...d.....(V.INVITE sip:12123089 at 2.22.222.10 > SIP/2.0 > Via: SIP/2.0/UDP 2.22.222.20:5080;rport;branch=z9hG4bKDZQUXBgeDHZpg > Max-Forwards: 69 > From: "18009889888" > >;tag=DK6NFKH46tFam > To: > > Call-ID: 5b2d87f7-f4f9-1230-e4b8-00505681534d > CSeq: 40290074 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.2.6+git~20130104T154559Z~a4247651ca > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, > REFER, NOTIFY > Supported: timer, precondition, path, replaces > Allow-Events: talk, hold, conference, refer > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 203 > X-FS-Support: update_display,send_info > Remote-Party-ID: "18009889888" >;party=calling;screen=yes;privacy=off > > -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org http://www.SwitchPI.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130218/7b837f4c/attachment-0001.html From shayne.alone at gmail.com Tue Feb 19 09:10:31 2013 From: shayne.alone at gmail.com (shayne.alone at gmail.com) Date: Tue, 19 Feb 2013 09:40:31 +0330 Subject: [Freeswitch-users] outbound caller ID In-Reply-To: References: Message-ID: :-) perfect On Tue, Feb 19, 2013 at 9:26 AM, Ken Rice wrote: > See http://wiki.freeswitch.org/wiki/Clarification:gateways#Caller_ID > > > On 2/18/13 11:47 PM, "shayne.alone at gmail.com" > wrote: > > Hi all; > how can I change the green "FREESWITCH" string on outbound calls? > > > > > 09:07:57.171420 IP 2.22.222.20.5080 > 2.22.222.10.5060: SIP, length: 1056 > E..<.9.. at .@..-.E...d.....(V.INVITE sip:12123089 at 2.22.222.10 < > mailto:sip%3A12123089 at 2.22.222.10 > SIP/2.0 > > Via: SIP/2.0/UDP 2.22.222.20:5080;rport;branch=z9hG4bKDZQUXBgeDHZpg > Max-Forwards: 69 > From: "18009889888" > >;tag=DK6NFKH46tFam > To: > > > > > Call-ID: 5b2d87f7-f4f9-1230-e4b8-00505681534d > CSeq: 40290074 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.2.6+git~20130104T154559Z~a4247651ca > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, > REFER, NOTIFY > Supported: timer, precondition, path, replaces > Allow-Events: talk, hold, conference, refer > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 203 > X-FS-Support: update_display,send_info > Remote-Party-ID: "18009889888" mailto:sip%3A18009889888 at 2.22.222.10 > > >;party=calling;screen=yes;privacy=off > > > > -- > Ken > *http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > http://www.SwitchPI.org > *irc.freenode.net #freeswitch > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Regards, Ali R. Taleghani -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130219/7fb3aac8/attachment.html From lloyd.aloysius at gmail.com Tue Feb 19 10:16:40 2013 From: lloyd.aloysius at gmail.com (Lloyd Aloysius) Date: Tue, 19 Feb 2013 02:16:40 -0500 Subject: [Freeswitch-users] mod_directory - Cannot locate domain In-Reply-To: References: Message-ID: Brian/Mike, Thank you for the reply. When I copy and paste into the mailing list.. the section tag missed There is already a section tag end
When I reload the modules from cli. switch identify the profiles. But when I execute the .. session:execute("directory", domain_name.." "..domain_name) I got the 2013-02-16 21:43:04.621318 [ERR] switch_xml.c:1650 Error[[error near line 1]: unclosed tag ] 2013-02-16 21:43:04.621318 [WARNING] mod_directory.c:545 Cannot locate domain mydomain.com domain_name and profile name is same Thanks Lloyd On Sun, Feb 17, 2013 at 11:44 PM, Michael Collins wrote: > You forgot to close the
tag... > -MC > > On Sat, Feb 16, 2013 at 6:58 PM, Lloyd Aloysius wrote: > >> Hi All >> >> I have a strange problem with the mod_directory. I use xml_curl for all >> the applications. Only mod_directory giving this trouble. >> >> >> EXECUTE sofia/sipinterface_1/marv at mydomain.com directory(mydomain.com >> mydomain.com) >> 2013-02-16 21:43:04.601315 [DEBUG] mod_directory.c:518 [mydomain.com] >> rwlock >> >> 2013-02-16 21:43:04.621318 [ERR] switch_xml.c:1650 Error[[error near line >> 1]: unclosed tag ] >> 2013-02-16 21:43:04.621318 [WARNING] mod_directory.c:545 Cannot locate >> domain mydomain.com >> >> -- >> >> my direcotroy.conf xml curl output >> >> >>
>> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> Any help is appreciated. >> >> Thanks >> Lloyd >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130219/ef643547/attachment.html From avi at avimarcus.net Tue Feb 19 10:26:54 2013 From: avi at avimarcus.net (Avi Marcus) Date: Tue, 19 Feb 2013 09:26:54 +0200 Subject: [Freeswitch-users] Pre-loading Voicemail prompts and names In-Reply-To: References: Message-ID: Steve, you can open up the voicemail_prefs database. If you set up ODBC for VM you know where it is, otherwise, it's in a sqlite DB. Then set the name_path to the full path of the file for each account. (You could have done that for greeting path, too...) -Avi Marcus BestFone On Tue, Feb 19, 2013 at 3:17 AM, Steven Schoch < schoch+freeswitch.org at xwin32.com> wrote: > We're switching from an old phone system to Freeswitch, so I converted > everyone's voicemail greetings and recorded names using sox, and dropped > the WAV files to greeting_1.wav and recorded_name.wav in their respective > voicemail directories. > > The voicemail module didn't automatically recognize these (even on > restart), but I was able to set everyone's greeting by connecting to their > voice mailbox from their extension and selecting greeting 1. > > I don't know how to do this to make it take their recorded name. I want > the directory module to play their name instead of spelling it. How to I > tell Freeswitch that there are recorded_name.wav files sitting there ready > to be used? > > -- > Steve > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130219/36c2e687/attachment-0001.html From steveayre at gmail.com Tue Feb 19 11:47:46 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 19 Feb 2013 08:47:46 +0000 Subject: [Freeswitch-users] Changing codec during calls In-Reply-To: <35AF9E79-771D-440A-BEBB-A856D831161C@kavun.ch> References: <6EF8B8E9-4CC5-480C-B1C1-09CBC533470C@kavun.ch> <35AF9E79-771D-440A-BEBB-A856D831161C@kavun.ch> Message-ID: Try: uuid_media_reneg 49f428e5-1d72-42d8-b0e1-5b3f29f9a4ca =PCMU (ie add = before PCMU) It appears to affect something in the code... -Steve On 19 February 2013 05:54, Emrah wrote: > Hey, > > In short, nothing seems to happen. I tried several phones. > I also tried enabling codec renegotiation on hold/unhold thinking that the change in SDP with uuid_media_reneg would do it. > > This is what happens: > freeswitch at internal> show channels > uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,sent_callee_name,sent_callee_num > 49f428e5-1d72-42d8-b0e1-5b3f29f9a4ca,inbound,2013-02-19 00:44:35,1361252675,sofia/internal/10000 at sip.domain.net,CS_EXECUTE,Emrah - Macbook Pro,10000,100.100.100.100,conf-3111,conference,3111-sip.domain.net at ek-custom-16k,XML,ek-conference,G722,16000,64000,G722,16000,64000,,cornflakes.domain.net,10000 at sip.domain.net,,ACTIVE,,,,,, > > 1 total. > > freeswitch at internal> uuid_media_reneg 49f428e5-1d72-42d8-b0e1-5b3f29f9a4ca PCMU > +OK Success > > 2013-02-19 00:45:25.172659 [DEBUG] sofia_glue.c:2647 Local SDP: > v=0 > o=FreeSWITCH 1361234787 1361234790 IN IP4 1.2.3.4 > s=FreeSWITCH > c=IN IP4 1.2.3.4 > t=0 0 > m=audio 17888 RTP/AVP 9 0 101 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=sendrecv > > 2013-02-19 00:45:25.172659 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/10000 at sip.domain.net [BREAK] > 2013-02-19 00:45:25.182652 [DEBUG] sofia.c:5574 Channel sofia/internal/10000 at sip.domain.net entering state [calling][0] > 2013-02-19 00:45:25.292661 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/10000 at sip.domain.net [BREAK] > 2013-02-19 00:45:25.292661 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/10000 at sip.domain.net [BREAK] > 2013-02-19 00:45:25.302683 [INFO] sofia.c:931 sofia/internal/10000 at sip.domain.net Update Callee ID to "10000" <10000> > 2013-02-19 00:45:25.302683 [DEBUG] sofia.c:5574 Channel sofia/internal/10000 at sip.domain.net entering state [completing][200] > 2013-02-19 00:45:25.302683 [DEBUG] sofia.c:5585 Remote SDP: > v=0 > o=- 3570241475 3570241476 IN IP4 10.0.0.131 > s=pjmedia > c=IN IP4 10.0.0.131 > t=0 0 > a=X-nat:0 > m=audio 4010 RTP/AVP 9 101 > a=rtcp:4011 IN IP4 10.0.0.131 > a=rtpmap:9 G722/8000 > a=sendrecv > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > > 2013-02-19 00:45:25.302683 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/10000 at sip.domain.net [BREAK] > 2013-02-19 00:45:25.302683 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/10000 at sip.domain.net [BREAK] > 2013-02-19 00:45:25.322657 [DEBUG] sofia.c:5574 Channel sofia/internal/10000 at sip.domain.net entering state [ready][200] > 2013-02-19 00:45:25.322657 [DEBUG] sofia.c:5582 Duplicate SDP > v=0 > o=- 3570241475 3570241476 IN IP4 10.0.0.131 > s=pjmedia > c=IN IP4 10.0.0.131 > t=0 0 > a=X-nat:0 > m=audio 4010 RTP/AVP 9 101 > a=rtcp:4011 IN IP4 10.0.0.131 > a=rtpmap:9 G722/8000 > a=sendrecv > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > > 2013-02-19 00:45:25.322657 [DEBUG] sofia_glue.c:5137 Audio Codec Compare [G722:9:8000:20:64000]/[G722:9:8000:20:64000] > 2013-02-19 00:45:25.322657 [DEBUG] sofia_glue.c:3022 Already using G722 > 2013-02-19 00:45:25.322657 [DEBUG] sofia_glue.c:5266 Set 2833 dtmf send/recv payload to 101 > > > Thanks for any help, > Emrah > On Feb 18, 2013, at 1:38 PM, Michael Collins wrote: > >> What happened with the uuid_media_reneg command? >> -MC >> >> On Sat, Feb 16, 2013 at 3:10 PM, Emrah wrote: >> Hi all, >> >> How do we go about renegotiating the codecs of a call that is already established? >> I tried uuid_media_reneg to no avail. >> E.g.: Upgrade someone who's entered a 16k conference from PCMU to G722. >> >> Cheers and thanks, >> Emrah >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 steveayre at gmail.com Tue Feb 19 12:16:24 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 19 Feb 2013 09:16:24 +0000 Subject: [Freeswitch-users] mod_directory - Cannot locate domain In-Reply-To: References: Message-ID: That XML looks valid... Use the command "xml_curl debug_on" It'll log a filename where the XML CURL result has been saved to a file, which will let you examine what FS has actually received. If you run 'xmllint $filename' on it (it's in libxml2-utils) then you'll have another XML parser check it which might also give you a hint where the error is from its error message(s). -Steve On 19 February 2013 07:16, Lloyd Aloysius wrote: > Brian/Mike, >ly > > There is already a section tag end > > >
> > > > > > > > > > > > > > > > > > > > >
> > > > When I reload the modules from cli. switch identify the profiles. But when I > execute the .. > > session:execute("directory", domain_name.." "..domain_name) > > I got the > > 2013-02-16 21:43:04.621318 [ERR] switch_xml.c:1650 Error[[error near line > 1]: unclosed tag ] > 2013-02-16 21:43:04.621318 [WARNING] mod_directory.c:545 Cannot locate > domain mydomain.com > > domain_name and profile name is same > > Thanks > Lloyd > > > On Sun, Feb 17, 2013 at 11:44 PM, Michael Collins > wrote: >> >> You forgot to close the
tag... >> -MC >> >> On Sat, Feb 16, 2013 at 6:58 PM, Lloyd Aloysius >> wrote: >>> >>> Hi All >>> >>> I have a strange problem with the mod_directory. I use xml_curl for all >>> the applications. Only mod_directory giving this trouble. >>> >>> >>> EXECUTE sofia/sipinterface_1/marv at mydomain.com directory(mydomain.com >>> mydomain.com) >>> 2013-02-16 21:43:04.601315 [DEBUG] mod_directory.c:518 [mydomain.com] >>> rwlock >>> >>> 2013-02-16 21:43:04.621318 [ERR] switch_xml.c:1650 Error[[error near line >>> 1]: unclosed tag ] >>> 2013-02-16 21:43:04.621318 [WARNING] mod_directory.c:545 Cannot locate >>> domain mydomain.com >>> >>> -- >>> >>> my direcotroy.conf xml curl output >>> >>> >>>
>>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> Any help is appreciated. >>> >>> Thanks >>> Lloyd >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 julf at julf.com Tue Feb 19 12:22:02 2013 From: julf at julf.com (Johan Helsingius) Date: Tue, 19 Feb 2013 10:22:02 +0100 Subject: [Freeswitch-users] Problems with sip provider In-Reply-To: References: <51210FA4.1010200@julf.com> <51211478.1080108@julf.com> <5121CEBC.2020707@julf.com> Message-ID: <5123443A.704@julf.com> > Ask them for a list of devices that they support. Once you have that we can > probably help you figure out how to fool them into thinking that they are > talking to one of those "supported" devices. Well, the only "supported" devices are ADSL gateways. For anything else, they just give this set-up advice that looks pretty normal to me: SIP number: 020NNNNNNN Username: 020NNNNNNN Password: XXXXXXXX SIP proxy: sip.telefoniedienst.nl SIP service location: sip.telefoniedienst.nl SIP signaling port: UDP 5060 SIP register service location: tel.telefoniedienst.nl SIP register signaling port: UDP 5060 SIP domain: tel.telefoniedienst.nl URL type coding: SIP Registration interval: 6000 seconds Session expires: 1800 seconds Reserved RTP port range: UDP 16384-32767 Codecs: First negotiated G.729 secondly negotiated G.711 a-law type DTMF mode: PCM Sending INVITE Anonymous: Not permitted Echo Cancellation G.168: In use Voice Activation Detection: Not in use SIP SUBSCRIBES messages: Not permitted Early Media acknowledge: Not in Use This is what works on the ZyXEL P-2601HN0F1 gateway: SIP Account: 020NNNNNNN Username: 020NNNNNNN Password: XXXXXXXX Compression Type: G711ALaw, G.729, G.722 SIP Service Provider Name : telfort SIP Local Port : 5060 SIP Server Address : sip.telefoniedienst.nl SIP Server Port : 5060 REGISTER Server Address : sip.telefoniedienst.nl REGISTER Server Port : 5060 SIP Service Domain : tel.telefoniedienst.nl (different than SIP server) RTP Port Range: 16384-32767 Expiration Duration: 6000 s Register Re-send timer: 512 s Session Expires: 1800 s Min-SE: 1800 s Enable Early Media: Yes And this is what I have in my Freeswitch SIP gateway profile: It seems like the registration goes through OK, but when I try calling out, I get 013-02-19 10:00:37.236188 [NOTICE] sofia.c:6385 Hangup sofia/external/0651142219 [CS_EXCHANGE_MEDIA] [CALL_REJECTED] > In cases where the provider is so > totally foolish like this you have two choices: > * Use their foolishness against them and make them think your FreeSWITCH server > is really something else Well, I tried to claim the problem was with the ADSL gateway as well, and they just kept sending the same setup instructions and advice to reboot the gateway. Not helpful. > * Switch providers I agree, and I am seriously considering it - problem is that their Internet connection is pretty good and keenly priced, and the VoIP add-on to that is priced aggressively. Julf From julf at julf.com Tue Feb 19 12:23:46 2013 From: julf at julf.com (Johan Helsingius) Date: Tue, 19 Feb 2013 10:23:46 +0100 Subject: [Freeswitch-users] Problems with sip provider In-Reply-To: References: <51210FA4.1010200@julf.com> <51211478.1080108@julf.com> <5121CEBC.2020707@julf.com> Message-ID: <512344A2.1080800@julf.com> > It's totally your choice if you want to continue trying to get it fixed, but > most of the time, it's not worth the time/effort. Have to agree - just annoying that it does work when using their ADSL gateway :( Julf From julf at julf.com Tue Feb 19 12:27:02 2013 From: julf at julf.com (Johan Helsingius) Date: Tue, 19 Feb 2013 10:27:02 +0100 Subject: [Freeswitch-users] making group of phones ring from the command line In-Reply-To: References: <51224C37.4090902@julf.com> <5122592A.5060604@julf.com> <51225FB3.9060304@julf.com> Message-ID: <51234566.7050304@julf.com> > expand originate ${group_call(group at domain.com+A)} &playback(alarm.wav) OK, that works great, but is there any way to actually place the call to a dialplan extension rather than a SIP address or group of addresses? Julf From avi at avimarcus.net Tue Feb 19 12:36:44 2013 From: avi at avimarcus.net (Avi Marcus) Date: Tue, 19 Feb 2013 11:36:44 +0200 Subject: [Freeswitch-users] making group of phones ring from the command line In-Reply-To: <51234566.7050304@julf.com> References: <51224C37.4090902@julf.com> <5122592A.5060604@julf.com> <51225FB3.9060304@julf.com> <51234566.7050304@julf.com> Message-ID: originate user/1000 &playback(alarm.wav) .. will work. It's not the 1000 extension though, but it is the actual registered user. -Avi On Tue, Feb 19, 2013 at 11:27 AM, Johan Helsingius wrote: > > expand originate ${group_call(group at domain.com+A)} &playback(alarm.wav) > > OK, that works great, but is there any way to actually > place the call to a dialplan extension rather than a > SIP address or group of addresses? > > Julf > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130219/51b4cf05/attachment-0001.html From steveayre at gmail.com Tue Feb 19 12:38:13 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 19 Feb 2013 09:38:13 +0000 Subject: [Freeswitch-users] making group of phones ring from the command line In-Reply-To: <51234566.7050304@julf.com> References: <51224C37.4090902@julf.com> <5122592A.5060604@julf.com> <51225FB3.9060304@julf.com> <51234566.7050304@julf.com> Message-ID: You need an outgoing session to send to the dialplan, so need a dialstring. The 2nd parameter can send those calls into the dialplan: expand originate ${group_call(group at domain.com+A)} extension You could also use the loopback endpoint to create a session that goes into dialplan. expand originate loopback/extension &playback(alarm.wav) http://wiki.freeswitch.org/wiki/Loopback%20endpoint -Steve On 19 February 2013 09:27, Johan Helsingius wrote: >> expand originate ${group_call(group at domain.com+A)} &playback(alarm.wav) > > OK, that works great, but is there any way to actually > place the call to a dialplan extension rather than a > SIP address or group of addresses? > > Julf > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 steveayre at gmail.com Tue Feb 19 12:48:14 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 19 Feb 2013 09:48:14 +0000 Subject: [Freeswitch-users] Problems with sip provider In-Reply-To: <512344A2.1080800@julf.com> References: <51210FA4.1010200@julf.com> <51211478.1080108@julf.com> <5121CEBC.2020707@julf.com> <512344A2.1080800@julf.com> Message-ID: Is it possible to get a SIP trace of the calls from the gateway and FS and compare them? -Steve On 19 February 2013 09:23, Johan Helsingius wrote: >> It's totally your choice if you want to continue trying to get it fixed, but >> most of the time, it's not worth the time/effort. > > Have to agree - just annoying that it does work when using their > ADSL gateway :( > > Julf > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 muqtheear.s at gmail.com Tue Feb 19 12:53:09 2013 From: muqtheear.s at gmail.com (Muqtheear S) Date: Tue, 19 Feb 2013 15:23:09 +0530 Subject: [Freeswitch-users] How SIP clients able to establish a call with out Freeswitch Server? Message-ID: 1. I stopped Freeswitch server. User Agent client 1 was able to call User Agent client 2. 2. What configurations I need to change , so that Sip clients can use Freeswitch server.? Thanks & Regards, Muqtheear.S -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130219/01c71101/attachment.html From julf at julf.com Tue Feb 19 12:56:42 2013 From: julf at julf.com (Johan Helsingius) Date: Tue, 19 Feb 2013 10:56:42 +0100 Subject: [Freeswitch-users] Problems with sip provider In-Reply-To: References: <51210FA4.1010200@julf.com> <51211478.1080108@julf.com> <5121CEBC.2020707@julf.com> <512344A2.1080800@julf.com> Message-ID: <51234C5A.70904@julf.com> > Is it possible to get a SIP trace of the calls from the gateway and FS > and compare them? Unfortunately there is no way to get into the traffic between the gateway and the ADSL connection, as it also contains the ADSL modem. Julf From julf at julf.com Tue Feb 19 13:00:18 2013 From: julf at julf.com (Johan Helsingius) Date: Tue, 19 Feb 2013 11:00:18 +0100 Subject: [Freeswitch-users] making group of phones ring from the command line In-Reply-To: References: <51224C37.4090902@julf.com> <5122592A.5060604@julf.com> <51225FB3.9060304@julf.com> <51234566.7050304@julf.com> Message-ID: <51234D32.6090403@julf.com> > You could also use the loopback endpoint to create a session that goes > into dialplan. > expand originate loopback/extension &playback(alarm.wav) Perfect! Thanks! Julf From freeswitch-list at puzzled.xs4all.nl Tue Feb 19 14:59:58 2013 From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists) Date: Tue, 19 Feb 2013 12:59:58 +0100 Subject: [Freeswitch-users] Problems with sip provider In-Reply-To: <5123443A.704@julf.com> References: <51210FA4.1010200@julf.com> <51211478.1080108@julf.com> <5121CEBC.2020707@julf.com> <5123443A.704@julf.com> Message-ID: <5123693E.6060107@puzzled.xs4all.nl> On 02/19/2013 10:22 AM, Johan Helsingius wrote: > > > > > > > > > > > It seems like the registration goes through OK, but when I try > calling out, I get > > 013-02-19 10:00:37.236188 [NOTICE] sofia.c:6385 Hangup sofia/external/0651142219 > [CS_EXCHANGE_MEDIA] [CALL_REJECTED] I can't be of much help here other than that I found an Asterisk config that seems to work at http://www.van-sluis.nl/?p=18 [tiscali] type=peer insecure=very context=tiscali-in username=0201234567 fromuser=0201234567 secret=password qualify=yes host=sip.telefoniedienst.nl fromdomain=tel.telefoniedienst.nl dtmf=inband dtmfmode=inband canreinvite=yes nat=yes The difference with the Asterisk config is the use of realm and proxy in your config. Does your box use/offer alaw (and g729) as specified by the sip provider? Regards, Patrick From shayne.alone at gmail.com Tue Feb 19 15:11:33 2013 From: shayne.alone at gmail.com (shayne.alone at gmail.com) Date: Tue, 19 Feb 2013 15:41:33 +0330 Subject: [Freeswitch-users] multiple inbound trunks Message-ID: Hi all; I need to run a switch will handle calls from 30 different provides ( some of them have more that IPaddress as originator ) to two gate. 30 SIP providers =====> FS ==> 2 SIP gate cos there is no registration, first way is to open them in acl.conf.xml. but I like to know if there any other way ( for incoming trunks without registration ) to open / accept there calls and distinct them before dial plan to route them better and of course easily log parsing! simply management is there any idea? tnx -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130219/d55f0568/attachment.html From steveayre at gmail.com Tue Feb 19 15:24:13 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 19 Feb 2013 12:24:13 +0000 Subject: [Freeswitch-users] multiple inbound trunks In-Reply-To: References: Message-ID: http://wiki.freeswitch.org/wiki/Acl#Users On 19 February 2013 12:11, shayne.alone at gmail.com wrote: > Hi all; > > I need to run a switch will handle calls from 30 different provides ( some > of them have more that IPaddress as originator ) to two gate. > > > 30 SIP providers =====> FS ==> 2 SIP gate > > cos there is no registration, first way is to open them in acl.conf.xml. > but I like to know if there any other way ( for incoming trunks without > registration ) to open / accept there calls and distinct them before dial > plan to route them better and of course easily log parsing! simply > management > > is there any idea? > > tnx > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 vitaliy.davudov at vts24.ru Tue Feb 19 15:31:09 2013 From: vitaliy.davudov at vts24.ru (=?UTF-8?B?0JLQuNGC0LDQu9C40Lkg0JTQsNCy0YPQtNC+0LI=?=) Date: Tue, 19 Feb 2013 16:31:09 +0400 Subject: [Freeswitch-users] Memory growth In-Reply-To: References: Message-ID: <5123708D.80101@vts24.ru> My FS installed is in production network. If I'll upgrade it to latest git, and then anything goes wrong, can somebody to explain to me: how safely and fast I can back to my previous version? 18.02.2013 23:22, Ken Rice ?????: > Re: [Freeswitch-users] Memory growth We?re trying to figure out where > this last bit of leaking is coming from... Looks like it might be > related to Registrations some how... > > Once we squash that there will be some more info on the 1.2 branch coming > > > On 2/18/13 12:19 PM, "Mario G" wrote: > > FYI/in case it's related: I had a substantial memory leak on the > Oct 25 though Jan 29 Head versions and 1.2.3. After Feb 11 head, > now a much smaller leak I can't pinpoint, see > http://jira.freeswitch.org/browse/FS-5095#comment-39379. Updating > to HEAD will probably fix it, I hope someone else sees the slow > leak since I am getting nowhere real fast on it..... > Mario G > > On Feb 18, 2013, at 3:16 AM, Steven Ayre wrote: > > Are you able to reproduce it on a newer version? Try current > head of the master branch. > > If you can, then try running FS through valgrind which may > reveal the source of any memory leaks. > > -Steve > > > > On 18 February 2013 11:00, ??????? ??????? > wrote: > > Hi, list! > I've installed FreeSWITCH Version > 1.2.3+git~20120920T220849Z~f718a5e8e6 > (1.2.3; git at commit f718a5e8e6 on Thu, 20 Sep 2012 > 22:08:49 Z) > > I noticed that FreeSwtich uses a lot of memory of my > system, and percent > of used memory is gradually growing regardless of the > number of the > simultaneous calls: > > # uname -a > Linux freeswitch1 2.6.18-308.13.1.el5 #1 SMP Tue Aug 21 > 17:10:18 EDT > 2012 x86_64 x86_64 x86_64 GNU/Linux > > # top > top - 14:21:39 up 140 days, 29 min, 2 users, load > average: 0.11, 0.14, > 0.10 > Tasks: 99 total, 1 running, 96 sleeping, 2 stopped, > 0 zombie > Cpu(s): 1.2%us, 0.8%sy, 0.0%ni, 96.9%id, 0.1%wa, > 0.2%hi, 0.7%si, > 0.0%st > Mem: 3781680k total, 3748360k used, 33320k free, > 120708k buffers > Swap: 2096472k total, 17224k used, 2079248k free, > 1838356k cached > > PID USER PR NI VIRT RES SHR S %CPU %MEM > TIME+ COMMAND > 9584 root -2 -10 1792m 1.3g 8252 S 2.7 36.8 > 935:07.85 freeswitch > 3095 mysql 15 0 329m 28m 4588 S 0.7 0.8 > 713:07.03 mysqld > > At this moment FS handles no more 15 simultaneous calls, > and has about > 117 registered users. > Is there a tool to monitor, which of processes or modules > in FS is using > memory and how much? > > -- > Best regards, > Vitaly. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > ------------------------------------------------------------------------ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > -- > Ken > _http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > _irc.freenode.net #freeswitch > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -- ? ?????????? ???????????, ??????? ??????? ????????? ??? "???-???????-??????" (?????? ???????? "ETERIA") http://www.vts24.ru ???: (495) 989-47-00 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130219/6a2e195b/attachment-0001.html From sos at sokhapkin.dyndns.org Tue Feb 19 15:35:15 2013 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Tue, 19 Feb 2013 07:35:15 -0500 Subject: [Freeswitch-users] Problems with sip provider In-Reply-To: <51234C5A.70904@julf.com> References: <51210FA4.1010200@julf.com> <51234C5A.70904@julf.com> Message-ID: <2065762.1B7GdqJvp0@sos> Well, at least provide a SIP trace which INVITE request FS sends to the provider and which response the provider sends back. On Tuesday 19 February 2013 10:56:42 Johan Helsingius wrote: > > Is it possible to get a SIP trace of the calls from the gateway and FS > > and compare them? > > Unfortunately there is no way to get into the traffic between the > gateway and the ADSL connection, as it also contains the ADSL modem. > > Julf > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 avi at avimarcus.net Tue Feb 19 15:35:08 2013 From: avi at avimarcus.net (Avi Marcus) Date: Tue, 19 Feb 2013 14:35:08 +0200 Subject: [Freeswitch-users] multiple inbound trunks In-Reply-To: References: Message-ID: You can create ACLs for a user, see: http://wiki.freeswitch.org/wiki/Acl#Users In short: you can specify multiple IPs that auth to a specific user and then you can route & log accordingly. -Avi Marcus BestFone On Tue, Feb 19, 2013 at 2:11 PM, shayne.alone at gmail.com < shayne.alone at gmail.com> wrote: > Hi all; > > I need to run a switch will handle calls from 30 different provides ( some > of them have more that IPaddress as originator ) to two gate. > > > 30 SIP providers =====> FS ==> 2 SIP gate > > cos there is no registration, first way is to open them in acl.conf.xml. > but I like to know if there any other way ( for incoming trunks without > registration ) to open / accept there calls and distinct them before dial > plan to route them better and of course easily log parsing! simply > management > > is there any idea? > > tnx > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130219/63bc04a8/attachment.html From steveayre at gmail.com Tue Feb 19 15:38:15 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 19 Feb 2013 12:38:15 +0000 Subject: [Freeswitch-users] Memory growth In-Reply-To: <5123708D.80101@vts24.ru> References: <5123708D.80101@vts24.ru> Message-ID: If you're installed to /usr/local/freeswitch make a backup copy of that path. A little harder if it's installed to the LSB paths. Personally I build the debian packages and can then install the old packages. -Steve On 19 February 2013 12:31, ??????? ??????? wrote: > My FS installed is in production network. If I'll upgrade it to latest git, > and then anything goes wrong, can somebody to explain to me: how safely and > fast I can back to my previous version? > > > 18.02.2013 23:22, Ken Rice ?????: > > We're trying to figure out where this last bit of leaking is coming from... > Looks like it might be related to Registrations some how... > > Once we squash that there will be some more info on the 1.2 branch coming > > > On 2/18/13 12:19 PM, "Mario G" wrote: > > FYI/in case it's related: I had a substantial memory leak on the Oct 25 > though Jan 29 Head versions and 1.2.3. After Feb 11 head, now a much smaller > leak I can't pinpoint, see > http://jira.freeswitch.org/browse/FS-5095#comment-39379. Updating to HEAD > will probably fix it, I hope someone else sees the slow leak since I am > getting nowhere real fast on it..... > Mario G > > On Feb 18, 2013, at 3:16 AM, Steven Ayre wrote: > > Are you able to reproduce it on a newer version? Try current head of the > master branch. > > If you can, then try running FS through valgrind which may reveal the source > of any memory leaks. > > -Steve > > > > On 18 February 2013 11:00, ??????? ??????? wrote: > > Hi, list! > I've installed FreeSWITCH Version 1.2.3+git~20120920T220849Z~f718a5e8e6 > (1.2.3; git at commit f718a5e8e6 on Thu, 20 Sep 2012 22:08:49 Z) > > I noticed that FreeSwtich uses a lot of memory of my system, and percent > of used memory is gradually growing regardless of the number of the > simultaneous calls: > > # uname -a > Linux freeswitch1 2.6.18-308.13.1.el5 #1 SMP Tue Aug 21 17:10:18 EDT > 2012 x86_64 x86_64 x86_64 GNU/Linux > > # top > top - 14:21:39 up 140 days, 29 min, 2 users, load average: 0.11, 0.14, > 0.10 > Tasks: 99 total, 1 running, 96 sleeping, 2 stopped, 0 zombie > Cpu(s): 1.2%us, 0.8%sy, 0.0%ni, 96.9%id, 0.1%wa, 0.2%hi, 0.7%si, > 0.0%st > Mem: 3781680k total, 3748360k used, 33320k free, 120708k buffers > Swap: 2096472k total, 17224k used, 2079248k free, 1838356k cached > > PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND > 9584 root -2 -10 1792m 1.3g 8252 S 2.7 36.8 935:07.85 freeswitch > 3095 mysql 15 0 329m 28m 4588 S 0.7 0.8 713:07.03 mysqld > > At this moment FS handles no more 15 simultaneous calls, and has about > 117 registered users. > Is there a tool to monitor, which of processes or modules in FS is using > memory and how much? > > -- > Best regards, > Vitaly. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > ________________________________ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > -- > Ken > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > irc.freenode.net #freeswitch > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > -- > ? ?????????? ???????????, > ??????? ??????? ????????? > ??? "???-???????-??????" > (?????? ???????? "ETERIA") > http://www.vts24.ru > ???: (495) 989-47-00 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 julf at julf.com Tue Feb 19 16:03:12 2013 From: julf at julf.com (Johan Helsingius) Date: Tue, 19 Feb 2013 14:03:12 +0100 Subject: [Freeswitch-users] Problems with sip provider In-Reply-To: <2065762.1B7GdqJvp0@sos> References: <51210FA4.1010200@julf.com> <51234C5A.70904@julf.com> <2065762.1B7GdqJvp0@sos> Message-ID: <51237810.4080800@julf.com> > Well, at least provide a SIP trace which INVITE request FS sends to the > provider and which response the provider sends back. Here: http://pastebin.freeswitch.org/20605 I wonder it the problem is the Remote-Party-ID: "FreeSWITCH" in the INVITE? Julf From shaheryarkh at gmail.com Tue Feb 19 16:35:04 2013 From: shaheryarkh at gmail.com (Muhammad Shahzad) Date: Tue, 19 Feb 2013 14:35:04 +0100 Subject: [Freeswitch-users] Problems with sip provider In-Reply-To: <51237810.4080800@julf.com> References: <51210FA4.1010200@julf.com> <51234C5A.70904@julf.com> <2065762.1B7GdqJvp0@sos> <51237810.4080800@julf.com> Message-ID: To get a SIP trace from ADSL, is a little tricky but if you manage it here are the steps which may help, 1. Setup an OpenSIPS server as stateless proxy. 2. In ADSL set that server as Outbound Proxy. 3. Make a call, it will pass through OpenSIPS where you can do a SIP trace. 4. You can even modify SIP requests and replies there and get a hint of their restrictions. For example change user-agent in INVITE from ADSL to FS's user-agent, and see what happens? Does the INVITE gets rejected? etc. etc. Some security paranoid providers even do restriction based on your ADSL device's MAC address, which is issued and activated by them. Still there are ways to get into their network. All that matters is how badly you need that provider. ;-) Thank you. On Tue, Feb 19, 2013 at 2:03 PM, Johan Helsingius wrote: > > Well, at least provide a SIP trace which INVITE request FS sends to the > > provider and which response the provider sends back. > > Here: > > http://pastebin.freeswitch.org/20605 > > I wonder it the problem is the Remote-Party-ID: "FreeSWITCH" > > in the INVITE? > > Julf > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +49 176 99 83 10 85 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130219/278a6cf5/attachment-0001.html From sos at sokhapkin.dyndns.org Tue Feb 19 16:48:38 2013 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Tue, 19 Feb 2013 08:48:38 -0500 Subject: [Freeswitch-users] Problems with sip provider In-Reply-To: <51237810.4080800@julf.com> References: <51210FA4.1010200@julf.com> <2065762.1B7GdqJvp0@sos> <51237810.4080800@julf.com> Message-ID: <2362461.fVmeII454V@sos> I suspect the problem is codec-related. The provider behaves badly when g729 is not offered. On Tuesday 19 February 2013 14:03:12 Johan Helsingius wrote: > > Well, at least provide a SIP trace which INVITE request FS sends to the > > provider and which response the provider sends back. > > Here: > > http://pastebin.freeswitch.org/20605 > > I wonder it the problem is the Remote-Party-ID: "FreeSWITCH" > > in the INVITE? > > Julf > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 shaheryarkh at gmail.com Tue Feb 19 16:53:21 2013 From: shaheryarkh at gmail.com (Muhammad Shahzad) Date: Tue, 19 Feb 2013 14:53:21 +0100 Subject: [Freeswitch-users] How SIP clients able to establish a call with out Freeswitch Server? In-Reply-To: References: Message-ID: Well, perhaps they were not using FS at all. Some clients such as XLite can make call to another XLite client as long as you know SIP address of destination XLite, for example if caller has callee in its address book (where direct address of callee is stored) and both caller and callee are on same LAN OR ICE is enabled in both phones. Thank you. On Tue, Feb 19, 2013 at 10:53 AM, Muqtheear S wrote: > 1. I stopped Freeswitch server. User Agent client 1 was able to call User > Agent client 2. > 2. What configurations I need to change , so that Sip clients can use > Freeswitch server.? > > Thanks & Regards, > Muqtheear.S > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +49 176 99 83 10 85 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130219/7080c17c/attachment.html From julf at julf.com Tue Feb 19 16:55:02 2013 From: julf at julf.com (Johan Helsingius) Date: Tue, 19 Feb 2013 14:55:02 +0100 Subject: [Freeswitch-users] Problems with sip provider In-Reply-To: References: <51210FA4.1010200@julf.com> <51234C5A.70904@julf.com> <2065762.1B7GdqJvp0@sos> <51237810.4080800@julf.com> Message-ID: <51238436.9010506@julf.com> > To get a SIP trace from ADSL, is a little tricky but if you manage it here are > the steps which may help, > > 1. Setup an OpenSIPS server as stateless proxy. Yes, that would probably do it, but if I really have to go trough all that trouble, I think I might just dump that provider. > Some security paranoid providers even do restriction based on your ADSL device's > MAC address, which is issued and activated by them. Still there are ways to get > into their network. All that matters is how badly you need that provider. ;-) They do state that they do allow third-party equipment and systems, they just don't provide any customer support for them, so I assume they aren't blocking based on MAC address. But you never know... Julf From julf at julf.com Tue Feb 19 16:57:57 2013 From: julf at julf.com (Johan Helsingius) Date: Tue, 19 Feb 2013 14:57:57 +0100 Subject: [Freeswitch-users] Problems with sip provider In-Reply-To: <2362461.fVmeII454V@sos> References: <51210FA4.1010200@julf.com> <2065762.1B7GdqJvp0@sos> <51237810.4080800@julf.com> <2362461.fVmeII454V@sos> Message-ID: <512384E5.6030503@julf.com> > I suspect the problem is codec-related. The provider behaves badly when g729 > is not offered. Ah. And I guess the way to deal with that is to purchase a g729 license? Julf From sos at sokhapkin.dyndns.org Tue Feb 19 17:05:27 2013 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Tue, 19 Feb 2013 09:05:27 -0500 Subject: [Freeswitch-users] Problems with sip provider In-Reply-To: <512384E5.6030503@julf.com> References: <51210FA4.1010200@julf.com> <2362461.fVmeII454V@sos> <512384E5.6030503@julf.com> Message-ID: <2227539.Sa58f9dUxv@sos> g729 license will never hurt. On Tuesday 19 February 2013 14:57:57 Johan Helsingius wrote: > > I suspect the problem is codec-related. The provider behaves badly when > > g729 is not offered. > > Ah. And I guess the way to deal with that is to purchase a g729 license? > > Julf > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 shaheryarkh at gmail.com Tue Feb 19 17:31:33 2013 From: shaheryarkh at gmail.com (Muhammad Shahzad) Date: Tue, 19 Feb 2013 15:31:33 +0100 Subject: [Freeswitch-users] Problems with sip provider In-Reply-To: <2227539.Sa58f9dUxv@sos> References: <51210FA4.1010200@julf.com> <2362461.fVmeII454V@sos> <512384E5.6030503@julf.com> <2227539.Sa58f9dUxv@sos> Message-ID: No, 603 is generic error, codec related problems are usually 488 Not Acceptable Here. Also they are using Digitalk SBC which has no codec based restriction as far as i know. http://www.digitalk.com/carrier/Session_Border_Controller/ Thank you. On Tue, Feb 19, 2013 at 3:05 PM, Sergey Okhapkin wrote: > g729 license will never hurt. > > On Tuesday 19 February 2013 14:57:57 Johan Helsingius wrote: > > > I suspect the problem is codec-related. The provider behaves badly when > > > g729 is not offered. > > > > Ah. And I guess the way to deal with that is to purchase a g729 license? > > > > Julf > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +49 176 99 83 10 85 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130219/d835f697/attachment-0001.html From steveayre at gmail.com Tue Feb 19 17:47:09 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 19 Feb 2013 14:47:09 +0000 Subject: [Freeswitch-users] Problems with sip provider In-Reply-To: References: <51210FA4.1010200@julf.com> <2362461.fVmeII454V@sos> <512384E5.6030503@julf.com> <2227539.Sa58f9dUxv@sos> Message-ID: In theory and according to the standards yes, but they could send anything they want in practice. It'd be incorrect and annoying, but not impossible. As for the SBC, the limitation could be in their configuration rather than the software, or further along the call path. -Steve On 19 February 2013 14:31, Muhammad Shahzad wrote: > No, 603 is generic error, codec related problems are usually 488 Not > Acceptable Here. > > Also they are using Digitalk SBC which has no codec based restriction as far > as i know. > > http://www.digitalk.com/carrier/Session_Border_Controller/ > > Thank you. > > > On Tue, Feb 19, 2013 at 3:05 PM, Sergey Okhapkin > wrote: >> >> g729 license will never hurt. >> >> On Tuesday 19 February 2013 14:57:57 Johan Helsingius wrote: >> > > I suspect the problem is codec-related. The provider behaves badly >> > > when >> > > g729 is not offered. >> > >> > Ah. And I guess the way to deal with that is to purchase a g729 license? >> > >> > Julf >> > >> > >> > >> > >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > 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 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 > > > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +49 176 99 83 10 85 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 julf at julf.com Tue Feb 19 17:58:24 2013 From: julf at julf.com (Johan Helsingius) Date: Tue, 19 Feb 2013 15:58:24 +0100 Subject: [Freeswitch-users] Problems with sip provider In-Reply-To: References: <51210FA4.1010200@julf.com> <2362461.fVmeII454V@sos> <512384E5.6030503@julf.com> <2227539.Sa58f9dUxv@sos> Message-ID: <51239310.6050906@julf.com> > No, 603 is generic error, codec related problems are usually 488 Not Acceptable > Here. Right - and I verified it by installing a g729 license and the mod_com_g729 module - no difference. Julf From ctroncoso at redvoiss.net Tue Feb 19 18:12:16 2013 From: ctroncoso at redvoiss.net (Camila Troncoso) Date: Tue, 19 Feb 2013 12:12:16 -0300 Subject: [Freeswitch-users] Kamailio+Freeswitch SBC Message-ID: Hi all, I?m working for a while with Kamailio+Freeswitch as SBC. I have this structure: * * * * * * * * * * * * * * * * When I make a call, from one client coneccted to LCR, it is route to my SBC and afterwards to his destiny in the cloud passing thourgh a gateway. When the destiny is unreachable, the LCR reroutes the call to another gateway. Sometimes the LCR send this retry again to the SBC , becouse the second gateway is also in the cloud, but when this happens, Fresswitch answer with : ?482 Request merged?. Is there a way to solve this? Regards, * * *Camila Troncoso **|* Ingeniero de Desarrollo RedVoiss *|*ctroncoso at redvoiss.net Santiago - Chile *|* +56 2 2408535 www.redvoiss.net -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130219/e5c505b3/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/octet-stream Size: 119905 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130219/e5c505b3/attachment-0002.obj -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/octet-stream Size: 54790 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130219/e5c505b3/attachment-0003.obj From steveayre at gmail.com Tue Feb 19 18:26:37 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 19 Feb 2013 15:26:37 +0000 Subject: [Freeswitch-users] Kamailio+Freeswitch SBC In-Reply-To: References: Message-ID: This probably means the INVITE FS receives is a duplicate of one it has already received, so it's assuming the client is resending a duplicate INVITE for a call it has already handled. -Steve On 19 February 2013 15:12, Camila Troncoso wrote: > Hi all, > > > > I?m working for a while with Kamailio+Freeswitch as SBC. > > of s > > I have this structure: > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > * * > > * * > > * * > > * * > > * * > > * * > > * * > > * * > > When I make a call, from one client coneccted to LCR, it is route to my > SBC and afterwards to his destiny in the cloud passing thourgh a gateway. > When the destiny is unreachable, the LCR reroutes the call to another > gateway. Sometimes the LCR send this retry again to the SBC , becouse the > second gateway is also in the cloud, but when this happens, Fresswitch > answer with : > > > > ?482 Request merged?. > > > > Is there a way to solve this? > > > > Regards, > > * * > > *Camila Troncoso **|* Ingeniero de Desarrollo > > RedVoiss *|*ctroncoso at redvoiss.net > > Santiago - Chile *|* +56 2 2408535 > > www.redvoiss.net > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130219/389f00d3/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/octet-stream Size: 54790 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130219/389f00d3/attachment-0001.obj From ctroncoso at redvoiss.net Tue Feb 19 18:35:25 2013 From: ctroncoso at redvoiss.net (Camila Troncoso) Date: Tue, 19 Feb 2013 12:35:25 -0300 Subject: [Freeswitch-users] Kamailio+Freeswitch SBC In-Reply-To: References: Message-ID: <1f3504ddd1087b68aa2a77bd76755a2e@mail.gmail.com> Steve, The Invite is the same , because the LCR retries the call to the second route he has. The only change is in the *Ruri* part. How can I handle this. I want FreeSWITCH to handle this second invite for the call to reach his destiny. Regards, Camila T. *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Steven Ayre *Sent:* martes, 19 de febrero de 2013 12:27 *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Kamailio+Freeswitch SBC This probably means the INVITE FS receives is a duplicate of one it has already received, so it's assuming the client is resending a duplicate INVITE for a call it has already handled. -Steve On 19 February 2013 15:12, Camila Troncoso wrote: Hi all, I?m working for a while with Kamailio+Freeswitch as SBC. of s I have this structure: * * * * * * * * * * * * * * * * When I make a call, from one client coneccted to LCR, it is route to my SBC and afterwards to his destiny in the cloud passing thourgh a gateway. When the destiny is unreachable, the LCR reroutes the call to another gateway. Sometimes the LCR send this retry again to the SBC , becouse the second gateway is also in the cloud, but when this happens, Fresswitch answer with : ?482 Request merged?. Is there a way to solve this? Regards, * * *Camila Troncoso **|* Ingeniero de Desarrollo RedVoiss *|*ctroncoso at redvoiss.net Santiago - Chile *|* +56 2 2408535 www.redvoiss.net _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130219/8cadec88/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/octet-stream Size: 54790 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130219/8cadec88/attachment-0001.obj From shaheryarkh at gmail.com Tue Feb 19 18:42:01 2013 From: shaheryarkh at gmail.com (Muhammad Shahzad) Date: Tue, 19 Feb 2013 16:42:01 +0100 Subject: [Freeswitch-users] Problems with sip provider In-Reply-To: <51239310.6050906@julf.com> References: <51210FA4.1010200@julf.com> <2362461.fVmeII454V@sos> <512384E5.6030503@julf.com> <2227539.Sa58f9dUxv@sos> <51239310.6050906@julf.com> Message-ID: Just talked to a friend, he says though Digitalk SBC is extremely configurable, but most providers specially in NL (due to its telecom and legal interception laws etc.), force customers to provide both caller and callee ids correctly. So, you need at least to make sure, 1. Destination number is in correct format, usually E.164 number but provider may choose differently, e.g. give you some dial prefix to dial out etc. 2. You are using Remote Part ID in correct format, usually the format is same as destination number format. 3. If using anonymous caller id then make sure to set Privacy, P-Asserted Identity and / or P-Preferred Identity. Try these if work then good otherwise just dump the carrier. Thank you. On Tue, Feb 19, 2013 at 3:58 PM, Johan Helsingius wrote: > > No, 603 is generic error, codec related problems are usually 488 Not > Acceptable > > Here. > > Right - and I verified it by installing a g729 license and the > mod_com_g729 module - no difference. > > Julf > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +49 176 99 83 10 85 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130219/1d75fd4d/attachment.html From steveayre at gmail.com Tue Feb 19 18:50:44 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 19 Feb 2013 15:50:44 +0000 Subject: [Freeswitch-users] Kamailio+Freeswitch SBC In-Reply-To: <1f3504ddd1087b68aa2a77bd76755a2e@mail.gmail.com> References: <1f3504ddd1087b68aa2a77bd76755a2e@mail.gmail.com> Message-ID: You should increment the CSeq for the next INVITE. FreeSWITCH is complying with the SIP standard which does this to cope with packet loss. -Steve On 19 February 2013 15:35, Camila Troncoso wrote: > Steve, > > > > The Invite is the same , because the LCR retries the call to the second > route he has. The only change is in the *Ruri* part. How can I handle > this. I want FreeSWITCH to handle this second invite for the call to reach > his destiny. > > > > Regards, > > > > Camila T. > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Steven Ayre > *Sent:* martes, 19 de febrero de 2013 12:27 > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Kamailio+Freeswitch SBC > > > > This probably means the INVITE FS receives is a duplicate of one it has > already received, so it's assuming the client is resending a duplicate > INVITE for a call it has already handled. > > > > -Steve > > > > > > On 19 February 2013 15:12, Camila Troncoso wrote: > > Hi all, > > > > I?m working for a while with Kamailio+Freeswitch as SBC. > > of s > > I have this structure: > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > * * > > * * > > * * > > * * > > * * > > * * > > * * > > * * > > When I make a call, from one client coneccted to LCR, it is route to my > SBC and afterwards to his destiny in the cloud passing thourgh a gateway. > When the destiny is unreachable, the LCR reroutes the call to another > gateway. Sometimes the LCR send this retry again to the SBC , becouse the > second gateway is also in the cloud, but when this happens, Fresswitch > answer with : > > > > ?482 Request merged?. > > > > Is there a way to solve this? > > > > Regards, > > * * > > *Camila Troncoso **|* Ingeniero de Desarrollo > > RedVoiss *|*ctroncoso at redvoiss.net > > Santiago - Chile *|* +56 2 2408535 > > www.redvoiss.net > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130219/f5e8ce0b/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/octet-stream Size: 54790 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130219/f5e8ce0b/attachment-0001.obj From a.venugopan at mundio.com Tue Feb 19 19:04:26 2013 From: a.venugopan at mundio.com (Archana Venugopan) Date: Tue, 19 Feb 2013 16:04:26 +0000 Subject: [Freeswitch-users] CHANNEL_DATA variables Message-ID: <592A9CF93E12394E8472A6CC66E66BF23AC429@Mail-Kilo.squay.com> Hi, I could see that I have been getting the below variables in logs when I dial my voicemail number. But when I make internal or external calls, I don't see these variables in logs. 910672e4-6ec8-11e2-8027-7dea0d27034c 2013-02-04 12:44:08.027128 [INFO] mod_dptools.c:1421 CHANNEL_DATA: Channel-State: [CS_EXECUTE] Channel-Call-State: [RINGING] Channel-State-Number: [4] Channel-Name: [sofia/internal/367 at vectone.uk01.net] Unique-ID: [910672e4-6ec8-11e2-8027-7dea0d27034c] Call-Direction: [inbound] Presence-Call-Direction: [inbound] Channel-HIT-Dialplan: [true] Channel-Presence-ID: [367 at vectone.uk01.net] Channel-Call-UUID: [910672e4-6ec8-11e2-8027-7dea0d27034c] Answer-State: [ringing] Caller-Direction: [inbound] Caller-Username: [367] Caller-Dialplan: [XML] Caller-Caller-ID-Name: [Test] Caller-Caller-ID-Number: [367] Caller-Network-Addr: [192.168.5.129] Can anyone please tell me how am I to get these variables if i make internal or external calls as well as like my voicemail call? Thanks Regards, Archana -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130219/bbd55566/attachment.html From shayne.alone at gmail.com Tue Feb 19 19:05:35 2013 From: shayne.alone at gmail.com (shayne.alone at gmail.com) Date: Tue, 19 Feb 2013 19:35:35 +0330 Subject: [Freeswitch-users] multiple inbound trunks In-Reply-To: References: Message-ID: nice :-) On Tue, Feb 19, 2013 at 4:05 PM, Avi Marcus wrote: > You can create ACLs for a user, see: > http://wiki.freeswitch.org/wiki/Acl#Users > In short: you can specify multiple IPs that auth to a specific user and > then you can route & log accordingly. > > -Avi Marcus > BestFone > > On Tue, Feb 19, 2013 at 2:11 PM, shayne.alone at gmail.com < > shayne.alone at gmail.com> wrote: > >> Hi all; >> >> I need to run a switch will handle calls from 30 different provides ( >> some of them have more that IPaddress as originator ) to two gate. >> >> >> 30 SIP providers =====> FS ==> 2 SIP gate >> >> cos there is no registration, first way is to open them in acl.conf.xml. >> but I like to know if there any other way ( for incoming trunks without >> registration ) to open / accept there calls and distinct them before dial >> plan to route them better and of course easily log parsing! simply >> management >> >> is there any idea? >> >> tnx >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Regards, Ali R. 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URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130219/2541da19/attachment.html From kaothics at hotmail.com Tue Feb 19 19:07:16 2013 From: kaothics at hotmail.com (PolyMorphiK) Date: Tue, 19 Feb 2013 08:07:16 -0800 (PST) Subject: [Freeswitch-users] troubles with Mod_gsmopen and Huawei E153 In-Reply-To: <1361243861053-7587637.post@n2.nabble.com> References: <1361204375768-7587585.post@n2.nabble.com> <1361223294004-7587628.post@n2.nabble.com> <1361228596913-7587633.post@n2.nabble.com> <1361243861053-7587637.post@n2.nabble.com> Message-ID: <1361290036149-7587679.post@n2.nabble.com> last tries with Oktell SIP-GSM, works fine with huawei E153 and Freeswitch. on Oktell setup to make it work i use Router Mode settings SIP account is a basic extension setup default port i use for SIP server 5060 nd local port 5070 outbound-proxy check As a SIP server Default SIP-Number (the extension number you wish to receive the incoming calls when in gateway mode) CallerID Send Caller ID nd Send in "From" default GSM number blank (it's used when in gateway mode) Codecs : 1 G.711Ulaw 2 G.711Alaw 3 G.729 4 GSM 610 additional setting DTMF type RFC2833 to dial out ring the extension , wait the tonality , dial and it's done i didn't try yet the gateway mode. else i wonder if i can make a lua script for dialing. P.S: about gsmopen com port and how to find them on Oktell SIP-GSM you can find log files between the modem and the computer to have the right adresses but still doesnt work on GSMOPEN Cordialy -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/troubles-with-Mod-gsmopen-and-Huawei-E153-tp7587585p7587679.html Sent from the freeswitch-users mailing list archive at Nabble.com. From ctroncoso at redvoiss.net Tue Feb 19 19:05:31 2013 From: ctroncoso at redvoiss.net (Camila Troncoso) Date: Tue, 19 Feb 2013 13:05:31 -0300 Subject: [Freeswitch-users] Kamailio+Freeswitch SBC In-Reply-To: References: <1f3504ddd1087b68aa2a77bd76755a2e@mail.gmail.com> Message-ID: Thanks fot the reply, The problem is that Kamailio LCR doesn?t increment the Cseq by itself in the second invite , I will try to do this by configuration. Regards, Camila T *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Steven Ayre *Sent:* martes, 19 de febrero de 2013 12:51 *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Kamailio+Freeswitch SBC You should increment the CSeq for the next INVITE. FreeSWITCH is complying with the SIP standard which does this to cope with packet loss. -Steve On 19 February 2013 15:35, Camila Troncoso wrote: Steve, The Invite is the same , because the LCR retries the call to the second route he has. The only change is in the *Ruri* part. How can I handle this. I want FreeSWITCH to handle this second invite for the call to reach his destiny. Regards, Camila T. *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Steven Ayre *Sent:* martes, 19 de febrero de 2013 12:27 *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Kamailio+Freeswitch SBC This probably means the INVITE FS receives is a duplicate of one it has already received, so it's assuming the client is resending a duplicate INVITE for a call it has already handled. -Steve On 19 February 2013 15:12, Camila Troncoso wrote: Hi all, I?m working for a while with Kamailio+Freeswitch as SBC. of s I have this structure: * * * * * * * * * * * * * * * * When I make a call, from one client coneccted to LCR, it is route to my SBC and afterwards to his destiny in the cloud passing thourgh a gateway. When the destiny is unreachable, the LCR reroutes the call to another gateway. Sometimes the LCR send this retry again to the SBC , becouse the second gateway is also in the cloud, but when this happens, Fresswitch answer with : ?482 Request merged?. Is there a way to solve this? Regards, * * *Camila Troncoso **|* Ingeniero de Desarrollo RedVoiss *|*ctroncoso at redvoiss.net Santiago - Chile *|* +56 2 2408535 www.redvoiss.net _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130219/78928d58/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/octet-stream Size: 54790 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130219/78928d58/attachment-0001.obj From gmaruzz at gmail.com Tue Feb 19 19:23:49 2013 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Tue, 19 Feb 2013 17:23:49 +0100 Subject: [Freeswitch-users] troubles with Mod_gsmopen and Huawei E153 In-Reply-To: <1361290036149-7587679.post@n2.nabble.com> References: <1361204375768-7587585.post@n2.nabble.com> <1361223294004-7587628.post@n2.nabble.com> <1361228596913-7587633.post@n2.nabble.com> <1361243861053-7587637.post@n2.nabble.com> <1361290036149-7587679.post@n2.nabble.com> Message-ID: On Tue, Feb 19, 2013 at 5:07 PM, PolyMorphiK wrote: > P.S: about gsmopen com port and how to find them on Oktell SIP-GSM you can > find log files between the modem and the computer to have the right > adresses > but still doesnt work on GSMOPEN > > Maybe (but just maybe, shooting in the dark here) is a problem of ownership/permission on the ports (eg: need to have dredentials or run as to use the port)? -giovanni -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130219/f1752cea/attachment.html From alex at digitalmail.com Tue Feb 19 19:21:05 2013 From: alex at digitalmail.com (Alex Lake) Date: Tue, 19 Feb 2013 16:21:05 +0000 Subject: [Freeswitch-users] Running sox from within lua Message-ID: <5123A671.4040608@digitalmail.com> Having a spot of bother with this bit of lua: result = os.execute(string.format([[sox --norm %s %s.new]], file_to_normalise, file_to_normalise)) It's returning -1. I believe freeswitch is running as root and that the sox file is in that location and is runnable. Any ideas? From emamirazavi at gmail.com Tue Feb 19 19:12:45 2013 From: emamirazavi at gmail.com (Sayyed Mohammad Emami Razavi) Date: Tue, 19 Feb 2013 19:42:45 +0330 Subject: [Freeswitch-users] Benchmark and making session and bridging virtual devices.... Message-ID: How to test server capacity to load many freeswitch sessions with ready voice recording and ... features? I want to make at least 2000 virtual sessions(1000 concurrent calls). How to do this? Does you have any idea? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130219/e3a13643/attachment.html From lists at kavun.ch Tue Feb 19 19:37:44 2013 From: lists at kavun.ch (Emrah) Date: Tue, 19 Feb 2013 11:37:44 -0500 Subject: [Freeswitch-users] Changing codec during calls In-Reply-To: References: <6EF8B8E9-4CC5-480C-B1C1-09CBC533470C@kavun.ch> <35AF9E79-771D-440A-BEBB-A856D831161C@kavun.ch> Message-ID: Excellent, that worked. However I am having a one way audio issue, where the peer that has its codecs changed can hear but cannot be heard. Even if both bridged peers are changed to a common codec it doesn't work. Any idea? Thanks a bunch for your help. On Feb 19, 2013, at 3:47 AM, Steven Ayre wrote: > Try: > uuid_media_reneg 49f428e5-1d72-42d8-b0e1-5b3f29f9a4ca =PCMU > > (ie add = before PCMU) > > It appears to affect something in the code... > > -Steve > > > On 19 February 2013 05:54, Emrah wrote: >> Hey, >> >> In short, nothing seems to happen. I tried several phones. >> I also tried enabling codec renegotiation on hold/unhold thinking that the change in SDP with uuid_media_reneg would do it. >> >> This is what happens: >> freeswitch at internal> show channels >> uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,sent_callee_name,sent_callee_num >> 49f428e5-1d72-42d8-b0e1-5b3f29f9a4ca,inbound,2013-02-19 00:44:35,1361252675,sofia/internal/10000 at sip.domain.net,CS_EXECUTE,Emrah - Macbook Pro,10000,100.100.100.100,conf-3111,conference,3111-sip.domain.net at ek-custom-16k,XML,ek-conference,G722,16000,64000,G722,16000,64000,,cornflakes.domain.net,10000 at sip.domain.net,,ACTIVE,,,,,, >> >> 1 total. >> >> freeswitch at internal> uuid_media_reneg 49f428e5-1d72-42d8-b0e1-5b3f29f9a4ca PCMU >> +OK Success >> >> 2013-02-19 00:45:25.172659 [DEBUG] sofia_glue.c:2647 Local SDP: >> v=0 >> o=FreeSWITCH 1361234787 1361234790 IN IP4 1.2.3.4 >> s=FreeSWITCH >> c=IN IP4 1.2.3.4 >> t=0 0 >> m=audio 17888 RTP/AVP 9 0 101 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=silenceSupp:off - - - - >> a=ptime:20 >> a=sendrecv >> >> 2013-02-19 00:45:25.172659 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/10000 at sip.domain.net [BREAK] >> 2013-02-19 00:45:25.182652 [DEBUG] sofia.c:5574 Channel sofia/internal/10000 at sip.domain.net entering state [calling][0] >> 2013-02-19 00:45:25.292661 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/10000 at sip.domain.net [BREAK] >> 2013-02-19 00:45:25.292661 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/10000 at sip.domain.net [BREAK] >> 2013-02-19 00:45:25.302683 [INFO] sofia.c:931 sofia/internal/10000 at sip.domain.net Update Callee ID to "10000" <10000> >> 2013-02-19 00:45:25.302683 [DEBUG] sofia.c:5574 Channel sofia/internal/10000 at sip.domain.net entering state [completing][200] >> 2013-02-19 00:45:25.302683 [DEBUG] sofia.c:5585 Remote SDP: >> v=0 >> o=- 3570241475 3570241476 IN IP4 10.0.0.131 >> s=pjmedia >> c=IN IP4 10.0.0.131 >> t=0 0 >> a=X-nat:0 >> m=audio 4010 RTP/AVP 9 101 >> a=rtcp:4011 IN IP4 10.0.0.131 >> a=rtpmap:9 G722/8000 >> a=sendrecv >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-15 >> >> 2013-02-19 00:45:25.302683 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/10000 at sip.domain.net [BREAK] >> 2013-02-19 00:45:25.302683 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/10000 at sip.domain.net [BREAK] >> 2013-02-19 00:45:25.322657 [DEBUG] sofia.c:5574 Channel sofia/internal/10000 at sip.domain.net entering state [ready][200] >> 2013-02-19 00:45:25.322657 [DEBUG] sofia.c:5582 Duplicate SDP >> v=0 >> o=- 3570241475 3570241476 IN IP4 10.0.0.131 >> s=pjmedia >> c=IN IP4 10.0.0.131 >> t=0 0 >> a=X-nat:0 >> m=audio 4010 RTP/AVP 9 101 >> a=rtcp:4011 IN IP4 10.0.0.131 >> a=rtpmap:9 G722/8000 >> a=sendrecv >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-15 >> >> 2013-02-19 00:45:25.322657 [DEBUG] sofia_glue.c:5137 Audio Codec Compare [G722:9:8000:20:64000]/[G722:9:8000:20:64000] >> 2013-02-19 00:45:25.322657 [DEBUG] sofia_glue.c:3022 Already using G722 >> 2013-02-19 00:45:25.322657 [DEBUG] sofia_glue.c:5266 Set 2833 dtmf send/recv payload to 101 >> >> >> Thanks for any help, >> Emrah >> On Feb 18, 2013, at 1:38 PM, Michael Collins wrote: >> >>> What happened with the uuid_media_reneg command? >>> -MC >>> >>> On Sat, Feb 16, 2013 at 3:10 PM, Emrah wrote: >>> Hi all, >>> >>> How do we go about renegotiating the codecs of a call that is already established? >>> I tried uuid_media_reneg to no avail. >>> E.g.: Upgrade someone who's entered a 16k conference from PCMU to G722. >>> >>> Cheers and thanks, >>> Emrah >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >>> >>> -- >>> Michael S Collins >>> Twitter: @mercutioviz >>> http://www.FreeSWITCH.org >>> http://www.ClueCon.com >>> http://www.OSTAG.org >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 sdevoy at bizfocused.com Tue Feb 19 19:40:05 2013 From: sdevoy at bizfocused.com (Sean Devoy) Date: Tue, 19 Feb 2013 11:40:05 -0500 Subject: [Freeswitch-users] Determine if a number is valid/working Message-ID: <072701ce0ebf$c5808640$508192c0$@bizfocused.com> HI All, My web site customer's call center claims that 1/3 of their web leads have nonworking phone numbers. Short of calling it, is there any way with FS (or even a paid service) to determine if a given phone number is a working number? I would like to make an Ajax SOAP call to a PHP page that checks (or dials) the number and returns a status for the validity of the phone number. I have an LUA script that will call, play a short message and hang up, but the customer thinks that may take too long. Any ideas? Thanks, Sean -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130219/c2b2de28/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 70 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130219/c2b2de28/attachment.gif From msc at freeswitch.org Tue Feb 19 19:44:12 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 19 Feb 2013 08:44:12 -0800 Subject: [Freeswitch-users] CHANNEL_DATA variables In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF23AC429@Mail-Kilo.squay.com> References: <592A9CF93E12394E8472A6CC66E66BF23AC429@Mail-Kilo.squay.com> Message-ID: off the top of my head I don't know why, although the first thing I would try is setting verbose_events to true and retesting. -MC On Tue, Feb 19, 2013 at 8:04 AM, Archana Venugopan wrote: > Hi,**** > > ** ** > > I could see that I have been getting the below variables in logs when I > dial my voicemail number. But when I make internal or external calls, I > don?t see these variables in logs. **** > > ** ** > > 910672e4-6ec8-11e2-8027-7dea0d27034c 2013-02-04 12:44:08.027128 [INFO] > mod_dptools.c:1421 CHANNEL_DATA:**** > > Channel-State: [CS_EXECUTE]**** > > Channel-Call-State: [RINGING]**** > > Channel-State-Number: [4]**** > > Channel-Name: [sofia/internal/367 at vectone.uk01.net]**** > > Unique-ID: [910672e4-6ec8-11e2-8027-7dea0d27034c]**** > > Call-Direction: [inbound]**** > > Presence-Call-Direction: [inbound]**** > > Channel-HIT-Dialplan: [true]**** > > Channel-Presence-ID: [367 at vectone.uk01.net]**** > > Channel-Call-UUID: [910672e4-6ec8-11e2-8027-7dea0d27034c]**** > > Answer-State: [ringing]**** > > Caller-Direction: [inbound]**** > > Caller-Username: [367]**** > > Caller-Dialplan: [XML]**** > > Caller-Caller-ID-Name: [Test]**** > > Caller-Caller-ID-Number: [367]**** > > Caller-Network-Addr: [192.168.5.129]**** > > ** ** > > Can anyone please tell me how am I to get these variables if i make > internal or external calls as well as like my voicemail call? Thanks**** > > ** ** > > ** ** > > ** ** > > Regards,**** > > Archana**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130219/685b1ba6/attachment-0001.html From msc at freeswitch.org Tue Feb 19 19:47:17 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 19 Feb 2013 08:47:17 -0800 Subject: [Freeswitch-users] Running sox from within lua In-Reply-To: <5123A671.4040608@digitalmail.com> References: <5123A671.4040608@digitalmail.com> Message-ID: Can you run a simple Lua script and try something like this: res=os.execute('/usr/bin/sox --version') Just to make absolutely certain that you are able to call sox from within a Lua script. -MC On Tue, Feb 19, 2013 at 8:21 AM, Alex Lake wrote: > Having a spot of bother with this bit of lua: > > result = os.execute(string.format([[sox --norm %s %s.new]], > file_to_normalise, file_to_normalise)) > > It's returning -1. I believe freeswitch is running as root and that the > sox file is in that location and is runnable. > > Any ideas? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130219/ca4b6f43/attachment.html From steveayre at gmail.com Tue Feb 19 19:46:53 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 19 Feb 2013 16:46:53 +0000 Subject: [Freeswitch-users] Changing codec during calls In-Reply-To: References: <6EF8B8E9-4CC5-480C-B1C1-09CBC533470C@kavun.ch> <35AF9E79-771D-440A-BEBB-A856D831161C@kavun.ch> Message-ID: Great. The code seems to show that the default is to append it to the current codec list, but = replaces the current codec list instead. I'll add a comment to that effect to the Wiki. As far the bridged peers goes, if you do a packet trace of the RTP do you see media with the correct codec being sent to the client? That might shed light on whether it's no media sent, wrong codec, or the some problem/bug on the peer. Steve On 19 February 2013 16:37, Emrah wrote: > Excellent, that worked. > > However I am having a one way audio issue, where the peer that has its codecs changed can hear but cannot be heard. > Even if both bridged peers are changed to a common codec it doesn't work. > > Any idea? > > Thanks a bunch for your help. > On Feb 19, 2013, at 3:47 AM, Steven Ayre wrote: > >> Try: >> uuid_media_reneg 49f428e5-1d72-42d8-b0e1-5b3f29f9a4ca =PCMU >> >> (ie add = before PCMU) >> >> It appears to affect something in the code... >> >> -Steve >> >> >> On 19 February 2013 05:54, Emrah wrote: >>> Hey, >>> >>> In short, nothing seems to happen. I tried several phones. >>> I also tried enabling codec renegotiation on hold/unhold thinking that the change in SDP with uuid_media_reneg would do it. >>> >>> This is what happens: >>> freeswitch at internal> show channels >>> uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,sent_callee_name,sent_callee_num >>> 49f428e5-1d72-42d8-b0e1-5b3f29f9a4ca,inbound,2013-02-19 00:44:35,1361252675,sofia/internal/10000 at sip.domain.net,CS_EXECUTE,Emrah - Macbook Pro,10000,100.100.100.100,conf-3111,conference,3111-sip.domain.net at ek-custom-16k,XML,ek-conference,G722,16000,64000,G722,16000,64000,,cornflakes.domain.net,10000 at sip.domain.net,,ACTIVE,,,,,, >>> >>> 1 total. >>> >>> freeswitch at internal> uuid_media_reneg 49f428e5-1d72-42d8-b0e1-5b3f29f9a4ca PCMU >>> +OK Success >>> >>> 2013-02-19 00:45:25.172659 [DEBUG] sofia_glue.c:2647 Local SDP: >>> v=0 >>> o=FreeSWITCH 1361234787 1361234790 IN IP4 1.2.3.4 >>> s=FreeSWITCH >>> c=IN IP4 1.2.3.4 >>> t=0 0 >>> m=audio 17888 RTP/AVP 9 0 101 >>> a=rtpmap:101 telephone-event/8000 >>> a=fmtp:101 0-16 >>> a=silenceSupp:off - - - - >>> a=ptime:20 >>> a=sendrecv >>> >>> 2013-02-19 00:45:25.172659 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/10000 at sip.domain.net [BREAK] >>> 2013-02-19 00:45:25.182652 [DEBUG] sofia.c:5574 Channel sofia/internal/10000 at sip.domain.net entering state [calling][0] >>> 2013-02-19 00:45:25.292661 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/10000 at sip.domain.net [BREAK] >>> 2013-02-19 00:45:25.292661 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/10000 at sip.domain.net [BREAK] >>> 2013-02-19 00:45:25.302683 [INFO] sofia.c:931 sofia/internal/10000 at sip.domain.net Update Callee ID to "10000" <10000> >>> 2013-02-19 00:45:25.302683 [DEBUG] sofia.c:5574 Channel sofia/internal/10000 at sip.domain.net entering state [completing][200] >>> 2013-02-19 00:45:25.302683 [DEBUG] sofia.c:5585 Remote SDP: >>> v=0 >>> o=- 3570241475 3570241476 IN IP4 10.0.0.131 >>> s=pjmedia >>> c=IN IP4 10.0.0.131 >>> t=0 0 >>> a=X-nat:0 >>> m=audio 4010 RTP/AVP 9 101 >>> a=rtcp:4011 IN IP4 10.0.0.131 >>> a=rtpmap:9 G722/8000 >>> a=sendrecv >>> a=rtpmap:101 telephone-event/8000 >>> a=fmtp:101 0-15 >>> >>> 2013-02-19 00:45:25.302683 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/10000 at sip.domain.net [BREAK] >>> 2013-02-19 00:45:25.302683 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/10000 at sip.domain.net [BREAK] >>> 2013-02-19 00:45:25.322657 [DEBUG] sofia.c:5574 Channel sofia/internal/10000 at sip.domain.net entering state [ready][200] >>> 2013-02-19 00:45:25.322657 [DEBUG] sofia.c:5582 Duplicate SDP >>> v=0 >>> o=- 3570241475 3570241476 IN IP4 10.0.0.131 >>> s=pjmedia >>> c=IN IP4 10.0.0.131 >>> t=0 0 >>> a=X-nat:0 >>> m=audio 4010 RTP/AVP 9 101 >>> a=rtcp:4011 IN IP4 10.0.0.131 >>> a=rtpmap:9 G722/8000 >>> a=sendrecv >>> a=rtpmap:101 telephone-event/8000 >>> a=fmtp:101 0-15 >>> >>> 2013-02-19 00:45:25.322657 [DEBUG] sofia_glue.c:5137 Audio Codec Compare [G722:9:8000:20:64000]/[G722:9:8000:20:64000] >>> 2013-02-19 00:45:25.322657 [DEBUG] sofia_glue.c:3022 Already using G722 >>> 2013-02-19 00:45:25.322657 [DEBUG] sofia_glue.c:5266 Set 2833 dtmf send/recv payload to 101 >>> >>> >>> Thanks for any help, >>> Emrah >>> On Feb 18, 2013, at 1:38 PM, Michael Collins wrote: >>> >>>> What happened with the uuid_media_reneg command? >>>> -MC >>>> >>>> On Sat, Feb 16, 2013 at 3:10 PM, Emrah wrote: >>>> Hi all, >>>> >>>> How do we go about renegotiating the codecs of a call that is already established? >>>> I tried uuid_media_reneg to no avail. >>>> E.g.: Upgrade someone who's entered a 16k conference from PCMU to G722. >>>> >>>> Cheers and thanks, >>>> Emrah >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>>> >>>> >>>> >>>> -- >>>> Michael S Collins >>>> Twitter: @mercutioviz >>>> http://www.FreeSWITCH.org >>>> http://www.ClueCon.com >>>> http://www.OSTAG.org >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 msc at freeswitch.org Tue Feb 19 19:48:30 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 19 Feb 2013 08:48:30 -0800 Subject: [Freeswitch-users] Changing codec during calls In-Reply-To: References: <6EF8B8E9-4CC5-480C-B1C1-09CBC533470C@kavun.ch> <35AF9E79-771D-440A-BEBB-A856D831161C@kavun.ch> Message-ID: pastebin the log. I'll bet Steven can diagnose it! ;) -MC On Tue, Feb 19, 2013 at 8:37 AM, Emrah wrote: > Excellent, that worked. > > However I am having a one way audio issue, where the peer that has its > codecs changed can hear but cannot be heard. > Even if both bridged peers are changed to a common codec it doesn't work. > > Any idea? > > Thanks a bunch for your help. > On Feb 19, 2013, at 3:47 AM, Steven Ayre wrote: > > > Try: > > uuid_media_reneg 49f428e5-1d72-42d8-b0e1-5b3f29f9a4ca =PCMU > > > > (ie add = before PCMU) > > > > It appears to affect something in the code... > > > > -Steve > > > > > > On 19 February 2013 05:54, Emrah wrote: > >> Hey, > >> > >> In short, nothing seems to happen. I tried several phones. > >> I also tried enabling codec renegotiation on hold/unhold thinking that > the change in SDP with uuid_media_reneg would do it. > >> > >> This is what happens: > >> freeswitch at internal> show channels > >> > uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,sent_callee_name,sent_callee_num > >> 49f428e5-1d72-42d8-b0e1-5b3f29f9a4ca,inbound,2013-02-19 > 00:44:35,1361252675,sofia/internal/10000 at sip.domain.net,CS_EXECUTE,Emrah > - Macbook > Pro,10000,100.100.100.100,conf-3111,conference,3111-sip.domain.net at ek-custom-16k > ,XML,ek-conference,G722,16000,64000,G722,16000,64000,, > cornflakes.domain.net,10000 at sip.domain.net,,ACTIVE,,,,,, > >> > >> 1 total. > >> > >> freeswitch at internal> uuid_media_reneg > 49f428e5-1d72-42d8-b0e1-5b3f29f9a4ca PCMU > >> +OK Success > >> > >> 2013-02-19 00:45:25.172659 [DEBUG] sofia_glue.c:2647 Local SDP: > >> v=0 > >> o=FreeSWITCH 1361234787 1361234790 IN IP4 1.2.3.4 > >> s=FreeSWITCH > >> c=IN IP4 1.2.3.4 > >> t=0 0 > >> m=audio 17888 RTP/AVP 9 0 101 > >> a=rtpmap:101 telephone-event/8000 > >> a=fmtp:101 0-16 > >> a=silenceSupp:off - - - - > >> a=ptime:20 > >> a=sendrecv > >> > >> 2013-02-19 00:45:25.172659 [DEBUG] switch_core_session.c:975 Send > signal sofia/internal/10000 at sip.domain.net [BREAK] > >> 2013-02-19 00:45:25.182652 [DEBUG] sofia.c:5574 Channel sofia/internal/ > 10000 at sip.domain.net entering state [calling][0] > >> 2013-02-19 00:45:25.292661 [DEBUG] switch_core_session.c:975 Send > signal sofia/internal/10000 at sip.domain.net [BREAK] > >> 2013-02-19 00:45:25.292661 [DEBUG] switch_core_session.c:975 Send > signal sofia/internal/10000 at sip.domain.net [BREAK] > >> 2013-02-19 00:45:25.302683 [INFO] sofia.c:931 sofia/internal/ > 10000 at sip.domain.net Update Callee ID to "10000" <10000> > >> 2013-02-19 00:45:25.302683 [DEBUG] sofia.c:5574 Channel sofia/internal/ > 10000 at sip.domain.net entering state [completing][200] > >> 2013-02-19 00:45:25.302683 [DEBUG] sofia.c:5585 Remote SDP: > >> v=0 > >> o=- 3570241475 3570241476 IN IP4 10.0.0.131 > >> s=pjmedia > >> c=IN IP4 10.0.0.131 > >> t=0 0 > >> a=X-nat:0 > >> m=audio 4010 RTP/AVP 9 101 > >> a=rtcp:4011 IN IP4 10.0.0.131 > >> a=rtpmap:9 G722/8000 > >> a=sendrecv > >> a=rtpmap:101 telephone-event/8000 > >> a=fmtp:101 0-15 > >> > >> 2013-02-19 00:45:25.302683 [DEBUG] switch_core_session.c:975 Send > signal sofia/internal/10000 at sip.domain.net [BREAK] > >> 2013-02-19 00:45:25.302683 [DEBUG] switch_core_session.c:975 Send > signal sofia/internal/10000 at sip.domain.net [BREAK] > >> 2013-02-19 00:45:25.322657 [DEBUG] sofia.c:5574 Channel sofia/internal/ > 10000 at sip.domain.net entering state [ready][200] > >> 2013-02-19 00:45:25.322657 [DEBUG] sofia.c:5582 Duplicate SDP > >> v=0 > >> o=- 3570241475 3570241476 IN IP4 10.0.0.131 > >> s=pjmedia > >> c=IN IP4 10.0.0.131 > >> t=0 0 > >> a=X-nat:0 > >> m=audio 4010 RTP/AVP 9 101 > >> a=rtcp:4011 IN IP4 10.0.0.131 > >> a=rtpmap:9 G722/8000 > >> a=sendrecv > >> a=rtpmap:101 telephone-event/8000 > >> a=fmtp:101 0-15 > >> > >> 2013-02-19 00:45:25.322657 [DEBUG] sofia_glue.c:5137 Audio Codec > Compare [G722:9:8000:20:64000]/[G722:9:8000:20:64000] > >> 2013-02-19 00:45:25.322657 [DEBUG] sofia_glue.c:3022 Already using G722 > >> 2013-02-19 00:45:25.322657 [DEBUG] sofia_glue.c:5266 Set 2833 dtmf > send/recv payload to 101 > >> > >> > >> Thanks for any help, > >> Emrah > >> On Feb 18, 2013, at 1:38 PM, Michael Collins > wrote: > >> > >>> What happened with the uuid_media_reneg command? > >>> -MC > >>> > >>> On Sat, Feb 16, 2013 at 3:10 PM, Emrah wrote: > >>> Hi all, > >>> > >>> How do we go about renegotiating the codecs of a call that is already > established? > >>> I tried uuid_media_reneg to no avail. > >>> E.g.: Upgrade someone who's entered a 16k conference from PCMU to G722. > >>> > >>> Cheers and thanks, > >>> Emrah > >>> > _________________________________________________________________________ > >>> Professional FreeSWITCH Consulting Services: > >>> consulting at freeswitch.org > >>> http://www.freeswitchsolutions.com > >>> > >>> > >>> > >>> > >>> Official FreeSWITCH Sites > >>> http://www.freeswitch.org > >>> http://wiki.freeswitch.org > >>> http://www.cluecon.com > >>> > >>> 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 > >>> > >>> > >>> > >>> -- > >>> Michael S Collins > >>> Twitter: @mercutioviz > >>> http://www.FreeSWITCH.org > >>> http://www.ClueCon.com > >>> http://www.OSTAG.org > >>> > >>> > _________________________________________________________________________ > >>> Professional FreeSWITCH Consulting Services: > >>> consulting at freeswitch.org > >>> http://www.freeswitchsolutions.com > >>> > >>> > >>> > >>> > >>> Official FreeSWITCH Sites > >>> http://www.freeswitch.org > >>> http://wiki.freeswitch.org > >>> http://www.cluecon.com > >>> > >>> 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 > >> > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> 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 > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130219/c3f3eec3/attachment-0001.html From julf at julf.com Tue Feb 19 19:50:41 2013 From: julf at julf.com (Johan Helsingius) Date: Tue, 19 Feb 2013 17:50:41 +0100 Subject: [Freeswitch-users] Problems with sip provider In-Reply-To: References: <51210FA4.1010200@julf.com> <2362461.fVmeII454V@sos> <512384E5.6030503@julf.com> <2227539.Sa58f9dUxv@sos> <51239310.6050906@julf.com> Message-ID: <5123AD61.5070905@julf.com> Muhammad, > Just talked to a friend, he says though Digitalk SBC is extremely configurable, > but most providers specially in NL (due to its telecom and legal interception > laws etc.), force customers to provide both caller and callee ids correctly. So, > you need at least to make sure, > > 1. Destination number is in correct format, usually E.164 number but provider > may choose differently, e.g. give you some dial prefix to dial out etc. > 2. You are using Remote Part ID in correct format, usually the format is same as > destination number format. Ah! Yes! Setting caller-id-in-from to true and effective-outbound-caller-id-number to the proper E.164 format of the assigned number did it! Many thanks! Julf From areski at gmail.com Tue Feb 19 19:51:20 2013 From: areski at gmail.com (Areski) Date: Tue, 19 Feb 2013 17:51:20 +0100 Subject: [Freeswitch-users] Running sox from within lua In-Reply-To: <5123A671.4040608@digitalmail.com> References: <5123A671.4040608@digitalmail.com> Message-ID: Here an example of how I execute sox from lua, to get the recording length : https://github.com/Star2Billing/newfies-dialer/blob/master/lua/libs/texttospeech.lua#L113 You might need something like this : function simple_command(command) -- local file = assert(io.popen(command, 'r')) local output = file:read('*all') file:close() return output end res = simple_command(string.format([[sox --norm %s %s.new]], file_to_normalise, file_to_normalise)) On Tue, Feb 19, 2013 at 5:21 PM, Alex Lake wrote: > Having a spot of bother with this bit of lua: > > result = os.execute(string.format([[sox --norm %s %s.new]], > file_to_normalise, file_to_normalise)) > > It's returning -1. I believe freeswitch is running as root and that the > sox file is in that location and is runnable. > > Any ideas? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -- Kind regards, /Areski ---- Arezqui Belaid, areski at gmail.com / +34650784355 Founder at Star2Billing (www.star2billing.com) Author @ A2Billing (www.a2billing.net), @ Newfies-Dialer (www.newfies-dialer.org), @ CDR-Stats (www.cdr-stats.org) ---- Twitter : http://twitter.com/areskib / LinkedIN : http://www.linkedin.com/in/areski From a.venugopan at mundio.com Tue Feb 19 19:56:01 2013 From: a.venugopan at mundio.com (Archana Venugopan) Date: Tue, 19 Feb 2013 16:56:01 +0000 Subject: [Freeswitch-users] CHANNEL_DATA variables In-Reply-To: References: <592A9CF93E12394E8472A6CC66E66BF23AC429@Mail-Kilo.squay.com> Message-ID: <592A9CF93E12394E8472A6CC66E66BF23AC4F1@Mail-Kilo.squay.com> Hi, Tried making verbose_Events to true even then I don't see these variables in logs. These CHANNEL_DATA are being picked from mod_dptools.c file and my internal or external call as well pass through that c file but it does not pick up CHANNEL_DATA. Please let me know. Many thanks. Regards, Archana From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 19 February 2013 16:44 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] CHANNEL_DATA variables off the top of my head I don't know why, although the first thing I would try is setting verbose_events to true and retesting. -MC On Tue, Feb 19, 2013 at 8:04 AM, Archana Venugopan > wrote: Hi, I could see that I have been getting the below variables in logs when I dial my voicemail number. But when I make internal or external calls, I don't see these variables in logs. 910672e4-6ec8-11e2-8027-7dea0d27034c 2013-02-04 12:44:08.027128 [INFO] mod_dptools.c:1421 CHANNEL_DATA: Channel-State: [CS_EXECUTE] Channel-Call-State: [RINGING] Channel-State-Number: [4] Channel-Name: [sofia/internal/367 at vectone.uk01.net] Unique-ID: [910672e4-6ec8-11e2-8027-7dea0d27034c] Call-Direction: [inbound] Presence-Call-Direction: [inbound] Channel-HIT-Dialplan: [true] Channel-Presence-ID: [367 at vectone.uk01.net] Channel-Call-UUID: [910672e4-6ec8-11e2-8027-7dea0d27034c] Answer-State: [ringing] Caller-Direction: [inbound] Caller-Username: [367] Caller-Dialplan: [XML] Caller-Caller-ID-Name: [Test] Caller-Caller-ID-Number: [367] Caller-Network-Addr: [192.168.5.129] Can anyone please tell me how am I to get these variables if i make internal or external calls as well as like my voicemail call? Thanks Regards, Archana _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130219/3b0fbc7a/attachment.html From alex at digitalmail.com Tue Feb 19 20:08:38 2013 From: alex at digitalmail.com (Alex Lake) Date: Tue, 19 Feb 2013 17:08:38 +0000 Subject: [Freeswitch-users] Running sox from within lua In-Reply-To: References: <5123A671.4040608@digitalmail.com> Message-ID: <5123B196.4020409@digitalmail.com> That also returns -1 On 19/02/2013 16:47, Michael Collins wrote: > res=os.execute('/usr/bin/sox --version') From alex at digitalmail.com Tue Feb 19 20:07:53 2013 From: alex at digitalmail.com (Alex Lake) Date: Tue, 19 Feb 2013 17:07:53 +0000 Subject: [Freeswitch-users] Running sox from within lua In-Reply-To: <5123A671.4040608@digitalmail.com> References: <5123A671.4040608@digitalmail.com> Message-ID: <5123B169.9000401@digitalmail.com> Actually, using os.execute even with totally innocuous commands seems to fail. Maybe io.popen would be more appropriate? Freeswitch-specific documentation/examples for this seem somewhat thin on the ground, though. In particular, dealing with the return value. > Having a spot of bother with this bit of lua: > > result = os.execute(string.format([[sox --norm %s %s.new]], > file_to_normalise, file_to_normalise)) > > It's returning -1. I believe freeswitch is running as root and that the > sox file is in that location and is runnable. > > Any ideas? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > ----- > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.2238 / Virus Database: 2639/5614 - Release Date: 02/18/13 > > From alex at digitalmail.com Tue Feb 19 20:18:58 2013 From: alex at digitalmail.com (Alex Lake) Date: Tue, 19 Feb 2013 17:18:58 +0000 Subject: [Freeswitch-users] Running sox from within lua In-Reply-To: References: <5123A671.4040608@digitalmail.com> Message-ID: <5123B402.4000402@digitalmail.com> Now THAT does run sox - at least it shows the output of sox on the screen... Thanks. Not sure, though, that --norm does quite what I was hoping! Alex > Here an example of how I execute sox from lua, to get the recording length : > https://github.com/Star2Billing/newfies-dialer/blob/master/lua/libs/texttospeech.lua#L113 > > You might need something like this : > > function simple_command(command) > -- > local file = assert(io.popen(command, 'r')) > local output = file:read('*all') > file:close() > return output > end > > res = simple_command(string.format([[sox --norm %s %s.new]], > file_to_normalise, file_to_normalise)) > > > On Tue, Feb 19, 2013 at 5:21 PM, Alex Lake wrote: >> Having a spot of bother with this bit of lua: >> >> result = os.execute(string.format([[sox --norm %s %s.new]], >> file_to_normalise, file_to_normalise)) >> >> It's returning -1. I believe freeswitch is running as root and that the >> sox file is in that location and is runnable. >> >> Any ideas? >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 a.venugopan at mundio.com Tue Feb 19 20:52:48 2013 From: a.venugopan at mundio.com (Archana Venugopan) Date: Tue, 19 Feb 2013 17:52:48 +0000 Subject: [Freeswitch-users] CHANNEL_DATA variables In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF23AC4F1@Mail-Kilo.squay.com> References: <592A9CF93E12394E8472A6CC66E66BF23AC429@Mail-Kilo.squay.com> <592A9CF93E12394E8472A6CC66E66BF23AC4F1@Mail-Kilo.squay.com> Message-ID: <592A9CF93E12394E8472A6CC66E66BF23AC52C@Mail-Kilo.squay.com> Thanks. Found it. Need to add 'info' Regards, Archana From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Archana Venugopan Sent: 19 February 2013 16:56 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] CHANNEL_DATA variables Hi, Tried making verbose_Events to true even then I don't see these variables in logs. These CHANNEL_DATA are being picked from mod_dptools.c file and my internal or external call as well pass through that c file but it does not pick up CHANNEL_DATA. Please let me know. Many thanks. Regards, Archana From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 19 February 2013 16:44 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] CHANNEL_DATA variables off the top of my head I don't know why, although the first thing I would try is setting verbose_events to true and retesting. -MC On Tue, Feb 19, 2013 at 8:04 AM, Archana Venugopan > wrote: Hi, I could see that I have been getting the below variables in logs when I dial my voicemail number. But when I make internal or external calls, I don't see these variables in logs. 910672e4-6ec8-11e2-8027-7dea0d27034c 2013-02-04 12:44:08.027128 [INFO] mod_dptools.c:1421 CHANNEL_DATA: Channel-State: [CS_EXECUTE] Channel-Call-State: [RINGING] Channel-State-Number: [4] Channel-Name: [sofia/internal/367 at vectone.uk01.net] Unique-ID: [910672e4-6ec8-11e2-8027-7dea0d27034c] Call-Direction: [inbound] Presence-Call-Direction: [inbound] Channel-HIT-Dialplan: [true] Channel-Presence-ID: [367 at vectone.uk01.net] Channel-Call-UUID: [910672e4-6ec8-11e2-8027-7dea0d27034c] Answer-State: [ringing] Caller-Direction: [inbound] Caller-Username: [367] Caller-Dialplan: [XML] Caller-Caller-ID-Name: [Test] Caller-Caller-ID-Number: [367] Caller-Network-Addr: [192.168.5.129] Can anyone please tell me how am I to get these variables if i make internal or external calls as well as like my voicemail call? Thanks Regards, Archana _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130219/9ff1f757/attachment.html From kaothics at hotmail.com Tue Feb 19 21:14:26 2013 From: kaothics at hotmail.com (PolyMorphiK) Date: Tue, 19 Feb 2013 10:14:26 -0800 (PST) Subject: [Freeswitch-users] troubles with Mod_gsmopen and Huawei E153 In-Reply-To: References: <1361204375768-7587585.post@n2.nabble.com> <1361223294004-7587628.post@n2.nabble.com> <1361228596913-7587633.post@n2.nabble.com> <1361243861053-7587637.post@n2.nabble.com> <1361290036149-7587679.post@n2.nabble.com> Message-ID: <1361297666918-7587699.post@n2.nabble.com> Thanks to keep trying to find out Giovanni , On Oktell that's true i've to run as Admin to make it work but i also tried yesterday to do the same running freeswitch as admin but i had the same result. is it possible to identify the port as for exemple "USB\VID_12D1&PID_14AC&MI_02\7&3491CC90&6&0002" in the gsmopen xml config file ? in same time i try to add a gateway profile for a better use of Oktell but not much support available and translating from russian to english or french is somewhat very fun lol :-D i saw something about Oktell and Asterisk : http://vk.it-sfera.com.ua/439.html did someone worked around too ? Nice day all Cordialy -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/troubles-with-Mod-gsmopen-and-Huawei-E153-tp7587585p7587699.html Sent from the freeswitch-users mailing list archive at Nabble.com. From msc at freeswitch.org Tue Feb 19 21:19:34 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 19 Feb 2013 10:19:34 -0800 Subject: [Freeswitch-users] CHANNEL_DATA variables In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF23AC4F1@Mail-Kilo.squay.com> References: <592A9CF93E12394E8472A6CC66E66BF23AC429@Mail-Kilo.squay.com> <592A9CF93E12394E8472A6CC66E66BF23AC4F1@Mail-Kilo.squay.com> Message-ID: Can you pastebin the events for a sample call? I'm curious if all of these are populated at the time the event is fired. If for some reason those values are not already populated then we won't see them in the event headers. -MC On Tue, Feb 19, 2013 at 8:56 AM, Archana Venugopan wrote: > Hi,**** > > ** ** > > Tried making verbose_Events to true even then I don?t see these variables > in logs. These CHANNEL_DATA are being picked from mod_dptools.c file and my > internal or external call as well pass through that c file but it does not > pick up CHANNEL_DATA.**** > > Please let me know. Many thanks.**** > > ** ** > > Regards,**** > > Archana**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael > Collins > *Sent:* 19 February 2013 16:44 > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] CHANNEL_DATA variables**** > > ** ** > > off the top of my head I don't know why, although the first thing I would > try is setting verbose_events to true and retesting. > -MC**** > > On Tue, Feb 19, 2013 at 8:04 AM, Archana Venugopan > wrote:**** > > Hi,**** > > **** > > I could see that I have been getting the below variables in logs when I > dial my voicemail number. But when I make internal or external calls, I > don?t see these variables in logs. **** > > **** > > 910672e4-6ec8-11e2-8027-7dea0d27034c 2013-02-04 12:44:08.027128 [INFO] > mod_dptools.c:1421 CHANNEL_DATA:**** > > Channel-State: [CS_EXECUTE]**** > > Channel-Call-State: [RINGING]**** > > Channel-State-Number: [4]**** > > Channel-Name: [sofia/internal/367 at vectone.uk01.net]**** > > Unique-ID: [910672e4-6ec8-11e2-8027-7dea0d27034c]**** > > Call-Direction: [inbound]**** > > Presence-Call-Direction: [inbound]**** > > Channel-HIT-Dialplan: [true]**** > > Channel-Presence-ID: [367 at vectone.uk01.net]**** > > Channel-Call-UUID: [910672e4-6ec8-11e2-8027-7dea0d27034c]**** > > Answer-State: [ringing]**** > > Caller-Direction: [inbound]**** > > Caller-Username: [367]**** > > Caller-Dialplan: [XML]**** > > Caller-Caller-ID-Name: [Test]**** > > Caller-Caller-ID-Number: [367]**** > > Caller-Network-Addr: [192.168.5.129]**** > > **** > > Can anyone please tell me how am I to get these variables if i make > internal or external calls as well as like my voicemail call? Thanks**** > > **** > > **** > > **** > > Regards,**** > > Archana**** > > **** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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**** > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130219/8991174c/attachment-0001.html From msc at freeswitch.org Tue Feb 19 21:21:38 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 19 Feb 2013 10:21:38 -0800 Subject: [Freeswitch-users] Problems with sip provider In-Reply-To: <5123AD61.5070905@julf.com> References: <51210FA4.1010200@julf.com> <2362461.fVmeII454V@sos> <512384E5.6030503@julf.com> <2227539.Sa58f9dUxv@sos> <51239310.6050906@julf.com> <5123AD61.5070905@julf.com> Message-ID: > > Ah! Yes! > > Setting caller-id-in-from to true and effective-outbound-caller-id-number > to the proper E.164 format of the assigned number did it! > > In other words, carriers in the USA don't have a monopoly on their support reps not knowing how their own systems work! -MC > Many thanks! > > Julf > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130219/4f0965a6/attachment.html From julf at julf.com Tue Feb 19 21:40:26 2013 From: julf at julf.com (Johan Helsingius) Date: Tue, 19 Feb 2013 19:40:26 +0100 Subject: [Freeswitch-users] Problems with sip provider In-Reply-To: References: <51210FA4.1010200@julf.com> <2362461.fVmeII454V@sos> <512384E5.6030503@julf.com> <2227539.Sa58f9dUxv@sos> <51239310.6050906@julf.com> <5123AD61.5070905@julf.com> Message-ID: <5123C71A.5080501@julf.com> > In other words, carriers in the USA don't have a monopoly on their support reps > not knowing how their own systems work! They are probably all sharing the same call center in some low-wage country. Made possible by VoIP, of course... :) Julf From msc at freeswitch.org Tue Feb 19 21:54:12 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 19 Feb 2013 10:54:12 -0800 Subject: [Freeswitch-users] Problems with sip provider In-Reply-To: <5123C71A.5080501@julf.com> References: <51210FA4.1010200@julf.com> <2362461.fVmeII454V@sos> <512384E5.6030503@julf.com> <2227539.Sa58f9dUxv@sos> <51239310.6050906@julf.com> <5123AD61.5070905@julf.com> <5123C71A.5080501@julf.com> Message-ID: On Tue, Feb 19, 2013 at 10:40 AM, Johan Helsingius wrote: > > In other words, carriers in the USA don't have a monopoly on their > support reps > > not knowing how their own systems work! > > They are probably all sharing the same call center in some low-wage > country. > Made possible by VoIP, of course... :) > Haha - we're our own worst enemy - the OSS telephony geeks are subverted by the work of their own hands! -MC > > Julf > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130219/77e35eb4/attachment.html From gmaruzz at gmail.com Tue Feb 19 21:55:28 2013 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Tue, 19 Feb 2013 19:55:28 +0100 Subject: [Freeswitch-users] troubles with Mod_gsmopen and Huawei E153 In-Reply-To: <1361297666918-7587699.post@n2.nabble.com> References: <1361204375768-7587585.post@n2.nabble.com> <1361223294004-7587628.post@n2.nabble.com> <1361228596913-7587633.post@n2.nabble.com> <1361243861053-7587637.post@n2.nabble.com> <1361290036149-7587679.post@n2.nabble.com> <1361297666918-7587699.post@n2.nabble.com> Message-ID: On Tue, Feb 19, 2013 at 7:14 PM, PolyMorphiK wrote: > > is it possible to identify the port as for exemple > "USB\VID_12D1&PID_14AC&MI_02\7&3491CC90&6&0002" in the gsmopen xml config > file ? > > no, and would not be accepted by the underlining library -giovanni -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130219/54c92358/attachment.html From msc at freeswitch.org Tue Feb 19 22:37:02 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 19 Feb 2013 11:37:02 -0800 Subject: [Freeswitch-users] Determine if a number is valid/working In-Reply-To: <072701ce0ebf$c5808640$508192c0$@bizfocused.com> References: <072701ce0ebf$c5808640$508192c0$@bizfocused.com> Message-ID: Well, pretty much the only way to guarantee that a number is working is to call it and see what happens. I suspect Ken Rice has some experience in this area and I'd definitely like to hear it. If anyone else has thoughts I'd like to hear them as well. -MC P.S. - Why does the customer think that calling the number and playing a short message will take too long? Also, what is the customer's primary concern about the web leads having non-working numbers? Is it because they have humans manually dialing them and it's a waste of time when the person comes across a bad number? On Tue, Feb 19, 2013 at 8:40 AM, Sean Devoy wrote: > HI All,**** > > ** ** > > My web site customer?s call center claims that 1/3 of their web leads have > nonworking phone numbers. Short of calling it, is there any way with FS > (or even a paid service) to determine if a given phone number is a working > number? I would like to make an Ajax SOAP call to a PHP page that checks > (or dials) the number and returns a status for the validity of the phone > number.**** > > ** ** > > I have an LUA script that will call, play a short message and hang up, but > the customer thinks that may take too long.**** > > ** ** > > Any ideas?**** > > ** ** > > Thanks,**** > > Sean**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130219/138527f8/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 70 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130219/138527f8/attachment.gif From msc at freeswitch.org Tue Feb 19 22:50:32 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 19 Feb 2013 11:50:32 -0800 Subject: [Freeswitch-users] Benchmark and making session and bridging virtual devices.... In-Reply-To: References: Message-ID: One of the most effective ways to do this is with a second FreeSWITCH server of equal or greater CPU/RAM/disk i/o power. You can use a shell/Perl/Python/PHP/whatever script to generate lots of "originate" commands from the test server to the real server. The real server just needs to route those incoming calls to whatever extension(s) handle call recording, etc. While all those test calls are running you can then make manual calls through the server and listen to how the audio sounds. Keep in mind that FreeSWITCH has some sanity limits in the sample configs: 30 sessions per second and 1000 concurrent sessions total. Be sure to modify those values in switch.conf.xml when you want to test beyond those values. All that being said, the absolute best way to test your configuration is to have it try to handle 1000 calls of real traffic. One of the problems with testing is that it is very difficult to truly simulate 1000 different people all calling in/out through your system. -MC On Tue, Feb 19, 2013 at 8:12 AM, Sayyed Mohammad Emami Razavi < emamirazavi at gmail.com> wrote: > How to test server capacity to load many freeswitch sessions with ready > voice recording and ... features? > I want to make at least 2000 virtual sessions(1000 concurrent calls). How > to do this? Does you have any idea? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130219/12c8eda0/attachment-0001.html From th982a at googlemail.com Tue Feb 19 23:54:01 2013 From: th982a at googlemail.com (Tamer Higazi) Date: Tue, 19 Feb 2013 21:54:01 +0100 Subject: [Freeswitch-users] portaudio on a system with pulseaudio Message-ID: <5123E669.8060701@googlemail.com> Hi people! I am interisted to make use of portaudio on my system. But my gentoo system along with gnome make use of pulseaudio by default. Is there a way to solve that?! for any advise I thank you all Question:Beyond that, what is mod_alsa ?! would it solve my problems ??? Tamer From vipkilla at gmail.com Wed Feb 20 00:18:35 2013 From: vipkilla at gmail.com (Vik Killa) Date: Tue, 19 Feb 2013 16:18:35 -0500 Subject: [Freeswitch-users] portaudio on a system with pulseaudio In-Reply-To: <5123E669.8060701@googlemail.com> References: <5123E669.8060701@googlemail.com> Message-ID: I tried sending audio to/from pulseaudio to portaudio... I had no success on several distros of linux (centos, debian and ubuntu). In my opinion, mod_portaudio needs to be updated or a mod_pulseaudio needs to be developed. I've never been able to get mod_alsa to work properly. From shaheryarkh at gmail.com Wed Feb 20 00:28:21 2013 From: shaheryarkh at gmail.com (Muhammad Shahzad) Date: Tue, 19 Feb 2013 22:28:21 +0100 Subject: [Freeswitch-users] Problems with sip provider In-Reply-To: References: <51210FA4.1010200@julf.com> <2362461.fVmeII454V@sos> <512384E5.6030503@julf.com> <2227539.Sa58f9dUxv@sos> <51239310.6050906@julf.com> <5123AD61.5070905@julf.com> <5123C71A.5080501@julf.com> Message-ID: Great. Glad to know it worked out for you. Thank you. On Tue, Feb 19, 2013 at 7:54 PM, Michael Collins wrote: > > > On Tue, Feb 19, 2013 at 10:40 AM, Johan Helsingius wrote: > >> > In other words, carriers in the USA don't have a monopoly on their >> support reps >> > not knowing how their own systems work! >> >> They are probably all sharing the same call center in some low-wage >> country. >> Made possible by VoIP, of course... :) >> > Haha - we're our own worst enemy - the OSS telephony geeks are subverted > by the work of their own hands! > -MC > > >> >> Julf >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +49 176 99 83 10 85 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130219/6b318665/attachment.html From lloyd.aloysius at gmail.com Wed Feb 20 01:00:48 2013 From: lloyd.aloysius at gmail.com (Lloyd Aloysius) Date: Tue, 19 Feb 2013 17:00:48 -0500 Subject: [Freeswitch-users] valet_parking - aastra phones Message-ID: Hi All: I have a problem with the valet parking and Aastra phones. There is no issue with Polycom 1. Use A Dial User B 2. User B answer the call 3. User B attn xfer to the parking lot 6000 4. User B hear the lot number and complete the transfer 5. *User A also hear the Lot Number . Is there anyway to avoid the announcement for the User A.* http://pastebin.freeswitch.org/20610 The 6000 Extension execute two times, by User B and User A I try with the blind xfer. User A hear the announcement. Any help is appreciated. Thanks Lloyd -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130219/2d329858/attachment.html From gabe at gundy.org Wed Feb 20 01:03:12 2013 From: gabe at gundy.org (Gabriel Gunderson) Date: Tue, 19 Feb 2013 15:03:12 -0700 Subject: [Freeswitch-users] Determine if a number is valid/working In-Reply-To: References: <072701ce0ebf$c5808640$508192c0$@bizfocused.com> Message-ID: On Tue, Feb 19, 2013 at 12:37 PM, Michael Collins wrote: > Well, pretty much the only way to guarantee that a number is working is to > call it and see what happens. I suspect Ken Rice has some experience in > this area and I'd definitely like to hear it. If anyone else has thoughts > I'd like to hear them as well. > I would also love to hear from a few of the callcenter guys/gals on the list about this topic. > P.S. - Why does the customer think that calling the number and playing a > short message will take too long? Also, what is the customer's primary > concern about the web leads having non-working numbers? Is it because they > have humans manually dialing them and it's a waste of time when the person > comes across a bad number? > I think it's because it counts as a short-call on the account. The provider has to do all the work of setting up the call and tearing it down without getting any billable minutes for it. Just a guess. Best, Gabe -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130219/bbfd8c11/attachment.html From victor.chukalovskiy at gmail.com Wed Feb 20 01:25:22 2013 From: victor.chukalovskiy at gmail.com (Victor Chukalovskiy) Date: Tue, 19 Feb 2013 17:25:22 -0500 Subject: [Freeswitch-users] NPDI tel URI parameters appearing in SIP URI Message-ID: <5123FBD2.7070602@gmail.com> Hello, SIP call comes on leg A as following: INVITE sip:6135555555*;npdi=yes*@1.2.3.4:5060 SIP/2.0 ${destination_number} var becomes "6135555555;npdi=yes" On leg B I send the call to gateway using unaltered ${destination_number}: I see the following on leg B: INVITE sip:6135555555 at 5.6.7.8:5060*;npdi=yes* SIP/2.0 So, FS moved ";npdi=yes" which is a tel URI parameter into the very end as if it was a SIP URI parameter. Such behavior does not seem correct. Is this a bug? Found the following implementation discussion: http://comments.gmane.org/gmane.ietf.sip-implementors/21554 Please help to make FS send tel URI parameters unaltered using bridge application and a gateway. WiKi for bridge dial-plan tool has some examples but it does not make use of gateway and I need to use gateway. Thank you, Victor -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130219/52148976/attachment-0001.html From steveayre at gmail.com Wed Feb 20 01:26:36 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 19 Feb 2013 22:26:36 +0000 Subject: [Freeswitch-users] Determine if a number is valid/working In-Reply-To: References: <072701ce0ebf$c5808640$508192c0$@bizfocused.com> Message-ID: > I think it's because it counts as a short-call on the account. The provider > has to do all the work of setting up the call and tearing it down without > getting any billable minutes for it. Just a guess. Then use a provider that bills by the second. Personally I'd think it might be because it could be annoying to the customers... but it really is the only way to check it. Or the other way around to receive a call from it and assume no callerid spoofing. -Steve On 19 February 2013 22:03, Gabriel Gunderson wrote: > On Tue, Feb 19, 2013 at 12:37 PM, Michael Collins > wrote: >> >> Well, pretty much the only way to guarantee that a number is working is to >> call it and see what happens. I suspect Ken Rice has some experience in this >> area and I'd definitely like to hear it. If anyone else has thoughts I'd >> like to hear them as well. > > > I would also love to hear from a few of the callcenter guys/gals on the list > about this topic. > > > >> >> P.S. - Why does the customer think that calling the number and playing a >> short message will take too long? Also, what is the customer's primary >> concern about the web leads having non-working numbers? Is it because they >> have humans manually dialing them and it's a waste of time when the person >> comes across a bad number? > > > I think it's because it counts as a short-call on the account. The provider > has to do all the work of setting up the call and tearing it down without > getting any billable minutes for it. Just a guess. > > > > Best, > Gabe > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 krice at freeswitch.org Wed Feb 20 01:30:20 2013 From: krice at freeswitch.org (Ken Rice) Date: Tue, 19 Feb 2013 16:30:20 -0600 Subject: [Freeswitch-users] Determine if a number is valid/working In-Reply-To: Message-ID: Theres only a few ways to know for sure, a) dial the number see if you get a reject or not lol b) lidb lookup (not always possible) c) look in lerg to see if the NPA-NXX is even valid in the first place, you can also try LNP (and maybe CNAM) lookups to see if you get a valid result there, it is possible to get a CNAM on a disco?d number... On 2/19/13 4:03 PM, "Gabriel Gunderson" wrote: > On Tue, Feb 19, 2013 at 12:37 PM, Michael Collins wrote: >> Well, pretty much the only way to guarantee that a number is working is to >> call it and see what happens. I suspect Ken Rice has some experience in this >> area and I'd definitely like to hear it. If anyone else has thoughts I'd like >> to hear them as well. > > I would also love to hear from a few of the callcenter guys/gals on the list > about this topic. > > > ? >> P.S. - Why does the customer think that calling the number and playing a >> short message will take too long? Also, what is the customer's primary >> concern about the web leads having non-working numbers? Is it because they >> have humans manually dialing them and it's a waste of time when the person >> comes across a bad number? > > I think it's because it counts as a short-call on the account. The provider > has to do all the work of setting up the call and tearing it down without > getting any billable minutes for it. Just a guess. > > > > Best, > Gabe > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130219/913dfcd9/attachment.html From krice at freeswitch.org Wed Feb 20 01:37:57 2013 From: krice at freeswitch.org (Ken Rice) Date: Tue, 19 Feb 2013 16:37:57 -0600 Subject: [Freeswitch-users] Determine if a number is valid/working In-Reply-To: Message-ID: Lol provider that bills by the second will slaughter you on these calls, they have short call penalties now... You can have a bad ASR on dialer traffic or you can have a bad ACD, you cant have bad both, and if either are bad enough they don't care what the other is... Not to mention in the states people still pay per lookup on LRN data, its not a lot but if you start hitting them with dialer traffic vs non-dialer traffic, even with a 20sec ACD, when compared to the 3 minute ACD on non-dialer this ends up costing them 9 times the resources to route a call... That means 9 times the LRN/DIP/Whatever costs per minute then what they are paying before, and with margins where they are these days that cuts into it quickly K On 2/19/13 4:26 PM, "Steven Ayre" wrote: >> I think it's because it counts as a short-call on the account. The provider >> has to do all the work of setting up the call and tearing it down without >> getting any billable minutes for it. Just a guess. > > Then use a provider that bills by the second. > > Personally I'd think it might be because it could be annoying to the > customers... but it really is the only way to check it. Or the other > way around to receive a call from it and assume no callerid spoofing. > > -Steve > > > > On 19 February 2013 22:03, Gabriel Gunderson wrote: >> On Tue, Feb 19, 2013 at 12:37 PM, Michael Collins >> wrote: >>> >>> Well, pretty much the only way to guarantee that a number is working is to >>> call it and see what happens. I suspect Ken Rice has some experience in this >>> area and I'd definitely like to hear it. If anyone else has thoughts I'd >>> like to hear them as well. >> >> >> I would also love to hear from a few of the callcenter guys/gals on the list >> about this topic. >> >> >> >>> >>> P.S. - Why does the customer think that calling the number and playing a >>> short message will take too long? Also, what is the customer's primary >>> concern about the web leads having non-working numbers? Is it because they >>> have humans manually dialing them and it's a waste of time when the person >>> comes across a bad number? >> >> >> I think it's because it counts as a short-call on the account. The provider >> has to do all the work of setting up the call and tearing it down without >> getting any billable minutes for it. Just a guess. >> >> >> >> Best, >> Gabe >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch From avi at avimarcus.net Wed Feb 20 03:25:12 2013 From: avi at avimarcus.net (Avi Marcus) Date: Wed, 20 Feb 2013 02:25:12 +0200 Subject: [Freeswitch-users] NPDI tel URI parameters appearing in SIP URI In-Reply-To: <5123FBD2.7070602@gmail.com> References: <5123FBD2.7070602@gmail.com> Message-ID: You'll have to use a regex and capture it manually. See: http://wiki.freeswitch.org/wiki/Regular_Expression#LNP_.28Local_Number_Portability.29 -Avi On Wed, Feb 20, 2013 at 12:25 AM, Victor Chukalovskiy < victor.chukalovskiy at gmail.com> wrote: > Hello, > > SIP call comes on leg A as following: > INVITE sip:6135555555*;npdi=yes*@1.2.3.4:5060 SIP/2.0 > > ${destination_number} var becomes "6135555555;npdi=yes" > > On leg B I send the call to gateway using unaltered ${destination_number}: > data="sofia/gateway/test_gw/${destination_number}"/> > > I see the following on leg B: > INVITE sip:6135555555 at 5.6.7.8:5060*;npdi=yes* SIP/2.0 > > So, FS moved ";npdi=yes" which is a tel URI parameter into the very end as > if it was a SIP URI parameter. Such behavior does not seem correct. Is this > a bug? Found the following implementation discussion: > http://comments.gmane.org/gmane.ietf.sip-implementors/21554 > > Please help to make FS send tel URI parameters unaltered using bridge > application and a gateway. WiKi for bridge dial-plan tool has some examples > but it does not make use of gateway and I need to use gateway. > > Thank you, > Victor > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130220/2901ebb3/attachment.html From krice at freeswitch.org Wed Feb 20 03:57:00 2013 From: krice at freeswitch.org (Ken Rice) Date: Tue, 19 Feb 2013 18:57:00 -0600 Subject: [Freeswitch-users] NPDI tel URI parameters appearing in SIP URI In-Reply-To: References: <5123FBD2.7070602@gmail.com> Message-ID: If its actually moving the ndpi to the host side thats a bug open a jira and attach debug logs with sip trace on Sent from my iPhone On Feb 19, 2013, at 6:25 PM, Avi Marcus wrote: > You'll have to use a regex and capture it manually. > See: http://wiki.freeswitch.org/wiki/Regular_Expression#LNP_.28Local_Number_Portability.29 > > -Avi > > On Wed, Feb 20, 2013 at 12:25 AM, Victor Chukalovskiy wrote: >> Hello, >> >> SIP call comes on leg A as following: >> INVITE sip:6135555555;npdi=yes at 1.2.3.4:5060 SIP/2.0 >> >> ${destination_number} var becomes "6135555555;npdi=yes" >> >> On leg B I send the call to gateway using unaltered ${destination_number}: >> >> >> I see the following on leg B: >> INVITE sip:6135555555 at 5.6.7.8:5060;npdi=yes SIP/2.0 >> >> So, FS moved ";npdi=yes" which is a tel URI parameter into the very end as if it was a SIP URI parameter. Such behavior does not seem correct. Is this a bug? Found the following implementation discussion: >> http://comments.gmane.org/gmane.ietf.sip-implementors/21554 >> >> Please help to make FS send tel URI parameters unaltered using bridge application and a gateway. WiKi for bridge dial-plan tool has some examples but it does not make use of gateway and I need to use gateway. >> >> Thank you, >> Victor >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130219/e707e438/attachment-0001.html From schoch+freeswitch.org at xwin32.com Wed Feb 20 05:16:45 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Tue, 19 Feb 2013 18:16:45 -0800 Subject: [Freeswitch-users] Pre-loading Voicemail prompts and names In-Reply-To: References: Message-ID: On Mon, Feb 18, 2013 at 11:26 PM, Avi Marcus wrote: > Steve, you can open up the voicemail_prefs database. > If you set up ODBC for VM you know where it is, otherwise, it's in a > sqlite DB. > That was it! A simple query solved the problem: update voicemail_prefs set name_path = '/usr/local/freeswitch/storage/voicemail/default/starnet.com/' || username || '/recorded_name.wav' where username in (......); -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130219/bb7902fe/attachment.html From victor.chukalovskiy at gmail.com Wed Feb 20 07:43:30 2013 From: victor.chukalovskiy at gmail.com (Victor Chukalovskiy) Date: Tue, 19 Feb 2013 23:43:30 -0500 Subject: [Freeswitch-users] NPDI tel URI parameters appearing in SIP URI In-Reply-To: References: <5123FBD2.7070602@gmail.com> Message-ID: <51245472.90105@gmail.com> Avi, I know this part of the WiKi....thank you but I don't think it helps here. -Victor On 02/19/2013 07:25 PM, Avi Marcus wrote: > You'll have to use a regex and capture it manually. > See: > http://wiki.freeswitch.org/wiki/Regular_Expression#LNP_.28Local_Number_Portability.29 > > -Avi > > On Wed, Feb 20, 2013 at 12:25 AM, Victor Chukalovskiy > > > wrote: > > Hello, > > SIP call comes on leg A as following: > INVITE sip:6135555555 *;npdi=yes*@1.2.3.4:5060 > SIP/2.0 > > ${destination_number} var becomes "6135555555 > ;npdi=yes" > > On leg B I send the call to gateway using unaltered > ${destination_number}: > data="sofia/gateway/test_gw/${destination_number}"/> > > I see the following on leg B: > INVITE sip:6135555555 at 5.6.7.8:5060 > *;npdi=yes* SIP/2.0 > > So, FS moved ";npdi=yes" which is a tel URI parameter into the > very end as if it was a SIP URI parameter. Such behavior does not > seem correct. Is this a bug? Found the following implementation > discussion: > http://comments.gmane.org/gmane.ietf.sip-implementors/21554 > > Please help to make FS send tel URI parameters unaltered using > bridge application and a gateway. WiKi for bridge dial-plan tool > has some examples but it does not make use of gateway and I need > to use gateway. > > Thank you, > Victor > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130219/fb632c42/attachment.html From victor.chukalovskiy at gmail.com Wed Feb 20 07:46:28 2013 From: victor.chukalovskiy at gmail.com (Victor Chukalovskiy) Date: Tue, 19 Feb 2013 23:46:28 -0500 Subject: [Freeswitch-users] NPDI tel URI parameters appearing in SIP URI In-Reply-To: References: <5123FBD2.7070602@gmail.com> Message-ID: <51245524.6070002@gmail.com> Yes, it moves it to the right of the host. Thank you for confirming, looks like I'll have to open a jira. On 02/19/2013 07:57 PM, Ken Rice wrote: > If its actually moving the ndpi to the host side thats a bug open a > jira and attach debug logs with sip trace on > > Sent from my iPhone > > On Feb 19, 2013, at 6:25 PM, Avi Marcus > wrote: > >> You'll have to use a regex and capture it manually. >> See: >> http://wiki.freeswitch.org/wiki/Regular_Expression#LNP_.28Local_Number_Portability.29 >> >> -Avi >> >> On Wed, Feb 20, 2013 at 12:25 AM, Victor Chukalovskiy >> > > wrote: >> >> Hello, >> >> SIP call comes on leg A as following: >> INVITE sip:6135555555 *;npdi=yes*@1.2.3.4:5060 >> SIP/2.0 >> >> ${destination_number} var becomes "6135555555 >> ;npdi=yes" >> >> On leg B I send the call to gateway using unaltered >> ${destination_number}: >> > data="sofia/gateway/test_gw/${destination_number}"/> >> >> I see the following on leg B: >> INVITE sip:6135555555 at 5.6.7.8:5060 >> *;npdi=yes* SIP/2.0 >> >> So, FS moved ";npdi=yes" which is a tel URI parameter into the >> very end as if it was a SIP URI parameter. Such behavior does not >> seem correct. Is this a bug? Found the following implementation >> discussion: >> http://comments.gmane.org/gmane.ietf.sip-implementors/21554 >> >> Please help to make FS send tel URI parameters unaltered using >> bridge application and a gateway. WiKi for bridge dial-plan tool >> has some examples but it does not make use of gateway and I need >> to use gateway. >> >> Thank you, >> Victor >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130219/234978db/attachment-0001.html From kaothics at hotmail.com Wed Feb 20 08:37:42 2013 From: kaothics at hotmail.com (PolyMorphiK) Date: Tue, 19 Feb 2013 21:37:42 -0800 (PST) Subject: [Freeswitch-users] troubles with Mod_gsmopen and Huawei E153 In-Reply-To: References: <1361223294004-7587628.post@n2.nabble.com> <1361228596913-7587633.post@n2.nabble.com> <1361243861053-7587637.post@n2.nabble.com> <1361290036149-7587679.post@n2.nabble.com> <1361297666918-7587699.post@n2.nabble.com> Message-ID: <1361338662323-7587721.post@n2.nabble.com> Could it comes from the libctb , check there http://translate.google.com/translate?hl=fr&sl=auto&tl=en&u=http%3A%2F%2Fblog.xtao.me%2F2012%2F01%2Fa-problem-with-libctb%2F beside that Oktell SIP-GSM woks really well for now and also in Gateway mode with a basic gateway setting on freeswitch so i can work on some lua scripts. i wonder if it could be possible to use an other library instead of libctb Cordialy -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/troubles-with-Mod-gsmopen-and-Huawei-E153-tp7587585p7587721.html Sent from the freeswitch-users mailing list archive at Nabble.com. From vbvbrj at gmail.com Wed Feb 20 10:02:58 2013 From: vbvbrj at gmail.com (Mimiko) Date: Wed, 20 Feb 2013 09:02:58 +0200 Subject: [Freeswitch-users] Return from lua two values. Message-ID: <51247522.5020801@gmail.com> Hello. If using I get one value from lua to MY_VAR. Can I get two values to two variables in one call? Thanks. -- Mimiko desu. From julf at julf.com Wed Feb 20 10:17:04 2013 From: julf at julf.com (Johan Helsingius) Date: Wed, 20 Feb 2013 08:17:04 +0100 Subject: [Freeswitch-users] Problems with sip provider In-Reply-To: References: <51210FA4.1010200@julf.com> <2362461.fVmeII454V@sos> <512384E5.6030503@julf.com> <2227539.Sa58f9dUxv@sos> <51239310.6050906@julf.com> <5123AD61.5070905@julf.com> <5123C71A.5080501@julf.com> Message-ID: <51247870.3060500@julf.com> > Great. Glad to know it worked out for you. Worked out thanks to the help from all you guys! Many thanks! Julf From mario at ims.bg Wed Feb 20 10:28:59 2013 From: mario at ims.bg (Mario Karakanovski) Date: Wed, 20 Feb 2013 09:28:59 +0200 Subject: [Freeswitch-users] freeswitch hack In-Reply-To: <1361338662323-7587721.post@n2.nabble.com> References: <1361223294004-7587628.post@n2.nabble.com><1361228596913-7587633.post@n2.nabble.com><1361243861053-7587637.post@n2.nabble.com><1361290036149-7587679.post@n2.nabble.com><1361297666918-7587699.post@n2.nabble.com> <1361338662323-7587721.post@n2.nabble.com> Message-ID: <6E50138BFCF3498F82BA51CFC29896E9@imsdeveloper> Hi all, For some days i noticed that somebody was able to register to my freeswitch and trying to call international numbers. The attack is very clever as the hacker logs at the night, trying to call international number 10-15 times while changing the prefix and go away. The sip profile is connected directly to the internet and require authentication: auth-calls = true auth-all-packets = true There is no IP filtering as the service does not allow setting some. Firewall blokes all port except TCP and UDP 5060 and required UDP media ports. The authentication is made by directory. What I wonder is how ones can authenticated with extension that not exist and not described anywhere. Can it be some security issue with freeswitch? Any ideas how to solve the problem? Regards, Mario From steveayre at gmail.com Wed Feb 20 11:14:50 2013 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 20 Feb 2013 08:14:50 +0000 Subject: [Freeswitch-users] freeswitch hack In-Reply-To: <6E50138BFCF3498F82BA51CFC29896E9@imsdeveloper> References: <1361223294004-7587628.post@n2.nabble.com> <1361228596913-7587633.post@n2.nabble.com> <1361243861053-7587637.post@n2.nabble.com> <1361290036149-7587679.post@n2.nabble.com> <1361297666918-7587699.post@n2.nabble.com> <1361338662323-7587721.post@n2.nabble.com> <6E50138BFCF3498F82BA51CFC29896E9@imsdeveloper> Message-ID: Not unusual at all, and not even clever... there are lots of bots that just randomly search the net for IP addresses that're open to allowing calls. First, are you sure the profile is actually requiring authentication (a simple packet trace will reveal that - the first INVITE should get a 401 reply). Second, do you have blind auth enabled, in which case it'd be accepting any username/password? Third, are they getting authenticated via an ACL or user CIDR? Finally, is it possible that you're loading CDRs for calls which have been rejected? -Steve On 20 February 2013 07:28, Mario Karakanovski wrote: > Hi all, > > For some days i noticed that somebody was able to register to my > freeswitch and trying to call international numbers. The attack is very > clever as the hacker logs at the night, trying to call international number > 10-15 times while changing the prefix and go away. > > The sip profile is connected directly to the internet and require > authentication: > auth-calls = true > auth-all-packets = true > > There is no IP filtering as the service does not allow setting some. > Firewall blokes all port except TCP and UDP 5060 and required UDP media > ports. The authentication is made by directory. > What I wonder is how ones can authenticated with extension that not exist > and not described anywhere. > > Can it be some security issue with freeswitch? Any ideas how to solve the > problem? > > Regards, > Mario > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 steveayre at gmail.com Wed Feb 20 11:15:54 2013 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 20 Feb 2013 08:15:54 +0000 Subject: [Freeswitch-users] Return from lua two values. In-Reply-To: <51247522.5020801@gmail.com> References: <51247522.5020801@gmail.com> Message-ID: Set the variables from the LUA script itself. http://wiki.freeswitch.org/wiki/Mod_lua#session:setVariable -Steve On 20 February 2013 07:02, Mimiko wrote: > Hello. > > If using > > I get one value from lua to MY_VAR. > > Can I get two values to two variables in one call? > > Thanks. > > -- > Mimiko desu. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 avi at avimarcus.net Wed Feb 20 11:32:15 2013 From: avi at avimarcus.net (Avi Marcus) Date: Wed, 20 Feb 2013 10:32:15 +0200 Subject: [Freeswitch-users] NPDI tel URI parameters appearing in SIP URI In-Reply-To: <51245472.90105@gmail.com> References: <5123FBD2.7070602@gmail.com> <51245472.90105@gmail.com> Message-ID: Ah sorry, I should have read your question more carefully! -Avi On Wed, Feb 20, 2013 at 6:43 AM, Victor Chukalovskiy < victor.chukalovskiy at gmail.com> wrote: > Avi, > > I know this part of the WiKi....thank you but I don't think it helps here. > > -Victor > > > On 02/19/2013 07:25 PM, Avi Marcus wrote: > > You'll have to use a regex and capture it manually. > See: > http://wiki.freeswitch.org/wiki/Regular_Expression#LNP_.28Local_Number_Portability.29 > > -Avi > > On Wed, Feb 20, 2013 at 12:25 AM, Victor Chukalovskiy < > victor.chukalovskiy at gmail.com> wrote: > >> Hello, >> >> SIP call comes on leg A as following: >> INVITE sip:6135555555*;npdi=yes*@1.2.3.4:5060 SIP/2.0 >> >> ${destination_number} var becomes "6135555555;npdi=yes" >> >> On leg B I send the call to gateway using unaltered ${destination_number}: >> > data="sofia/gateway/test_gw/${destination_number}"/> >> >> I see the following on leg B: >> INVITE sip:6135555555 at 5.6.7.8:5060*;npdi=yes* SIP/2.0 >> >> So, FS moved ";npdi=yes" which is a tel URI parameter into the very end >> as if it was a SIP URI parameter. Such behavior does not seem correct. Is >> this a bug? Found the following implementation discussion: >> http://comments.gmane.org/gmane.ietf.sip-implementors/21554 >> >> Please help to make FS send tel URI parameters unaltered using bridge >> application and a gateway. WiKi for bridge dial-plan tool has some examples >> but it does not make use of gateway and I need to use gateway. >> >> Thank you, >> Victor >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130220/4d6ca055/attachment-0001.html From vbvbrj at gmail.com Wed Feb 20 11:32:27 2013 From: vbvbrj at gmail.com (Mimiko) Date: Wed, 20 Feb 2013 10:32:27 +0200 Subject: [Freeswitch-users] Return from lua two values. In-Reply-To: References: <51247522.5020801@gmail.com> Message-ID: <51248A1B.1000402@gmail.com> On 20.02.2013 10:15, Steven Ayre wrote: > Set the variables from the LUA script itself. > > http://wiki.freeswitch.org/wiki/Mod_lua#session:setVariable > Thank you. I thought that was for other purposes. -- Mimiko desu. From avi at avimarcus.net Wed Feb 20 11:35:52 2013 From: avi at avimarcus.net (Avi Marcus) Date: Wed, 20 Feb 2013 10:35:52 +0200 Subject: [Freeswitch-users] freeswitch hack In-Reply-To: References: <1361223294004-7587628.post@n2.nabble.com> <1361228596913-7587633.post@n2.nabble.com> <1361243861053-7587637.post@n2.nabble.com> <1361290036149-7587679.post@n2.nabble.com> <1361297666918-7587699.post@n2.nabble.com> <1361338662323-7587721.post@n2.nabble.com> <6E50138BFCF3498F82BA51CFC29896E9@imsdeveloper> Message-ID: Personally, I've seen the last option -- CDRs for calls that have been rejected. They come in on the public profile, attempt to call a single number with several prefix types... but they are indeed unauthenticated so FS just hangs up on them. -Avi On Wed, Feb 20, 2013 at 10:14 AM, Steven Ayre wrote: > Not unusual at all, and not even clever... there are lots of bots that > just randomly search the net for IP addresses that're open to allowing > calls. > > First, are you sure the profile is actually requiring authentication > (a simple packet trace will reveal that - the first INVITE should get > a 401 reply). > > Second, do you have blind auth enabled, in which case it'd be > accepting any username/password? > > Third, are they getting authenticated via an ACL or user CIDR? > > Finally, is it possible that you're loading CDRs for calls which have > been rejected? > > -Steve > > > > > On 20 February 2013 07:28, Mario Karakanovski wrote: > > Hi all, > > > > For some days i noticed that somebody was able to register to my > > freeswitch and trying to call international numbers. The attack is very > > clever as the hacker logs at the night, trying to call international > number > > 10-15 times while changing the prefix and go away. > > > > The sip profile is connected directly to the internet and require > > authentication: > > auth-calls = true > > auth-all-packets = true > > > > There is no IP filtering as the service does not allow setting some. > > Firewall blokes all port except TCP and UDP 5060 and required UDP media > > ports. The authentication is made by directory. > > What I wonder is how ones can authenticated with extension that not exist > > and not described anywhere. > > > > Can it be some security issue with freeswitch? Any ideas how to solve the > > problem? > > > > Regards, > > Mario > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130220/28853595/attachment.html From krice at freeswitch.org Wed Feb 20 11:46:11 2013 From: krice at freeswitch.org (Ken Rice) Date: Wed, 20 Feb 2013 02:46:11 -0600 Subject: [Freeswitch-users] freeswitch hack In-Reply-To: Message-ID: They are probably hitting the public/external interface which by default accepts calls for any number but will reject them as the public dialplan doesn?t know what to do with them. I see this on a regular basis on my PBX... The real question is, are you setup to allow anyone to call you, and if you are, are you sure to block calls not destined for your box. As a previous response pointed out this is a very common attack, I see it several times a day from many different IPs as its just a bot.... There is also sipvicious attacks which I don?t see much of as I block them on the firewall level (they almost always include the string ?friendly-scanner? and a quick google with give you an iptables command to drop them). These bots exist to exploit not just freeswitch, but any SIP server, and to exploit sip users with bad/common passwords... Good System Admin practices and double checking your FreeSWITCH configs will generally stop them in their tracks. If you need assistance stop by the FreeSWITCH IRC channel on Freenode #freeswitch and ask around... If you need professional help email consulting at freeswitch.org and they will help you out. K On 2/20/13 2:35 AM, "Avi Marcus" wrote: > Personally, I've seen the last option -- CDRs for calls that have been > rejected. > > They come in on the public profile, attempt to call a single number with > several prefix types... but they are indeed unauthenticated so FS just hangs > up on them. > > -Avi > > On Wed, Feb 20, 2013 at 10:14 AM, Steven Ayre wrote: >> Not unusual at all, and not even clever... there are lots of bots that >> just randomly search the net for IP addresses that're open to allowing >> calls. >> >> First, are you sure the profile is actually requiring authentication >> (a simple packet trace will reveal that - the first INVITE should get >> a 401 reply). >> >> Second, do you have blind auth enabled, in which case it'd be >> accepting any username/password? >> >> Third, are they getting authenticated via an ACL or user CIDR? >> >> Finally, is it possible that you're loading CDRs for calls which have >> been rejected? >> >> -Steve >> >> >> >> >> On 20 February 2013 07:28, Mario Karakanovski wrote: >>> > Hi all, >>> > >>> > ? ? ? ? For some days i noticed that somebody was able to register to my >>> > freeswitch and trying to call international numbers. The attack is very >>> > clever as the hacker logs at the night, trying to call international >>> number >>> > 10-15 times while changing the prefix and go away. >>> > >>> > The sip profile is connected directly to the internet and require >>> > authentication: >>> > ? ? ? ? auth-calls = true >>> > ? ? ? ? auth-all-packets = true >>> > >>> > There is no IP filtering as the service does not allow setting some. >>> > Firewall blokes all port except TCP and UDP 5060 and required UDP media >>> > ports. The authentication is made by directory. >>> > What I wonder is how ones can authenticated with extension that not exist >>> > and not described anywhere. >>> > >>> > Can it be some security issue with freeswitch? Any ideas how to solve the >>> > problem? >>> > >>> > Regards, >>> > ? ? ? ? Mario >>> > >>> > >>> > _________________________________________________________________________ >>> > Professional FreeSWITCH Consulting Services: >>> > consulting at freeswitch.org >>> > http://www.freeswitchsolutions.com >>> > >>> > >>> > >>> > >>> > Official FreeSWITCH Sites >>> > http://www.freeswitch.org >>> > http://wiki.freeswitch.org >>> > http://www.cluecon.com >>> > >>> > 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 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130220/602080f0/attachment-0001.html From mario at ims.bg Wed Feb 20 12:53:38 2013 From: mario at ims.bg (Mario Karakanovski) Date: Wed, 20 Feb 2013 11:53:38 +0200 Subject: [Freeswitch-users] freeswitch hack In-Reply-To: References: Message-ID: <958ABE902C1F47B09FFAAE7EDEE074B1@imsdeveloper> Thanks Ken, It is helpful, but I still think there is some security issue. I've double check configuration. I've try to reproduce the issue trying to do direct call (TCP and UDP) or authenticate with invalid user, but everything works as expected - calls/authentication was rejected. I've decide to log the traffic - maybe I will be able to see where is the problem. Regards, Mario _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice Sent: Wednesday, February 20, 2013 10:46 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] freeswitch hack They are probably hitting the public/external interface which by default accepts calls for any number but will reject them as the public dialplan doesn't know what to do with them. I see this on a regular basis on my PBX... The real question is, are you setup to allow anyone to call you, and if you are, are you sure to block calls not destined for your box. As a previous response pointed out this is a very common attack, I see it several times a day from many different IPs as its just a bot.... There is also sipvicious attacks which I don't see much of as I block them on the firewall level (they almost always include the string 'friendly-scanner' and a quick google with give you an iptables command to drop them). These bots exist to exploit not just freeswitch, but any SIP server, and to exploit sip users with bad/common passwords... Good System Admin practices and double checking your FreeSWITCH configs will generally stop them in their tracks. If you need assistance stop by the FreeSWITCH IRC channel on Freenode #freeswitch and ask around... If you need professional help email consulting at freeswitch.org and they will help you out. K On 2/20/13 2:35 AM, "Avi Marcus" wrote: Personally, I've seen the last option -- CDRs for calls that have been rejected. They come in on the public profile, attempt to call a single number with several prefix types... but they are indeed unauthenticated so FS just hangs up on them. -Avi On Wed, Feb 20, 2013 at 10:14 AM, Steven Ayre wrote: Not unusual at all, and not even clever... there are lots of bots that just randomly search the net for IP addresses that're open to allowing calls. First, are you sure the profile is actually requiring authentication (a simple packet trace will reveal that - the first INVITE should get a 401 reply). Second, do you have blind auth enabled, in which case it'd be accepting any username/password? Third, are they getting authenticated via an ACL or user CIDR? Finally, is it possible that you're loading CDRs for calls which have been rejected? -Steve On 20 February 2013 07:28, Mario Karakanovski wrote: > Hi all, > > For some days i noticed that somebody was able to register to my > freeswitch and trying to call international numbers. The attack is very > clever as the hacker logs at the night, trying to call international number > 10-15 times while changing the prefix and go away. > > The sip profile is connected directly to the internet and require > authentication: > auth-calls = true > auth-all-packets = true > > There is no IP filtering as the service does not allow setting some. > Firewall blokes all port except TCP and UDP 5060 and required UDP media > ports. The authentication is made by directory. > What I wonder is how ones can authenticated with extension that not exist > and not described anywhere. > > Can it be some security issue with freeswitch? Any ideas how to solve the > problem? > > Regards, > Mario > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 _____ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130220/9870ed3c/attachment.html From a.venugopan at mundio.com Wed Feb 20 15:37:53 2013 From: a.venugopan at mundio.com (Archana Venugopan) Date: Wed, 20 Feb 2013 12:37:53 +0000 Subject: [Freeswitch-users] Event pick Message-ID: <592A9CF93E12394E8472A6CC66E66BF23AC6B5@Mail-Kilo.squay.com> Hi, I need to capture this ORIGINATOR _CANCEL event in my lua script. From google I found the below code. But we establish our calls from dialplan XML as like this, . Since am not that well versed in lua script, I am not sure in the place of obsession below what should I pass. As far I understood the =sofia/external/0${msc_prefix}$1$2 at 10.30.3.27, this is what I should put there but we already have this in our XML. Please suggest me on how should I proceed. Many thanks. -- Initiate an outbound call obSession = freeswitch.Session("sofia/192.168.0.4/1002") -- Check to see if the call was answered if obSession:ready() then -- Do something good here else -- This means the call was not answered ... Check for the reason local obCause = obSession:hangupCause() freeswitch.consoleLog("info", "obSession:hangupCause() = " .. obCause ) if ( obCause == "USER_BUSY" ) then -- SIP 486 -- For BUSY you may reschedule the call for later elseif ( obCause == "NO_ANSWER" ) then -- Call them back in an hour elseif ( obCause == "ORIGINATOR_CANCEL" ) then -- SIP 487 -- May need to check for network congestion or problems else -- Log these issues end end 2013-02-19 17:56:02.277173 [NOTICE] sofia.c:7082 Hangup sofia/internal/206 at fsfailover.uk01.com [CS_EXECUTE] [ORIGINATOR_CANCEL] Regards, Archana -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130220/24a7f005/attachment-0001.html From dvl36.ripe.nick at gmail.com Wed Feb 20 16:04:24 2013 From: dvl36.ripe.nick at gmail.com (Dmitry Lysenko) Date: Wed, 20 Feb 2013 15:04:24 +0200 Subject: [Freeswitch-users] Changing codec during calls In-Reply-To: References: <6EF8B8E9-4CC5-480C-B1C1-09CBC533470C@kavun.ch> <35AF9E79-771D-440A-BEBB-A856D831161C@kavun.ch> Message-ID: Wow! This great feature is really work. I have the same issue - one way audio. And another: "show channels" always return first negotiated codec. When switching codec back, audio working fine. Tested with 3 different softphones. If someone interested, please confirm. Thanks. 2013/2/19 Michael Collins > pastebin the log. I'll bet Steven can diagnose it! ;) > -MC > > > On Tue, Feb 19, 2013 at 8:37 AM, Emrah wrote: > >> Excellent, that worked. >> >> However I am having a one way audio issue, where the peer that has its >> codecs changed can hear but cannot be heard. >> Even if both bridged peers are changed to a common codec it doesn't work. >> >> Any idea? >> >> Thanks a bunch for your help. >> On Feb 19, 2013, at 3:47 AM, Steven Ayre wrote: >> >> > Try: >> > uuid_media_reneg 49f428e5-1d72-42d8-b0e1-5b3f29f9a4ca =PCMU >> > >> > (ie add = before PCMU) >> > >> > It appears to affect something in the code... >> > >> > -Steve >> > >> > >> > On 19 February 2013 05:54, Emrah wrote: >> >> Hey, >> >> >> >> In short, nothing seems to happen. I tried several phones. >> >> I also tried enabling codec renegotiation on hold/unhold thinking that >> the change in SDP with uuid_media_reneg would do it. >> >> >> >> This is what happens: >> >> freeswitch at internal> show channels >> >> >> uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,sent_callee_name,sent_callee_num >> >> 49f428e5-1d72-42d8-b0e1-5b3f29f9a4ca,inbound,2013-02-19 >> 00:44:35,1361252675,sofia/internal/10000 at sip.domain.net,CS_EXECUTE,Emrah >> - Macbook >> Pro,10000,100.100.100.100,conf-3111,conference,3111-sip.domain.net at ek-custom-16k >> ,XML,ek-conference,G722,16000,64000,G722,16000,64000,, >> cornflakes.domain.net,10000 at sip.domain.net,,ACTIVE,,,,,, >> >> >> >> 1 total. >> >> >> >> freeswitch at internal> uuid_media_reneg >> 49f428e5-1d72-42d8-b0e1-5b3f29f9a4ca PCMU >> >> +OK Success >> >> >> >> 2013-02-19 00:45:25.172659 [DEBUG] sofia_glue.c:2647 Local SDP: >> >> v=0 >> >> o=FreeSWITCH 1361234787 1361234790 IN IP4 1.2.3.4 >> >> s=FreeSWITCH >> >> c=IN IP4 1.2.3.4 >> >> t=0 0 >> >> m=audio 17888 RTP/AVP 9 0 101 >> >> a=rtpmap:101 telephone-event/8000 >> >> a=fmtp:101 0-16 >> >> a=silenceSupp:off - - - - >> >> a=ptime:20 >> >> a=sendrecv >> >> >> >> 2013-02-19 00:45:25.172659 [DEBUG] switch_core_session.c:975 Send >> signal sofia/internal/10000 at sip.domain.net [BREAK] >> >> 2013-02-19 00:45:25.182652 [DEBUG] sofia.c:5574 Channel sofia/internal/ >> 10000 at sip.domain.net entering state [calling][0] >> >> 2013-02-19 00:45:25.292661 [DEBUG] switch_core_session.c:975 Send >> signal sofia/internal/10000 at sip.domain.net [BREAK] >> >> 2013-02-19 00:45:25.292661 [DEBUG] switch_core_session.c:975 Send >> signal sofia/internal/10000 at sip.domain.net [BREAK] >> >> 2013-02-19 00:45:25.302683 [INFO] sofia.c:931 sofia/internal/ >> 10000 at sip.domain.net Update Callee ID to "10000" <10000> >> >> 2013-02-19 00:45:25.302683 [DEBUG] sofia.c:5574 Channel sofia/internal/ >> 10000 at sip.domain.net entering state [completing][200] >> >> 2013-02-19 00:45:25.302683 [DEBUG] sofia.c:5585 Remote SDP: >> >> v=0 >> >> o=- 3570241475 3570241476 IN IP4 10.0.0.131 >> >> s=pjmedia >> >> c=IN IP4 10.0.0.131 >> >> t=0 0 >> >> a=X-nat:0 >> >> m=audio 4010 RTP/AVP 9 101 >> >> a=rtcp:4011 IN IP4 10.0.0.131 >> >> a=rtpmap:9 G722/8000 >> >> a=sendrecv >> >> a=rtpmap:101 telephone-event/8000 >> >> a=fmtp:101 0-15 >> >> >> >> 2013-02-19 00:45:25.302683 [DEBUG] switch_core_session.c:975 Send >> signal sofia/internal/10000 at sip.domain.net [BREAK] >> >> 2013-02-19 00:45:25.302683 [DEBUG] switch_core_session.c:975 Send >> signal sofia/internal/10000 at sip.domain.net [BREAK] >> >> 2013-02-19 00:45:25.322657 [DEBUG] sofia.c:5574 Channel sofia/internal/ >> 10000 at sip.domain.net entering state [ready][200] >> >> 2013-02-19 00:45:25.322657 [DEBUG] sofia.c:5582 Duplicate SDP >> >> v=0 >> >> o=- 3570241475 3570241476 IN IP4 10.0.0.131 >> >> s=pjmedia >> >> c=IN IP4 10.0.0.131 >> >> t=0 0 >> >> a=X-nat:0 >> >> m=audio 4010 RTP/AVP 9 101 >> >> a=rtcp:4011 IN IP4 10.0.0.131 >> >> a=rtpmap:9 G722/8000 >> >> a=sendrecv >> >> a=rtpmap:101 telephone-event/8000 >> >> a=fmtp:101 0-15 >> >> >> >> 2013-02-19 00:45:25.322657 [DEBUG] sofia_glue.c:5137 Audio Codec >> Compare [G722:9:8000:20:64000]/[G722:9:8000:20:64000] >> >> 2013-02-19 00:45:25.322657 [DEBUG] sofia_glue.c:3022 Already using G722 >> >> 2013-02-19 00:45:25.322657 [DEBUG] sofia_glue.c:5266 Set 2833 dtmf >> send/recv payload to 101 >> >> >> >> >> >> Thanks for any help, >> >> Emrah >> >> On Feb 18, 2013, at 1:38 PM, Michael Collins >> wrote: >> >> >> >>> What happened with the uuid_media_reneg command? >> >>> -MC >> >>> >> >>> On Sat, Feb 16, 2013 at 3:10 PM, Emrah wrote: >> >>> Hi all, >> >>> >> >>> How do we go about renegotiating the codecs of a call that is already >> established? >> >>> I tried uuid_media_reneg to no avail. >> >>> E.g.: Upgrade someone who's entered a 16k conference from PCMU to >> G722. >> >>> >> >>> Cheers and thanks, >> >>> Emrah >> >>> >> _________________________________________________________________________ >> >>> Professional FreeSWITCH Consulting Services: >> >>> consulting at freeswitch.org >> >>> http://www.freeswitchsolutions.com >> >>> >> >>> >> >>> >> >>> >> >>> Official FreeSWITCH Sites >> >>> http://www.freeswitch.org >> >>> http://wiki.freeswitch.org >> >>> http://www.cluecon.com >> >>> >> >>> 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 >> >>> >> >>> >> >>> >> >>> -- >> >>> Michael S Collins >> >>> Twitter: @mercutioviz >> >>> http://www.FreeSWITCH.org >> >>> http://www.ClueCon.com >> >>> http://www.OSTAG.org >> >>> >> >>> >> _________________________________________________________________________ >> >>> Professional FreeSWITCH Consulting Services: >> >>> consulting at freeswitch.org >> >>> http://www.freeswitchsolutions.com >> >>> >> >>> >> >>> >> >>> >> >>> Official FreeSWITCH Sites >> >>> http://www.freeswitch.org >> >>> http://wiki.freeswitch.org >> >>> http://www.cluecon.com >> >>> >> >>> 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 >> >> >> >> >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://wiki.freeswitch.org >> >> http://www.cluecon.com >> >> >> >> 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 >> > >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > 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 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130220/f5b50bb5/attachment.html From david.villasmil.work at gmail.com Wed Feb 20 16:10:57 2013 From: david.villasmil.work at gmail.com (David Villasmil) Date: Wed, 20 Feb 2013 14:10:57 +0100 Subject: [Freeswitch-users] nibblebill custom_sql_save In-Reply-To: References: Message-ID: Anyone about this? Thanks On Mon, Feb 18, 2013 at 2:07 PM, David Villasmil < david.villasmil.work at gmail.com> wrote: > Hello guys, > > I have the following config for nibbebill: > > > > > I see this on the log, which is obviously failing: > > > [UPDATE cards_table SET balance=balance- WHERE pin=12345;] > > is ${nibble_increment} not the correct variable to use? It looks like it > is empty! > > Thanks! > > David > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130220/ccff942e/attachment-0001.html From shaheryarkh at gmail.com Wed Feb 20 16:23:40 2013 From: shaheryarkh at gmail.com (Muhammad Shahzad) Date: Wed, 20 Feb 2013 14:23:40 +0100 Subject: [Freeswitch-users] nibblebill custom_sql_save In-Reply-To: References: Message-ID: Did you set nibble_increment variable? e.g. Here is its details, http://wiki.freeswitch.org/wiki/Mod_nibblebill#Nibble_Method_.28Default.29 Thank you. On Wed, Feb 20, 2013 at 2:10 PM, David Villasmil < david.villasmil.work at gmail.com> wrote: > Anyone about this? > > Thanks > > > On Mon, Feb 18, 2013 at 2:07 PM, David Villasmil < > david.villasmil.work at gmail.com> wrote: > >> Hello guys, >> >> I have the following config for nibbebill: >> >> >> >> >> I see this on the log, which is obviously failing: >> >> >> [UPDATE cards_table SET balance=balance- WHERE pin=12345;] >> >> is ${nibble_increment} not the correct variable to use? It looks like it >> is empty! >> >> Thanks! >> >> David >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +49 176 99 83 10 85 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130220/e4b51d27/attachment.html From mehroz.ashraf85 at gmail.com Wed Feb 20 16:33:07 2013 From: mehroz.ashraf85 at gmail.com (mehroz) Date: Wed, 20 Feb 2013 05:33:07 -0800 (PST) Subject: [Freeswitch-users] FS with SSL/TLS issues! Message-ID: <1361367187947-7587736.post@n2.nabble.com> Hi, it seems like i am unable to get it done. Here is what i have done 1. created CA (Root) certificates: ./gentls_cert setup -cn pbx.freeswitch.org -alt DNS:pbx.freeswitch.org -org freeswitch.org 2. created Server Certificate: ./gentls_cert create_server -cn pbx.freeswitch.org -alt DNS:pbx.freeswitch.org -org freeswitch.org *I replaced pbx.freeswitch.org and freeswitch.org with my FS IP address in which i am listening.* 3. Enabled /internal_ssl_enable=true/ and/ ip_tls_version=tlsv1/ in vars.xml . 4. When i try to Register 3CX on TLS over port 5061, all i see is : tport_tls.c:869 tls_connect() tls_connect(0x7fbd94e7a5e0): events NEGOTIATING tport_tls.c:869 tls_connect() tls_connect(0x7fbd94e7a5e0): events NEGOTIATING tport_tls.c:958 tls_connect() tls_connect(0x7fbd94e7a5e0): TLS setup failed (error:00000001:lib(0):func(0):reason(1)) tport.c:2084 tport_close() tport_close(0x7fbd94e7a5e0): tls/59.103.76.82:56863/sips tport.c:2730 tport_wakeup_pri() tport_wakeup_pri(0x7fbd94005090): events IN tport.c:869 tport_alloc_secondary() tport_alloc_secondary(0x7fbd94005090): new secondary tport 0x7fbd94f53c70 tport_type_tls.c:603 tport_tls_accept() tport_tls_accept(0x7fbd94f53c70): new connection from tls/59.103.76.82:56864/sips. PLease help!! -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FS-with-SSL-TLS-issues-tp7587736.html Sent from the freeswitch-users mailing list archive at Nabble.com. From mail at dmatthews.org Wed Feb 20 16:42:32 2013 From: mail at dmatthews.org (David Matthews) Date: Wed, 20 Feb 2013 13:42:32 +0000 Subject: [Freeswitch-users] Reading incoming skype chat messges Message-ID: <20130220134232.GA7101@dmatthews.org> in autoload_configs/skypopen.conf.xml was the problem, fixed by changing that to "true" -- David Matthews mail at dmatthews.org From gmaruzz at gmail.com Wed Feb 20 16:56:59 2013 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Wed, 20 Feb 2013 14:56:59 +0100 Subject: [Freeswitch-users] Reading incoming skype chat messges In-Reply-To: <20130220134232.GA7101@dmatthews.org> References: <20130220134232.GA7101@dmatthews.org> Message-ID: On Wed, Feb 20, 2013 at 2:42 PM, David Matthews wrote: > > > in autoload_configs/skypopen.conf.xml was the problem, fixed by changing > that > to "true" > > I was about to write that, but you was quick! :) -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130220/d2ad8d54/attachment.html From abaci64 at gmail.com Wed Feb 20 17:14:19 2013 From: abaci64 at gmail.com (Abaci) Date: Wed, 20 Feb 2013 09:14:19 -0500 Subject: [Freeswitch-users] Return from lua two values. In-Reply-To: <51248A1B.1000402@gmail.com> References: <51247522.5020801@gmail.com> <51248A1B.1000402@gmail.com> Message-ID: <5124DA3B.9000109@gmail.com> or use the multiset application http://wiki.freeswitch.org/wiki/Multiset On 2/20/2013 3:32 AM, Mimiko wrote: > On 20.02.2013 10:15, Steven Ayre wrote: >> Set the variables from the LUA script itself. >> >> http://wiki.freeswitch.org/wiki/Mod_lua#session:setVariable >> > > Thank you. I thought that was for other purposes. > From sdevoy at bizfocused.com Wed Feb 20 17:14:58 2013 From: sdevoy at bizfocused.com (Sean Devoy) Date: Wed, 20 Feb 2013 09:14:58 -0500 Subject: [Freeswitch-users] Determine if a number is valid/working In-Reply-To: References: <072701ce0ebf$c5808640$508192c0$@bizfocused.com> Message-ID: <0d0401ce0f74$aa9fb7a0$ffdf26e0$@bizfocused.com> Thank you for your responses, everyone. MC: This customer has rejected the idea of calling with a message as bad marketing!! In my experience marketing people have their own "logic rules" and I can never follow their thinking. And yes, they have 30 customer sales reps calling these "leads" all day long. They claim 1/3 of the leads have invalid phone numbers. My customer suspects fraud and that someone is submitting false leads to boost their lead rate and fees. Ken: I am not sure we have any language in common - lol. Where can I learn what "ASR and ACD" are? I looked up lidb and lnp already! Also, the cost of a call compared to the cost of a fraudulent lead (over $20) is insignificant. The ideas I have suggested to my customer, for what it is worth, are comparing the State (we are in US) for the IP address of the submitter, to the State on the form and the State of the area code on the form. In the situations where they do not agree, prompt the user for permission to robo call for verification. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Tuesday, February 19, 2013 2:37 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Determine if a number is valid/working Well, pretty much the only way to guarantee that a number is working is to call it and see what happens. I suspect Ken Rice has some experience in this area and I'd definitely like to hear it. If anyone else has thoughts I'd like to hear them as well. -MC P.S. - Why does the customer think that calling the number and playing a short message will take too long? Also, what is the customer's primary concern about the web leads having non-working numbers? Is it because they have humans manually dialing them and it's a waste of time when the person comes across a bad number? On Tue, Feb 19, 2013 at 8:40 AM, Sean Devoy wrote: HI All, My web site customer's call center claims that 1/3 of their web leads have nonworking phone numbers. Short of calling it, is there any way with FS (or even a paid service) to determine if a given phone number is a working number? I would like to make an Ajax SOAP call to a PHP page that checks (or dials) the number and returns a status for the validity of the phone number. I have an LUA script that will call, play a short message and hang up, but the customer thinks that may take too long. Any ideas? Thanks, Sean _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130220/3d70b024/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 70 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130220/3d70b024/attachment-0001.gif From steveayre at gmail.com Wed Feb 20 17:55:19 2013 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 20 Feb 2013 14:55:19 +0000 Subject: [Freeswitch-users] Determine if a number is valid/working In-Reply-To: <0d0401ce0f74$aa9fb7a0$ffdf26e0$@bizfocused.com> References: <072701ce0ebf$c5808640$508192c0$@bizfocused.com> <0d0401ce0f74$aa9fb7a0$ffdf26e0$@bizfocused.com> Message-ID: > > Ken: I am not sure we have any language in common ? lol. Where can I > learn what ?ASR and ACD? are? I looked up lidb and lnp already! They're measurements of call quality. ASR is the % of terminated calls that're answered. ACD is average call length. Short of calling the numbers LNP-type databases would be your only bet I think, but as Ken points out even they won't be reliable - you may get hits for invalid numbers if the prefix is valid. And access to them is not free. "Bad marketing" sounds to be that they are indeed worried about annoying the customer with robo-calls. They'll be less annoying if they actually route the call to a sales rep rather than just playing a message and hanging up, via a queue. -Steve On 20 February 2013 14:14, Sean Devoy wrote: > Thank you for your responses, ev eryone.**** > > ** ** > > MC: This customer has rejected the idea of calling with a message as bad > marketing!! In my experience marketing people have their own ?logic rules? > and I can never follow their thinking. And yes, they have 30 customer > sales reps calling these ?leads? all day long. They claim 1/3 of the leads > have invalid phone numbers. My customer suspects fraud and that someone is > submitting false leads to boost their lead rate and fees.**** > > ** ** > > Ken: I am not sure we have any language in common ? lol. Where can I > learn what ?ASR and ACD? are? I looked up lidb and lnp already!**** > > ** ** > > Also, the cost of a call compared to the cost of a fraudulent lead (over > $20) is insignificant. **** > > ** ** > > The ideas I have suggested to my customer, for what it is worth, are > comparing the State (we are in US) for the IP address of the submitter, to > the State on the form and the State of the area code on the form. In the > situations where they do not agree, prompt the user for permission to robo > call for verification.**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael > Collins > *Sent:* Tuesday, February 19, 2013 2:37 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Determine if a number is valid/working** > ** > > ** ** > > Well, pretty much the only way to guarantee that a number is working is to > call it and see what happens. I suspect Ken Rice has some experience in > this area and I'd definitely like to hear it. If anyone else has thoughts > I'd like to hear them as well. > > -MC > > P.S. - Why does the customer think that calling the number and playing a > short message will take too long? Also, what is the customer's primary > concern about the web leads having non-working numbers? Is it because they > have humans manually dialing them and it's a waste of time when the person > comes across a bad number?**** > > On Tue, Feb 19, 2013 at 8:40 AM, Sean Devoy wrote: > **** > > **** > > HI All,**** > > **** > > My web site customer?s call center claims that 1/3 of their web leads have > nonworking phone numbers. Short of calling it, is there any way with FS > (or even a paid service) to determine if a given phone number is a working > number? I would like to make an Ajax SOAP call to a PHP page that checks > (or dials) the number and returns a status for the validity of the phone > number.**** > > **** > > I have an LUA script that will call, play a short message and hang up, but > the customer thinks that may take too long.**** > > **** > > Any ideas?**** > > **** > > Thanks,**** > > Sean**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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**** > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130220/e575102b/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 70 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130220/e575102b/attachment.gif From vbvbrj at gmail.com Wed Feb 20 18:04:43 2013 From: vbvbrj at gmail.com (Mimiko) Date: Wed, 20 Feb 2013 17:04:43 +0200 Subject: [Freeswitch-users] Adtran Total Access 916e 4242916L1 Message-ID: <5124E60B.4010505@gmail.com> Hi. Who knows about this device: Adtran Total Access 916e 4242916L1 http://www.newegg.com/Product/Product.aspx?Item=N82E16833706027 How stable is it? Does it use LAN ports in failover mode? Does it use T1/FT1 WAN ports in failover manner connected to a PBX with two PRI ports? How good it is working with FS? How much concurrent connection does it support? I asked to the company via email a week ago, and no response till now. -- Mimiko desu. From vbvbrj at gmail.com Wed Feb 20 18:06:35 2013 From: vbvbrj at gmail.com (Mimiko) Date: Wed, 20 Feb 2013 17:06:35 +0200 Subject: [Freeswitch-users] Return from lua two values. In-Reply-To: <5124DA3B.9000109@gmail.com> References: <51247522.5020801@gmail.com> <51248A1B.1000402@gmail.com> <5124DA3B.9000109@gmail.com> Message-ID: <5124E67B.3040701@gmail.com> On 20.02.2013 16:14, Abaci wrote: > or use the multiset application > http://wiki.freeswitch.org/wiki/Multiset How it can set values to two variables calling one lua script only and only once? Doesn't seems to be so. -- Mimiko desu. From krice at freeswitch.org Wed Feb 20 18:12:52 2013 From: krice at freeswitch.org (Ken Rice) Date: Wed, 20 Feb 2013 09:12:52 -0600 Subject: [Freeswitch-users] Adtran Total Access 916e 4242916L1 In-Reply-To: <5124E60B.4010505@gmail.com> Message-ID: The Adtran TA9XX series are pretty good units. Check out their website at adtran.com to find tech manuals for these things... I cant answer all the questions below, but these are very stable units designed to be sold to carriers (and used by many carriers) How many concurrent connections depends on the specific model you are getting. On 2/20/13 9:04 AM, "Mimiko" wrote: > Hi. > > Who knows about this device: Adtran Total Access 916e 4242916L1 > http://www.newegg.com/Product/Product.aspx?Item=N82E16833706027 > > How stable is it? Does it use LAN ports in failover mode? Does it use > T1/FT1 WAN ports in failover manner connected to a PBX with two PRI > ports? How good it is working with FS? How much concurrent connection > does it support? > > I asked to the company via email a week ago, and no response till now. -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch From steveayre at gmail.com Wed Feb 20 18:19:53 2013 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 20 Feb 2013 15:19:53 +0000 Subject: [Freeswitch-users] Return from lua two values. In-Reply-To: <5124E67B.3040701@gmail.com> References: <51247522.5020801@gmail.com> <51248A1B.1000402@gmail.com> <5124DA3B.9000109@gmail.com> <5124E67B.3040701@gmail.com> Message-ID: Return a single string of "name1=value1 name2=value2" -Steve On 20 February 2013 15:06, Mimiko wrote: > On 20.02.2013 16:14, Abaci wrote: >> or use the multiset application >> http://wiki.freeswitch.org/wiki/Multiset > > How it can set values to two variables calling one lua script only and > only once? > Doesn't seems to be so. > > -- > Mimiko desu. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 royce3 at gmail.com Wed Feb 20 18:20:51 2013 From: royce3 at gmail.com (Royce Mitchell III) Date: Wed, 20 Feb 2013 09:20:51 -0600 Subject: [Freeswitch-users] Adtran Total Access 916e 4242916L1 In-Reply-To: References: <5124E60B.4010505@gmail.com> Message-ID: I have a few 908e's, and I'm experiencing a serious incompatibility between FreeSWITCH, the 908e, and the AT&T wireless network. I'm confident I will get it fixed eventually, but just sharing what I know about the device. On Wed, Feb 20, 2013 at 9:12 AM, Ken Rice wrote: > The Adtran TA9XX series are pretty good units. Check out their website at > adtran.com to find tech manuals for these things... > > I cant answer all the questions below, but these are very stable units > designed to be sold to carriers (and used by many carriers) > > How many concurrent connections depends on the specific model you are > getting. > > > On 2/20/13 9:04 AM, "Mimiko" wrote: > >> Hi. >> >> Who knows about this device: Adtran Total Access 916e 4242916L1 >> http://www.newegg.com/Product/Product.aspx?Item=N82E16833706027 >> >> How stable is it? Does it use LAN ports in failover mode? Does it use >> T1/FT1 WAN ports in failover manner connected to a PBX with two PRI >> ports? How good it is working with FS? How much concurrent connection >> does it support? >> >> I asked to the company via email a week ago, and no response till now. > > -- > Ken > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > irc.freenode.net #freeswitch > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -- Royce Mitchell IT Consultant ITAS Solutions royce3 at itas-solutions.com From krice at freeswitch.org Wed Feb 20 18:27:11 2013 From: krice at freeswitch.org (Ken Rice) Date: Wed, 20 Feb 2013 09:27:11 -0600 Subject: [Freeswitch-users] Adtran Total Access 916e 4242916L1 In-Reply-To: Message-ID: What kind of incompatibility? As this does not make a lot of sense for a TDM/SIP device to have problems with ATT Wireless a Cellular network, they cant talk directly to each other On 2/20/13 9:20 AM, "Royce Mitchell III" wrote: > I have a few 908e's, and I'm experiencing a serious incompatibility > between FreeSWITCH, the 908e, and the AT&T wireless network. I'm > confident I will get it fixed eventually, but just sharing what I know > about the device. > > On Wed, Feb 20, 2013 at 9:12 AM, Ken Rice wrote: >> The Adtran TA9XX series are pretty good units. Check out their website at >> adtran.com to find tech manuals for these things... >> >> I cant answer all the questions below, but these are very stable units >> designed to be sold to carriers (and used by many carriers) >> >> How many concurrent connections depends on the specific model you are >> getting. >> >> >> On 2/20/13 9:04 AM, "Mimiko" wrote: >> >>> Hi. >>> >>> Who knows about this device: Adtran Total Access 916e 4242916L1 >>> http://www.newegg.com/Product/Product.aspx?Item=N82E16833706027 >>> >>> How stable is it? Does it use LAN ports in failover mode? Does it use >>> T1/FT1 WAN ports in failover manner connected to a PBX with two PRI >>> ports? How good it is working with FS? How much concurrent connection >>> does it support? >>> >>> I asked to the company via email a week ago, and no response till now. >> >> -- >> Ken >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> irc.freenode.net #freeswitch >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 > > -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch From royce3 at gmail.com Wed Feb 20 18:45:48 2013 From: royce3 at gmail.com (Royce Mitchell III) Date: Wed, 20 Feb 2013 09:45:48 -0600 Subject: [Freeswitch-users] Adtran Total Access 916e 4242916L1 In-Reply-To: References: Message-ID: when placing an outbound call through the 908e, via PRI hosted by TWTelecom and dialing a destination on the AT&T wireless network, a "different" series of ISDN codes comes back than from most other destinations. This series of ISDN codes causes the Adtran to send a 183+early media followed by a 180 w/o media, which causes the caller to hear nothing during most of the ringing process. All work-arounds I have tried to employ so far ( like ignore early media ) have had disastrous side-effects, we've almost lost clients over them. I put out a request to pay for a FreeSWITCH expert to help fix the issue, but I've been too busy with other projects to follow up. On Wed, Feb 20, 2013 at 9:27 AM, Ken Rice wrote: > What kind of incompatibility? As this does not make a lot of sense for a > TDM/SIP device to have problems with ATT Wireless a Cellular network, they > cant talk directly to each other > > On 2/20/13 9:20 AM, "Royce Mitchell III" wrote: > >> I have a few 908e's, and I'm experiencing a serious incompatibility >> between FreeSWITCH, the 908e, and the AT&T wireless network. I'm >> confident I will get it fixed eventually, but just sharing what I know >> about the device. >> >> On Wed, Feb 20, 2013 at 9:12 AM, Ken Rice wrote: >>> The Adtran TA9XX series are pretty good units. Check out their website at >>> adtran.com to find tech manuals for these things... >>> >>> I cant answer all the questions below, but these are very stable units >>> designed to be sold to carriers (and used by many carriers) >>> >>> How many concurrent connections depends on the specific model you are >>> getting. >>> >>> >>> On 2/20/13 9:04 AM, "Mimiko" wrote: >>> >>>> Hi. >>>> >>>> Who knows about this device: Adtran Total Access 916e 4242916L1 >>>> http://www.newegg.com/Product/Product.aspx?Item=N82E16833706027 >>>> >>>> How stable is it? Does it use LAN ports in failover mode? Does it use >>>> T1/FT1 WAN ports in failover manner connected to a PBX with two PRI >>>> ports? How good it is working with FS? How much concurrent connection >>>> does it support? >>>> >>>> I asked to the company via email a week ago, and no response till now. >>> >>> -- >>> Ken >>> http://www.FreeSWITCH.org >>> http://www.ClueCon.com >>> http://www.OSTAG.org >>> irc.freenode.net #freeswitch >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >> >> > > -- > Ken > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > irc.freenode.net #freeswitch > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -- Royce Mitchell IT Consultant ITAS Solutions royce3 at itas-solutions.com From yehavi.bourvine at gmail.com Wed Feb 20 19:02:31 2013 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Wed, 20 Feb 2013 18:02:31 +0200 Subject: [Freeswitch-users] FS with SSL/TLS issues! In-Reply-To: <1361367187947-7587736.post@n2.nabble.com> References: <1361367187947-7587736.post@n2.nabble.com> Message-ID: Hi, As a first attempt try setting the organization name and the pbx name to be the same: The fully qualified DNS name of the FreeSwitch server (name, not dotted IP). Then, set the phones to register to the name and not the IP. I know that with Polycom you must use names. Try catching this connection with wireshark - maybe it will tell you some more details. __Yehavi: 2013/2/20 mehroz > Hi, > > it seems like i am unable to get it done. Here is what i have done > > 1. created CA (Root) certificates: > ./gentls_cert setup -cn pbx.freeswitch.org -alt DNS:pbx.freeswitch.org-org > freeswitch.org > > 2. created Server Certificate: > ./gentls_cert create_server -cn pbx.freeswitch.org -alt > DNS:pbx.freeswitch.org -org freeswitch.org > > *I replaced pbx.freeswitch.org and freeswitch.org with my FS IP address in > which i am listening.* > > 3. Enabled /internal_ssl_enable=true/ and/ ip_tls_version=tlsv1/ in > vars.xml . > > 4. When i try to Register 3CX on TLS over port 5061, all i see is : > tport_tls.c:869 tls_connect() tls_connect(0x7fbd94e7a5e0): events > NEGOTIATING > tport_tls.c:869 tls_connect() tls_connect(0x7fbd94e7a5e0): events > NEGOTIATING > tport_tls.c:958 tls_connect() tls_connect(0x7fbd94e7a5e0): TLS setup failed > (error:00000001:lib(0):func(0):reason(1)) > tport.c:2084 tport_close() tport_close(0x7fbd94e7a5e0): > tls/59.103.76.82:56863/sips > tport.c:2730 tport_wakeup_pri() tport_wakeup_pri(0x7fbd94005090): events IN > tport.c:869 tport_alloc_secondary() tport_alloc_secondary(0x7fbd94005090): > new secondary tport 0x7fbd94f53c70 > tport_type_tls.c:603 tport_tls_accept() tport_tls_accept(0x7fbd94f53c70): > new connection from tls/59.103.76.82:56864/sips. > > > > PLease help!! > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/FS-with-SSL-TLS-issues-tp7587736.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130220/0c76c718/attachment.html From tknchris at gmail.com Wed Feb 20 14:57:04 2013 From: tknchris at gmail.com (chris) Date: Wed, 20 Feb 2013 06:57:04 -0500 Subject: [Freeswitch-users] DID providers - any thoughts? (jan 2013) In-Reply-To: References: Message-ID: +1 here, flowroute has exceeded my expectations in many areas. The fact that they actually have humans available via phone, and even humans who understand the technology quite well. Reliability has been great but you should always plan for a rainy day and have a backup plan, even the biggest and best have issues from time to time chris On Thu, Jan 31, 2013 at 5:34 PM, Steven Schoch < schoch+freeswitch.org at xwin32.com> wrote: > On Wed, Jan 30, 2013 at 4:59 AM, Cal Leeming [Simplicity Media Ltd] < > cal.leeming at simplicitymedialtd.co.uk> wrote: > >> >> Flowroute - Any else able to offer some reviews on this company - >> specifically relating to US inbound?? >> > > I'm just starting my VoIP venture and I have chosen Flowroute as our > provider. So far, mostly as a test, I have transferred one of our AT&T > lines to Flowroute. It worked very smoothly. One thing I discovered, > which you all probably know, but I'm new with VoIP, was that when I called > the number from a POTS line, it went through the PSTN, through Flowroute, > through FreeSwitch, and rang on my Polycom phone even before I heard the > first ring from the calling phone. > > Also, the Flowroute tech support team responds very quickly and helpfully. > > No complaints. > > -- > Steve > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130220/8450beb4/attachment-0001.html From khorsmann at gmail.com Wed Feb 20 15:03:19 2013 From: khorsmann at gmail.com (Karsten Horsmann) Date: Wed, 20 Feb 2013 13:03:19 +0100 Subject: [Freeswitch-users] originate to loopback/answer and park a lua script from another lua script In-Reply-To: References: Message-ID: Hello Michael, i create this loop.lua to update the sip username spelling (register is not casesensitiv vs. fs is casesensitiv). Sure i could start the loop.lua with mod_lua as start script. With the loopback hack i want to simulate an "job control" and maybe do some nagios check to be sure that the loop.lua is alive (simple grep output of "show channels"). Is there a other way to check that the loop.lua is alive from outside? I play around with an start.lua to originate the loop.lua at startup. But my start.lua didnt work. Also an freeswitch.api call from start.lua didnt do the trick. Any hints? my "start.lua": -- new_session = freeswitch.Session("loopback/set:job=1234\,answer\,park/default/inline lua:loop.lua inline", session) new_session = freeswitch.Session("loopback/set:job=1234", session) new_session:answer(); new_session:execute("park/default/inline lua:loop.lua inline") Got this FS-Log output http://pastebin.freeswitch.org/20595 Kind Regards Karsten 2013/2/15 Michael Collins : > Can you share with us a small Lua code sample demonstrating what you are > trying to do? That might make it easier for us to help. > > -MC > > On Fri, Feb 15, 2013 at 3:26 AM, Karsten Horsmann > wrote: >> >> Hello! >> >> as described in Anthonys mail >> >> http://lists.freeswitch.org/pipermail/freeswitch-users/2013-January/091769.html >> i want also to originate my loop.lua to an loopback channel. >> >> No problem from fs_cli. I want to originate the loop.lua from my >> mod_lua start lua script. >> >> How can i do this? The freeswitch.Session() didnt work as aspected. >> >> >> -- >> Kind Regards >> *Karsten Horsmann* > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org -- Mit freundlichen Gr??en *Karsten Horsmann* -------------- next part -------------- package.path = package.path .. ";/usr/share/lua/5.1/?.lua" package.cpath = package.cpath .. ";/usr/lib64/lua/5.1/?.so" -- mysql Config require ('DBI') mydb,myuser,mypass,myhost = "yourdb,youruser,yourpass,yourhost" -- Logging LOGLEVEL = "info" -- Progname PROGNAME = "loop.lua" -- functions function logger(message) freeswitch.consoleLog(LOGLEVEL,"["..PROGNAME.."] "..message.."\n") end function mysqlabfrage(u) if u == nil then return end local dbh = assert(DBI.Connect('MySQL', mydb, myuser, mypass, myhost)) local sth = assert(dbh:prepare('SELECT username, domain, password, fs_username FROM subscriber WHERE username=? LIMIT 1')) sth:execute(u) -- print all rows, the rows will be indexed by field names row = sth:fetch ({}, "a") if not row then return end username = row.username fs_username = row.fs_username domain = row.domain password = row.password dbh:close() sth:close() return username,fs_username,domain,password end function mysqlupdate(u) if u == nil then logger("Execute Update failed - no username given") return end local dbh = assert(DBI.Connect('MySQL', mydb, myuser, mypass, myhost)) dbh:autocommit(true) local wrt = assert(dbh:prepare("UPDATE subscriber SET fs_username=? WHERE username=? LIMIT 1")) logger("Execute Update fs_username to [" .. u .. "] !") success,err = wrt:execute(u,u) if success == false then logger("Execute Update ERROR") end dbh:close() wrt:close() end -- Send Stop Event on arg[1] == stop if argv[1] then i=1 while argv[i] do if argv[i] == "stop" then --Send Stop message local event = freeswitch.Event("custom", "lua::stop_event") event:addHeader("Action", "stop") event:fire() logger("Sent stop message to lua script") return end i=i+1 end return end -- main loop logger("Starting script and loop forever") con = freeswitch.EventConsumer("CUSTOM"); run = true while run do for e in (function() return con:pop(1,1000) end) do -- logger("event\n" .. e:serialize("xml")) element = e:getHeader("Event-Subclass") or "" if (element == "lua::stop_event") then action = e:getHeader("Action") or "" if (action == "stop") then logger("got stop message, Exiting") run = false break end end if (element == "sofia::register") then registerstatus = e:getHeader("status") if (registerstatus:find("Registered",1,true)) then req_username = e:getHeader("username") if req_username then logger("user registerd " .. req_username) mysqlabfrage(req_username) if req_username ~= fs_username then mysqlupdate(req_username) end end end end end end From mario_fs at mgtech.com Wed Feb 20 19:57:42 2013 From: mario_fs at mgtech.com (Mario G) Date: Wed, 20 Feb 2013 08:57:42 -0800 Subject: [Freeswitch-users] Return from lua two values. In-Reply-To: References: <51247522.5020801@gmail.com> <51248A1B.1000402@gmail.com> <5124DA3B.9000109@gmail.com> <5124E67B.3040701@gmail.com> Message-ID: <925F78EB-437F-4657-9E1A-45F9F2B6E11E@mgtech.com> Hmm, not sure if I understand what exactly you need but here is how I send three parms/vars to lua: And I return 3 vars from lua to FS: session:setVariable("lua_ringback",ringback) session:setVariable("lua_vmmsg",vmmsg) session:setVariable("lua_ringtone",ringtone) FS then uses $lua_ringback, etc... Mario G On Feb 20, 2013, at 7:19 AM, Steven Ayre wrote: > Return a single string of "name1=value1 name2=value2" > > -Steve > > > On 20 February 2013 15:06, Mimiko wrote: >> On 20.02.2013 16:14, Abaci wrote: >>> or use the multiset application >>> http://wiki.freeswitch.org/wiki/Multiset >> >> How it can set values to two variables calling one lua script only and >> only once? >> Doesn't seems to be so. >> >> -- >> Mimiko desu. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130220/d22e7d6a/attachment.html From msc at freeswitch.org Wed Feb 20 19:58:28 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 20 Feb 2013 08:58:28 -0800 Subject: [Freeswitch-users] FreeSWITCH Weekly Conference Call Message-ID: Hey folks, today's conference call agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2013_02_20 We will be doing a few news updates, including a recap of last night's "FreeSWITCH HA" conference call. Then I will be doing a tour of the CudaTel Communications Server. Talk to you at 1PM EST/10AM PST -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130220/b9185012/attachment.html From msc at freeswitch.org Wed Feb 20 20:08:35 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 20 Feb 2013 09:08:35 -0800 Subject: [Freeswitch-users] freeswitch hack In-Reply-To: <958ABE902C1F47B09FFAAE7EDEE074B1@imsdeveloper> References: <958ABE902C1F47B09FFAAE7EDEE074B1@imsdeveloper> Message-ID: On Wed, Feb 20, 2013 at 1:53 AM, Mario Karakanovski wrote: > Thanks Ken,**** > > ** ** > > It is helpful, but I still think there is some security issue. I?ve double > check configuration. I?ve try to reproduce the issue trying to do direct > call (TCP and UDP) or authenticate with invalid user, but everything works > as expected ? calls/authentication was rejected. I?ve decide to log the > traffic ? maybe I will be able to see where is the problem. > What "security issue"? You said that they cannot make calls with the passwords that they've guessed, correct? About the only thing left to do is set up fail2ban and just shut the door on them when they fail too many times. -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130220/f346250a/attachment.html From msc at freeswitch.org Wed Feb 20 20:11:46 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 20 Feb 2013 09:11:46 -0800 Subject: [Freeswitch-users] Event pick In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF23AC6B5@Mail-Kilo.squay.com> References: <592A9CF93E12394E8472A6CC66E66BF23AC6B5@Mail-Kilo.squay.com> Message-ID: What value is in obCause when the originator cancels the call? Print the value in the else block and find out. -MC On Wed, Feb 20, 2013 at 4:37 AM, Archana Venugopan wrote: > Hi,**** > > I need to capture this ORIGINATOR _CANCEL event in my lua script. From > google I found the below code. But we establish our calls from dialplan XML > as like this, data="sofia/external/0${msc_prefix}$1$2 at 10.30.3.27"/>.**** > > Since am not that well versed in lua script, I am not sure in the place of > obsession below what should I pass. As far I understood the = > sofia/external/0${msc_prefix}$1$2 at 10.30.3.27, this is what I should put > there but we already have this in our XML.**** > > Please suggest me on how should I proceed. Many thanks.**** > > ** ** > > -- Initiate an outbound call**** > > ** ** > > obSession = freeswitch.Session("sofia/192.168.0.4/1002")**** > > ** ** > > -- Check to see if the call was answered**** > > ** ** > > if obSession:ready() then**** > > -- Do something good here**** > > ** ** > > else -- This means the call was not answered ... Check for the reason** > ** > > ** ** > > local obCause = obSession:hangupCause()**** > > ** ** > > freeswitch.consoleLog("info", "obSession:hangupCause() = " .. obCause ) > **** > > ** ** > > if ( obCause == "USER_BUSY" ) then -- SIP 486**** > > -- For BUSY you may reschedule the call for later**** > > elseif ( obCause == "NO_ANSWER" ) then**** > > -- Call them back in an hour**** > > elseif ( obCause == "ORIGINATOR_CANCEL" ) then -- SIP 487**** > > -- May need to check for network congestion or problems**** > > else**** > > -- Log these issues**** > > end**** > > end**** > > ** ** > > ** ** > > ** ** > > ** ** > > 2013-02-19 17:56:02.277173 [NOTICE] sofia.c:7082 Hangup sofia/internal/ > 206 at fsfailover.uk01.com [CS_EXECUTE] [ORIGINATOR_CANCEL]**** > > ** ** > > Regards,**** > > Archana**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130220/125a1dac/attachment-0001.html From msc at freeswitch.org Wed Feb 20 20:12:35 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 20 Feb 2013 09:12:35 -0800 Subject: [Freeswitch-users] Changing codec during calls In-Reply-To: References: <6EF8B8E9-4CC5-480C-B1C1-09CBC533470C@kavun.ch> <35AF9E79-771D-440A-BEBB-A856D831161C@kavun.ch> Message-ID: Definitely get the console log and SIP trace of those problem calls and put them in pastein.freeswitch.org. -MC On Wed, Feb 20, 2013 at 5:04 AM, Dmitry Lysenko wrote: > Wow! This great feature is really work. I have the same issue - one way > audio. And another: "show channels" always return first negotiated codec. > When switching codec back, audio working fine. Tested with 3 different > softphones. > If someone interested, please confirm. > Thanks. > > > 2013/2/19 Michael Collins > >> pastebin the log. I'll bet Steven can diagnose it! ;) >> -MC >> >> >> On Tue, Feb 19, 2013 at 8:37 AM, Emrah wrote: >> >>> Excellent, that worked. >>> >>> However I am having a one way audio issue, where the peer that has its >>> codecs changed can hear but cannot be heard. >>> Even if both bridged peers are changed to a common codec it doesn't work. >>> >>> Any idea? >>> >>> Thanks a bunch for your help. >>> On Feb 19, 2013, at 3:47 AM, Steven Ayre wrote: >>> >>> > Try: >>> > uuid_media_reneg 49f428e5-1d72-42d8-b0e1-5b3f29f9a4ca =PCMU >>> > >>> > (ie add = before PCMU) >>> > >>> > It appears to affect something in the code... >>> > >>> > -Steve >>> > >>> > >>> > On 19 February 2013 05:54, Emrah wrote: >>> >> Hey, >>> >> >>> >> In short, nothing seems to happen. I tried several phones. >>> >> I also tried enabling codec renegotiation on hold/unhold thinking >>> that the change in SDP with uuid_media_reneg would do it. >>> >> >>> >> This is what happens: >>> >> freeswitch at internal> show channels >>> >> >>> uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,sent_callee_name,sent_callee_num >>> >> 49f428e5-1d72-42d8-b0e1-5b3f29f9a4ca,inbound,2013-02-19 >>> 00:44:35,1361252675,sofia/internal/10000 at sip.domain.net,CS_EXECUTE,Emrah >>> - Macbook >>> Pro,10000,100.100.100.100,conf-3111,conference,3111-sip.domain.net at ek-custom-16k >>> ,XML,ek-conference,G722,16000,64000,G722,16000,64000,, >>> cornflakes.domain.net,10000 at sip.domain.net,,ACTIVE,,,,,, >>> >> >>> >> 1 total. >>> >> >>> >> freeswitch at internal> uuid_media_reneg >>> 49f428e5-1d72-42d8-b0e1-5b3f29f9a4ca PCMU >>> >> +OK Success >>> >> >>> >> 2013-02-19 00:45:25.172659 [DEBUG] sofia_glue.c:2647 Local SDP: >>> >> v=0 >>> >> o=FreeSWITCH 1361234787 1361234790 IN IP4 1.2.3.4 >>> >> s=FreeSWITCH >>> >> c=IN IP4 1.2.3.4 >>> >> t=0 0 >>> >> m=audio 17888 RTP/AVP 9 0 101 >>> >> a=rtpmap:101 telephone-event/8000 >>> >> a=fmtp:101 0-16 >>> >> a=silenceSupp:off - - - - >>> >> a=ptime:20 >>> >> a=sendrecv >>> >> >>> >> 2013-02-19 00:45:25.172659 [DEBUG] switch_core_session.c:975 Send >>> signal sofia/internal/10000 at sip.domain.net [BREAK] >>> >> 2013-02-19 00:45:25.182652 [DEBUG] sofia.c:5574 Channel >>> sofia/internal/10000 at sip.domain.net entering state [calling][0] >>> >> 2013-02-19 00:45:25.292661 [DEBUG] switch_core_session.c:975 Send >>> signal sofia/internal/10000 at sip.domain.net [BREAK] >>> >> 2013-02-19 00:45:25.292661 [DEBUG] switch_core_session.c:975 Send >>> signal sofia/internal/10000 at sip.domain.net [BREAK] >>> >> 2013-02-19 00:45:25.302683 [INFO] sofia.c:931 sofia/internal/ >>> 10000 at sip.domain.net Update Callee ID to "10000" <10000> >>> >> 2013-02-19 00:45:25.302683 [DEBUG] sofia.c:5574 Channel >>> sofia/internal/10000 at sip.domain.net entering state [completing][200] >>> >> 2013-02-19 00:45:25.302683 [DEBUG] sofia.c:5585 Remote SDP: >>> >> v=0 >>> >> o=- 3570241475 3570241476 IN IP4 10.0.0.131 >>> >> s=pjmedia >>> >> c=IN IP4 10.0.0.131 >>> >> t=0 0 >>> >> a=X-nat:0 >>> >> m=audio 4010 RTP/AVP 9 101 >>> >> a=rtcp:4011 IN IP4 10.0.0.131 >>> >> a=rtpmap:9 G722/8000 >>> >> a=sendrecv >>> >> a=rtpmap:101 telephone-event/8000 >>> >> a=fmtp:101 0-15 >>> >> >>> >> 2013-02-19 00:45:25.302683 [DEBUG] switch_core_session.c:975 Send >>> signal sofia/internal/10000 at sip.domain.net [BREAK] >>> >> 2013-02-19 00:45:25.302683 [DEBUG] switch_core_session.c:975 Send >>> signal sofia/internal/10000 at sip.domain.net [BREAK] >>> >> 2013-02-19 00:45:25.322657 [DEBUG] sofia.c:5574 Channel >>> sofia/internal/10000 at sip.domain.net entering state [ready][200] >>> >> 2013-02-19 00:45:25.322657 [DEBUG] sofia.c:5582 Duplicate SDP >>> >> v=0 >>> >> o=- 3570241475 3570241476 IN IP4 10.0.0.131 >>> >> s=pjmedia >>> >> c=IN IP4 10.0.0.131 >>> >> t=0 0 >>> >> a=X-nat:0 >>> >> m=audio 4010 RTP/AVP 9 101 >>> >> a=rtcp:4011 IN IP4 10.0.0.131 >>> >> a=rtpmap:9 G722/8000 >>> >> a=sendrecv >>> >> a=rtpmap:101 telephone-event/8000 >>> >> a=fmtp:101 0-15 >>> >> >>> >> 2013-02-19 00:45:25.322657 [DEBUG] sofia_glue.c:5137 Audio Codec >>> Compare [G722:9:8000:20:64000]/[G722:9:8000:20:64000] >>> >> 2013-02-19 00:45:25.322657 [DEBUG] sofia_glue.c:3022 Already using >>> G722 >>> >> 2013-02-19 00:45:25.322657 [DEBUG] sofia_glue.c:5266 Set 2833 dtmf >>> send/recv payload to 101 >>> >> >>> >> >>> >> Thanks for any help, >>> >> Emrah >>> >> On Feb 18, 2013, at 1:38 PM, Michael Collins >>> wrote: >>> >> >>> >>> What happened with the uuid_media_reneg command? >>> >>> -MC >>> >>> >>> >>> On Sat, Feb 16, 2013 at 3:10 PM, Emrah wrote: >>> >>> Hi all, >>> >>> >>> >>> How do we go about renegotiating the codecs of a call that is >>> already established? >>> >>> I tried uuid_media_reneg to no avail. >>> >>> E.g.: Upgrade someone who's entered a 16k conference from PCMU to >>> G722. >>> >>> >>> >>> Cheers and thanks, >>> >>> Emrah >>> >>> >>> _________________________________________________________________________ >>> >>> Professional FreeSWITCH Consulting Services: >>> >>> consulting at freeswitch.org >>> >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> >>> http://www.freeswitch.org >>> >>> http://wiki.freeswitch.org >>> >>> http://www.cluecon.com >>> >>> >>> >>> 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 >>> >>> >>> >>> >>> >>> >>> >>> -- >>> >>> Michael S Collins >>> >>> Twitter: @mercutioviz >>> >>> http://www.FreeSWITCH.org >>> >>> http://www.ClueCon.com >>> >>> http://www.OSTAG.org >>> >>> >>> >>> >>> _________________________________________________________________________ >>> >>> Professional FreeSWITCH Consulting Services: >>> >>> consulting at freeswitch.org >>> >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> >>> http://www.freeswitch.org >>> >>> http://wiki.freeswitch.org >>> >>> http://www.cluecon.com >>> >>> >>> >>> 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 >>> >> >>> >> >>> >> >>> _________________________________________________________________________ >>> >> Professional FreeSWITCH Consulting Services: >>> >> consulting at freeswitch.org >>> >> http://www.freeswitchsolutions.com >>> >> >>> >> >>> >> >>> >> >>> >> Official FreeSWITCH Sites >>> >> http://www.freeswitch.org >>> >> http://wiki.freeswitch.org >>> >> http://www.cluecon.com >>> >> >>> >> 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 >>> > >>> > >>> _________________________________________________________________________ >>> > Professional FreeSWITCH Consulting Services: >>> > consulting at freeswitch.org >>> > http://www.freeswitchsolutions.com >>> > >>> > >>> > >>> > >>> > Official FreeSWITCH Sites >>> > http://www.freeswitch.org >>> > http://wiki.freeswitch.org >>> > http://www.cluecon.com >>> > >>> > 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 >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130220/81b380d5/attachment-0001.html From msc at freeswitch.org Wed Feb 20 20:26:01 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 20 Feb 2013 09:26:01 -0800 Subject: [Freeswitch-users] Determine if a number is valid/working In-Reply-To: <0d0401ce0f74$aa9fb7a0$ffdf26e0$@bizfocused.com> References: <072701ce0ebf$c5808640$508192c0$@bizfocused.com> <0d0401ce0f74$aa9fb7a0$ffdf26e0$@bizfocused.com> Message-ID: On Wed, Feb 20, 2013 at 6:14 AM, Sean Devoy wrote: > Thank you for your responses, everyone.**** > > ** ** > > MC: This customer has rejected the idea of calling with a message as bad > marketing!! In my experience marketing people have their own ?logic rules? > and I can never follow their thinking. And yes, they have 30 customer > sales reps calling these ?leads? all day long. They claim 1/3 of the leads > have invalid phone numbers. My customer suspects fraud and that someone is > submitting false leads to boost their lead rate and fees. > I agree with Steven's assessment (elsewhere in this thread). > **** > > ** ** > > Ken: I am not sure we have any language in common ? lol. Where can I > learn what ?ASR and ACD? are? I looked up lidb and lnp already!**** > > ** ** > > Also, the cost of a call compared to the cost of a fraudulent lead (over > $20) is insignificant. > How is that cost calculated and to whom does the money go? Maybe the "verification call" needs to take place at the time that the form is filled out. > **** > > ** ** > > The ideas I have suggested to my customer, for what it is worth, are > comparing the State (we are in US) for the IP address of the submitter, to > the State on the form and the State of the area code on the form. In the > situations where they do not agree, prompt the user for permission to robo > call for verification.**** > > ** > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130220/08f89727/attachment.html From a.venugopan at mundio.com Wed Feb 20 20:33:33 2013 From: a.venugopan at mundio.com (Archana Venugopan) Date: Wed, 20 Feb 2013 17:33:33 +0000 Subject: [Freeswitch-users] FW: Event pick Message-ID: <592A9CF93E12394E8472A6CC66E66BF23AC83A@Mail-Kilo.squay.com> Hi, I should define a obsession before going to obcause. But my outbound call is already initiated from my dialplan XML []. Am not sure how to give obsession in lua script as my outbound call is initiated from dialplan XML. Can you please guide me. Many thanks From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 20 February 2013 17:12 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Event pick What value is in obCause when the originator cancels the call? Print the value in the else block and find out. -MC On Wed, Feb 20, 2013 at 4:37 AM, Archana Venugopan > wrote: Hi, I need to capture this ORIGINATOR _CANCEL event in my lua script. From google I found the below code. But we establish our calls from dialplan XML as like this, . Since am not that well versed in lua script, I am not sure in the place of obsession below what should I pass. As far I understood the =sofia/external/0${msc_prefix}$1$2 at 10.30.3.27, this is what I should put there but we already have this in our XML. Please suggest me on how should I proceed. Many thanks. -- Initiate an outbound call obSession = freeswitch.Session("sofia/192.168.0.4/1002") -- Check to see if the call was answered if obSession:ready() then -- Do something good here else -- This means the call was not answered ... Check for the reason local obCause = obSession:hangupCause() freeswitch.consoleLog("info", "obSession:hangupCause() = " .. obCause ) if ( obCause == "USER_BUSY" ) then -- SIP 486 -- For BUSY you may reschedule the call for later elseif ( obCause == "NO_ANSWER" ) then -- Call them back in an hour elseif ( obCause == "ORIGINATOR_CANCEL" ) then -- SIP 487 -- May need to check for network congestion or problems else -- Log these issues end end 2013-02-19 17:56:02.277173 [NOTICE] sofia.c:7082 Hangup sofia/internal/206 at fsfailover.uk01.com [CS_EXECUTE] [ORIGINATOR_CANCEL] Regards, Archana _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130220/ae02622e/attachment.html From msc at freeswitch.org Wed Feb 20 20:36:18 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 20 Feb 2013 09:36:18 -0800 Subject: [Freeswitch-users] originate to loopback/answer and park a lua script from another lua script In-Reply-To: References: Message-ID: I think your whole dialplan needs to be in the call to freeswitch.session like you have in the line that's commented out. It looks to me from the log that FS is looking for a destination number of "set:job=1234" because there's no mention of inline when you create the session. -MC On Wed, Feb 20, 2013 at 4:03 AM, Karsten Horsmann wrote: > Hello Michael, > > i create this loop.lua to update the sip username spelling (register > is not casesensitiv vs. fs is casesensitiv). > > Sure i could start the loop.lua with mod_lua as start script. > > With the loopback hack i want to simulate an "job control" and maybe > do some nagios check to be sure > that the loop.lua is alive (simple grep output of "show channels"). > > Is there a other way to check that the loop.lua is alive from outside? > > I play around with an start.lua to originate the loop.lua at startup. > But my start.lua didnt work. > Also an freeswitch.api call from start.lua didnt do the trick. Any hints? > > my "start.lua": > -- new_session = > freeswitch.Session("loopback/set:job=1234\,answer\,park/default/inline > lua:loop.lua inline", session) > new_session = freeswitch.Session("loopback/set:job=1234", session) > new_session:answer(); > new_session:execute("park/default/inline lua:loop.lua inline") > > Got this FS-Log output http://pastebin.freeswitch.org/20595 > > > Kind Regards > Karsten > > 2013/2/15 Michael Collins : > > Can you share with us a small Lua code sample demonstrating what you are > > trying to do? That might make it easier for us to help. > > > > -MC > > > > On Fri, Feb 15, 2013 at 3:26 AM, Karsten Horsmann > > wrote: > >> > >> Hello! > >> > >> as described in Anthonys mail > >> > >> > http://lists.freeswitch.org/pipermail/freeswitch-users/2013-January/091769.html > >> i want also to originate my loop.lua to an loopback channel. > >> > >> No problem from fs_cli. I want to originate the loop.lua from my > >> mod_lua start lua script. > >> > >> How can i do this? The freeswitch.Session() didnt work as aspected. > >> > >> > >> -- > >> Kind Regards > >> *Karsten Horsmann* > > > > > > -- > > Michael S Collins > > Twitter: @mercutioviz > > http://www.FreeSWITCH.org > > http://www.ClueCon.com > > http://www.OSTAG.org > > > > -- > Mit freundlichen Gr??en > *Karsten Horsmann* > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130220/354f7263/attachment-0001.html From msc at freeswitch.org Wed Feb 20 20:48:33 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 20 Feb 2013 09:48:33 -0800 Subject: [Freeswitch-users] Adtran Total Access 916e 4242916L1 In-Reply-To: References: Message-ID: Do you have d-channel traces of working vs. problematic calls? I'd love to see those. -MC On Wed, Feb 20, 2013 at 7:45 AM, Royce Mitchell III wrote: > when placing an outbound call through the 908e, via PRI hosted by > TWTelecom and dialing a destination on the AT&T wireless network, a > "different" series of ISDN codes comes back than from most other > destinations. This series of ISDN codes causes the Adtran to send a > 183+early media followed by a 180 w/o media, which causes the caller > to hear nothing during most of the ringing process. All work-arounds I > have tried to employ so far ( like ignore early media ) have had > disastrous side-effects, we've almost lost clients over them. I put > out a request to pay for a FreeSWITCH expert to help fix the issue, > but I've been too busy with other projects to follow up. > > On Wed, Feb 20, 2013 at 9:27 AM, Ken Rice wrote: > > What kind of incompatibility? As this does not make a lot of sense for a > > TDM/SIP device to have problems with ATT Wireless a Cellular network, > they > > cant talk directly to each other > > > > On 2/20/13 9:20 AM, "Royce Mitchell III" wrote: > > > >> I have a few 908e's, and I'm experiencing a serious incompatibility > >> between FreeSWITCH, the 908e, and the AT&T wireless network. I'm > >> confident I will get it fixed eventually, but just sharing what I know > >> about the device. > >> > >> On Wed, Feb 20, 2013 at 9:12 AM, Ken Rice wrote: > >>> The Adtran TA9XX series are pretty good units. Check out their website > at > >>> adtran.com to find tech manuals for these things... > >>> > >>> I cant answer all the questions below, but these are very stable units > >>> designed to be sold to carriers (and used by many carriers) > >>> > >>> How many concurrent connections depends on the specific model you are > >>> getting. > >>> > >>> > >>> On 2/20/13 9:04 AM, "Mimiko" wrote: > >>> > >>>> Hi. > >>>> > >>>> Who knows about this device: Adtran Total Access 916e 4242916L1 > >>>> http://www.newegg.com/Product/Product.aspx?Item=N82E16833706027 > >>>> > >>>> How stable is it? Does it use LAN ports in failover mode? Does it use > >>>> T1/FT1 WAN ports in failover manner connected to a PBX with two PRI > >>>> ports? How good it is working with FS? How much concurrent connection > >>>> does it support? > >>>> > >>>> I asked to the company via email a week ago, and no response till now. > >>> > >>> -- > >>> Ken > >>> http://www.FreeSWITCH.org > >>> http://www.ClueCon.com > >>> http://www.OSTAG.org > >>> irc.freenode.net #freeswitch > >>> > >>> > >>> > >>> > _________________________________________________________________________ > >>> Professional FreeSWITCH Consulting Services: > >>> consulting at freeswitch.org > >>> http://www.freeswitchsolutions.com > >>> > >>> > >>> > >>> > >>> Official FreeSWITCH Sites > >>> http://www.freeswitch.org > >>> http://wiki.freeswitch.org > >>> http://www.cluecon.com > >>> > >>> 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 > >> > >> > > > > -- > > Ken > > http://www.FreeSWITCH.org > > http://www.ClueCon.com > > http://www.OSTAG.org > > irc.freenode.net #freeswitch > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > 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 > > > > -- > Royce Mitchell > IT Consultant > ITAS Solutions > royce3 at itas-solutions.com > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130220/37b8f1d8/attachment.html From dgarcia at anew.com.ve Wed Feb 20 21:04:37 2013 From: dgarcia at anew.com.ve (Saugort Dario Garcia Tovar) Date: Wed, 20 Feb 2013 13:34:37 -0430 Subject: [Freeswitch-users] Determine if a number is valid/working In-Reply-To: References: <072701ce0ebf$c5808640$508192c0$@bizfocused.com> Message-ID: <51251035.4010307@anew.com.ve> Hi, I think you count with several tools to "clean-up" the calling list. You could use call progress detection to detect if the number is active. Also, you could use answer machine detection to verify if a real person is at the other end. Even you could use ASR and TTS in a script to get some interaction from the called person. However, you could never get sorted all cases, for example, the number could be a good one but belong to another person or it belong to the main number of a company. FS give some tools. I quite sure you can do some tone detection (busy, no answer, sit, reorder, disconnect, etc) , take a look at: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_tone_detect http://wiki.freeswitch.org/wiki/Early_Media http://wiki.freeswitch.org/wiki/Mod_spandsp About answer machine detection, I not sure how good is FS. Take a look to: http://wiki.freeswitch.org/wiki/Mod_avmd http://wiki.freeswitch.org/wiki/Mod_vmd You could combine a third party tool to make the work, check this link http://www.sangoma.com/products/call_center_software/netborder_call_analyzer.html I hope this helps you On 2/19/2013 3:07 PM, Michael Collins wrote: > Well, pretty much the only way to guarantee that a number is working > is to call it and see what happens. I suspect Ken Rice has some > experience in this area and I'd definitely like to hear it. If anyone > else has thoughts I'd like to hear them as well. > > -MC > > P.S. - Why does the customer think that calling the number and playing > a short message will take too long? Also, what is the customer's > primary concern about the web leads having non-working numbers? Is it > because they have humans manually dialing them and it's a waste of > time when the person comes across a bad number? > > On Tue, Feb 19, 2013 at 8:40 AM, Sean Devoy > wrote: > > HI All, > > My web site customer?s call center claims that 1/3 of their web > leads have nonworking phone numbers. Short of calling it, is > there any way with FS (or even a paid service) to determine if a > given phone number is a working number? I would like to make an > Ajax SOAP call to a PHP page that checks (or dials) the number and > returns a status for the validity of the phone number. > > I have an LUA script that will call, play a short message and hang > up, but the customer thinks that may take too long. > > Any ideas? > > Thanks, > > Sean > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.2238 / Virus Database: 2639/5616 - Release Date: 02/19/13 > -- Atentamente, *Dario Garc?a* Consultor. CCCT, Nivel C2, Sector Yarey, Mz, Ofc. MZ03a. Caracas-Venezuela. Tel?fono: +58 212 9081842 Cel: +58 412 2221515 dgarcia at anew.com.ve http://www.anew.com.ve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130220/c70d770d/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 70 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130220/c70d770d/attachment-0001.gif From dgarcia at anew.com.ve Wed Feb 20 21:10:29 2013 From: dgarcia at anew.com.ve (Saugort Dario Garcia Tovar) Date: Wed, 20 Feb 2013 13:40:29 -0430 Subject: [Freeswitch-users] Determine if a number is valid/working In-Reply-To: References: <072701ce0ebf$c5808640$508192c0$@bizfocused.com> Message-ID: <51251195.2000903@anew.com.ve> Hi, I think you count with several tools to "clean-up" the calling list. You could use call progress detection to detect if the number is active. Also, you could use answer machine detection to verify if a real person is at the other end. Even you could use ASR and TTS in a script to get some interaction from the called person. However, you could never get sorted all cases, for example, the number could be a good one but belong to another person or it belong to the main number of a company. FS give some tools. I quite sure you can do some tone detection (busy, no answer, sit, reorder, disconnect, etc) , take a look at: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_tone_detect http://wiki.freeswitch.org/wiki/Early_Media http://wiki.freeswitch.org/wiki/Mod_spandsp About answer machine detection, I not sure how good is FS. Take a look to: http://wiki.freeswitch.org/wiki/Mod_avmd http://wiki.freeswitch.org/wiki/Mod_vmd You could combine a third party tool to make the work, check this link http://www.sangoma.com/products/call_center_software/netborder_call_analyzer.html I hope this helps you On 2/19/2013 3:07 PM, Michael Collins wrote: > Well, pretty much the only way to guarantee that a number is working > is to call it and see what happens. I suspect Ken Rice has some > experience in this area and I'd definitely like to hear it. If anyone > else has thoughts I'd like to hear them as well. > > -MC > > P.S. - Why does the customer think that calling the number and playing > a short message will take too long? Also, what is the customer's > primary concern about the web leads having non-working numbers? Is it > because they have humans manually dialing them and it's a waste of > time when the person comes across a bad number? > > On Tue, Feb 19, 2013 at 8:40 AM, Sean Devoy > wrote: > > HI All, > > My web site customer?s call center claims that 1/3 of their web > leads have nonworking phone numbers. Short of calling it, is > there any way with FS (or even a paid service) to determine if a > given phone number is a working number? I would like to make an > Ajax SOAP call to a PHP page that checks (or dials) the number and > returns a status for the validity of the phone number. > > I have an LUA script that will call, play a short message and hang > up, but the customer thinks that may take too long. > > Any ideas? > > Thanks, > > Sean > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.2238 / Virus Database: 2639/5616 - Release Date: 02/19/13 > -- Atentamente, *Dario Garc?a* Consultor. CCCT, Nivel C2, Sector Yarey, Mz, Ofc. MZ03a. Caracas-Venezuela. Tel?fono: +58 212 9081842 Cel: +58 412 2221515 dgarcia at anew.com.ve http://www.anew.com.ve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130220/79ba0670/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 70 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130220/79ba0670/attachment.gif From msc at freeswitch.org Wed Feb 20 21:07:26 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 20 Feb 2013 10:07:26 -0800 Subject: [Freeswitch-users] FW: Event pick In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF23AC83A@Mail-Kilo.squay.com> References: <592A9CF93E12394E8472A6CC66E66BF23AC83A@Mail-Kilo.squay.com> Message-ID: If you bridge in the dialplan then your Lua script won't run until the bridge app has terminated. -MC On Wed, Feb 20, 2013 at 9:33 AM, Archana Venugopan wrote: > Hi,**** > > ** ** > > I should define a obsession before going to obcause. But my outbound call > is already initiated from my dialplan XML [ data="sofia/external/0${msc_prefix}$1$2 at 10.30.3.27"/>].**** > > Am not sure how to give obsession in lua script as my outbound call is > initiated from dialplan XML.**** > > Can you please guide me. Many thanks**** > > ** ** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael > Collins > *Sent:* 20 February 2013 17:12 > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Event pick**** > > ** ** > > What value is in obCause when the originator cancels the call? Print the > value in the else block and find out. > -MC**** > > On Wed, Feb 20, 2013 at 4:37 AM, Archana Venugopan > wrote:**** > > Hi,**** > > I need to capture this ORIGINATOR _CANCEL event in my lua script. >From > google I found the below code. But we establish our calls from dialplan XML > as like this, data="sofia/external/0${msc_prefix}$1$2 at 10.30.3.27"/>.**** > > Since am not that well versed in lua script, I am not sure in the place of > obsession below what should I pass. As far I understood the = > sofia/external/0${msc_prefix}$1$2 at 10.30.3.27, this is what I should put > there but we already have this in our XML.**** > > Please suggest me on how should I proceed. Many thanks.**** > > **** > > -- Initiate an outbound call**** > > **** > > obSession = freeswitch.Session("sofia/192.168.0.4/1002")**** > > **** > > -- Check to see if the call was answered**** > > **** > > if obSession:ready() then**** > > -- Do something good here**** > > **** > > else -- This means the call was not answered ... Check for the reason** > ** > > **** > > local obCause = obSession:hangupCause()**** > > **** > > freeswitch.consoleLog("info", "obSession:hangupCause() = " .. obCause ) > **** > > **** > > if ( obCause == "USER_BUSY" ) then -- SIP 486**** > > -- For BUSY you may reschedule the call for later**** > > elseif ( obCause == "NO_ANSWER" ) then**** > > -- Call them back in an hour**** > > elseif ( obCause == "ORIGINATOR_CANCEL" ) then -- SIP 487**** > > -- May need to check for network congestion or problems**** > > else**** > > -- Log these issues**** > > end**** > > end**** > > **** > > **** > > **** > > **** > > 2013-02-19 17:56:02.277173 [NOTICE] sofia.c:7082 Hangup sofia/internal/ > 206 at fsfailover.uk01.com [CS_EXECUTE] [ORIGINATOR_CANCEL]**** > > **** > > Regards,**** > > Archana**** > > **** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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**** > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130220/2682b4ce/attachment-0001.html From th982a at googlemail.com Wed Feb 20 21:11:13 2013 From: th982a at googlemail.com (Tamer Higazi) Date: Wed, 20 Feb 2013 19:11:13 +0100 Subject: [Freeswitch-users] portaudio on a system with pulseaudio In-Reply-To: References: <5123E669.8060701@googlemail.com> Message-ID: <512511C1.9030100@googlemail.com> Hi Vik! portaudio is for me still a riddle..... don't get it worked proparly at any way.... mod_alsa also as well makes me a lot of headache! I can speak through the mic, and the opponent party can listen me very well, but I am not able to hear anything on my side. totally strange..... Tamer Am 19.02.2013 22:18, schrieb Vik Killa: > I tried sending audio to/from pulseaudio to portaudio... I had no > success on several distros of linux (centos, debian and ubuntu). In my > opinion, mod_portaudio needs to be updated or a mod_pulseaudio needs > to be developed. I've never been able to get mod_alsa to work > properly. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 bpriddy at bryantschools.org Wed Feb 20 21:51:26 2013 From: bpriddy at bryantschools.org (Blake Priddy) Date: Wed, 20 Feb 2013 12:51:26 -0600 Subject: [Freeswitch-users] Adtran Total Access 916e 4242916L1 In-Reply-To: References: Message-ID: The 908e looks pretty :) I am going to be installing mine for the district tonight. On Wed, Feb 20, 2013 at 11:48 AM, Michael Collins wrote: > Do you have d-channel traces of working vs. problematic calls? I'd love to > see those. > -MC > > > On Wed, Feb 20, 2013 at 7:45 AM, Royce Mitchell III wrote: > >> when placing an outbound call through the 908e, via PRI hosted by >> TWTelecom and dialing a destination on the AT&T wireless network, a >> "different" series of ISDN codes comes back than from most other >> destinations. This series of ISDN codes causes the Adtran to send a >> 183+early media followed by a 180 w/o media, which causes the caller >> to hear nothing during most of the ringing process. All work-arounds I >> have tried to employ so far ( like ignore early media ) have had >> disastrous side-effects, we've almost lost clients over them. I put >> out a request to pay for a FreeSWITCH expert to help fix the issue, >> but I've been too busy with other projects to follow up. >> >> On Wed, Feb 20, 2013 at 9:27 AM, Ken Rice wrote: >> > What kind of incompatibility? As this does not make a lot of sense for a >> > TDM/SIP device to have problems with ATT Wireless a Cellular network, >> they >> > cant talk directly to each other >> > >> > On 2/20/13 9:20 AM, "Royce Mitchell III" wrote: >> > >> >> I have a few 908e's, and I'm experiencing a serious incompatibility >> >> between FreeSWITCH, the 908e, and the AT&T wireless network. I'm >> >> confident I will get it fixed eventually, but just sharing what I know >> >> about the device. >> >> >> >> On Wed, Feb 20, 2013 at 9:12 AM, Ken Rice >> wrote: >> >>> The Adtran TA9XX series are pretty good units. Check out their >> website at >> >>> adtran.com to find tech manuals for these things... >> >>> >> >>> I cant answer all the questions below, but these are very stable units >> >>> designed to be sold to carriers (and used by many carriers) >> >>> >> >>> How many concurrent connections depends on the specific model you are >> >>> getting. >> >>> >> >>> >> >>> On 2/20/13 9:04 AM, "Mimiko" wrote: >> >>> >> >>>> Hi. >> >>>> >> >>>> Who knows about this device: Adtran Total Access 916e 4242916L1 >> >>>> http://www.newegg.com/Product/Product.aspx?Item=N82E16833706027 >> >>>> >> >>>> How stable is it? Does it use LAN ports in failover mode? Does it use >> >>>> T1/FT1 WAN ports in failover manner connected to a PBX with two PRI >> >>>> ports? How good it is working with FS? How much concurrent connection >> >>>> does it support? >> >>>> >> >>>> I asked to the company via email a week ago, and no response till >> now. >> >>> >> >>> -- >> >>> Ken >> >>> http://www.FreeSWITCH.org >> >>> http://www.ClueCon.com >> >>> http://www.OSTAG.org >> >>> irc.freenode.net #freeswitch >> >>> >> >>> >> >>> >> >>> >> _________________________________________________________________________ >> >>> Professional FreeSWITCH Consulting Services: >> >>> consulting at freeswitch.org >> >>> http://www.freeswitchsolutions.com >> >>> >> >>> >> >>> >> >>> >> >>> Official FreeSWITCH Sites >> >>> http://www.freeswitch.org >> >>> http://wiki.freeswitch.org >> >>> http://www.cluecon.com >> >>> >> >>> 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 >> >> >> >> >> > >> > -- >> > Ken >> > http://www.FreeSWITCH.org >> > http://www.ClueCon.com >> > http://www.OSTAG.org >> > irc.freenode.net #freeswitch >> > >> > >> > >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > 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 >> >> >> >> -- >> Royce Mitchell >> IT Consultant >> ITAS Solutions >> royce3 at itas-solutions.com >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> > > > > -- > Michael S Collins > Twitter: @mercutioviz > > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- *Blakelund Priddy* Network Systems Engineer Bryant Public School District Bryant, Arkansas 72022 http://www.bryantschools.org p 501-653-5038 f 501-847-5656 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130220/cf1751e4/attachment.html From royce3 at gmail.com Wed Feb 20 22:07:51 2013 From: royce3 at gmail.com (Royce Mitchell III) Date: Wed, 20 Feb 2013 13:07:51 -0600 Subject: [Freeswitch-users] Adtran Total Access 916e 4242916L1 In-Reply-To: References: Message-ID: I'll recreate them and send them On Wed, Feb 20, 2013 at 11:48 AM, Michael Collins wrote: > Do you have d-channel traces of working vs. problematic calls? I'd love to > see those. > -MC > > > On Wed, Feb 20, 2013 at 7:45 AM, Royce Mitchell III > wrote: >> >> when placing an outbound call through the 908e, via PRI hosted by >> TWTelecom and dialing a destination on the AT&T wireless network, a >> "different" series of ISDN codes comes back than from most other >> destinations. This series of ISDN codes causes the Adtran to send a >> 183+early media followed by a 180 w/o media, which causes the caller >> to hear nothing during most of the ringing process. All work-arounds I >> have tried to employ so far ( like ignore early media ) have had >> disastrous side-effects, we've almost lost clients over them. I put >> out a request to pay for a FreeSWITCH expert to help fix the issue, >> but I've been too busy with other projects to follow up. >> >> On Wed, Feb 20, 2013 at 9:27 AM, Ken Rice wrote: >> > What kind of incompatibility? As this does not make a lot of sense for a >> > TDM/SIP device to have problems with ATT Wireless a Cellular network, >> > they >> > cant talk directly to each other >> > >> > On 2/20/13 9:20 AM, "Royce Mitchell III" wrote: >> > >> >> I have a few 908e's, and I'm experiencing a serious incompatibility >> >> between FreeSWITCH, the 908e, and the AT&T wireless network. I'm >> >> confident I will get it fixed eventually, but just sharing what I know >> >> about the device. >> >> >> >> On Wed, Feb 20, 2013 at 9:12 AM, Ken Rice wrote: >> >>> The Adtran TA9XX series are pretty good units. Check out their website >> >>> at >> >>> adtran.com to find tech manuals for these things... >> >>> >> >>> I cant answer all the questions below, but these are very stable units >> >>> designed to be sold to carriers (and used by many carriers) >> >>> >> >>> How many concurrent connections depends on the specific model you are >> >>> getting. >> >>> >> >>> >> >>> On 2/20/13 9:04 AM, "Mimiko" wrote: >> >>> >> >>>> Hi. >> >>>> >> >>>> Who knows about this device: Adtran Total Access 916e 4242916L1 >> >>>> http://www.newegg.com/Product/Product.aspx?Item=N82E16833706027 >> >>>> >> >>>> How stable is it? Does it use LAN ports in failover mode? Does it use >> >>>> T1/FT1 WAN ports in failover manner connected to a PBX with two PRI >> >>>> ports? How good it is working with FS? How much concurrent connection >> >>>> does it support? >> >>>> >> >>>> I asked to the company via email a week ago, and no response till >> >>>> now. >> >>> >> >>> -- >> >>> Ken >> >>> http://www.FreeSWITCH.org >> >>> http://www.ClueCon.com >> >>> http://www.OSTAG.org >> >>> irc.freenode.net #freeswitch >> >>> >> >>> >> >>> >> >>> >> >>> _________________________________________________________________________ >> >>> Professional FreeSWITCH Consulting Services: >> >>> consulting at freeswitch.org >> >>> http://www.freeswitchsolutions.com >> >>> >> >>> >> >>> >> >>> >> >>> Official FreeSWITCH Sites >> >>> http://www.freeswitch.org >> >>> http://wiki.freeswitch.org >> >>> http://www.cluecon.com >> >>> >> >>> 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 >> >> >> >> >> > >> > -- >> > Ken >> > http://www.FreeSWITCH.org >> > http://www.ClueCon.com >> > http://www.OSTAG.org >> > irc.freenode.net #freeswitch >> > >> > >> > >> > >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > 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 >> >> >> >> -- >> Royce Mitchell >> IT Consultant >> ITAS Solutions >> royce3 at itas-solutions.com >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 > > > > > -- > Michael S Collins > Twitter: @mercutioviz > > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -- Royce Mitchell IT Consultant ITAS Solutions royce3 at itas-solutions.com From emamirazavi at gmail.com Wed Feb 20 22:07:31 2013 From: emamirazavi at gmail.com (Sayyed Mohammad Emami Razavi) Date: Wed, 20 Feb 2013 22:37:31 +0330 Subject: [Freeswitch-users] Config several FS for routing and reading unique configs Message-ID: How to config several FS systems for 4 FS servers after one LVS(load balancer) so that all agents on all systems defined correctly and route without any problem?! You know that i should have one DB [for e.g. blue.box] and one CONF DIR for my FS XML config files! What is your idea? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130220/75a32041/attachment.html From emamirazavi at gmail.com Wed Feb 20 22:10:53 2013 From: emamirazavi at gmail.com (Sayyed Mohammad Emami Razavi) Date: Wed, 20 Feb 2013 22:40:53 +0330 Subject: [Freeswitch-users] DSN for callcenter, what does it do? Message-ID: DSN and SQLite and update mode helps in which needs in call center module? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130220/05f81b92/attachment.html From bpriddy at bryantschools.org Wed Feb 20 22:19:12 2013 From: bpriddy at bryantschools.org (Blake Priddy) Date: Wed, 20 Feb 2013 13:19:12 -0600 Subject: [Freeswitch-users] freeswitch hack In-Reply-To: References: <958ABE902C1F47B09FFAAE7EDEE074B1@imsdeveloper> Message-ID: I have also had the situation that they are calls getting rejected. On Wed, Feb 20, 2013 at 11:08 AM, Michael Collins wrote: > > > On Wed, Feb 20, 2013 at 1:53 AM, Mario Karakanovski wrote: > >> Thanks Ken,**** >> >> ** ** >> >> It is helpful, but I still think there is some security issue. I?ve >> double check configuration. I?ve try to reproduce the issue trying to do >> direct call (TCP and UDP) or authenticate with invalid user, but everything >> works as expected ? calls/authentication was rejected. I?ve decide to log >> the traffic ? maybe I will be able to see where is the problem. >> > > What "security issue"? You said that they cannot make calls with the > passwords that they've guessed, correct? About the only thing left to do is > set up fail2ban and just shut > the door on them when they fail too many times. > > -Michael > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- *Blakelund Priddy* Network Systems Engineer Bryant Public School District Bryant, Arkansas 72022 http://www.bryantschools.org p 501-653-5038 f 501-847-5656 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130220/f8384457/attachment.html From victor.chukalovskiy at gmail.com Wed Feb 20 22:26:46 2013 From: victor.chukalovskiy at gmail.com (Victor Chukalovskiy) Date: Wed, 20 Feb 2013 14:26:46 -0500 Subject: [Freeswitch-users] NPDI tel URI parameters appearing in SIP URI In-Reply-To: References: <5123FBD2.7070602@gmail.com> <51245472.90105@gmail.com> Message-ID: <51252376.8090601@gmail.com> Did a bit more testing today. gives what I need as per WiKi: INVITE sip:6135555555;npdi=yes at 5.6.7.8:5060 SIP/2.0 But here is a caveat: this is not gateway! I do need to use gateway in the bridge app (because of OPTIONS PING that determines if gateway is up or down). So, I try: This is no good, ;npdi=yes moved in to the domain part: INVITE sip:6135555555 at 5.6.7.8:5060*;npdi=yes* SIP/2.0 Also, tried this: But this is no good either. Looks like sip: syntax does not work with gateways: INVITE *sip:sip:*6135555555 at 5.6.7.8:5060*;npdi=yes* SIP/2.0 ---------------- So, my immediate question is, is there a way to use sip: syntax when bridging to a gateway? ---------------- And on a broader scale, something along these lines: http://freeswitch-users.2379917.n2.nabble.com/generating-RFC-3966-and-RFC-4694-calls-td2544806.html Was there much work done since then on RFC3966 and RFC4694 support in FS? Ideally, that would be something that's parsed as channel variable on incoming call and can be manipulated as a channel variable on the outgoing call. FS would take care of parsing tel URI on incoming and putting it together on the outgoing. -Victor On 02/20/2013 03:32 AM, Avi Marcus wrote: > Ah sorry, I should have read your question more carefully! > -Avi > > > On Wed, Feb 20, 2013 at 6:43 AM, Victor Chukalovskiy > > > wrote: > > Avi, > > I know this part of the WiKi....thank you but I don't think it > helps here. > > -Victor > > > On 02/19/2013 07:25 PM, Avi Marcus wrote: >> You'll have to use a regex and capture it manually. >> See: >> http://wiki.freeswitch.org/wiki/Regular_Expression#LNP_.28Local_Number_Portability.29 >> >> -Avi >> >> On Wed, Feb 20, 2013 at 12:25 AM, Victor Chukalovskiy >> > > wrote: >> >> Hello, >> >> SIP call comes on leg A as following: >> INVITE sip:6135555555 >> *;npdi=yes*@1.2.3.4:5060 >> SIP/2.0 >> >> ${destination_number} var becomes "6135555555 >> ;npdi=yes" >> >> On leg B I send the call to gateway using unaltered >> ${destination_number}: >> > data="sofia/gateway/test_gw/${destination_number}"/> >> >> I see the following on leg B: >> INVITE sip:6135555555 at 5.6.7.8:5060 >> *;npdi=yes* SIP/2.0 >> >> So, FS moved ";npdi=yes" which is a tel URI parameter into >> the very end as if it was a SIP URI parameter. Such behavior >> does not seem correct. Is this a bug? Found the following >> implementation discussion: >> http://comments.gmane.org/gmane.ietf.sip-implementors/21554 >> >> Please help to make FS send tel URI parameters unaltered >> using bridge application and a gateway. WiKi for bridge >> dial-plan tool has some examples but it does not make use of >> gateway and I need to use gateway. >> >> Thank you, >> Victor >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130220/fb5f7688/attachment-0001.html From dvl36.ripe.nick at gmail.com Wed Feb 20 22:57:26 2013 From: dvl36.ripe.nick at gmail.com (Dmitry Lysenko) Date: Wed, 20 Feb 2013 21:57:26 +0200 Subject: [Freeswitch-users] Changing codec during calls In-Reply-To: References: <6EF8B8E9-4CC5-480C-B1C1-09CBC533470C@kavun.ch> <35AF9E79-771D-440A-BEBB-A856D831161C@kavun.ch> Message-ID: I do not have account on pastebin.freeswitch.org See: http://jira.freeswitch.org/browse/FS-5116 Thanks. 2013/2/20 Michael Collins > Definitely get the console log and SIP trace of those problem calls and > put them in pastein.freeswitch.org. > > -MC > > > On Wed, Feb 20, 2013 at 5:04 AM, Dmitry Lysenko > wrote: > >> Wow! This great feature is really work. I have the same issue - one way >> audio. And another: "show channels" always return first negotiated codec. >> When switching codec back, audio working fine. Tested with 3 different >> softphones. >> If someone interested, please confirm. >> Thanks. >> >> >> 2013/2/19 Michael Collins >> >>> pastebin the log. I'll bet Steven can diagnose it! ;) >>> -MC >>> >>> >>> On Tue, Feb 19, 2013 at 8:37 AM, Emrah wrote: >>> >>>> Excellent, that worked. >>>> >>>> However I am having a one way audio issue, where the peer that has its >>>> codecs changed can hear but cannot be heard. >>>> Even if both bridged peers are changed to a common codec it doesn't >>>> work. >>>> >>>> Any idea? >>>> >>>> Thanks a bunch for your help. >>>> On Feb 19, 2013, at 3:47 AM, Steven Ayre wrote: >>>> >>>> > Try: >>>> > uuid_media_reneg 49f428e5-1d72-42d8-b0e1-5b3f29f9a4ca =PCMU >>>> > >>>> > (ie add = before PCMU) >>>> > >>>> > It appears to affect something in the code... >>>> > >>>> > -Steve >>>> > >>>> > >>>> > On 19 February 2013 05:54, Emrah wrote: >>>> >> Hey, >>>> >> >>>> >> In short, nothing seems to happen. I tried several phones. >>>> >> I also tried enabling codec renegotiation on hold/unhold thinking >>>> that the change in SDP with uuid_media_reneg would do it. >>>> >> >>>> >> This is what happens: >>>> >> freeswitch at internal> show channels >>>> >> >>>> uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,sent_callee_name,sent_callee_num >>>> >> 49f428e5-1d72-42d8-b0e1-5b3f29f9a4ca,inbound,2013-02-19 >>>> 00:44:35,1361252675,sofia/internal/10000 at sip.domain.net,CS_EXECUTE,Emrah >>>> - Macbook >>>> Pro,10000,100.100.100.100,conf-3111,conference,3111-sip.domain.net at ek-custom-16k >>>> ,XML,ek-conference,G722,16000,64000,G722,16000,64000,, >>>> cornflakes.domain.net,10000 at sip.domain.net,,ACTIVE,,,,,, >>>> >> >>>> >> 1 total. >>>> >> >>>> >> freeswitch at internal> uuid_media_reneg >>>> 49f428e5-1d72-42d8-b0e1-5b3f29f9a4ca PCMU >>>> >> +OK Success >>>> >> >>>> >> 2013-02-19 00:45:25.172659 [DEBUG] sofia_glue.c:2647 Local SDP: >>>> >> v=0 >>>> >> o=FreeSWITCH 1361234787 1361234790 IN IP4 1.2.3.4 >>>> >> s=FreeSWITCH >>>> >> c=IN IP4 1.2.3.4 >>>> >> t=0 0 >>>> >> m=audio 17888 RTP/AVP 9 0 101 >>>> >> a=rtpmap:101 telephone-event/8000 >>>> >> a=fmtp:101 0-16 >>>> >> a=silenceSupp:off - - - - >>>> >> a=ptime:20 >>>> >> a=sendrecv >>>> >> >>>> >> 2013-02-19 00:45:25.172659 [DEBUG] switch_core_session.c:975 Send >>>> signal sofia/internal/10000 at sip.domain.net [BREAK] >>>> >> 2013-02-19 00:45:25.182652 [DEBUG] sofia.c:5574 Channel >>>> sofia/internal/10000 at sip.domain.net entering state [calling][0] >>>> >> 2013-02-19 00:45:25.292661 [DEBUG] switch_core_session.c:975 Send >>>> signal sofia/internal/10000 at sip.domain.net [BREAK] >>>> >> 2013-02-19 00:45:25.292661 [DEBUG] switch_core_session.c:975 Send >>>> signal sofia/internal/10000 at sip.domain.net [BREAK] >>>> >> 2013-02-19 00:45:25.302683 [INFO] sofia.c:931 sofia/internal/ >>>> 10000 at sip.domain.net Update Callee ID to "10000" <10000> >>>> >> 2013-02-19 00:45:25.302683 [DEBUG] sofia.c:5574 Channel >>>> sofia/internal/10000 at sip.domain.net entering state [completing][200] >>>> >> 2013-02-19 00:45:25.302683 [DEBUG] sofia.c:5585 Remote SDP: >>>> >> v=0 >>>> >> o=- 3570241475 3570241476 IN IP4 10.0.0.131 >>>> >> s=pjmedia >>>> >> c=IN IP4 10.0.0.131 >>>> >> t=0 0 >>>> >> a=X-nat:0 >>>> >> m=audio 4010 RTP/AVP 9 101 >>>> >> a=rtcp:4011 IN IP4 10.0.0.131 >>>> >> a=rtpmap:9 G722/8000 >>>> >> a=sendrecv >>>> >> a=rtpmap:101 telephone-event/8000 >>>> >> a=fmtp:101 0-15 >>>> >> >>>> >> 2013-02-19 00:45:25.302683 [DEBUG] switch_core_session.c:975 Send >>>> signal sofia/internal/10000 at sip.domain.net [BREAK] >>>> >> 2013-02-19 00:45:25.302683 [DEBUG] switch_core_session.c:975 Send >>>> signal sofia/internal/10000 at sip.domain.net [BREAK] >>>> >> 2013-02-19 00:45:25.322657 [DEBUG] sofia.c:5574 Channel >>>> sofia/internal/10000 at sip.domain.net entering state [ready][200] >>>> >> 2013-02-19 00:45:25.322657 [DEBUG] sofia.c:5582 Duplicate SDP >>>> >> v=0 >>>> >> o=- 3570241475 3570241476 IN IP4 10.0.0.131 >>>> >> s=pjmedia >>>> >> c=IN IP4 10.0.0.131 >>>> >> t=0 0 >>>> >> a=X-nat:0 >>>> >> m=audio 4010 RTP/AVP 9 101 >>>> >> a=rtcp:4011 IN IP4 10.0.0.131 >>>> >> a=rtpmap:9 G722/8000 >>>> >> a=sendrecv >>>> >> a=rtpmap:101 telephone-event/8000 >>>> >> a=fmtp:101 0-15 >>>> >> >>>> >> 2013-02-19 00:45:25.322657 [DEBUG] sofia_glue.c:5137 Audio Codec >>>> Compare [G722:9:8000:20:64000]/[G722:9:8000:20:64000] >>>> >> 2013-02-19 00:45:25.322657 [DEBUG] sofia_glue.c:3022 Already using >>>> G722 >>>> >> 2013-02-19 00:45:25.322657 [DEBUG] sofia_glue.c:5266 Set 2833 dtmf >>>> send/recv payload to 101 >>>> >> >>>> >> >>>> >> Thanks for any help, >>>> >> Emrah >>>> >> On Feb 18, 2013, at 1:38 PM, Michael Collins >>>> wrote: >>>> >> >>>> >>> What happened with the uuid_media_reneg command? >>>> >>> -MC >>>> >>> >>>> >>> On Sat, Feb 16, 2013 at 3:10 PM, Emrah wrote: >>>> >>> Hi all, >>>> >>> >>>> >>> How do we go about renegotiating the codecs of a call that is >>>> already established? >>>> >>> I tried uuid_media_reneg to no avail. >>>> >>> E.g.: Upgrade someone who's entered a 16k conference from PCMU to >>>> G722. >>>> >>> >>>> >>> Cheers and thanks, >>>> >>> Emrah >>>> >>> >>>> _________________________________________________________________________ >>>> >>> Professional FreeSWITCH Consulting Services: >>>> >>> consulting at freeswitch.org >>>> >>> http://www.freeswitchsolutions.com >>>> >>> >>>> >>> >>>> >>> >>>> >>> >>>> >>> Official FreeSWITCH Sites >>>> >>> http://www.freeswitch.org >>>> >>> http://wiki.freeswitch.org >>>> >>> http://www.cluecon.com >>>> >>> >>>> >>> 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 >>>> >>> >>>> >>> >>>> >>> >>>> >>> -- >>>> >>> Michael S Collins >>>> >>> Twitter: @mercutioviz >>>> >>> http://www.FreeSWITCH.org >>>> >>> http://www.ClueCon.com >>>> >>> http://www.OSTAG.org >>>> >>> >>>> >>> >>>> _________________________________________________________________________ >>>> >>> Professional FreeSWITCH Consulting Services: >>>> >>> consulting at freeswitch.org >>>> >>> http://www.freeswitchsolutions.com >>>> >>> >>>> >>> >>>> >>> >>>> >>> >>>> >>> Official FreeSWITCH Sites >>>> >>> http://www.freeswitch.org >>>> >>> http://wiki.freeswitch.org >>>> >>> http://www.cluecon.com >>>> >>> >>>> >>> 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 >>>> >> >>>> >> >>>> >> >>>> _________________________________________________________________________ >>>> >> Professional FreeSWITCH Consulting Services: >>>> >> consulting at freeswitch.org >>>> >> http://www.freeswitchsolutions.com >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> Official FreeSWITCH Sites >>>> >> http://www.freeswitch.org >>>> >> http://wiki.freeswitch.org >>>> >> http://www.cluecon.com >>>> >> >>>> >> 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 >>>> > >>>> > >>>> _________________________________________________________________________ >>>> > Professional FreeSWITCH Consulting Services: >>>> > consulting at freeswitch.org >>>> > http://www.freeswitchsolutions.com >>>> > >>>> > >>>> > >>>> > >>>> > Official FreeSWITCH Sites >>>> > http://www.freeswitch.org >>>> > http://wiki.freeswitch.org >>>> > http://www.cluecon.com >>>> > >>>> > 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 >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>>> >>> >>> >>> >>> -- >>> Michael S Collins >>> Twitter: @mercutioviz >>> http://www.FreeSWITCH.org >>> http://www.ClueCon.com >>> http://www.OSTAG.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130220/d6aa67b9/attachment-0001.html From max at nysolutions.com Wed Feb 20 22:30:48 2013 From: max at nysolutions.com (Moishe Grunstein) Date: Wed, 20 Feb 2013 19:30:48 +0000 Subject: [Freeswitch-users] Cudatel Message-ID: Is these an updated cudatel demo link? https://join.me/411-332-153 not working. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130220/9c3a8812/attachment.html From msc at freeswitch.org Wed Feb 20 23:40:30 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 20 Feb 2013 12:40:30 -0800 Subject: [Freeswitch-users] freeswitch hack In-Reply-To: References: <958ABE902C1F47B09FFAAE7EDEE074B1@imsdeveloper> Message-ID: Aren't they supposed to be rejected? On Wed, Feb 20, 2013 at 11:19 AM, Blake Priddy wrote: > I have also had the situation that they are calls getting rejected. > > > On Wed, Feb 20, 2013 at 11:08 AM, Michael Collins wrote: > >> >> >> On Wed, Feb 20, 2013 at 1:53 AM, Mario Karakanovski wrote: >> >>> Thanks Ken,**** >>> >>> ** ** >>> >>> It is helpful, but I still think there is some security issue. I?ve >>> double check configuration. I?ve try to reproduce the issue trying to do >>> direct call (TCP and UDP) or authenticate with invalid user, but everything >>> works as expected ? calls/authentication was rejected. I?ve decide to log >>> the traffic ? maybe I will be able to see where is the problem. >>> >> >> What "security issue"? You said that they cannot make calls with the >> passwords that they've guessed, correct? About the only thing left to do is >> set up fail2ban and just shut >> the door on them when they fail too many times. >> >> -Michael >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > > -- > > *Blakelund Priddy* > Network Systems Engineer > Bryant Public School District > Bryant, Arkansas 72022 > http://www.bryantschools.org > p 501-653-5038 > f 501-847-5656 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130220/08f0d396/attachment.html From msc at freeswitch.org Wed Feb 20 23:42:07 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 20 Feb 2013 12:42:07 -0800 Subject: [Freeswitch-users] Changing codec during calls In-Reply-To: References: <6EF8B8E9-4CC5-480C-B1C1-09CBC533470C@kavun.ch> <35AF9E79-771D-440A-BEBB-A856D831161C@kavun.ch> Message-ID: Haha, you don't need an account. Re-read the challenge dialog on pastebin.freeswitch.org. -MC On Wed, Feb 20, 2013 at 11:57 AM, Dmitry Lysenko wrote: > I do not have account on pastebin.freeswitch.org > See: http://jira.freeswitch.org/browse/FS-5116 > Thanks. > > > 2013/2/20 Michael Collins > >> Definitely get the console log and SIP trace of those problem calls and >> put them in pastein.freeswitch.org. >> >> -MC >> >> >> On Wed, Feb 20, 2013 at 5:04 AM, Dmitry Lysenko < >> dvl36.ripe.nick at gmail.com> wrote: >> >>> Wow! This great feature is really work. I have the same issue - one way >>> audio. And another: "show channels" always return first negotiated codec. >>> When switching codec back, audio working fine. Tested with 3 different >>> softphones. >>> If someone interested, please confirm. >>> Thanks. >>> >>> >>> 2013/2/19 Michael Collins >>> >>>> pastebin the log. I'll bet Steven can diagnose it! ;) >>>> -MC >>>> >>>> >>>> On Tue, Feb 19, 2013 at 8:37 AM, Emrah wrote: >>>> >>>>> Excellent, that worked. >>>>> >>>>> However I am having a one way audio issue, where the peer that has its >>>>> codecs changed can hear but cannot be heard. >>>>> Even if both bridged peers are changed to a common codec it doesn't >>>>> work. >>>>> >>>>> Any idea? >>>>> >>>>> Thanks a bunch for your help. >>>>> On Feb 19, 2013, at 3:47 AM, Steven Ayre wrote: >>>>> >>>>> > Try: >>>>> > uuid_media_reneg 49f428e5-1d72-42d8-b0e1-5b3f29f9a4ca =PCMU >>>>> > >>>>> > (ie add = before PCMU) >>>>> > >>>>> > It appears to affect something in the code... >>>>> > >>>>> > -Steve >>>>> > >>>>> > >>>>> > On 19 February 2013 05:54, Emrah wrote: >>>>> >> Hey, >>>>> >> >>>>> >> In short, nothing seems to happen. I tried several phones. >>>>> >> I also tried enabling codec renegotiation on hold/unhold thinking >>>>> that the change in SDP with uuid_media_reneg would do it. >>>>> >> >>>>> >> This is what happens: >>>>> >> freeswitch at internal> show channels >>>>> >> >>>>> uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,sent_callee_name,sent_callee_num >>>>> >> 49f428e5-1d72-42d8-b0e1-5b3f29f9a4ca,inbound,2013-02-19 >>>>> 00:44:35,1361252675,sofia/internal/10000 at sip.domain.net,CS_EXECUTE,Emrah >>>>> - Macbook >>>>> Pro,10000,100.100.100.100,conf-3111,conference,3111-sip.domain.net at ek-custom-16k >>>>> ,XML,ek-conference,G722,16000,64000,G722,16000,64000,, >>>>> cornflakes.domain.net,10000 at sip.domain.net,,ACTIVE,,,,,, >>>>> >> >>>>> >> 1 total. >>>>> >> >>>>> >> freeswitch at internal> uuid_media_reneg >>>>> 49f428e5-1d72-42d8-b0e1-5b3f29f9a4ca PCMU >>>>> >> +OK Success >>>>> >> >>>>> >> 2013-02-19 00:45:25.172659 [DEBUG] sofia_glue.c:2647 Local SDP: >>>>> >> v=0 >>>>> >> o=FreeSWITCH 1361234787 1361234790 IN IP4 1.2.3.4 >>>>> >> s=FreeSWITCH >>>>> >> c=IN IP4 1.2.3.4 >>>>> >> t=0 0 >>>>> >> m=audio 17888 RTP/AVP 9 0 101 >>>>> >> a=rtpmap:101 telephone-event/8000 >>>>> >> a=fmtp:101 0-16 >>>>> >> a=silenceSupp:off - - - - >>>>> >> a=ptime:20 >>>>> >> a=sendrecv >>>>> >> >>>>> >> 2013-02-19 00:45:25.172659 [DEBUG] switch_core_session.c:975 Send >>>>> signal sofia/internal/10000 at sip.domain.net [BREAK] >>>>> >> 2013-02-19 00:45:25.182652 [DEBUG] sofia.c:5574 Channel >>>>> sofia/internal/10000 at sip.domain.net entering state [calling][0] >>>>> >> 2013-02-19 00:45:25.292661 [DEBUG] switch_core_session.c:975 Send >>>>> signal sofia/internal/10000 at sip.domain.net [BREAK] >>>>> >> 2013-02-19 00:45:25.292661 [DEBUG] switch_core_session.c:975 Send >>>>> signal sofia/internal/10000 at sip.domain.net [BREAK] >>>>> >> 2013-02-19 00:45:25.302683 [INFO] sofia.c:931 sofia/internal/ >>>>> 10000 at sip.domain.net Update Callee ID to "10000" <10000> >>>>> >> 2013-02-19 00:45:25.302683 [DEBUG] sofia.c:5574 Channel >>>>> sofia/internal/10000 at sip.domain.net entering state [completing][200] >>>>> >> 2013-02-19 00:45:25.302683 [DEBUG] sofia.c:5585 Remote SDP: >>>>> >> v=0 >>>>> >> o=- 3570241475 3570241476 IN IP4 10.0.0.131 >>>>> >> s=pjmedia >>>>> >> c=IN IP4 10.0.0.131 >>>>> >> t=0 0 >>>>> >> a=X-nat:0 >>>>> >> m=audio 4010 RTP/AVP 9 101 >>>>> >> a=rtcp:4011 IN IP4 10.0.0.131 >>>>> >> a=rtpmap:9 G722/8000 >>>>> >> a=sendrecv >>>>> >> a=rtpmap:101 telephone-event/8000 >>>>> >> a=fmtp:101 0-15 >>>>> >> >>>>> >> 2013-02-19 00:45:25.302683 [DEBUG] switch_core_session.c:975 Send >>>>> signal sofia/internal/10000 at sip.domain.net [BREAK] >>>>> >> 2013-02-19 00:45:25.302683 [DEBUG] switch_core_session.c:975 Send >>>>> signal sofia/internal/10000 at sip.domain.net [BREAK] >>>>> >> 2013-02-19 00:45:25.322657 [DEBUG] sofia.c:5574 Channel >>>>> sofia/internal/10000 at sip.domain.net entering state [ready][200] >>>>> >> 2013-02-19 00:45:25.322657 [DEBUG] sofia.c:5582 Duplicate SDP >>>>> >> v=0 >>>>> >> o=- 3570241475 3570241476 IN IP4 10.0.0.131 >>>>> >> s=pjmedia >>>>> >> c=IN IP4 10.0.0.131 >>>>> >> t=0 0 >>>>> >> a=X-nat:0 >>>>> >> m=audio 4010 RTP/AVP 9 101 >>>>> >> a=rtcp:4011 IN IP4 10.0.0.131 >>>>> >> a=rtpmap:9 G722/8000 >>>>> >> a=sendrecv >>>>> >> a=rtpmap:101 telephone-event/8000 >>>>> >> a=fmtp:101 0-15 >>>>> >> >>>>> >> 2013-02-19 00:45:25.322657 [DEBUG] sofia_glue.c:5137 Audio Codec >>>>> Compare [G722:9:8000:20:64000]/[G722:9:8000:20:64000] >>>>> >> 2013-02-19 00:45:25.322657 [DEBUG] sofia_glue.c:3022 Already using >>>>> G722 >>>>> >> 2013-02-19 00:45:25.322657 [DEBUG] sofia_glue.c:5266 Set 2833 dtmf >>>>> send/recv payload to 101 >>>>> >> >>>>> >> >>>>> >> Thanks for any help, >>>>> >> Emrah >>>>> >> On Feb 18, 2013, at 1:38 PM, Michael Collins >>>>> wrote: >>>>> >> >>>>> >>> What happened with the uuid_media_reneg command? >>>>> >>> -MC >>>>> >>> >>>>> >>> On Sat, Feb 16, 2013 at 3:10 PM, Emrah wrote: >>>>> >>> Hi all, >>>>> >>> >>>>> >>> How do we go about renegotiating the codecs of a call that is >>>>> already established? >>>>> >>> I tried uuid_media_reneg to no avail. >>>>> >>> E.g.: Upgrade someone who's entered a 16k conference from PCMU to >>>>> G722. >>>>> >>> >>>>> >>> Cheers and thanks, >>>>> >>> Emrah >>>>> >>> >>>>> _________________________________________________________________________ >>>>> >>> Professional FreeSWITCH Consulting Services: >>>>> >>> consulting at freeswitch.org >>>>> >>> http://www.freeswitchsolutions.com >>>>> >>> >>>>> >>> >>>>> >>> >>>>> >>> >>>>> >>> Official FreeSWITCH Sites >>>>> >>> http://www.freeswitch.org >>>>> >>> http://wiki.freeswitch.org >>>>> >>> http://www.cluecon.com >>>>> >>> >>>>> >>> 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 >>>>> >>> >>>>> >>> >>>>> >>> >>>>> >>> -- >>>>> >>> Michael S Collins >>>>> >>> Twitter: @mercutioviz >>>>> >>> http://www.FreeSWITCH.org >>>>> >>> http://www.ClueCon.com >>>>> >>> http://www.OSTAG.org >>>>> >>> >>>>> >>> >>>>> _________________________________________________________________________ >>>>> >>> Professional FreeSWITCH Consulting Services: >>>>> >>> consulting at freeswitch.org >>>>> >>> http://www.freeswitchsolutions.com >>>>> >>> >>>>> >>> >>>>> >>> >>>>> >>> >>>>> >>> Official FreeSWITCH Sites >>>>> >>> http://www.freeswitch.org >>>>> >>> http://wiki.freeswitch.org >>>>> >>> http://www.cluecon.com >>>>> >>> >>>>> >>> 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 >>>>> >> >>>>> >> >>>>> >> >>>>> _________________________________________________________________________ >>>>> >> Professional FreeSWITCH Consulting Services: >>>>> >> consulting at freeswitch.org >>>>> >> http://www.freeswitchsolutions.com >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> Official FreeSWITCH Sites >>>>> >> http://www.freeswitch.org >>>>> >> http://wiki.freeswitch.org >>>>> >> http://www.cluecon.com >>>>> >> >>>>> >> 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 >>>>> > >>>>> > >>>>> _________________________________________________________________________ >>>>> > Professional FreeSWITCH Consulting Services: >>>>> > consulting at freeswitch.org >>>>> > http://www.freeswitchsolutions.com >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > Official FreeSWITCH Sites >>>>> > http://www.freeswitch.org >>>>> > http://wiki.freeswitch.org >>>>> > http://www.cluecon.com >>>>> > >>>>> > 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 >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> 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 >>>>> >>>> >>>> >>>> >>>> -- >>>> Michael S Collins >>>> Twitter: @mercutioviz >>>> http://www.FreeSWITCH.org >>>> http://www.ClueCon.com >>>> http://www.OSTAG.org >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org 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URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130220/32d7334a/attachment-0001.html From dvl36.ripe.nick at gmail.com Thu Feb 21 00:01:26 2013 From: dvl36.ripe.nick at gmail.com (Dmitry Lysenko) Date: Wed, 20 Feb 2013 23:01:26 +0200 Subject: [Freeswitch-users] Changing codec during calls In-Reply-To: References: <6EF8B8E9-4CC5-480C-B1C1-09CBC533470C@kavun.ch> <35AF9E79-771D-440A-BEBB-A856D831161C@kavun.ch> Message-ID: PB-20616 2013/2/20 Michael Collins > Haha, you don't need an account. Re-read the challenge dialog on > pastebin.freeswitch.org. > > -MC > > > On Wed, Feb 20, 2013 at 11:57 AM, Dmitry Lysenko < > dvl36.ripe.nick at gmail.com> wrote: > >> I do not have account on pastebin.freeswitch.org >> See: http://jira.freeswitch.org/browse/FS-5116 >> Thanks. >> >> >> 2013/2/20 Michael Collins >> >>> Definitely get the console log and SIP trace of those problem calls and >>> put them in pastein.freeswitch.org. >>> >>> -MC >>> >>> >>> On Wed, Feb 20, 2013 at 5:04 AM, Dmitry Lysenko < >>> dvl36.ripe.nick at gmail.com> wrote: >>> >>>> Wow! This great feature is really work. I have the same issue - one way >>>> audio. And another: "show channels" always return first negotiated codec. >>>> When switching codec back, audio working fine. Tested with 3 different >>>> softphones. >>>> If someone interested, please confirm. >>>> Thanks. >>>> >>>> >>>> 2013/2/19 Michael Collins >>>> >>>>> pastebin the log. I'll bet Steven can diagnose it! ;) >>>>> -MC >>>>> >>>>> >>>>> On Tue, Feb 19, 2013 at 8:37 AM, Emrah wrote: >>>>> >>>>>> Excellent, that worked. >>>>>> >>>>>> However I am having a one way audio issue, where the peer that has >>>>>> its codecs changed can hear but cannot be heard. >>>>>> Even if both bridged peers are changed to a common codec it doesn't >>>>>> work. >>>>>> >>>>>> Any idea? >>>>>> >>>>>> Thanks a bunch for your help. >>>>>> On Feb 19, 2013, at 3:47 AM, Steven Ayre wrote: >>>>>> >>>>>> > Try: >>>>>> > uuid_media_reneg 49f428e5-1d72-42d8-b0e1-5b3f29f9a4ca =PCMU >>>>>> > >>>>>> > (ie add = before PCMU) >>>>>> > >>>>>> > It appears to affect something in the code... >>>>>> > >>>>>> > -Steve >>>>>> > >>>>>> > >>>>>> > On 19 February 2013 05:54, Emrah wrote: >>>>>> >> Hey, >>>>>> >> >>>>>> >> In short, nothing seems to happen. I tried several phones. >>>>>> >> I also tried enabling codec renegotiation on hold/unhold thinking >>>>>> that the change in SDP with uuid_media_reneg would do it. >>>>>> >> >>>>>> >> This is what happens: >>>>>> >> freeswitch at internal> show channels >>>>>> >> >>>>>> uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,sent_callee_name,sent_callee_num >>>>>> >> 49f428e5-1d72-42d8-b0e1-5b3f29f9a4ca,inbound,2013-02-19 >>>>>> 00:44:35,1361252675,sofia/internal/10000 at sip.domain.net,CS_EXECUTE,Emrah >>>>>> - Macbook >>>>>> Pro,10000,100.100.100.100,conf-3111,conference,3111-sip.domain.net at ek-custom-16k >>>>>> ,XML,ek-conference,G722,16000,64000,G722,16000,64000,, >>>>>> cornflakes.domain.net,10000 at sip.domain.net,,ACTIVE,,,,,, >>>>>> >> >>>>>> >> 1 total. >>>>>> >> >>>>>> >> freeswitch at internal> uuid_media_reneg >>>>>> 49f428e5-1d72-42d8-b0e1-5b3f29f9a4ca PCMU >>>>>> >> +OK Success >>>>>> >> >>>>>> >> 2013-02-19 00:45:25.172659 [DEBUG] sofia_glue.c:2647 Local SDP: >>>>>> >> v=0 >>>>>> >> o=FreeSWITCH 1361234787 1361234790 IN IP4 1.2.3.4 >>>>>> >> s=FreeSWITCH >>>>>> >> c=IN IP4 1.2.3.4 >>>>>> >> t=0 0 >>>>>> >> m=audio 17888 RTP/AVP 9 0 101 >>>>>> >> a=rtpmap:101 telephone-event/8000 >>>>>> >> a=fmtp:101 0-16 >>>>>> >> a=silenceSupp:off - - - - >>>>>> >> a=ptime:20 >>>>>> >> a=sendrecv >>>>>> >> >>>>>> >> 2013-02-19 00:45:25.172659 [DEBUG] switch_core_session.c:975 Send >>>>>> signal sofia/internal/10000 at sip.domain.net [BREAK] >>>>>> >> 2013-02-19 00:45:25.182652 [DEBUG] sofia.c:5574 Channel >>>>>> sofia/internal/10000 at sip.domain.net entering state [calling][0] >>>>>> >> 2013-02-19 00:45:25.292661 [DEBUG] switch_core_session.c:975 Send >>>>>> signal sofia/internal/10000 at sip.domain.net [BREAK] >>>>>> >> 2013-02-19 00:45:25.292661 [DEBUG] switch_core_session.c:975 Send >>>>>> signal sofia/internal/10000 at sip.domain.net [BREAK] >>>>>> >> 2013-02-19 00:45:25.302683 [INFO] sofia.c:931 sofia/internal/ >>>>>> 10000 at sip.domain.net Update Callee ID to "10000" <10000> >>>>>> >> 2013-02-19 00:45:25.302683 [DEBUG] sofia.c:5574 Channel >>>>>> sofia/internal/10000 at sip.domain.net entering state [completing][200] >>>>>> >> 2013-02-19 00:45:25.302683 [DEBUG] sofia.c:5585 Remote SDP: >>>>>> >> v=0 >>>>>> >> o=- 3570241475 3570241476 IN IP4 10.0.0.131 >>>>>> >> s=pjmedia >>>>>> >> c=IN IP4 10.0.0.131 >>>>>> >> t=0 0 >>>>>> >> a=X-nat:0 >>>>>> >> m=audio 4010 RTP/AVP 9 101 >>>>>> >> a=rtcp:4011 IN IP4 10.0.0.131 >>>>>> >> a=rtpmap:9 G722/8000 >>>>>> >> a=sendrecv >>>>>> >> a=rtpmap:101 telephone-event/8000 >>>>>> >> a=fmtp:101 0-15 >>>>>> >> >>>>>> >> 2013-02-19 00:45:25.302683 [DEBUG] switch_core_session.c:975 Send >>>>>> signal sofia/internal/10000 at sip.domain.net [BREAK] >>>>>> >> 2013-02-19 00:45:25.302683 [DEBUG] switch_core_session.c:975 Send >>>>>> signal sofia/internal/10000 at sip.domain.net [BREAK] >>>>>> >> 2013-02-19 00:45:25.322657 [DEBUG] sofia.c:5574 Channel >>>>>> sofia/internal/10000 at sip.domain.net entering state [ready][200] >>>>>> >> 2013-02-19 00:45:25.322657 [DEBUG] sofia.c:5582 Duplicate SDP >>>>>> >> v=0 >>>>>> >> o=- 3570241475 3570241476 IN IP4 10.0.0.131 >>>>>> >> s=pjmedia >>>>>> >> c=IN IP4 10.0.0.131 >>>>>> >> t=0 0 >>>>>> >> a=X-nat:0 >>>>>> >> m=audio 4010 RTP/AVP 9 101 >>>>>> >> a=rtcp:4011 IN IP4 10.0.0.131 >>>>>> >> a=rtpmap:9 G722/8000 >>>>>> >> a=sendrecv >>>>>> >> a=rtpmap:101 telephone-event/8000 >>>>>> >> a=fmtp:101 0-15 >>>>>> >> >>>>>> >> 2013-02-19 00:45:25.322657 [DEBUG] sofia_glue.c:5137 Audio Codec >>>>>> Compare [G722:9:8000:20:64000]/[G722:9:8000:20:64000] >>>>>> >> 2013-02-19 00:45:25.322657 [DEBUG] sofia_glue.c:3022 Already using >>>>>> G722 >>>>>> >> 2013-02-19 00:45:25.322657 [DEBUG] sofia_glue.c:5266 Set 2833 dtmf >>>>>> send/recv payload to 101 >>>>>> >> >>>>>> >> >>>>>> >> Thanks for any help, >>>>>> >> Emrah >>>>>> >> On Feb 18, 2013, at 1:38 PM, Michael Collins >>>>>> wrote: >>>>>> >> >>>>>> >>> What happened with the uuid_media_reneg command? >>>>>> >>> -MC >>>>>> >>> >>>>>> >>> On Sat, Feb 16, 2013 at 3:10 PM, Emrah wrote: >>>>>> >>> Hi all, >>>>>> >>> >>>>>> >>> How do we go about renegotiating the codecs of a call that is >>>>>> already established? >>>>>> >>> I tried uuid_media_reneg to no avail. >>>>>> >>> E.g.: Upgrade someone who's entered a 16k conference from PCMU to >>>>>> G722. >>>>>> >>> >>>>>> >>> Cheers and thanks, >>>>>> >>> Emrah >>>>>> >>> >>>>>> _________________________________________________________________________ >>>>>> >>> Professional FreeSWITCH Consulting Services: >>>>>> >>> consulting at freeswitch.org >>>>>> >>> http://www.freeswitchsolutions.com >>>>>> >>> >>>>>> >>> >>>>>> >>> >>>>>> >>> >>>>>> >>> Official FreeSWITCH Sites >>>>>> >>> http://www.freeswitch.org >>>>>> >>> http://wiki.freeswitch.org >>>>>> >>> http://www.cluecon.com >>>>>> >>> >>>>>> >>> 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 >>>>>> >>> >>>>>> >>> >>>>>> >>> >>>>>> >>> -- >>>>>> >>> Michael S Collins >>>>>> >>> Twitter: @mercutioviz >>>>>> >>> http://www.FreeSWITCH.org >>>>>> >>> http://www.ClueCon.com >>>>>> >>> http://www.OSTAG.org >>>>>> >>> >>>>>> >>> >>>>>> _________________________________________________________________________ >>>>>> >>> Professional FreeSWITCH Consulting Services: >>>>>> >>> consulting at freeswitch.org >>>>>> >>> http://www.freeswitchsolutions.com >>>>>> >>> >>>>>> >>> >>>>>> >>> >>>>>> >>> >>>>>> >>> Official FreeSWITCH Sites >>>>>> >>> http://www.freeswitch.org >>>>>> >>> http://wiki.freeswitch.org >>>>>> >>> http://www.cluecon.com >>>>>> >>> >>>>>> >>> 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 >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> _________________________________________________________________________ >>>>>> >> Professional FreeSWITCH Consulting Services: >>>>>> >> consulting at freeswitch.org >>>>>> >> http://www.freeswitchsolutions.com >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> Official FreeSWITCH Sites >>>>>> >> http://www.freeswitch.org >>>>>> >> http://wiki.freeswitch.org >>>>>> >> http://www.cluecon.com >>>>>> >> >>>>>> >> 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 >>>>>> > >>>>>> > >>>>>> _________________________________________________________________________ >>>>>> > Professional FreeSWITCH Consulting Services: >>>>>> > consulting at freeswitch.org >>>>>> > http://www.freeswitchsolutions.com >>>>>> > >>>>>> > >>>>>> > >>>>>> > >>>>>> > Official FreeSWITCH Sites >>>>>> > http://www.freeswitch.org >>>>>> > http://wiki.freeswitch.org >>>>>> > http://www.cluecon.com >>>>>> > >>>>>> > 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 >>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> 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 >>>>>> >>>>> >>>>> >>>>> >>>>> -- >>>>> Michael S Collins >>>>> Twitter: @mercutioviz >>>>> http://www.FreeSWITCH.org >>>>> http://www.ClueCon.com >>>>> http://www.OSTAG.org >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> 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 >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>>> >>>> >>> >>> >>> -- >>> Michael S Collins >>> Twitter: @mercutioviz >>> http://www.FreeSWITCH.org >>> http://www.ClueCon.com >>> http://www.OSTAG.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > 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URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130220/2ad628f0/attachment-0001.html From cal.leeming at simplicitymedialtd.co.uk Thu Feb 21 00:03:53 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Wed, 20 Feb 2013 21:03:53 +0000 Subject: [Freeswitch-users] DID providers - any thoughts? (jan 2013) In-Reply-To: References: Message-ID: Seems that flowroute is getting some good feedback.. If anyone else has comments/feedback about their experiences with this company, please feel free to update the thread. Cal On Wed, Feb 20, 2013 at 11:57 AM, chris wrote: > +1 here, flowroute has exceeded my expectations in many areas. The fact > that they actually have humans available via phone, and even humans who > understand the technology quite well. > > Reliability has been great but you should always plan for a rainy day and > have a backup plan, even the biggest and best have issues from time to time > > chris > > On Thu, Jan 31, 2013 at 5:34 PM, Steven Schoch < > schoch+freeswitch.org at xwin32.com> wrote: > >> On Wed, Jan 30, 2013 at 4:59 AM, Cal Leeming [Simplicity Media Ltd] < >> cal.leeming at simplicitymedialtd.co.uk> wrote: >> >>> >>> Flowroute - Any else able to offer some reviews on this company - >>> specifically relating to US inbound?? >>> >> >> I'm just starting my VoIP venture and I have chosen Flowroute as our >> provider. So far, mostly as a test, I have transferred one of our AT&T >> lines to Flowroute. It worked very smoothly. One thing I discovered, >> which you all probably know, but I'm new with VoIP, was that when I called >> the number from a POTS line, it went through the PSTN, through Flowroute, >> through FreeSwitch, and rang on my Polycom phone even before I heard the >> first ring from the calling phone. >> >> Also, the Flowroute tech support team responds very quickly and helpfully. >> >> No complaints. >> >> -- >> Steve >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130220/90787df0/attachment.html From shaheryarkh at gmail.com Thu Feb 21 00:03:59 2013 From: shaheryarkh at gmail.com (Muhammad Shahzad) Date: Wed, 20 Feb 2013 22:03:59 +0100 Subject: [Freeswitch-users] Config several FS for routing and reading unique configs In-Reply-To: References: Message-ID: Use xml_curl module to serve configs from single web server. Use ODBC to share FS db. Thank you. On Wed, Feb 20, 2013 at 8:07 PM, Sayyed Mohammad Emami Razavi < emamirazavi at gmail.com> wrote: > How to config several FS systems for 4 FS servers after one LVS(load > balancer) so that all agents on all systems defined correctly and route > without any problem?! > You know that i should have one DB [for e.g. blue.box] and one CONF DIR > for my FS XML config files! What is your idea? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +49 176 99 83 10 85 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130220/832ee023/attachment.html From craig.robert at hotmail.com Thu Feb 21 00:13:00 2013 From: craig.robert at hotmail.com (Robert Craig) Date: Wed, 20 Feb 2013 14:13:00 -0700 Subject: [Freeswitch-users] DID providers - any thoughts? (jan 2013) In-Reply-To: References: , , , , Message-ID: I can vouch for Flowroute. I use them as one of my primary providers and have had no problems. In the event you experience a glitch, you can submit a support ticket or call them and they will work with you. Very helpful techs. Rob Unlimited calling to the US and Canada.Visit www.craigcomm.net for more information. Date: Wed, 20 Feb 2013 21:03:53 +0000 From: cal.leeming at simplicitymedialtd.co.uk To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] DID providers - any thoughts? (jan 2013) Seems that flowroute is getting some good feedback.. If anyone else has comments/feedback about their experiences with this company, please feel free to update the thread. Cal On Wed, Feb 20, 2013 at 11:57 AM, chris wrote: +1 here, flowroute has exceeded my expectations in many areas. The fact that they actually have humans available via phone, and even humans who understand the technology quite well. Reliability has been great but you should always plan for a rainy day and have a backup plan, even the biggest and best have issues from time to time chris On Thu, Jan 31, 2013 at 5:34 PM, Steven Schoch wrote: On Wed, Jan 30, 2013 at 4:59 AM, Cal Leeming [Simplicity Media Ltd] wrote: Flowroute - Any else able to offer some reviews on this company - specifically relating to US inbound?? I'm just starting my VoIP venture and I have chosen Flowroute as our provider. So far, mostly as a test, I have transferred one of our AT&T lines to Flowroute. It worked very smoothly. One thing I discovered, which you all probably know, but I'm new with VoIP, was that when I called the number from a POTS line, it went through the PSTN, through Flowroute, through FreeSwitch, and rang on my Polycom phone even before I heard the first ring from the calling phone. Also, the Flowroute tech support team responds very quickly and helpfully. No complaints. -- Steve _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130220/33ed5cb1/attachment-0001.html From X.Liu at hw.ac.uk Thu Feb 21 02:05:39 2013 From: X.Liu at hw.ac.uk (Liu, Xingkun) Date: Wed, 20 Feb 2013 23:05:39 -0000 Subject: [Freeswitch-users] Weird record_session behaviours References: , , , , Message-ID: Hello, I am using record_session via ESL to record the user/system conversations of a spoken dialogue system. The audio is saved as a wav file. I am also doing the logging from the start of the recording until the user hangs up. I found the recorded audio length is shorter than the time length shown in the log file. Roughly it is 20 seconds shorter in a 10 minutes recording. Googled around, didn't find any clue. ( I did find a similar post which was a year ago and fixed in the newer FS version, I just tried latest FS git head today but still have the same problem as before). Does anybody have any idea about the possible reasons for that? Many thanks! Xing ----- Sunday Times Scottish University of the Year 2011-2013 Top in the UK for student experience Fourth university in the UK and top in Scotland (National Student Survey 2012) We invite research leaders and ambitious early career researchers to join us in leading and driving research in key inter-disciplinary themes. Please see www.hw.ac.uk/researchleaders for further information and how to apply. Heriot-Watt University is a Scottish charity registered under charity number SC000278. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130220/3cd36d0a/attachment.html From mario_fs at mgtech.com Thu Feb 21 02:42:17 2013 From: mario_fs at mgtech.com (Mario G) Date: Wed, 20 Feb 2013 15:42:17 -0800 Subject: [Freeswitch-users] Memory growth In-Reply-To: References: Message-ID: <7D4D7EF3-9889-4C06-A1E4-26225C86E791@mgtech.com> FYI, Anthony nailed my memory leak yesterday (see the jira below). 24 hours of testing aok so far. The leak varied between 300k-2M per hour, not sure if it was only OSX related so good idea for anyone with a leak to test with updated HEAD. Mario G On Feb 18, 2013, at 11:22 AM, Ken Rice wrote: > We?re trying to figure out where this last bit of leaking is coming from... Looks like it might be related to Registrations some how... > > Once we squash that there will be some more info on the 1.2 branch coming > > > On 2/18/13 12:19 PM, "Mario G" wrote: > >> FYI/in case it's related: I had a substantial memory leak on the Oct 25 though Jan 29 Head versions and 1.2.3. After Feb 11 head, now a much smaller leak I can't pinpoint, see http://jira.freeswitch.org/browse/FS-5095#comment-39379. Updating to HEAD will probably fix it, I hope someone else sees the slow leak since I am getting nowhere real fast on it..... >> Mario G >> >> On Feb 18, 2013, at 3:16 AM, Steven Ayre wrote: >> >>> Are you able to reproduce it on a newer version? Try current head of the master branch. >>> >>> If you can, then try running FS through valgrind which may reveal the source of any memory leaks. >>> >>> -Steve >>> >>> >>> >>> On 18 February 2013 11:00, ??????? ??????? wrote: >>>> Hi, list! >>>> I've installed FreeSWITCH Version 1.2.3+git~20120920T220849Z~f718a5e8e6 >>>> (1.2.3; git at commit f718a5e8e6 on Thu, 20 Sep 2012 22:08:49 Z) >>>> >>>> I noticed that FreeSwtich uses a lot of memory of my system, and percent >>>> of used memory is gradually growing regardless of the number of the >>>> simultaneous calls: >>>> >>>> # uname -a >>>> Linux freeswitch1 2.6.18-308.13.1.el5 #1 SMP Tue Aug 21 17:10:18 EDT >>>> 2012 x86_64 x86_64 x86_64 GNU/Linux >>>> >>>> # top >>>> top - 14:21:39 up 140 days, 29 min, 2 users, load average: 0.11, 0.14, >>>> 0.10 >>>> Tasks: 99 total, 1 running, 96 sleeping, 2 stopped, 0 zombie >>>> Cpu(s): 1.2%us, 0.8%sy, 0.0%ni, 96.9%id, 0.1%wa, 0.2%hi, 0.7%si, >>>> 0.0%st >>>> Mem: 3781680k total, 3748360k used, 33320k free, 120708k buffers >>>> Swap: 2096472k total, 17224k used, 2079248k free, 1838356k cached >>>> >>>> PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND >>>> 9584 root -2 -10 1792m 1.3g 8252 S 2.7 36.8 935:07.85 freeswitch >>>> 3095 mysql 15 0 329m 28m 4588 S 0.7 0.8 713:07.03 mysqld >>>> >>>> At this moment FS handles no more 15 simultaneous calls, and has about >>>> 117 registered users. >>>> Is there a tool to monitor, which of processes or modules in FS is using >>>> memory and how much? >>>> >>>> -- >>>> Best regards, >>>> Vitaly. >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 > > -- > Ken > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > irc.freenode.net #freeswitch > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130220/67ef3374/attachment.html From schoch+freeswitch.org at xwin32.com Thu Feb 21 02:56:08 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Wed, 20 Feb 2013 15:56:08 -0800 Subject: [Freeswitch-users] Is this an internal call? Message-ID: This must be a FAQ, but I can't find it. In the default dialplan, how can I tell if the call was from an internal phone, or from an external call that was transferred? One page said there was a variable ${internal_call}, but I can't find that anywhere in the source. Should I just look to see if a directory variable, like account_code or user_context is set? -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130220/a022c4ec/attachment.html From jaykris at gmail.com Thu Feb 21 03:02:59 2013 From: jaykris at gmail.com (JP) Date: Wed, 20 Feb 2013 16:02:59 -0800 Subject: [Freeswitch-users] mod-lua Vs mod-java Message-ID: Hi, I want to access a webservice that exposes a SOAP interface. I was wondering if this is better to do in Lua or in Java. Any suggestions will be greatly appreciated. Also in general is it better to use mod-java instead of mod-lua for doing advanced IVR functions like database access or web service access? Are there any drawbacks to using java instead of Lua? Thanks, JP -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130220/58a7986c/attachment-0001.html From vitaliy.davudov at vts24.ru Thu Feb 21 08:14:13 2013 From: vitaliy.davudov at vts24.ru (=?UTF-8?B?0JLQuNGC0LDQu9C40Lkg0JTQsNCy0YPQtNC+0LI=?=) Date: Thu, 21 Feb 2013 09:14:13 +0400 Subject: [Freeswitch-users] Memory growth In-Reply-To: <7D4D7EF3-9889-4C06-A1E4-26225C86E791@mgtech.com> References: <7D4D7EF3-9889-4C06-A1E4-26225C86E791@mgtech.com> Message-ID: <5125AD25.6020005@vts24.ru> I would be glad to do it, but my FS in git branch v1.2.stable (it is in production environment). I will have to wait patches to the 1.2 series. 21.02.2013 3:42, Mario G ?????: > FYI, Anthony nailed my memory leak yesterday (see the jira below). 24 > hours of testing aok so far. The leak varied between 300k-2M per hour, > not sure if it was only OSX related so good idea for anyone with a > leak to test with updated HEAD. > Mario G > > On Feb 18, 2013, at 11:22 AM, Ken Rice wrote: > >> Re: [Freeswitch-users] Memory growth >> We?re trying to figure out where this last bit of leaking is coming >> from... Looks like it might be related to Registrations some how... >> >> Once we squash that there will be some more info on the 1.2 branch coming >> >> >> On 2/18/13 12:19 PM, "Mario G" > > wrote: >> >>> FYI/in case it's related: I had a substantial memory leak on the Oct >>> 25 though Jan 29 Head versions and 1.2.3. After Feb 11 head, now a >>> much smaller leak I can't pinpoint, see >>> http://jira.freeswitch.org/browse/FS-5095#comment-39379. Updating to >>> HEAD will probably fix it, I hope someone else sees the slow leak >>> since I am getting nowhere real fast on it..... >>> Mario G >>> >>> On Feb 18, 2013, at 3:16 AM, Steven Ayre wrote: >>> >>>> Are you able to reproduce it on a newer version? Try current head >>>> of the master branch. >>>> >>>> If you can, then try running FS through valgrind which may reveal >>>> the source of any memory leaks. >>>> >>>> -Steve >>>> >>>> >>>> >>>> On 18 February 2013 11:00, ??????? ??????? >>>> > wrote: >>>>> Hi, list! >>>>> I've installed FreeSWITCH Version >>>>> 1.2.3+git~20120920T220849Z~f718a5e8e6 >>>>> (1.2.3; git at commit f718a5e8e6 on Thu, 20 Sep 2012 22:08:49 Z) >>>>> >>>>> I noticed that FreeSwtich uses a lot of memory of my system, and >>>>> percent >>>>> of used memory is gradually growing regardless of the number of the >>>>> simultaneous calls: >>>>> >>>>> # uname -a >>>>> Linux freeswitch1 2.6.18-308.13.1.el5 #1 SMP Tue Aug 21 17:10:18 EDT >>>>> 2012 x86_64 x86_64 x86_64 GNU/Linux >>>>> >>>>> # top >>>>> top - 14:21:39 up 140 days, 29 min, 2 users, load average: 0.11, >>>>> 0.14, >>>>> 0.10 >>>>> Tasks: 99 total, 1 running, 96 sleeping, 2 stopped, 0 zombie >>>>> Cpu(s): 1.2%us, 0.8%sy, 0.0%ni, 96.9%id, 0.1%wa, 0.2%hi, 0.7%si, >>>>> 0.0%st >>>>> Mem: 3781680k total, 3748360k used, 33320k free, 120708k >>>>> buffers >>>>> Swap: 2096472k total, 17224k used, 2079248k free, 1838356k >>>>> cached >>>>> >>>>> PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND >>>>> 9584 root -2 -10 1792m 1.3g 8252 S 2.7 36.8 935:07.85 >>>>> freeswitch >>>>> 3095 mysql 15 0 329m 28m 4588 S 0.7 0.8 713:07.03 mysqld >>>>> >>>>> At this moment FS handles no more 15 simultaneous calls, and has about >>>>> 117 registered users. >>>>> Is there a tool to monitor, which of processes or modules in FS is >>>>> using >>>>> memory and how much? >>>>> >>>>> -- >>>>> Best regards, >>>>> Vitaly. >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> >>>>> http://wiki.freeswitch.org >>>>> >>>>> http://www.cluecon.com >>>>> >>>>> >>>>> 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 >>>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>> >>> >>> ------------------------------------------------------------------------ >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >> >> -- >> Ken >> _http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> _irc.freenode.net #freeswitch >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -- ? ?????????? ???????????, ??????? ??????? ????????? ??? "???-???????-??????" (?????? ???????? "ETERIA") http://www.vts24.ru ???: (495) 989-47-00 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130221/72725869/attachment-0001.html From vbvbrj at gmail.com Thu Feb 21 09:21:24 2013 From: vbvbrj at gmail.com (Mimiko) Date: Thu, 21 Feb 2013 08:21:24 +0200 Subject: [Freeswitch-users] DSN for callcenter, what does it do? In-Reply-To: References: Message-ID: <5125BCE4.30105@gmail.com> On 20.02.2013 21:10, Sayyed Mohammad Emami Razavi wrote: > DSN and SQLite and update mode helps in which needs in call center module? mod_callcenter has its own DB where it stores statuses about agents and members. Agents can log in/log out dynamically. Members enters and leaves. Tiers can be created and deleted dynamically. -- Mimiko desu. From vbvbrj at gmail.com Thu Feb 21 09:24:36 2013 From: vbvbrj at gmail.com (Mimiko) Date: Thu, 21 Feb 2013 08:24:36 +0200 Subject: [Freeswitch-users] mod-lua Vs mod-java In-Reply-To: References: Message-ID: <5125BDA4.20505@gmail.com> On 21.02.2013 02:02, JP wrote: > Hi, > I want to access a webservice that exposes a SOAP interface. I was > wondering if this is better to do in Lua or in Java. Any suggestions > will be greatly appreciated. Also in general is it better to use > mod-java instead of mod-lua for doing advanced IVR functions like > database access or web service access? Are there any drawbacks to using > java instead of Lua? > > Thanks, > JP I had a conversation with a member on this list about what to use for scripting: lua, perl. java. Ended up that lua is the fastest in its own. For doing simple things, even IVR's, lua is enough. And lua is installed with FS. I use lua to connect to DB and don't see any performance drop. -- Mimiko desu. From sdevoy at bizfocused.com Thu Feb 21 09:25:46 2013 From: sdevoy at bizfocused.com (Sean Devoy) Date: Thu, 21 Feb 2013 01:25:46 -0500 Subject: [Freeswitch-users] Determine if a number is valid/working In-Reply-To: References: <072701ce0ebf$c5808640$508192c0$@bizfocused.com> <0d0401ce0f74$aa9fb7a0$ffdf26e0$@bizfocused.com> Message-ID: <12c801ce0ffc$48f84b10$dae8e130$@bizfocused.com> Hey MC. I like Steven's response as well. Can't quite do the connect to sales agent yet in the process, need to pull credit report and load DBMS values. The cost number is cost of lead from providers like google, msn, etc. It is a VERY competitive market. That does not count the cost of sales reps dialing non-working numbers on about 1/3 of the leads. I suspect we will offer a tiered approach like: 1. Area Code, "State" Field and IP Geo Data State Field all match - accept it. 2. Else try LNP Lookup for match 3. Else still fishy - Web Popup to explain and then Robo Dial and require a "pin" to be entered. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Wednesday, February 20, 2013 12:26 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Determine if a number is valid/working On Wed, Feb 20, 2013 at 6:14 AM, Sean Devoy wrote: Thank you for your responses, everyone. MC: This customer has rejected the idea of calling with a message as bad marketing!! In my experience marketing people have their own "logic rules" and I can never follow their thinking. And yes, they have 30 customer sales reps calling these "leads" all day long. They claim 1/3 of the leads have invalid phone numbers. My customer suspects fraud and that someone is submitting false leads to boost their lead rate and fees. I agree with Steven's assessment (elsewhere in this thread). Ken: I am not sure we have any language in common - lol. Where can I learn what "ASR and ACD" are? I looked up lidb and lnp already! Also, the cost of a call compared to the cost of a fraudulent lead (over $20) is insignificant. How is that cost calculated and to whom does the money go? Maybe the "verification call" needs to take place at the time that the form is filled out. The ideas I have suggested to my customer, for what it is worth, are comparing the State (we are in US) for the IP address of the submitter, to the State on the form and the State of the area code on the form. In the situations where they do not agree, prompt the user for permission to robo call for verification. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130221/01b464c1/attachment.html From mario at ims.bg Thu Feb 21 09:35:52 2013 From: mario at ims.bg (Mario Karakanovski) Date: Thu, 21 Feb 2013 08:35:52 +0200 Subject: [Freeswitch-users] freeswitch hack In-Reply-To: References: <958ABE902C1F47B09FFAAE7EDEE074B1@imsdeveloper> Message-ID: In my situation all calls are rejected, but I think it is because they are authenticated with invalid username. My concern is how ones can authenticate in freeswitch with user that not exists and never was configured. I was not able to reproduce that. What I found so far: they use a couple of IPs. They send OPTIONS (only one time) during the day and start try at the night. They tried a maximum of 100 calls. I am still waiting to log some packet Mario _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Wednesday, February 20, 2013 10:41 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] freeswitch hack Aren't they supposed to be rejected? On Wed, Feb 20, 2013 at 11:19 AM, Blake Priddy wrote: I have also had the situation that they are calls getting rejected. On Wed, Feb 20, 2013 at 11:08 AM, Michael Collins wrote: On Wed, Feb 20, 2013 at 1:53 AM, Mario Karakanovski wrote: Thanks Ken, It is helpful, but I still think there is some security issue. I've double check configuration. I've try to reproduce the issue trying to do direct call (TCP and UDP) or authenticate with invalid user, but everything works as expected - calls/authentication was rejected. I've decide to log the traffic - maybe I will be able to see where is the problem. What "security issue"? You said that they cannot make calls with the passwords that they've guessed, correct? About the only thing left to do is set up fail2ban and just shut the door on them when they fail too many times. -Michael _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -- Blakelund Priddy Network Systems Engineer Bryant Public School District Bryant, Arkansas 72022 http://www.bryantschools.org p 501-653-5038 f 501-847-5656 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130221/4116a8f4/attachment-0001.html From a.venugopan at mundio.com Thu Feb 21 12:10:46 2013 From: a.venugopan at mundio.com (Archana Venugopan) Date: Thu, 21 Feb 2013 09:10:46 +0000 Subject: [Freeswitch-users] make mod_voicemail Message-ID: <592A9CF93E12394E8472A6CC66E66BF23AC9F1@Mail-Kilo.squay.com> Hi, When I try to give 'make mod_voicemail' command, I see its compiling everything as like make command instead of just doing mod_voicemail. Can someone suggest why is this happening and what can be done to just compile mod_voicemail. Many thanks. Regards, Archana -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130221/a94998d5/attachment.html From avi at avimarcus.net Thu Feb 21 12:24:12 2013 From: avi at avimarcus.net (Avi Marcus) Date: Thu, 21 Feb 2013 11:24:12 +0200 Subject: [Freeswitch-users] make mod_voicemail In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF23AC9F1@Mail-Kilo.squay.com> References: <592A9CF93E12394E8472A6CC66E66BF23AC9F1@Mail-Kilo.squay.com> Message-ID: Try: make mod_voicemail-install -Avi Marcus On Thu, Feb 21, 2013 at 11:10 AM, Archana Venugopan wrote: > Hi,**** > > ** ** > > When I try to give ?make mod_voicemail? command, I see its compiling > everything as like make command instead of just doing mod_voicemail. **** > > Can someone suggest why is this happening and what can be done to just > compile mod_voicemail. Many thanks.**** > > ** ** > > Regards,**** > > Archana**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130221/8b5b4e7e/attachment.html From a.venugopan at mundio.com Thu Feb 21 12:30:10 2013 From: a.venugopan at mundio.com (Archana Venugopan) Date: Thu, 21 Feb 2013 09:30:10 +0000 Subject: [Freeswitch-users] make mod_voicemail In-Reply-To: References: <592A9CF93E12394E8472A6CC66E66BF23AC9F1@Mail-Kilo.squay.com> Message-ID: <592A9CF93E12394E8472A6CC66E66BF23ACA1B@Mail-Kilo.squay.com> I am afraid if that is as well installing everything so I just stopped it. [root at sme-xswitch freeswitch]# make mod_voicemail-install quiet_libtool: compile: gcc -DHAVE_CONFIG_H -I. -I./src/include -I./libs/xmlrpc-c -I/usr/local/src/freeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -pthread -DLINUX=2 -D_REENTRANT -D_GNU_SOURCE -D_LARGEFILE64_SOURCE -I/usr/local/src/freeswitch/libs/apr/include -I/usr/local/src/freeswitch/libs/apr-util/include -I/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib -I/usr/local/src/freeswitch/libs/libtpl-1.5/src -I/usr/local/src/freeswitch/libs/stfu -I/usr/local/src/freeswitch/libs/sqlite -I/usr/local/src/freeswitch/libs/pcre -I/usr/local/src/freeswitch/libs/speex/include -Ilibs/speex/include -I/usr/local/src/freeswitch/libs/srtp/include -I/usr/local/src/freeswitch/libs/srtp/crypto/include -Ilibs/srtp/crypto/include -I/usr/local/src/freeswitch/libs/spandsp/src -I/usr/local/src/freeswitch/libs/tiff-4.0.2/libtiff -DENABLE_SRTP -DSWITCH_HAVE_ODBC -I/usr/include -I/usr/local/src/freeswitch/libs/libedit/src -DSWITCH_HAVE_LIBEDIT -I/usr/local/src/freeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -Wall -std=c99 -pedantic -Wdeclaration-after-statement -g -O2 -MT libfreeswitch_la-switch_ivr.lo -MD -MP -MF .deps/libfreeswitch_la-switch_ivr.Tpo -c src/switch_ivr.c -fPIC -DPIC -o .libs/libfreeswitch_la-switch_ivr.o quiet_libtool: compile: gcc -DHAVE_CONFIG_H -I. -I./src/include -I./libs/xmlrpc-c -I/usr/local/src/freeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -pthread -DLINUX=2 -D_REENTRANT -D_GNU_SOURCE -D_LARGEFILE64_SOURCE -I/usr/local/src/freeswitch/libs/apr/include -I/usr/local/src/freeswitch/libs/apr-util/include -I/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib -I/usr/local/src/freeswitch/libs/libtpl-1.5/src -I/usr/local/src/freeswitch/libs/stfu -I/usr/local/src/freeswitch/libs/sqlite -I/usr/local/src/freeswitch/libs/pcre -I/usr/local/src/freeswitch/libs/speex/include -Ilibs/speex/include -I/usr/local/src/freeswitch/libs/srtp/include -I/usr/local/src/freeswitch/libs/srtp/crypto/include -Ilibs/srtp/crypto/include -I/usr/local/src/freeswitch/libs/spandsp/src -I/usr/local/src/freeswitch/libs/tiff-4.0.2/libtiff -DENABLE_SRTP -DSWITCH_HAVE_ODBC -I/usr/include -I/usr/local/src/freeswitch/libs/libedit/src -DSWITCH_HAVE_LIBEDIT -I/usr/local/src/freeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -Wall -std=c99 -pedantic -Wdeclaration-after-statement -g -O2 -MT libfreeswitch_la-switch_ivr.lo -MD -MP -MF .deps/libfreeswitch_la-switch_ivr.Tpo -c src/switch_ivr.c -o libfreeswitch_la-switch_ivr.o >/dev/null 2>&1 ^Cmake[1]: *** [libfreeswitch_la-switch_ivr.lo] Error 1 make: *** [mod_voicemail-install] Interrupt Regards, Archana From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi Marcus Sent: 21 February 2013 09:24 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] make mod_voicemail Try: make mod_voicemail-install -Avi Marcus On Thu, Feb 21, 2013 at 11:10 AM, Archana Venugopan > wrote: Hi, When I try to give 'make mod_voicemail' command, I see its compiling everything as like make command instead of just doing mod_voicemail. Can someone suggest why is this happening and what can be done to just compile mod_voicemail. Many thanks. Regards, Archana _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130221/6ebffe9b/attachment-0001.html From POlsson at enghouse.com Thu Feb 21 12:37:56 2013 From: POlsson at enghouse.com (Peter Olsson) Date: Thu, 21 Feb 2013 09:37:56 +0000 Subject: [Freeswitch-users] make mod_voicemail Message-ID: <1FFF97C269757C458224B7C895F35F1522A507@cantor.std.visionutv.se> So, was everything else built already? I believe FS will build the dependencies, if they were not built already - I'm not totally sure about that though. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Archana Venugopan Skickat: den 21 februari 2013 10:30 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] make mod_voicemail I am afraid if that is as well installing everything so I just stopped it. [root at sme-xswitch freeswitch]# make mod_voicemail-install quiet_libtool: compile: gcc -DHAVE_CONFIG_H -I. -I./src/include -I./libs/xmlrpc-c -I/usr/local/src/freeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -pthread -DLINUX=2 -D_REENTRANT -D_GNU_SOURCE -D_LARGEFILE64_SOURCE -I/usr/local/src/freeswitch/libs/apr/include -I/usr/local/src/freeswitch/libs/apr-util/include -I/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib -I/usr/local/src/freeswitch/libs/libtpl-1.5/src -I/usr/local/src/freeswitch/libs/stfu -I/usr/local/src/freeswitch/libs/sqlite -I/usr/local/src/freeswitch/libs/pcre -I/usr/local/src/freeswitch/libs/speex/include -Ilibs/speex/include -I/usr/local/src/freeswitch/libs/srtp/include -I/usr/local/src/freeswitch/libs/srtp/crypto/include -Ilibs/srtp/crypto/include -I/usr/local/src/freeswitch/libs/spandsp/src -I/usr/local/src/freeswitch/libs/tiff-4.0.2/libtiff -DENABLE_SRTP -DSWITCH_HAVE_ODBC -I/usr/include -I/usr/local/src/freeswitch/libs/libedit/src -DSWITCH_HAVE_LIBEDIT -I/usr/local/src/freeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -Wall -std=c99 -pedantic -Wdeclaration-after-statement -g -O2 -MT libfreeswitch_la-switch_ivr.lo -MD -MP -MF .deps/libfreeswitch_la-switch_ivr.Tpo -c src/switch_ivr.c -fPIC -DPIC -o .libs/libfreeswitch_la-switch_ivr.o quiet_libtool: compile: gcc -DHAVE_CONFIG_H -I. -I./src/include -I./libs/xmlrpc-c -I/usr/local/src/freeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -pthread -DLINUX=2 -D_REENTRANT -D_GNU_SOURCE -D_LARGEFILE64_SOURCE -I/usr/local/src/freeswitch/libs/apr/include -I/usr/local/src/freeswitch/libs/apr-util/include -I/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib -I/usr/local/src/freeswitch/libs/libtpl-1.5/src -I/usr/local/src/freeswitch/libs/stfu -I/usr/local/src/freeswitch/libs/sqlite -I/usr/local/src/freeswitch/libs/pcre -I/usr/local/src/freeswitch/libs/speex/include -Ilibs/speex/include -I/usr/local/src/freeswitch/libs/srtp/include -I/usr/local/src/freeswitch/libs/srtp/crypto/include -Ilibs/srtp/crypto/include -I/usr/local/src/freeswitch/libs/spandsp/src -I/usr/local/src/freeswitch/libs/tiff-4.0.2/libtiff -DENABLE_SRTP -DSWITCH_HAVE_ODBC -I/usr/include -I/usr/local/src/freeswitch/libs/libedit/src -DSWITCH_HAVE_LIBEDIT -I/usr/local/src/freeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -Wall -std=c99 -pedantic -Wdeclaration-after-statement -g -O2 -MT libfreeswitch_la-switch_ivr.lo -MD -MP -MF .deps/libfreeswitch_la-switch_ivr.Tpo -c src/switch_ivr.c -o libfreeswitch_la-switch_ivr.o >/dev/null 2>&1 ^Cmake[1]: *** [libfreeswitch_la-switch_ivr.lo] Error 1 make: *** [mod_voicemail-install] Interrupt Regards, Archana From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi Marcus Sent: 21 February 2013 09:24 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] make mod_voicemail Try: make mod_voicemail-install -Avi Marcus On Thu, Feb 21, 2013 at 11:10 AM, Archana Venugopan > wrote: Hi, When I try to give 'make mod_voicemail' command, I see its compiling everything as like make command instead of just doing mod_voicemail. Can someone suggest why is this happening and what can be done to just compile mod_voicemail. Many thanks. Regards, Archana _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 !DSPAM:5125e59032761278114654! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130221/87f1c664/attachment.html From a.venugopan at mundio.com Thu Feb 21 12:46:49 2013 From: a.venugopan at mundio.com (Archana Venugopan) Date: Thu, 21 Feb 2013 09:46:49 +0000 Subject: [Freeswitch-users] make mod_voicemail In-Reply-To: <1FFF97C269757C458224B7C895F35F1522A507@cantor.std.visionutv.se> References: <1FFF97C269757C458224B7C895F35F1522A507@cantor.std.visionutv.se> Message-ID: <592A9CF93E12394E8472A6CC66E66BF23ACA44@Mail-Kilo.squay.com> Ya, this is being carried out in our production server. And it was built years before I suppose. So am afraid to proceed as well with make mod_voicemail commands, if it compiles everything then our production set up will change completely. But I made few changes in voicemail and I need to compile mod_voicemail part alone. Please do help me in this. Regards, Archana From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peter Olsson Sent: 21 February 2013 09:38 To: 'FreeSWITCH Users Help' Subject: Re: [Freeswitch-users] make mod_voicemail So, was everything else built already? I believe FS will build the dependencies, if they were not built already - I'm not totally sure about that though. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Archana Venugopan Skickat: den 21 februari 2013 10:30 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] make mod_voicemail I am afraid if that is as well installing everything so I just stopped it. [root at sme-xswitch freeswitch]# make mod_voicemail-install quiet_libtool: compile: gcc -DHAVE_CONFIG_H -I. -I./src/include -I./libs/xmlrpc-c -I/usr/local/src/freeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -pthread -DLINUX=2 -D_REENTRANT -D_GNU_SOURCE -D_LARGEFILE64_SOURCE -I/usr/local/src/freeswitch/libs/apr/include -I/usr/local/src/freeswitch/libs/apr-util/include -I/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib -I/usr/local/src/freeswitch/libs/libtpl-1.5/src -I/usr/local/src/freeswitch/libs/stfu -I/usr/local/src/freeswitch/libs/sqlite -I/usr/local/src/freeswitch/libs/pcre -I/usr/local/src/freeswitch/libs/speex/include -Ilibs/speex/include -I/usr/local/src/freeswitch/libs/srtp/include -I/usr/local/src/freeswitch/libs/srtp/crypto/include -Ilibs/srtp/crypto/include -I/usr/local/src/freeswitch/libs/spandsp/src -I/usr/local/src/freeswitch/libs/tiff-4.0.2/libtiff -DENABLE_SRTP -DSWITCH_HAVE_ODBC -I/usr/include -I/usr/local/src/freeswitch/libs/libedit/src -DSWITCH_HAVE_LIBEDIT -I/usr/local/src/freeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -Wall -std=c99 -pedantic -Wdeclaration-after-statement -g -O2 -MT libfreeswitch_la-switch_ivr.lo -MD -MP -MF .deps/libfreeswitch_la-switch_ivr.Tpo -c src/switch_ivr.c -fPIC -DPIC -o .libs/libfreeswitch_la-switch_ivr.o quiet_libtool: compile: gcc -DHAVE_CONFIG_H -I. -I./src/include -I./libs/xmlrpc-c -I/usr/local/src/freeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -pthread -DLINUX=2 -D_REENTRANT -D_GNU_SOURCE -D_LARGEFILE64_SOURCE -I/usr/local/src/freeswitch/libs/apr/include -I/usr/local/src/freeswitch/libs/apr-util/include -I/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib -I/usr/local/src/freeswitch/libs/libtpl-1.5/src -I/usr/local/src/freeswitch/libs/stfu -I/usr/local/src/freeswitch/libs/sqlite -I/usr/local/src/freeswitch/libs/pcre -I/usr/local/src/freeswitch/libs/speex/include -Ilibs/speex/include -I/usr/local/src/freeswitch/libs/srtp/include -I/usr/local/src/freeswitch/libs/srtp/crypto/include -Ilibs/srtp/crypto/include -I/usr/local/src/freeswitch/libs/spandsp/src -I/usr/local/src/freeswitch/libs/tiff-4.0.2/libtiff -DENABLE_SRTP -DSWITCH_HAVE_ODBC -I/usr/include -I/usr/local/src/freeswitch/libs/libedit/src -DSWITCH_HAVE_LIBEDIT -I/usr/local/src/freeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -Wall -std=c99 -pedantic -Wdeclaration-after-statement -g -O2 -MT libfreeswitch_la-switch_ivr.lo -MD -MP -MF .deps/libfreeswitch_la-switch_ivr.Tpo -c src/switch_ivr.c -o libfreeswitch_la-switch_ivr.o >/dev/null 2>&1 ^Cmake[1]: *** [libfreeswitch_la-switch_ivr.lo] Error 1 make: *** [mod_voicemail-install] Interrupt Regards, Archana From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi Marcus Sent: 21 February 2013 09:24 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] make mod_voicemail Try: make mod_voicemail-install -Avi Marcus On Thu, Feb 21, 2013 at 11:10 AM, Archana Venugopan > wrote: Hi, When I try to give 'make mod_voicemail' command, I see its compiling everything as like make command instead of just doing mod_voicemail. Can someone suggest why is this happening and what can be done to just compile mod_voicemail. Many thanks. Regards, Archana _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 !DSPAM:5125e59032761278114654! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130221/48de8fac/attachment-0001.html From avi at avimarcus.net Thu Feb 21 12:54:03 2013 From: avi at avimarcus.net (Avi Marcus) Date: Thu, 21 Feb 2013 11:54:03 +0200 Subject: [Freeswitch-users] make mod_voicemail In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF23ACA44@Mail-Kilo.squay.com> References: <1FFF97C269757C458224B7C895F35F1522A507@cantor.std.visionutv.se> <592A9CF93E12394E8472A6CC66E66BF23ACA44@Mail-Kilo.squay.com> Message-ID: If you updated GIT since your last compile, then all the dependencies have changed... -Avi On Thu, Feb 21, 2013 at 11:46 AM, Archana Venugopan wrote: > Ya, this is being carried out in our production server. And it was built > years before I suppose. So am afraid to proceed as well with make > mod_voicemail commands, if it compiles everything then our production set > up will change completely.**** > > But I made few changes in voicemail and I need to compile mod_voicemail > part alone. Please do help me in this.**** > > ** ** > > Regards,**** > > Archana**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Peter Olsson > *Sent:* 21 February 2013 09:38 > > *To:* 'FreeSWITCH Users Help' > *Subject:* Re: [Freeswitch-users] make mod_voicemail**** > > ** ** > > So, was everything else built already? I believe FS will build the > dependencies, if they were not built already ? I?m not totally sure about > that though.**** > > **** > > /Peter**** > > **** > > *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *F?r *Archana Venugopan > *Skickat:* den 21 februari 2013 10:30 > *Till:* FreeSWITCH Users Help > *?mne:* Re: [Freeswitch-users] make mod_voicemail**** > > **** > > I am afraid if that is as well installing everything so I just stopped it.. > **** > > **** > > [root at sme-xswitch freeswitch]# make mod_voicemail-install**** > > quiet_libtool: compile: gcc -DHAVE_CONFIG_H -I. -I./src/include > -I./libs/xmlrpc-c -I/usr/local/src/freeswitch/libs/curl/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden > -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -pthread -DLINUX=2 > -D_REENTRANT -D_GNU_SOURCE -D_LARGEFILE64_SOURCE > -I/usr/local/src/freeswitch/libs/apr/include > -I/usr/local/src/freeswitch/libs/apr-util/include > -I/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib > -I/usr/local/src/freeswitch/libs/libtpl-1.5/src > -I/usr/local/src/freeswitch/libs/stfu > -I/usr/local/src/freeswitch/libs/sqlite > -I/usr/local/src/freeswitch/libs/pcre > -I/usr/local/src/freeswitch/libs/speex/include -Ilibs/speex/include > -I/usr/local/src/freeswitch/libs/srtp/include > -I/usr/local/src/freeswitch/libs/srtp/crypto/include > -Ilibs/srtp/crypto/include -I/usr/local/src/freeswitch/libs/spandsp/src > -I/usr/local/src/freeswitch/libs/tiff-4.0.2/libtiff -DENABLE_SRTP > -DSWITCH_HAVE_ODBC -I/usr/include > -I/usr/local/src/freeswitch/libs/libedit/src -DSWITCH_HAVE_LIBEDIT > -I/usr/local/src/freeswitch/libs/curl/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -Werror > -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb > -DHAVE_OPENSSL -Wall -std=c99 -pedantic -Wdeclaration-after-statement -g > -O2 -MT libfreeswitch_la-switch_ivr.lo -MD -MP -MF > .deps/libfreeswitch_la-switch_ivr.Tpo -c src/switch_ivr.c -fPIC -DPIC -o > .libs/libfreeswitch_la-switch_ivr.o**** > > quiet_libtool: compile: gcc -DHAVE_CONFIG_H -I. -I./src/include > -I./libs/xmlrpc-c -I/usr/local/src/freeswitch/libs/curl/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden > -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -pthread -DLINUX=2 > -D_REENTRANT -D_GNU_SOURCE -D_LARGEFILE64_SOURCE > -I/usr/local/src/freeswitch/libs/apr/include > -I/usr/local/src/freeswitch/libs/apr-util/include > -I/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib > -I/usr/local/src/freeswitch/libs/libtpl-1.5/src > -I/usr/local/src/freeswitch/libs/stfu > -I/usr/local/src/freeswitch/libs/sqlite > -I/usr/local/src/freeswitch/libs/pcre > -I/usr/local/src/freeswitch/libs/speex/include -Ilibs/speex/include > -I/usr/local/src/freeswitch/libs/srtp/include > -I/usr/local/src/freeswitch/libs/srtp/crypto/include > -Ilibs/srtp/crypto/include -I/usr/local/src/freeswitch/libs/spandsp/src > -I/usr/local/src/freeswitch/libs/tiff-4.0.2/libtiff -DENABLE_SRTP > -DSWITCH_HAVE_ODBC -I/usr/include > -I/usr/local/src/freeswitch/libs/libedit/src -DSWITCH_HAVE_LIBEDIT > -I/usr/local/src/freeswitch/libs/curl/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -Werror > -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb > -DHAVE_OPENSSL -Wall -std=c99 -pedantic -Wdeclaration-after-statement -g > -O2 -MT libfreeswitch_la-switch_ivr.lo -MD -MP -MF > .deps/libfreeswitch_la-switch_ivr.Tpo -c src/switch_ivr.c -o > libfreeswitch_la-switch_ivr.o >/dev/null 2>&1**** > > ^Cmake[1]: *** [libfreeswitch_la-switch_ivr.lo] Error 1**** > > make: *** [mod_voicemail-install] Interrupt**** > > **** > > Regards,**** > > Archana**** > > **** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [ > mailto:freeswitch-users-bounces at lists.freeswitch.org] > *On Behalf Of *Avi Marcus > *Sent:* 21 February 2013 09:24 > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] make mod_voicemail**** > > **** > > Try:**** > > > > > **** > > make mod_voicemail-install**** > > > **** > > -Avi Marcus**** > > **** > > On Thu, Feb 21, 2013 at 11:10 AM, Archana Venugopan < > a.venugopan at mundio.com> wrote:**** > > Hi,**** > > **** > > When I try to give ?make mod_voicemail? command, I see its compiling > everything as like make command instead of just doing mod_voicemail. **** > > Can someone suggest why is this happening and what can be done to just > compile mod_voicemail. Many thanks.**** > > **** > > Regards,**** > > Archana**** > > **** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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**** > > **** > > !DSPAM:5125e59032761278114654! **** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130221/21f2b80d/attachment.html From a.venugopan at mundio.com Thu Feb 21 13:03:27 2013 From: a.venugopan at mundio.com (Archana Venugopan) Date: Thu, 21 Feb 2013 10:03:27 +0000 Subject: [Freeswitch-users] make mod_voicemail In-Reply-To: References: <1FFF97C269757C458224B7C895F35F1522A507@cantor.std.visionutv.se> <592A9CF93E12394E8472A6CC66E66BF23ACA44@Mail-Kilo.squay.com> Message-ID: <592A9CF93E12394E8472A6CC66E66BF23ACA6D@Mail-Kilo.squay.com> Hi, I haven't updated GIT recently. Freeswitch version I am using is 1.0.6. The same version am using in my test environment as well and there if I give make mod_voicemail-install, only mod_voicemail is being compiled. I face this issue in my production server only. Regards, Archana From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi Marcus Sent: 21 February 2013 09:54 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] make mod_voicemail If you updated GIT since your last compile, then all the dependencies have changed... -Avi On Thu, Feb 21, 2013 at 11:46 AM, Archana Venugopan > wrote: Ya, this is being carried out in our production server. And it was built years before I suppose. So am afraid to proceed as well with make mod_voicemail commands, if it compiles everything then our production set up will change completely. But I made few changes in voicemail and I need to compile mod_voicemail part alone. Please do help me in this. Regards, Archana From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peter Olsson Sent: 21 February 2013 09:38 To: 'FreeSWITCH Users Help' Subject: Re: [Freeswitch-users] make mod_voicemail So, was everything else built already? I believe FS will build the dependencies, if they were not built already - I'm not totally sure about that though. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Archana Venugopan Skickat: den 21 februari 2013 10:30 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] make mod_voicemail I am afraid if that is as well installing everything so I just stopped it. [root at sme-xswitch freeswitch]# make mod_voicemail-install quiet_libtool: compile: gcc -DHAVE_CONFIG_H -I. -I./src/include -I./libs/xmlrpc-c -I/usr/local/src/freeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -pthread -DLINUX=2 -D_REENTRANT -D_GNU_SOURCE -D_LARGEFILE64_SOURCE -I/usr/local/src/freeswitch/libs/apr/include -I/usr/local/src/freeswitch/libs/apr-util/include -I/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib -I/usr/local/src/freeswitch/libs/libtpl-1.5/src -I/usr/local/src/freeswitch/libs/stfu -I/usr/local/src/freeswitch/libs/sqlite -I/usr/local/src/freeswitch/libs/pcre -I/usr/local/src/freeswitch/libs/speex/include -Ilibs/speex/include -I/usr/local/src/freeswitch/libs/srtp/include -I/usr/local/src/freeswitch/libs/srtp/crypto/include -Ilibs/srtp/crypto/include -I/usr/local/src/freeswitch/libs/spandsp/src -I/usr/local/src/freeswitch/libs/tiff-4.0.2/libtiff -DENABLE_SRTP -DSWITCH_HAVE_ODBC -I/usr/include -I/usr/local/src/freeswitch/libs/libedit/src -DSWITCH_HAVE_LIBEDIT -I/usr/local/src/freeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -Wall -std=c99 -pedantic -Wdeclaration-after-statement -g -O2 -MT libfreeswitch_la-switch_ivr.lo -MD -MP -MF .deps/libfreeswitch_la-switch_ivr.Tpo -c src/switch_ivr.c -fPIC -DPIC -o .libs/libfreeswitch_la-switch_ivr.o quiet_libtool: compile: gcc -DHAVE_CONFIG_H -I. -I./src/include -I./libs/xmlrpc-c -I/usr/local/src/freeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -pthread -DLINUX=2 -D_REENTRANT -D_GNU_SOURCE -D_LARGEFILE64_SOURCE -I/usr/local/src/freeswitch/libs/apr/include -I/usr/local/src/freeswitch/libs/apr-util/include -I/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib -I/usr/local/src/freeswitch/libs/libtpl-1.5/src -I/usr/local/src/freeswitch/libs/stfu -I/usr/local/src/freeswitch/libs/sqlite -I/usr/local/src/freeswitch/libs/pcre -I/usr/local/src/freeswitch/libs/speex/include -Ilibs/speex/include -I/usr/local/src/freeswitch/libs/srtp/include -I/usr/local/src/freeswitch/libs/srtp/crypto/include -Ilibs/srtp/crypto/include -I/usr/local/src/freeswitch/libs/spandsp/src -I/usr/local/src/freeswitch/libs/tiff-4.0.2/libtiff -DENABLE_SRTP -DSWITCH_HAVE_ODBC -I/usr/include -I/usr/local/src/freeswitch/libs/libedit/src -DSWITCH_HAVE_LIBEDIT -I/usr/local/src/freeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -Wall -std=c99 -pedantic -Wdeclaration-after-statement -g -O2 -MT libfreeswitch_la-switch_ivr.lo -MD -MP -MF .deps/libfreeswitch_la-switch_ivr.Tpo -c src/switch_ivr.c -o libfreeswitch_la-switch_ivr.o >/dev/null 2>&1 ^Cmake[1]: *** [libfreeswitch_la-switch_ivr.lo] Error 1 make: *** [mod_voicemail-install] Interrupt Regards, Archana From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi Marcus Sent: 21 February 2013 09:24 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] make mod_voicemail Try: make mod_voicemail-install -Avi Marcus On Thu, Feb 21, 2013 at 11:10 AM, Archana Venugopan > wrote: Hi, When I try to give 'make mod_voicemail' command, I see its compiling everything as like make command instead of just doing mod_voicemail. Can someone suggest why is this happening and what can be done to just compile mod_voicemail. Many thanks. Regards, Archana _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 !DSPAM:5125e59032761278114654! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130221/59cedc26/attachment-0001.html From emamirazavi at gmail.com Thu Feb 21 13:24:19 2013 From: emamirazavi at gmail.com (Sayyed Mohammad Emami Razavi) Date: Thu, 21 Feb 2013 13:54:19 +0330 Subject: [Freeswitch-users] mod_php Message-ID: How to active again mod_php for using in IVR and take it from recent FS versions that it has existed? ... just like mod_lua? LUA is very beautiful but programming in PHP is easier for me! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130221/48918373/attachment.html From sertys at gmail.com Thu Feb 21 14:34:23 2013 From: sertys at gmail.com (Daniel Ivanov) Date: Thu, 21 Feb 2013 12:34:23 +0100 Subject: [Freeswitch-users] Determine if a number is valid/working In-Reply-To: <12c801ce0ffc$48f84b10$dae8e130$@bizfocused.com> References: <072701ce0ebf$c5808640$508192c0$@bizfocused.com> <0d0401ce0f74$aa9fb7a0$ffdf26e0$@bizfocused.com> <12c801ce0ffc$48f84b10$dae8e130$@bizfocused.com> Message-ID: The passive approach is always better to marketing people. They are generally afraid of torturing a possible cuatomer too much. The data comparison is an amazing guess given the data you have. I wouldn't do any lookups if there's a mismatch, but jump to phone verification right away. On Feb 21, 2013 7:31 AM, "Sean Devoy" wrote: > Hey MC. I like Steven?s response as well. Can?t quite do the connect to > sales agent yet in the process, need to pull credit report and load DBMS > values.**** > > ** ** > > The cost number is cost of lead from providers like google, msn, etc. It > is a VERY competitive market. That does not count the cost of sales reps > dialing non-working numbers on about 1/3 of the leads.**** > > ** ** > > I suspect we will offer a tiered approach like:**** > > **1. **Area Code, ?State? Field and IP Geo Data State Field all match ? > accept it.**** > > **2. **Else try LNP Lookup for match**** > > **3. **Else still fishy ? Web Popup to explain and then Robo Dial and > require a ?pin? to be entered.**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael > Collins > *Sent:* Wednesday, February 20, 2013 12:26 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Determine if a number is valid/working** > ** > > ** ** > > ** ** > > On Wed, Feb 20, 2013 at 6:14 AM, Sean Devoy wrote: > **** > > Thank you for your responses, everyone.**** > > **** > > MC: This customer has rejected the idea of calling with a message as bad > marketing!! In my experience marketing people have their own ?logic rules? > and I can never follow their thinking. And yes, they have 30 customer > sales reps calling these ?leads? all day long. They claim 1/3 of the leads > have invalid phone numbers. My customer suspects fraud and that someone is > submitting false leads to boost their lead rate and fees.**** > > I agree with Steven's assessment (elsewhere in this thread). > **** > > **** > > Ken: I am not sure we have any language in common ? lol. Where can I > learn what ?ASR and ACD? are? I looked up lidb and lnp already!**** > > **** > > Also, the cost of a call compared to the cost of a fraudulent lead (over > $20) is insignificant. **** > > How is that cost calculated and to whom does the money go? Maybe the > "verification call" needs to take place at the time that the form is filled > out. > **** > > **** > > The ideas I have suggested to my customer, for what it is worth, are > comparing the State (we are in US) for the IP address of the submitter, to > the State on the form and the State of the area code on the form. In the > situations where they do not agree, prompt the user for permission to robo > call for verification.**** > > **** > > ** ** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130221/a54ad291/attachment.html From a.venugopan at mundio.com Thu Feb 21 15:13:35 2013 From: a.venugopan at mundio.com (Archana Venugopan) Date: Thu, 21 Feb 2013 12:13:35 +0000 Subject: [Freeswitch-users] FW: FW: Event pick Message-ID: <592A9CF93E12394E8472A6CC66E66BF23ACB84@Mail-Kilo.squay.com> Hi, Ya bridge is in dialplan XML only. In that case how am I to trace the termination event and run a DB query? Is there a way to do? Thanks. Regards, Archana From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 20 February 2013 18:07 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] FW: Event pick If you bridge in the dialplan then your Lua script won't run until the bridge app has terminated. -MC On Wed, Feb 20, 2013 at 9:33 AM, Archana Venugopan > wrote: Hi, I should define a obsession before going to obcause. But my outbound call is already initiated from my dialplan XML []. Am not sure how to give obsession in lua script as my outbound call is initiated from dialplan XML. Can you please guide me. Many thanks From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 20 February 2013 17:12 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Event pick What value is in obCause when the originator cancels the call? Print the value in the else block and find out. -MC On Wed, Feb 20, 2013 at 4:37 AM, Archana Venugopan > wrote: Hi, I need to capture this ORIGINATOR _CANCEL event in my lua script. >From google I found the below code. But we establish our calls from dialplan XML as like this, . Since am not that well versed in lua script, I am not sure in the place of obsession below what should I pass. As far I understood the =sofia/external/0${msc_prefix}$1$2 at 10.30.3.27, this is what I should put there but we already have this in our XML. Please suggest me on how should I proceed. Many thanks. -- Initiate an outbound call obSession = freeswitch.Session("sofia/192.168.0.4/1002") -- Check to see if the call was answered if obSession:ready() then -- Do something good here else -- This means the call was not answered ... Check for the reason local obCause = obSession:hangupCause() freeswitch.consoleLog("info", "obSession:hangupCause() = " .. obCause ) if ( obCause == "USER_BUSY" ) then -- SIP 486 -- For BUSY you may reschedule the call for later elseif ( obCause == "NO_ANSWER" ) then -- Call them back in an hour elseif ( obCause == "ORIGINATOR_CANCEL" ) then -- SIP 487 -- May need to check for network congestion or problems else -- Log these issues end end 2013-02-19 17:56:02.277173 [NOTICE] sofia.c:7082 Hangup sofia/internal/206 at fsfailover.uk01.com [CS_EXECUTE] [ORIGINATOR_CANCEL] Regards, Archana _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130221/328006e1/attachment-0001.html From david.villasmil.work at gmail.com Thu Feb 21 15:14:09 2013 From: david.villasmil.work at gmail.com (David Villasmil) Date: Thu, 21 Feb 2013 13:14:09 +0100 Subject: [Freeswitch-users] mod_php In-Reply-To: References: Message-ID: Hello, You should learn lua, it is VERY simple and very powerful. Also it is embeded in FS, so resource-wise it is better... IF you already program in PHP, lua will be no problem for you. David On Thu, Feb 21, 2013 at 11:24 AM, Sayyed Mohammad Emami Razavi < emamirazavi at gmail.com> wrote: > How to active again mod_php for using in IVR and take it from recent FS > versions that it has existed? ... just like mod_lua? LUA is very beautiful > but programming in PHP is easier for me! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130221/a42c479d/attachment.html From david.villasmil.work at gmail.com Thu Feb 21 15:17:56 2013 From: david.villasmil.work at gmail.com (David Villasmil) Date: Thu, 21 Feb 2013 13:17:56 +0100 Subject: [Freeswitch-users] mod-lua Vs mod-java In-Reply-To: <5125BDA4.20505@gmail.com> References: <5125BDA4.20505@gmail.com> Message-ID: lua is by far the best approach. I really don't know about accessing web services, but for db it is perfect. I wrote a whole wholesale and calling card app with lua, so you can pretty much do anything wit it. Also, lua has a SOAP interface: http://tomasguisasola.github.com/luasoap/ Good luck! David On Thu, Feb 21, 2013 at 7:24 AM, Mimiko wrote: > On 21.02.2013 02:02, JP wrote: > > Hi, > > I want to access a webservice that exposes a SOAP interface. I was > > wondering if this is better to do in Lua or in Java. Any suggestions > > will be greatly appreciated. Also in general is it better to use > > mod-java instead of mod-lua for doing advanced IVR functions like > > database access or web service access? Are there any drawbacks to using > > java instead of Lua? > > > > Thanks, > > JP > > I had a conversation with a member on this list about what to use for > scripting: lua, perl. java. Ended up that lua is the fastest in its own. > For doing simple things, even IVR's, lua is enough. And lua is installed > with FS. > > I use lua to connect to DB and don't see any performance drop. > > -- > Mimiko desu. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130221/7c0896e2/attachment.html From david.villasmil.work at gmail.com Thu Feb 21 15:49:08 2013 From: david.villasmil.work at gmail.com (David Villasmil) Date: Thu, 21 Feb 2013 13:49:08 +0100 Subject: [Freeswitch-users] nibblebill custom_sql_save In-Reply-To: References: Message-ID: Hello, Thanks for replying! It was not set, but i understood from the wiki that if you don't want any rounding, you dont't set it. I set it to "10", i want the user to be charged tvery 10 seconds, you nibblebil is charging 10 DOLLARS! I must be getting something wrong... Here's my nibblebill.conf.xml. (BTW, the two highlighted lines, is it possible to do that? I want the custom SQLs to use channel variables, is it possible?)
** * *
Here's the dialplan:
Thanks for your help! David On Wed, Feb 20, 2013 at 2:23 PM, Muhammad Shahzad wrote: > Did you set nibble_increment variable? e.g. > > > > Here is its details, > > http://wiki.freeswitch.org/wiki/Mod_nibblebill#Nibble_Method_.28Default.29 > > Thank you. > > > On Wed, Feb 20, 2013 at 2:10 PM, David Villasmil < > david.villasmil.work at gmail.com> wrote: > >> Anyone about this? >> >> Thanks >> >> >> On Mon, Feb 18, 2013 at 2:07 PM, David Villasmil < >> david.villasmil.work at gmail.com> wrote: >> >>> Hello guys, >>> >>> I have the following config for nibbebill: >>> >>> >>> >>> >>> I see this on the log, which is obviously failing: >>> >>> >>> [UPDATE cards_table SET balance=balance- WHERE pin=12345;] >>> >>> is ${nibble_increment} not the correct variable to use? It looks like it >>> is empty! >>> >>> Thanks! >>> >>> David >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +49 176 99 83 10 85 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130221/7ec39569/attachment-0001.html From shaheryarkh at gmail.com Thu Feb 21 18:43:23 2013 From: shaheryarkh at gmail.com (Muhammad Shahzad) Date: Thu, 21 Feb 2013 16:43:23 +0100 Subject: [Freeswitch-users] nibblebill custom_sql_save In-Reply-To: References: Message-ID: Not sure what you mean below. You want to charge every 10 seconds (billing interval) or you want to charge 10 cents per minute (billing rate)? Thank you. On Thu, Feb 21, 2013 at 1:49 PM, David Villasmil < david.villasmil.work at gmail.com> wrote: > Hello, > > Thanks for replying! > > It was not set, but i understood from the wiki that if you don't want any > rounding, you dont't set it. > > I set it to "10", i want the user to be charged tvery 10 seconds, you > nibblebil is charging 10 DOLLARS! > I must be getting something wrong... > > > Here's my nibblebill.conf.xml. > > (BTW, the two highlighted lines, is it possible to do that? I want the > custom SQLs to use channel variables, is it possible?) > > >
> > > > > > > > > ** > * * > > > > > > > > > >
> > Here's the dialplan: > >
> > > > data="continue_on_fail=NO_ROUTE_DESTINATION,UNALLOCATED_NUMBER,407" /> > > data="sip_auth_username=1002" /> > > data="hangup_after_bridge=true" /> > > > data="card_db_table=cards_table" /> > data="card_db_column_cash=balance" /> > data="card_db_column_account=pin" /> > /> > data="gw=${distributor(TEST_ROUTE)}" /> > > data="execute_on_answer=nibblebill adjust -0.1000" /> > data="{enable_heartbeat_events=10,nibble_rate=0.0500,nibble_account=12345}sofia/gateway/${gw}/0013058883456" > /> > > > >
> > > Thanks for your help! > > > David > > > > On Wed, Feb 20, 2013 at 2:23 PM, Muhammad Shahzad wrote: > >> Did you set nibble_increment variable? e.g. >> >> >> >> Here is its details, >> >> http://wiki.freeswitch.org/wiki/Mod_nibblebill#Nibble_Method_.28Default.29 >> >> Thank you. >> >> >> On Wed, Feb 20, 2013 at 2:10 PM, David Villasmil < >> david.villasmil.work at gmail.com> wrote: >> >>> Anyone about this? >>> >>> Thanks >>> >>> >>> On Mon, Feb 18, 2013 at 2:07 PM, David Villasmil < >>> david.villasmil.work at gmail.com> wrote: >>> >>>> Hello guys, >>>> >>>> I have the following config for nibbebill: >>>> >>>> >>>> >>>> >>>> I see this on the log, which is obviously failing: >>>> >>>> >>>> [UPDATE cards_table SET balance=balance- WHERE pin=12345;] >>>> >>>> is ${nibble_increment} not the correct variable to use? It looks like >>>> it is empty! >>>> >>>> Thanks! >>>> >>>> David >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> >> -- >> Muhammad Shahzad >> ----------------------------------- >> CISCO Rich Media Communication Specialist (CRMCS) >> CISCO Certified Network Associate (CCNA) >> Cell: +49 176 99 83 10 85 >> MSN: shari_786pk at hotmail.com >> Email: shaheryarkh at googlemail.com >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +49 176 99 83 10 85 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130221/73280056/attachment.html From shaheryarkh at gmail.com Thu Feb 21 18:49:47 2013 From: shaheryarkh at gmail.com (Muhammad Shahzad) Date: Thu, 21 Feb 2013 16:49:47 +0100 Subject: [Freeswitch-users] nibblebill custom_sql_save In-Reply-To: References: Message-ID: Usually you do VOIP billing in so called "Units", Each unit equals 0.01 cent, meaning 1 USD (or whatever currency you are using in multi-currency billing system) equal to 10,000 VOIP units. Therefore, if you want to bill a call at rate of 10 cents per minute then you should set increment 1,000 units and call interval to 60 seconds. If you shorten the interval to say 6 seconds, then you will charge 100 units per 6 seconds, resulting in same billing as per minute, i.e. 60 / 6 x 100 = 1,000 units = 10 cents Thank you. On Thu, Feb 21, 2013 at 4:43 PM, Muhammad Shahzad wrote: > Not sure what you mean below. You want to charge every 10 seconds (billing > interval) or you want to charge 10 cents per minute (billing rate)? > > Thank you. > > > On Thu, Feb 21, 2013 at 1:49 PM, David Villasmil < > david.villasmil.work at gmail.com> wrote: > >> Hello, >> >> Thanks for replying! >> >> It was not set, but i understood from the wiki that if you don't want any >> rounding, you dont't set it. >> >> I set it to "10", i want the user to be charged tvery 10 seconds, you >> nibblebil is charging 10 DOLLARS! >> I must be getting something wrong... >> >> >> Here's my nibblebill.conf.xml. >> >> (BTW, the two highlighted lines, is it possible to do that? I want the >> custom SQLs to use channel variables, is it possible?) >> >> >>
>> >> >> >> >> >> >> >> >> ** >> * * >> >> >> >> >> >> >> >> >> >>
>> >> Here's the dialplan: >> >>
>> >> >> >> > data="continue_on_fail=NO_ROUTE_DESTINATION,UNALLOCATED_NUMBER,407" /> >> >> > data="sip_auth_username=1002" /> >> >> > data="hangup_after_bridge=true" /> >> >> > /> >> > data="card_db_table=cards_table" /> >> > data="card_db_column_cash=balance" /> >> > data="card_db_column_account=pin" /> >> > /> >> > data="gw=${distributor(TEST_ROUTE)}" /> >> >> > data="execute_on_answer=nibblebill adjust -0.1000" /> >> > data="{enable_heartbeat_events=10,nibble_rate=0.0500,nibble_account=12345}sofia/gateway/${gw}/0013058883456" >> /> >> >> >> >>
>> >> >> Thanks for your help! >> >> >> David >> >> >> >> On Wed, Feb 20, 2013 at 2:23 PM, Muhammad Shahzad wrote: >> >>> Did you set nibble_increment variable? e.g. >>> >>> >>> >>> Here is its details, >>> >>> >>> http://wiki.freeswitch.org/wiki/Mod_nibblebill#Nibble_Method_.28Default.29 >>> >>> Thank you. >>> >>> >>> On Wed, Feb 20, 2013 at 2:10 PM, David Villasmil < >>> david.villasmil.work at gmail.com> wrote: >>> >>>> Anyone about this? >>>> >>>> Thanks >>>> >>>> >>>> On Mon, Feb 18, 2013 at 2:07 PM, David Villasmil < >>>> david.villasmil.work at gmail.com> wrote: >>>> >>>>> Hello guys, >>>>> >>>>> I have the following config for nibbebill: >>>>> >>>>> >>>>> >>>>> >>>>> I see this on the log, which is obviously failing: >>>>> >>>>> >>>>> [UPDATE cards_table SET balance=balance- WHERE pin=12345;] >>>>> >>>>> is ${nibble_increment} not the correct variable to use? It looks like >>>>> it is empty! >>>>> >>>>> Thanks! >>>>> >>>>> David >>>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>>> >>>> >>> >>> >>> -- >>> Muhammad Shahzad >>> ----------------------------------- >>> CISCO Rich Media Communication Specialist (CRMCS) >>> CISCO Certified Network Associate (CCNA) >>> Cell: +49 176 99 83 10 85 >>> MSN: shari_786pk at hotmail.com >>> Email: shaheryarkh at googlemail.com >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +49 176 99 83 10 85 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +49 176 99 83 10 85 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130221/3b571231/attachment-0001.html From stuart.mills3 at btopenworld.com Thu Feb 21 19:49:00 2013 From: stuart.mills3 at btopenworld.com (Stuart Mills) Date: Thu, 21 Feb 2013 16:49:00 -0000 Subject: [Freeswitch-users] Outbound mod event bug Message-ID: <68FC21E54C0D40E986A880956375D5FF@PBPC> Hi All, I know some of you may not consider this as a bug, especially if things are done in the correct way this would never happen, but please read on and I'll explain the issue as I was a little surprised at how badly FreeSwitch handled this user error, seen as though it?s such a good piece of software. Basically, a call arrives into Freeswitch and a connection is established to a remote server, then the following relay occurs (in it?s simplest terms) - connect sent into Freeswitch FreeSwitch replies OK myevents sent into FreeSwitch FreeSwitch replies OK answer sent into FreeSwitch FreeSwitch replies OK bridge with arguments sent into FreeSwitch FreeSwitch replies OK Now, everything is perfect up until here and I'd say that as a call switching platform it?s doing exactly what is being asked of it ? however here?s the bit most of you would agree is wrong and isn?t a ?real? bug - if you reply again with another bridge, be it by mistake or a deliberate attempt to fool FreeSwitch, the system crashes and causes a core dump. I can get this to happen every time, no warning, no nothing, just a crash dump and FreeSwitch dies in front of my eyes. Answers on a postcard please ? try not to be too harsh. I am particularly new to FreeSwitch and find it very very good, I was just really surprised to fin a bug that could crash it. Kind Regards, Stuart -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130221/9500664b/attachment.html From bh at udev.org Thu Feb 21 15:35:53 2013 From: bh at udev.org (Benjamin Henrion) Date: Thu, 21 Feb 2013 13:35:53 +0100 Subject: [Freeswitch-users] mod-lua Vs mod-java In-Reply-To: References: <5125BDA4.20505@gmail.com> Message-ID: On Thu, Feb 21, 2013 at 1:17 PM, David Villasmil wrote: > lua is by far the best approach. I really don't know about accessing web > services, but for db it is perfect. > I wrote a whole wholesale and calling card app with lua, so you can pretty > much do anything wit it. Number of reusable libraries is quite limited compared to other languages. Lua is well documented on FS wiki, other languages are just undocumented. -- Benjamin Henrion FFII Brussels - +32-484-566109 - +32-2-3500762 "In July 2005, after several failed attempts to legalise software patents in Europe, the patent establishment changed its strategy. Instead of explicitly seeking to sanction the patentability of software, they are now seeking to create a central European patent court, which would establish and enforce patentability rules in their favor, without any possibility of correction by competing courts or democratically elected legislators." From steveayre at gmail.com Thu Feb 21 20:01:18 2013 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 21 Feb 2013 17:01:18 +0000 Subject: [Freeswitch-users] Outbound mod event bug In-Reply-To: <68FC21E54C0D40E986A880956375D5FF@PBPC> References: <68FC21E54C0D40E986A880956375D5FF@PBPC> Message-ID: A core dump is always a bug. Check if it happens on the latest git head (it may already be fixed). If it does it there collect a backtrack and file a Jira. http://wiki.freeswitch.org/wiki/Reporting_Bugs#Creating_A_Backtrace_With_gdb_.28Linux.2FUnix.29 Steve On 21 Feb 2013, at 16:49, "Stuart Mills" wrote: > Hi All, > > I know some of you may not consider this as a bug, especially if things are done in the correct way this would never happen, but please read on and I'll explain the issue as I was a little surprised at how badly FreeSwitch handled this user error, seen as though it?s such a good piece of software. > > Basically, a call arrives into Freeswitch and a connection is established to a remote server, then the following relay occurs (in it?s simplest terms) - > > connect sent into Freeswitch > FreeSwitch replies OK > myevents sent into FreeSwitch > FreeSwitch replies OK > answer sent into FreeSwitch > FreeSwitch replies OK > bridge with arguments sent into FreeSwitch > FreeSwitch replies OK > > Now, everything is perfect up until here and I'd say that as a call switching platform it?s doing exactly what is being asked of it ? however here?s the bit most of you would agree is wrong and isn?t a ?real? bug - if you reply again with another bridge, be it by mistake or a deliberate attempt to fool FreeSwitch, the system crashes and causes a core dump. > > I can get this to happen every time, no warning, no nothing, just a crash dump and FreeSwitch dies in front of my eyes. > > Answers on a postcard please ? try not to be too harsh. I am particularly new to FreeSwitch and find it very very good, I was just really surprised to fin a bug that could crash it. > > Kind Regards, > > Stuart > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130221/a7318ae7/attachment.html From ira at connectmevoice.com Thu Feb 21 20:10:57 2013 From: ira at connectmevoice.com (Ira Tessler) Date: Thu, 21 Feb 2013 12:10:57 -0500 Subject: [Freeswitch-users] Nat Issue with AT&T Uverse Message-ID: We are running Freeswitch in a hosted environment. We have a customer that uses AT&T U-Verse as an ISP and they have theirstandard router. Attached is an example of a call that after about 1600 seconds the session timer re-invite does not get a response and the audio dies. Does anyone have any experience with hosted Freeswitch and A&T U-Verse that can help? Thanks, Ira Tessler Lead Software Engineer ConnectMe (732) 490-9007 x2 ira at connectmevoice.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130221/90398567/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: jason bad router.zip Type: application/zip Size: 20241007 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130221/90398567/attachment-0001.zip From david.villasmil.work at gmail.com Thu Feb 21 21:00:03 2013 From: david.villasmil.work at gmail.com (David Villasmil) Date: Thu, 21 Feb 2013 19:00:03 +0100 Subject: [Freeswitch-users] mod-lua Vs mod-java In-Reply-To: References: <5125BDA4.20505@gmail.com> Message-ID: Well that depends on what you intent to accomplish, I guess. The nicest thing about lua (in our case) is that it is embeded inside FS. David On Thu, Feb 21, 2013 at 1:35 PM, Benjamin Henrion wrote: > On Thu, Feb 21, 2013 at 1:17 PM, David Villasmil > wrote: > > lua is by far the best approach. I really don't know about accessing web > > services, but for db it is perfect. > > I wrote a whole wholesale and calling card app with lua, so you can > pretty > > much do anything wit it. > > Number of reusable libraries is quite limited compared to other languages. > > Lua is well documented on FS wiki, other languages are just undocumented. > > -- > Benjamin Henrion > FFII Brussels - +32-484-566109 - +32-2-3500762 > "In July 2005, after several failed attempts to legalise software > patents in Europe, the patent establishment changed its strategy. > Instead of explicitly seeking to sanction the patentability of > software, they are now seeking to create a central European patent > court, which would establish and enforce patentability rules in their > favor, without any possibility of correction by competing courts or > democratically elected legislators." > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130221/41528390/attachment.html From victor.chukalovskiy at gmail.com Thu Feb 21 21:10:58 2013 From: victor.chukalovskiy at gmail.com (Victor Chukalovskiy) Date: Thu, 21 Feb 2013 13:10:58 -0500 Subject: [Freeswitch-users] NPDI tel URI parameters appearing in SIP URI In-Reply-To: <51252376.8090601@gmail.com> References: <5123FBD2.7070602@gmail.com> <51245472.90105@gmail.com> <51252376.8090601@gmail.com> Message-ID: <51266332.6060701@gmail.com> Hello, I really appreciate if Brian or Anthony could have a look here. Looking for some help and advice to understand if I got it right and how to move forward: Is there a way to pass tel URI parameters (;npdi= ;rn=) correctly when bridging SIP call to a gateway? Or is it not possible without opening a Jira bug to fix? What is the current support of RFC3966 and RFC4694 in FS? Does it require a feature request with perhaps a bounty? Many thanks, Victor On 13-02-20 02:26 PM, Victor Chukalovskiy wrote: > Did a bit more testing today. > > data="sofia/test_profile/*sip:6135555555;npdi=yes*@24.114.15.50:5060"/> gives > what I need as per WiKi: > INVITE sip:6135555555;npdi=yes at 5.6.7.8:5060 SIP/2.0 > > But here is a caveat: this is not gateway! I do need to use gateway in > the bridge app (because of OPTIONS PING that determines if gateway is > up or down). So, I try: > > data="sofia/gateway/test_gateway/6135555555;npdi=yes"/> This is no > good, ;npdi=yes moved in to the domain part: > INVITE sip:6135555555 at 5.6.7.8:5060*;npdi=yes* SIP/2.0 > > Also, tried this: > data="sofia/gateway/test_gateway/sip:6135555555;npdi=yes"/> But this > is no good either. Looks like sip: syntax does not work with gateways: > INVITE *sip:sip:*6135555555 at 5.6.7.8:5060*;npdi=yes* SIP/2.0 > > ---------------- > So, my immediate question is, is there a way to use sip: syntax when > bridging to a gateway? > ---------------- > And on a broader scale, something along these lines: > http://freeswitch-users.2379917.n2.nabble.com/generating-RFC-3966-and-RFC-4694-calls-td2544806.html > Was there much work done since then on RFC3966 and RFC4694 support in > FS? Ideally, that would be something that's parsed as channel variable > on incoming call and can be manipulated as a channel variable on the > outgoing call. FS would take care of parsing tel URI on incoming and > putting it together on the outgoing. > > -Victor > > > On 02/20/2013 03:32 AM, Avi Marcus wrote: >> Ah sorry, I should have read your question more carefully! >> -Avi >> >> >> On Wed, Feb 20, 2013 at 6:43 AM, Victor Chukalovskiy >> > > wrote: >> >> Avi, >> >> I know this part of the WiKi....thank you but I don't think it >> helps here. >> >> -Victor >> >> >> On 02/19/2013 07:25 PM, Avi Marcus wrote: >>> You'll have to use a regex and capture it manually. >>> See: >>> http://wiki.freeswitch.org/wiki/Regular_Expression#LNP_.28Local_Number_Portability.29 >>> >>> -Avi >>> >>> On Wed, Feb 20, 2013 at 12:25 AM, Victor Chukalovskiy >>> >> > wrote: >>> >>> Hello, >>> >>> SIP call comes on leg A as following: >>> INVITE sip:6135555555 >>> *;npdi=yes*@1.2.3.4:5060 >>> SIP/2.0 >>> >>> ${destination_number} var becomes "6135555555 >>> ;npdi=yes" >>> >>> On leg B I send the call to gateway using unaltered >>> ${destination_number}: >>> >> data="sofia/gateway/test_gw/${destination_number}"/> >>> >>> I see the following on leg B: >>> INVITE sip:6135555555 at 5.6.7.8:5060 >>> *;npdi=yes* SIP/2.0 >>> >>> So, FS moved ";npdi=yes" which is a tel URI parameter into >>> the very end as if it was a SIP URI parameter. Such behavior >>> does not seem correct. Is this a bug? Found the following >>> implementation discussion: >>> http://comments.gmane.org/gmane.ietf.sip-implementors/21554 >>> >>> Please help to make FS send tel URI parameters unaltered >>> using bridge application and a gateway. WiKi for bridge >>> dial-plan tool has some examples but it does not make use of >>> gateway and I need to use gateway. >>> >>> Thank you, >>> Victor >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130221/562795da/attachment-0001.html From mario_fs at mgtech.com Thu Feb 21 21:22:58 2013 From: mario_fs at mgtech.com (Mario G) Date: Thu, 21 Feb 2013 10:22:58 -0800 Subject: [Freeswitch-users] [Solved] Can local_stream MOH have a time limit/max duration? In-Reply-To: References: <219BCB63-3A5A-419F-B374-BDD45FDE3FAB@mgtech.com> Message-ID: <006916BC-5985-4489-9515-7E13BDF0FDFC@mgtech.com> I updated the wiki with a new Execution Parameters section: http://wiki.freeswitch.org/wiki/Mod_local_stream#Execution_Parameters . Needless to say this only works with the latest HEAD for now. Thanks to Anthony! Mario G I am using this now: On Feb 12, 2013, at 10:43 AM, Mario G wrote: > I tried a few things with transfer_ringaback but no dice. The wiki also states transfer_ringback is only for after the call is answered, use ringback for early media, which is what I need it for. i looked at source code but did not find anything that would allow limiting moh duration, bummer. Thanks. > Mario G > > On Feb 11, 2013, at 12:57 PM, Michael Collins wrote: > >> For #2 use transfer_ringback channel variable and set it to MOH and it should work the same way, i.e. instead of ringing the caller will hear music. >> >> -MC >> >> On Fri, Feb 8, 2013 at 8:55 AM, Mario G wrote: >> I looked into chime-ins, but it only chimes in at fixed intervals, and it was not flexible enough for true customization. I could not find anything else on the wiki that would play moh (or any other file) for a specific time. See my previous post for a details of how it works and what I wanted to do. Thanks. >> Mario G >> >> On Feb 8, 2013, at 5:48 AM, Vik Killa wrote: >> >> > I don't quite understand what you are trying to do here. >> > mod_local_stream plays audio in a loop constantly with chime-ins >> > (announcements) set to play periodically. If you need something to >> > play from the beginning for a certain amount of time, you need to use >> > something other than mod_local_stream. >> > >> > >> >> On Wed, Feb 6, 2013 at 9:04 AM, Mario G wrote: >> >>> >> >>> Could not find anything in the wiki and this may not be possible but... I >> >>> have early media with instructions, is has voice, ring, voice, music. I want >> >>> to change the ring to music but I could not find a way to limit the time the >> >>> music plays. See the second example, I would like to limit the first moh to >> >>> a maximum time. If not possible, is this a "bounty" enhancement? Thanks for >> >>> any help. >> >>> Mario G >> >>> >> >>> >> >>> Currently: >> >>> > >>> data="ringback=file_string://${lua_ring1}!tone_stream://${us-ring};loops=3!file_string://${lua_ring2}!local_stream://moh"/> >> >>> >> >>> Would Like (add something to limit the first moh): >> >>> > >>> data="ringback=file_string://${lua_ring1}!local_stream://moh!file_string://${lua_ring2}!local_stream://moh"/> >> >>> >> >>> >> >>> ______________________________________________________________________ >> > >> > _________________________________________________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130221/3d7c08f5/attachment.html From mike at jerris.com Thu Feb 21 21:35:37 2013 From: mike at jerris.com (Michael Jerris) Date: Thu, 21 Feb 2013 13:35:37 -0500 Subject: [Freeswitch-users] NPDI tel URI parameters appearing in SIP URI In-Reply-To: <51266332.6060701@gmail.com> References: <5123FBD2.7070602@gmail.com> <51245472.90105@gmail.com> <51252376.8090601@gmail.com> <51266332.6060701@gmail.com> Message-ID: <8D2BE1B7-FC70-4135-8D13-FC6414285558@jerris.com> Did you ever open the jira that Ken requested you open on this? All of the information you found about this issue should be on a jira or its not going to get the attention you desire. Mike On Feb 21, 2013, at 1:10 PM, Victor Chukalovskiy wrote: > Hello, > > I really appreciate if Brian or Anthony could have a look here. Looking for some help and advice to understand if I got it right and how to move forward: > > Is there a way to pass tel URI parameters (;npdi= ;rn=) correctly when bridging SIP call to a gateway? Or is it not possible without opening a Jira bug to fix? > > What is the current support of RFC3966 and RFC4694 in FS? Does it require a feature request with perhaps a bounty? > > Many thanks, > Victor From notlikeme75 at yahoo.com Thu Feb 21 22:00:52 2013 From: notlikeme75 at yahoo.com (Rodney) Date: Thu, 21 Feb 2013 11:00:52 -0800 (PST) Subject: [Freeswitch-users] Rodney Message-ID: <1361473252.99311.YahooMailNeo@web164501.mail.gq1.yahoo.com> http://www.fliesenmosaik.eu/xmmxodq/a6ncli091hqz3zx3lqu5irc3r16.x7ussihn6    Rodney -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130221/2fba3e53/attachment.html From dvl36.ripe.nick at gmail.com Fri Feb 22 00:23:58 2013 From: dvl36.ripe.nick at gmail.com (Dmitry Lysenko) Date: Thu, 21 Feb 2013 23:23:58 +0200 Subject: [Freeswitch-users] Jira does not work any more? Message-ID: http://jira.freeswitch.org/ : "If you're experiencing a problem with one of our software products, please fill out, print, and submit the following bug report form *via FAX* to +1-918-420-9002. After an initial assessment by our support department, the resolution of the problem will be assigned to an appropriate engineer." Is this temporary? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130221/e12b092c/attachment.html From jnvines at gmail.com Fri Feb 22 00:33:09 2013 From: jnvines at gmail.com (Nick Vines) Date: Thu, 21 Feb 2013 13:33:09 -0800 Subject: [Freeswitch-users] Jira does not work any more? In-Reply-To: References: Message-ID: Looks like a simplified reporting interface. Clicking the link at the very bottom took me to the normal Jira. On Thu, Feb 21, 2013 at 1:23 PM, Dmitry Lysenko wrote: > http://jira.freeswitch.org/ : > > "If you're experiencing a problem with one of our software products, > please fill out, print, and submit the following bug report form *via FAX* to > +1-918-420-9002. After an initial assessment by our support department, > the resolution of the problem will be assigned to an appropriate engineer. > " > > Is this temporary? > Thanks. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130221/c3653c48/attachment.html From schoch+freeswitch.org at xwin32.com Fri Feb 22 00:36:10 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Thu, 21 Feb 2013 13:36:10 -0800 Subject: [Freeswitch-users] "Outgoing Call" on phone display Message-ID: When I place a call from my Polycom phone and the remote party answers, the display changes to say "Outgoing Call()" when the remote party answers. This is annoying because it makes the important bit of information (the actual number) scroll off the right of the screen. I'm trying to figure out where this "Outgoing Call" string is coming from. I ran an SIP trace and found some messages that get sent to the phone when the remote party answers. One of these messages includes this line: Remote-Party-ID: "Outbound Call" <1xxxxxxxxxx>;party=calling;privacy=off;screen=no A later message includes this line: P-Asserted-Identity: "Outbound Call" <1xxxxxxxxxx> Where are these coming from, and can I change them? -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130221/67379adf/attachment-0001.html From jnvines at gmail.com Fri Feb 22 00:36:38 2013 From: jnvines at gmail.com (Nick Vines) Date: Thu, 21 Feb 2013 13:36:38 -0800 Subject: [Freeswitch-users] Jira does not work any more? In-Reply-To: References: Message-ID: I spoke too soon. I have no idea why that page is there, but you can still get to the normal section. On Thu, Feb 21, 2013 at 1:33 PM, Nick Vines wrote: > Looks like a simplified reporting interface. Clicking the link at the very > bottom took me to the normal Jira. > > > > On Thu, Feb 21, 2013 at 1:23 PM, Dmitry Lysenko > wrote: > >> http://jira.freeswitch.org/ : >> >> "If you're experiencing a problem with one of our software products, >> please fill out, print, and submit the following bug report form *via FAX >> * to +1-918-420-9002. After an initial assessment by our support >> department, the resolution of the problem will be assigned to an >> appropriate engineer." >> >> Is this temporary? >> Thanks. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130221/7f94e057/attachment.html From victor.chukalovskiy at gmail.com Fri Feb 22 01:02:26 2013 From: victor.chukalovskiy at gmail.com (Victor Chukalovskiy) Date: Thu, 21 Feb 2013 17:02:26 -0500 Subject: [Freeswitch-users] NPDI tel URI parameters appearing in SIP URI In-Reply-To: <8D2BE1B7-FC70-4135-8D13-FC6414285558@jerris.com> References: <5123FBD2.7070602@gmail.com> <51245472.90105@gmail.com> <51252376.8090601@gmail.com> <51266332.6060701@gmail.com> <8D2BE1B7-FC70-4135-8D13-FC6414285558@jerris.com> Message-ID: <51269972.40606@gmail.com> Mike, Thanks I did now FS-5118. Sorry I thought better to double-check it here before opening a Jira (I'm not sure at this point weather it's a bug or a feature request). Thank you, Victor On 13-02-21 01:35 PM, Michael Jerris wrote: > Did you ever open the jira that Ken requested you open on this? All of the information you found about this issue should be on a jira or its not going to get the attention you desire. > > Mike > > On Feb 21, 2013, at 1:10 PM, Victor Chukalovskiy wrote: > >> Hello, >> >> I really appreciate if Brian or Anthony could have a look here. Looking for some help and advice to understand if I got it right and how to move forward: >> >> Is there a way to pass tel URI parameters (;npdi= ;rn=) correctly when bridging SIP call to a gateway? Or is it not possible without opening a Jira bug to fix? >> >> What is the current support of RFC3966 and RFC4694 in FS? Does it require a feature request with perhaps a bounty? >> >> Many thanks, >> Victor > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 brian at freeswitch.org Fri Feb 22 01:10:39 2013 From: brian at freeswitch.org (Brian West) Date: Thu, 21 Feb 2013 16:10:39 -0600 Subject: [Freeswitch-users] "Outgoing Call" on phone display In-Reply-To: References: Message-ID: <094868EE-812C-4E21-8551-0EC80F553D3D@freeswitch.org> effective_callee_id_name and effective_callee_id_number, You're telling you forgot what number you just dialed? LOL -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire T: +1.918.420.9266 | F: +1.918.420.9267 | M: +1.918.424.WEST iNUM: +883 5100 1420 9266 ISN: 410*543 On Feb 21, 2013, at 3:36 PM, Steven Schoch wrote: > When I place a call from my Polycom phone and the remote party answers, the display changes to say "Outgoing Call()" when the remote party answers. This is annoying because it makes the important bit of information (the actual number) scroll off the right of the screen. > > I'm trying to figure out where this "Outgoing Call" string is coming from. I ran an SIP trace and found some messages that get sent to the phone when the remote party answers. One of these messages includes this line: > > Remote-Party-ID: "Outbound Call" <1xxxxxxxxxx>;party=calling;privacy=off;screen=no > > A later message includes this line: > > P-Asserted-Identity: "Outbound Call" <1xxxxxxxxxx> > > Where are these coming from, and can I change them? > > -- > Steve From brian at freeswitch.org Fri Feb 22 01:12:40 2013 From: brian at freeswitch.org (Brian West) Date: Thu, 21 Feb 2013 16:12:40 -0600 Subject: [Freeswitch-users] Jira does not work any more? In-Reply-To: References: Message-ID: <3E87CDA5-B406-4999-9005-69F3F4058732@freeswitch.org> sigh, Come on guys we had to update JIRA due to a security issue. We had a little fun with it... give us that. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire T: +1.918.420.9266 | F: +1.918.420.9267 | M: +1.918.424.WEST iNUM: +883 5100 1420 9266 ISN: 410*543 On Feb 21, 2013, at 3:23 PM, Dmitry Lysenko wrote: > http://jira.freeswitch.org/ : > > "If you're experiencing a problem with one of our software products, please fill out, print, and submit the following bug report form via FAX to +1-918-420-9002. After an initial assessment by our support department, the resolution of the problem will be assigned to an appropriate engineer." > > Is this temporary? > Thanks. > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 dvl36.ripe.nick at gmail.com Fri Feb 22 01:14:25 2013 From: dvl36.ripe.nick at gmail.com (Dmitry Lysenko) Date: Fri, 22 Feb 2013 00:14:25 +0200 Subject: [Freeswitch-users] Jira does not work any more? In-Reply-To: References: Message-ID: It is already working. Closed. Thanks. 2013/2/21 Nick Vines > I spoke too soon. I have no idea why that page is there, but you can still > get to the normal section. > > > On Thu, Feb 21, 2013 at 1:33 PM, Nick Vines wrote: > >> Looks like a simplified reporting interface. Clicking the link at the >> very bottom took me to the normal Jira. >> >> >> >> On Thu, Feb 21, 2013 at 1:23 PM, Dmitry Lysenko < >> dvl36.ripe.nick at gmail.com> wrote: >> >>> http://jira.freeswitch.org/ : >>> >>> "If you're experiencing a problem with one of our software products, >>> please fill out, print, and submit the following bug report form *via >>> FAX* to +1-918-420-9002. After an initial assessment by our support >>> department, the resolution of the problem will be assigned to an >>> appropriate engineer." >>> >>> Is this temporary? >>> Thanks. >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130222/ba2efe7d/attachment.html From krice at freeswitch.org Fri Feb 22 01:18:25 2013 From: krice at freeswitch.org (Ken Rice) Date: Thu, 21 Feb 2013 16:18:25 -0600 Subject: [Freeswitch-users] Jira does not work any more? In-Reply-To: <3E87CDA5-B406-4999-9005-69F3F4058732@freeswitch.org> Message-ID: Some people are just to serious all the time On 2/21/13 4:12 PM, "Brian West" wrote: > sigh, Come on guys we had to update JIRA due to a security issue. We had a > little fun with it... give us that. > -- > > On Feb 21, 2013, at 3:23 PM, Dmitry Lysenko wrote: > >> http://jira.freeswitch.org/ : >> >> "If you're experiencing a problem with one of our software products, please >> fill out, print, and submit the following bug report form via FAX to >> +1-918-420-9002. After an initial assessment by our support department, the >> resolution of the problem will be assigned to an appropriate engineer." -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch From anthony.minessale at gmail.com Fri Feb 22 01:28:34 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 21 Feb 2013 16:28:34 -0600 Subject: [Freeswitch-users] Jira does not work any more? In-Reply-To: References: <3E87CDA5-B406-4999-9005-69F3F4058732@freeswitch.org> Message-ID: We still have our fail-safe option in place....... On Thu, Feb 21, 2013 at 4:18 PM, Ken Rice wrote: > Some people are just to serious all the time > > > On 2/21/13 4:12 PM, "Brian West" wrote: > > > sigh, Come on guys we had to update JIRA due to a security issue. We > had a > > little fun with it... give us that. > > -- > > > > On Feb 21, 2013, at 3:23 PM, Dmitry Lysenko > wrote: > > > >> http://jira.freeswitch.org/ : > >> > >> "If you're experiencing a problem with one of our software products, > please > >> fill out, print, and submit the following bug report form via FAX to > >> +1-918-420-9002. After an initial assessment by our support > department, the > >> resolution of the problem will be assigned to an appropriate engineer." > -- > Ken > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > irc.freenode.net #freeswitch > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130221/3b32d801/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: number.jpg Type: image/jpeg Size: 56001 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130221/3b32d801/attachment-0001.jpg From victor.chukalovskiy at gmail.com Fri Feb 22 01:29:21 2013 From: victor.chukalovskiy at gmail.com (Victor Chukalovskiy) Date: Thu, 21 Feb 2013 17:29:21 -0500 Subject: [Freeswitch-users] Jira does not work any more? In-Reply-To: References: Message-ID: <51269FC1.7030305@gmail.com> Funny! I really did fill in the form and was looking for a "print form" button :-) ! On 13-02-21 05:18 PM, Ken Rice wrote: > Some people are just to serious all the time > > > On 2/21/13 4:12 PM, "Brian West" wrote: > >> sigh, Come on guys we had to update JIRA due to a security issue. We had a >> little fun with it... give us that. >> -- >> >> On Feb 21, 2013, at 3:23 PM, Dmitry Lysenko wrote: >> >>> http://jira.freeswitch.org/ : >>> >>> "If you're experiencing a problem with one of our software products, please >>> fill out, print, and submit the following bug report form via FAX to >>> +1-918-420-9002. After an initial assessment by our support department, the >>> resolution of the problem will be assigned to an appropriate engineer." From dvl36.ripe.nick at gmail.com Fri Feb 22 01:35:57 2013 From: dvl36.ripe.nick at gmail.com (Dmitry Lysenko) Date: Fri, 22 Feb 2013 00:35:57 +0200 Subject: [Freeswitch-users] Jira does not work any more? In-Reply-To: <51269FC1.7030305@gmail.com> References: <51269FC1.7030305@gmail.com> Message-ID: How many fax pages was sended? ) 2013/2/22 Victor Chukalovskiy > Funny! I really did fill in the form and was looking for a "print form" > button :-) ! > > > On 13-02-21 05:18 PM, Ken Rice wrote: > > Some people are just to serious all the time > > > > > > On 2/21/13 4:12 PM, "Brian West" wrote: > > > >> sigh, Come on guys we had to update JIRA due to a security issue. We > had a > >> little fun with it... give us that. > >> -- > >> > >> On Feb 21, 2013, at 3:23 PM, Dmitry Lysenko > wrote: > >> > >>> http://jira.freeswitch.org/ : > >>> > >>> "If you're experiencing a problem with one of our software products, > please > >>> fill out, print, and submit the following bug report form via FAX to > >>> +1-918-420-9002. After an initial assessment by our support > department, the > >>> resolution of the problem will be assigned to an appropriate engineer." > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130222/5c819d2d/attachment.html From stuart.mills3 at btopenworld.com Fri Feb 22 01:39:15 2013 From: stuart.mills3 at btopenworld.com (Stuart Mills) Date: Thu, 21 Feb 2013 22:39:15 -0000 Subject: [Freeswitch-users] Outbound mod event bug In-Reply-To: References: <68FC21E54C0D40E986A880956375D5FF@PBPC> Message-ID: Cheers Steven, This was only built last Friday from the latest git head, I'll check to see if there?s been any latest release since then. If I am on the latest, how do I go about reporting the bug? Cheers, Stuart From: Steven Ayre Sent: Thursday, February 21, 2013 5:01 PM To: FreeSWITCH Users Help Cc: mailto:freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Outbound mod event bug A core dump is always a bug. Check if it happens on the latest git head (it may already be fixed). If it does it there collect a backtrack and file a Jira. http://wiki.freeswitch.org/wiki/Reporting_Bugs#Creating_A_Backtrace_With_gdb_.28Linux.2FUnix.29 Steve On 21 Feb 2013, at 16:49, "Stuart Mills" wrote: Hi All, I know some of you may not consider this as a bug, especially if things are done in the correct way this would never happen, but please read on and I'll explain the issue as I was a little surprised at how badly FreeSwitch handled this user error, seen as though it?s such a good piece of software. Basically, a call arrives into Freeswitch and a connection is established to a remote server, then the following relay occurs (in it?s simplest terms) - connect sent into Freeswitch FreeSwitch replies OK myevents sent into FreeSwitch FreeSwitch replies OK answer sent into FreeSwitch FreeSwitch replies OK bridge with arguments sent into FreeSwitch FreeSwitch replies OK Now, everything is perfect up until here and I'd say that as a call switching platform it?s doing exactly what is being asked of it ? however here?s the bit most of you would agree is wrong and isn?t a ?real? bug - if you reply again with another bridge, be it by mistake or a deliberate attempt to fool FreeSwitch, the system crashes and causes a core dump. I can get this to happen every time, no warning, no nothing, just a crash dump and FreeSwitch dies in front of my eyes. Answers on a postcard please ? try not to be too harsh. I am particularly new to FreeSwitch and find it very very good, I was just really surprised to fin a bug that could crash it. Kind Regards, Stuart _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -------------------------------------------------------------------------------- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130221/1ac0fc8d/attachment.html From dvl36.ripe.nick at gmail.com Fri Feb 22 01:44:38 2013 From: dvl36.ripe.nick at gmail.com (Dmitry Lysenko) Date: Fri, 22 Feb 2013 00:44:38 +0200 Subject: [Freeswitch-users] Jira does not work any more? In-Reply-To: References: <51269FC1.7030305@gmail.com> Message-ID: My first thought: It's time to think about T38. ) 2013/2/22 Dmitry Lysenko > How many fax pages was sended? ) > > > 2013/2/22 Victor Chukalovskiy > >> Funny! I really did fill in the form and was looking for a "print form" >> button :-) ! >> >> >> On 13-02-21 05:18 PM, Ken Rice wrote: >> > Some people are just to serious all the time >> > >> > >> > On 2/21/13 4:12 PM, "Brian West" wrote: >> > >> >> sigh, Come on guys we had to update JIRA due to a security issue. We >> had a >> >> little fun with it... give us that. >> >> -- >> >> >> >> On Feb 21, 2013, at 3:23 PM, Dmitry Lysenko >> wrote: >> >> >> >>> http://jira.freeswitch.org/ : >> >>> >> >>> "If you're experiencing a problem with one of our software products, >> please >> >>> fill out, print, and submit the following bug report form via FAX to >> >>> +1-918-420-9002. After an initial assessment by our support >> department, the >> >>> resolution of the problem will be assigned to an appropriate >> engineer." >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130222/494019ff/attachment-0001.html From covici at ccs.covici.com Fri Feb 22 01:58:32 2013 From: covici at ccs.covici.com (covici at ccs.covici.com) Date: Thu, 21 Feb 2013 17:58:32 -0500 Subject: [Freeswitch-users] DID providers - any thoughts? (jan 2013) In-Reply-To: References: , , , , Message-ID: <8024.1361487512@ccs.covici.com> Didforsale is very nice as well. Robert Craig wrote: > I can vouch for Flowroute. I use them as one of my primary providers and have had no problems. In the event you experience a glitch, you can submit a support ticket or call them and they will work with you. Very helpful techs. > > Rob > Unlimited calling to the US and Canada.Visit www.craigcomm.net for more information. > > Date: Wed, 20 Feb 2013 21:03:53 +0000 > From: cal.leeming at simplicitymedialtd.co.uk > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] DID providers - any thoughts? (jan 2013) > > Seems that flowroute is getting some good feedback.. > > If anyone else has comments/feedback about their experiences with this company, please feel free to update the thread. > > Cal > > On Wed, Feb 20, 2013 at 11:57 AM, chris wrote: > > +1 here, flowroute has exceeded my expectations in many areas. The fact that they actually have humans available via phone, and even humans who understand the technology quite well. > > Reliability has been great but you should always plan for a rainy day and have a backup plan, even the biggest and best have issues from time to time > > chris > On Thu, Jan 31, 2013 at 5:34 PM, Steven Schoch wrote: > > > On Wed, Jan 30, 2013 at 4:59 AM, Cal Leeming [Simplicity Media Ltd] wrote: > > > > > Flowroute - Any else able to offer some reviews on this company - specifically relating to US inbound?? > > > > I'm just starting my VoIP venture and I have chosen Flowroute as our provider. So far, mostly as a test, I have transferred one of our AT&T lines to Flowroute. It worked very smoothly. One thing I discovered, which you all probably know, but I'm new with VoIP, was that when I called the number from a POTS line, it went through the PSTN, through Flowroute, through FreeSwitch, and rang on my Polycom phone even before I heard the first ring from the calling phone. > > > > Also, the Flowroute tech support team responds very quickly and helpfully. > No complaints. > -- Steve > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > 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 > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > 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 > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > ---------------------------------------------------- > Alternatives: > > ---------------------------------------------------- > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici at ccs.covici.com From krice at freeswitch.org Fri Feb 22 02:05:12 2013 From: krice at freeswitch.org (Ken Rice) Date: Thu, 21 Feb 2013 17:05:12 -0600 Subject: [Freeswitch-users] Jira does not work any more? In-Reply-To: <51269FC1.7030305@gmail.com> Message-ID: Its under File -> Print in your browser On 2/21/13 4:29 PM, "Victor Chukalovskiy" wrote: > Funny! I really did fill in the form and was looking for a "print form" > button :-) ! > > > On 13-02-21 05:18 PM, Ken Rice wrote: >> Some people are just to serious all the time >> >> >> On 2/21/13 4:12 PM, "Brian West" wrote: >> >>> sigh, Come on guys we had to update JIRA due to a security issue. We had a >>> little fun with it... give us that. >>> -- >>> >>> On Feb 21, 2013, at 3:23 PM, Dmitry Lysenko >>> wrote: >>> >>>> http://jira.freeswitch.org/ : >>>> >>>> "If you're experiencing a problem with one of our software products, please >>>> fill out, print, and submit the following bug report form via FAX to >>>> +1-918-420-9002. After an initial assessment by our support department, the >>>> resolution of the problem will be assigned to an appropriate engineer." > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch From schoch+freeswitch.org at xwin32.com Fri Feb 22 02:11:45 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Thu, 21 Feb 2013 15:11:45 -0800 Subject: [Freeswitch-users] "Outgoing Call" on phone display In-Reply-To: <094868EE-812C-4E21-8551-0EC80F553D3D@freeswitch.org> References: <094868EE-812C-4E21-8551-0EC80F553D3D@freeswitch.org> Message-ID: On Thu, Feb 21, 2013 at 2:10 PM, Brian West wrote: > effective_callee_id_name and effective_callee_id_number, I never set effective_callee_id_name. Does this get set to "Outgoing Call" automatically? > You're telling you forgot what number you just dialed? LOL > The phone keeps a history. I probably dialed the number yesterday. :-) -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130221/284c6960/attachment.html From jaykris at gmail.com Fri Feb 22 02:17:03 2013 From: jaykris at gmail.com (JP) Date: Thu, 21 Feb 2013 15:17:03 -0800 Subject: [Freeswitch-users] mod-lua Vs mod-java In-Reply-To: References: <5125BDA4.20505@gmail.com> Message-ID: Thanks to all for the feedback. I guess what I learnt from this conversation is that... Use Lua for as far as possible, but for those exceptional scenarios when Lua falls short, it may make more sense to use a full fledged programming language like Java. -JP On Thu, Feb 21, 2013 at 10:00 AM, David Villasmil < david.villasmil.work at gmail.com> wrote: > Well that depends on what you intent to accomplish, I guess. > The nicest thing about lua (in our case) is that it is embeded inside FS. > > David > > > > On Thu, Feb 21, 2013 at 1:35 PM, Benjamin Henrion wrote: > >> On Thu, Feb 21, 2013 at 1:17 PM, David Villasmil >> wrote: >> > lua is by far the best approach. I really don't know about accessing web >> > services, but for db it is perfect. >> > I wrote a whole wholesale and calling card app with lua, so you can >> pretty >> > much do anything wit it. >> >> Number of reusable libraries is quite limited compared to other languages. >> >> Lua is well documented on FS wiki, other languages are just undocumented. >> >> -- >> Benjamin Henrion >> FFII Brussels - +32-484-566109 - +32-2-3500762 >> "In July 2005, after several failed attempts to legalise software >> patents in Europe, the patent establishment changed its strategy. >> Instead of explicitly seeking to sanction the patentability of >> software, they are now seeking to create a central European patent >> court, which would establish and enforce patentability rules in their >> favor, without any possibility of correction by competing courts or >> democratically elected legislators." >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130221/8134323a/attachment.html From brian at freeswitch.org Fri Feb 22 02:26:45 2013 From: brian at freeswitch.org (Brian West) Date: Thu, 21 Feb 2013 17:26:45 -0600 Subject: [Freeswitch-users] Jira does not work any more? In-Reply-To: References: <51269FC1.7030305@gmail.com> Message-ID: <5478EF41-5827-4538-BF3A-A214B7A9B98E@freeswitch.org> Well that number is REAL, its T.38 capable from the provider. So it would have worked thats kinda the FUNNY joke if it all LOL. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire T: +1.918.420.9266 | F: +1.918.420.9267 | M: +1.918.424.WEST iNUM: +883 5100 1420 9266 ISN: 410*543 On Feb 21, 2013, at 4:44 PM, Dmitry Lysenko wrote: > My first thought: It's time to think about T38. ) From brian at freeswitch.org Fri Feb 22 02:27:46 2013 From: brian at freeswitch.org (Brian West) Date: Thu, 21 Feb 2013 17:27:46 -0600 Subject: [Freeswitch-users] "Outgoing Call" on phone display In-Reply-To: References: <094868EE-812C-4E21-8551-0EC80F553D3D@freeswitch.org> Message-ID: <2F45CFD8-1F72-44AB-93D4-0C59BED92AD2@freeswitch.org> You know you can do CNAM lookups while dialing and set those values and update the phone during the ringing phase. It puts Outgoing Call in there when it has no other information. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire T: +1.918.420.9266 | F: +1.918.420.9267 | M: +1.918.424.WEST iNUM: +883 5100 1420 9266 ISN: 410*543 On Feb 21, 2013, at 5:11 PM, Steven Schoch wrote: > On Thu, Feb 21, 2013 at 2:10 PM, Brian West wrote: > effective_callee_id_name and effective_callee_id_number, > I never set effective_callee_id_name. Does this get set to "Outgoing Call" automatically? > > You're telling you forgot what number you just dialed? LOL > The phone keeps a history. I probably dialed the number yesterday. :-) > > -- > Steve From engrmuhammadshahzad at hotmail.com Fri Feb 22 02:09:44 2013 From: engrmuhammadshahzad at hotmail.com (Muhammad Shahzad) Date: Fri, 22 Feb 2013 04:09:44 +0500 Subject: [Freeswitch-users] FreeSwitch Skype installation and clients startup issue Message-ID: Hi Guys, I am trying to setup FreeSwitch for Skype calling, I have followed the steps mention in below URL and my Linux distribution is Centos 5.5 64bits.. http://wiki.freeswitch.org/wiki/Skypiax Face issue while starting the skype clients, pls help me out. [root at FreeSwitch install]# sh /usr/local/freeswitch/skypopen/skype-clients-startup-dir/start_skype_clients .sh ERROR: Module snd_pcm_oss does not exist in /proc/modules ERROR: Module snd_mixer_oss does not exist in /proc/modules ERROR: Module snd_seq_oss does not exist in /proc/modules mknod: `/dev/dsp': File exists insmod: error inserting '/usr/local/freeswitch/skypopen/skypopen-sound-driver-dir/skypopen.ko': -1 File exists Fatal server error: Server is already active for display 101 If this server is no longer running, remove /tmp/.X101-lock and start again. bash: /usr/local/freeswitch/skypopen/skype-clients-symlinks-dir/skype101: No such file or directory Fatal server error: Server is already active for display 102 If this server is no longer running, remove /tmp/.X102-lock and start again. bash: /usr/local/freeswitch/skypopen/skype-clients-symlinks-dir/skype102: No such file or directory Fatal server error: Server is already active for display 103 If this server is no longer running, remove /tmp/.X103-lock and start again. bash: /usr/local/freeswitch/skypopen/skype-clients-symlinks-dir/skype103: No such file or directory Fatal server error: Server is already active for display 104 If this server is no longer running, remove /tmp/.X104-lock and start again. bash: /usr/local/freeswitch/skypopen/skype-clients-symlinks-dir/skype104: No such file or directory Fatal server error: Server is already active for display 105 If this server is no longer running, remove /tmp/.X105-lock and start again. bash: /usr/local/freeswitch/skypopen/skype-clients-symlinks-dir/skype105: No such file or directory Regards, Shazi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130222/057d16ae/attachment.html From gmaruzz at gmail.com Fri Feb 22 03:15:37 2013 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Fri, 22 Feb 2013 01:15:37 +0100 Subject: [Freeswitch-users] FreeSwitch Skype installation and clients startup issue In-Reply-To: References: Message-ID: Ciao Muhammad, I'd use CentOS 6, or Debian Squeeze. That said, follow carefully all the steps, install the prerequisites, and use the installer.pl Other comments are below, between your lines On Fri, Feb 22, 2013 at 12:09 AM, Muhammad Shahzad < engrmuhammadshahzad at hotmail.com> wrote: > > > [root at FreeSwitch install]# sh > /usr/local/freeswitch/skypopen/skype-clients-startup-dir/start_skype_clients.sh > **** > > ERROR: Module snd_pcm_oss does not exist in /proc/modules**** > > ERROR: Module snd_mixer_oss does not exist in /proc/modules**** > > ERROR: Module snd_seq_oss does not exist in /proc/modules**** > > That's ok, you have not loaded those modules we're trying to unload, no problem > mknod: `/dev/dsp': File exists**** > > insmod: error inserting > '/usr/local/freeswitch/skypopen/skypopen-sound-driver-dir/skypopen.ko': -1 > File exists**** > > ** > that's ok too, it's just that you had loaded skypopen before (probably a previous run of the startup_clients script) > ** > > Fatal server error:**** > > Server is already active for display 101**** > > If this server is no longer running, remove /tmp/.X101-lock**** > > and start again. > That means you have running X servers, probably from a previous run of the startup script. Kill them with a killall -9 Xvfb (heck, I can add this to the beginning of that script, btw) > **** > > ** ** > > bash: /usr/local/freeswitch/skypopen/skype-clients-symlinks-dir/skype101: > No such file or directory**** > > ** > that's bad: seems the installer.pl has not created the symlink, or has not installed skype client, or maybe there are parts (eg: shared libraries, dynamic loadable libraries) the skype client do not find. Have you installed ALL the listed pre-requisites? They're needed ALL (and I mean ALL of them, exactly as listed in the wiki page) ! Have you used the installer.pl ? .... and it continues like that Let me know how it goes... -giovanni -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130222/9e882f3f/attachment.html From msc at freeswitch.org Fri Feb 22 03:49:18 2013 From: msc at freeswitch.org (Michael Collins) Date: Thu, 21 Feb 2013 16:49:18 -0800 Subject: [Freeswitch-users] Is this an internal call? In-Reply-To: References: Message-ID: I grep'd the entire source tree and found no mention of this variable. One thing you can do is look to see if various transfer_xxx variables are set in phone-to-phone calls vs. transferred calls. I'd start with "transfer_source" var and go from there. -MC On Wed, Feb 20, 2013 at 3:56 PM, Steven Schoch < schoch+freeswitch.org at xwin32.com> wrote: > This must be a FAQ, but I can't find it. > > In the default dialplan, how can I tell if the call was from an internal > phone, or from an external call that was transferred? > > One page said there was a variable ${internal_call}, but I can't find that > anywhere in the source. Should I just look to see if a directory variable, > like account_code or user_context is set? > > -- > Steve > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130221/b1b6e8f5/attachment.html From msc at freeswitch.org Fri Feb 22 04:04:57 2013 From: msc at freeswitch.org (Michael Collins) Date: Thu, 21 Feb 2013 17:04:57 -0800 Subject: [Freeswitch-users] NPDI tel URI parameters appearing in SIP URI In-Reply-To: <51269972.40606@gmail.com> References: <5123FBD2.7070602@gmail.com> <51245472.90105@gmail.com> <51252376.8090601@gmail.com> <51266332.6060701@gmail.com> <8D2BE1B7-FC70-4135-8D13-FC6414285558@jerris.com> <51269972.40606@gmail.com> Message-ID: On Thu, Feb 21, 2013 at 2:02 PM, Victor Chukalovskiy < victor.chukalovskiy at gmail.com> wrote: > Mike, > > Thanks I did now FS-5118. > > Sorry I thought better to double-check it here before opening a Jira > (I'm not sure at this point weather it's a bug or a feature request). > You never have to ask on the ML whether to open a Jira. Just open it and the interested parties will comment there. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130221/d7c11015/attachment-0001.html From msc at freeswitch.org Fri Feb 22 04:07:19 2013 From: msc at freeswitch.org (Michael Collins) Date: Thu, 21 Feb 2013 17:07:19 -0800 Subject: [Freeswitch-users] Outbound mod event bug In-Reply-To: References: <68FC21E54C0D40E986A880956375D5FF@PBPC> Message-ID: On Thu, Feb 21, 2013 at 2:39 PM, Stuart Mills wrote: > Cheers Steven, > > This was only built last Friday from the latest git head, I'll check to > see if there?s been any latest release since then. > > If I am on the latest, how do I go about reporting the bug? > All the information is on this page: http://wiki.freeswitch.org/wiki/Reporting_Bugs Skip the parts about how to test and go straight to the part about opening a Jira. -MC > > Cheers, > > Stuart > > *From:* Steven Ayre > *Sent:* Thursday, February 21, 2013 5:01 PM > *To:* FreeSWITCH Users Help > *Cc:* mailto:freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Outbound mod event bug > > A core dump is always a bug. > > Check if it happens on the latest git head (it may already be fixed). If > it does it there collect a backtrack and file a Jira. > > > http://wiki.freeswitch.org/wiki/Reporting_Bugs#Creating_A_Backtrace_With_gdb_.28Linux.2FUnix.29 > > > Steve > > > > On 21 Feb 2013, at 16:49, "Stuart Mills" > wrote: > > Hi All, > > I know some of you may not consider this as a bug, especially if things > are done in the correct way this would never happen, but please read on and > I'll explain the issue as I was a little surprised at how badly FreeSwitch > handled this user error, seen as though it?s such a good piece of software. > > Basically, a call arrives into Freeswitch and a connection is established > to a remote server, then the following relay occurs (in it?s simplest > terms) - > > connect sent into Freeswitch > FreeSwitch replies OK > myevents sent into FreeSwitch > FreeSwitch replies OK > answer sent into FreeSwitch > FreeSwitch replies OK > bridge with arguments sent into FreeSwitch > FreeSwitch replies OK > > Now, everything is perfect up until here and I'd say that as a call > switching platform it?s doing exactly what is being asked of it ? however > here?s the bit most of you would agree is wrong and isn?t a ?real? bug - if > you reply again with another bridge, be it by mistake or a deliberate > attempt to fool FreeSwitch, the system crashes and causes a core dump. > > I can get this to happen every time, no warning, no nothing, just a crash > dump and FreeSwitch dies in front of my eyes. > > Answers on a postcard please ? try not to be too harsh. I am particularly > new to FreeSwitch and find it very very good, I was just really surprised > to fin a bug that could crash it. > > Kind Regards, > > Stuart > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > ------------------------------ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130221/b178c765/attachment.html From steveayre at gmail.com Fri Feb 22 04:06:10 2013 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 22 Feb 2013 01:06:10 +0000 Subject: [Freeswitch-users] Is this an internal call? In-Reply-To: References: Message-ID: I would guess its an example of a custom variable being set in directory or in dialplan under certain conditions , and checked for later in dialplan Steve On 22 Feb 2013, at 00:49, Michael Collins wrote: > I grep'd the entire source tree and found no mention of this variable. > > One thing you can do is look to see if various transfer_xxx variables are set in phone-to-phone calls vs. transferred calls. I'd start with "transfer_source" var and go from there. > > -MC > > On Wed, Feb 20, 2013 at 3:56 PM, Steven Schoch wrote: >> This must be a FAQ, but I can't find it. >> >> In the default dialplan, how can I tell if the call was from an internal phone, or from an external call that was transferred? >> >> One page said there was a variable ${internal_call}, but I can't find that anywhere in the source. Should I just look to see if a directory variable, like account_code or user_context is set? >> >> -- >> Steve >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130222/17bef3a1/attachment.html From steveayre at gmail.com Fri Feb 22 04:12:27 2013 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 22 Feb 2013 01:12:27 +0000 Subject: [Freeswitch-users] Outbound mod event bug In-Reply-To: References: <68FC21E54C0D40E986A880956375D5FF@PBPC> Message-ID: <02A46C7C-EBF2-4F83-9D0C-EDADCE0FA6C9@gmail.com> Bug tracker is on http://jira.freeswitch.org - register and then create an issue. The wiki link I posted before has plenty of helpful information. It would be useful to add a description of how to reproduce the crash. If you can create a short simple script that triggers the crash that would be even better. Also a debug-level log and crucially the coredump which the wiki has instructions on collecting. The coredump shows exactly where/what triggered the crash. Steve On 21 Feb 2013, at 22:39, "Stuart Mills" wrote: > Cheers Steven, > > This was only built last Friday from the latest git head, I'll check to see if there?s been any latest release since then. > > If I am on the latest, how do I go about reporting the bug? > > Cheers, > > Stuart > > From: Steven Ayre > Sent: Thursday, February 21, 2013 5:01 PM > To: FreeSWITCH Users Help > Cc: mailto:freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Outbound mod event bug > > A core dump is always a bug. > > Check if it happens on the latest git head (it may already be fixed). If it does it there collect a backtrack and file a Jira. > > http://wiki.freeswitch.org/wiki/Reporting_Bugs#Creating_A_Backtrace_With_gdb_.28Linux.2FUnix.29 > > Steve > > > > On 21 Feb 2013, at 16:49, "Stuart Mills" wrote: > >> Hi All, >> >> I know some of you may not consider this as a bug, especially if things are done in the correct way this would never happen, but please read on and I'll explain the issue as I was a little surprised at how badly FreeSwitch handled this user error, seen as though it?s such a good piece of software. >> >> Basically, a call arrives into Freeswitch and a connection is established to a remote server, then the following relay occurs (in it?s simplest terms) - >> >> connect sent into Freeswitch >> FreeSwitch replies OK >> myevents sent into FreeSwitch >> FreeSwitch replies OK >> answer sent into FreeSwitch >> FreeSwitch replies OK >> bridge with arguments sent into FreeSwitch >> FreeSwitch replies OK >> >> Now, everything is perfect up until here and I'd say that as a call switching platform it?s doing exactly what is being asked of it ? however here?s the bit most of you would agree is wrong and isn?t a ?real? bug - if you reply again with another bridge, be it by mistake or a deliberate attempt to fool FreeSwitch, the system crashes and causes a core dump. >> >> I can get this to happen every time, no warning, no nothing, just a crash dump and FreeSwitch dies in front of my eyes. >> >> Answers on a postcard please ? try not to be too harsh. I am particularly new to FreeSwitch and find it very very good, I was just really surprised to fin a bug that could crash it. >> >> Kind Regards, >> >> Stuart >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130222/970fc95a/attachment-0001.html From 8f27e956 at gmail.com Fri Feb 22 04:19:38 2013 From: 8f27e956 at gmail.com (S. Scott) Date: Thu, 21 Feb 2013 20:19:38 -0500 Subject: [Freeswitch-users] Is this an internal call? In-Reply-To: References: Message-ID: <-240422102377626460@unknownmsgid> Doesn't that var (or its like) get created and set by operation of the dial plan xml (not source code)? OP, you can have your external-to-internal context set any such semaphore variable. ????? iThing: Big thumbs & little keys. Please excuse typo, spelling and grammar errors ? Thought of the Day ? "With all this manure, there must be a pony in here somewhere.? On 2013-02-21, at 19:52, Michael Collins wrote: I grep'd the entire source tree and found no mention of this variable. One thing you can do is look to see if various transfer_xxx variables are set in phone-to-phone calls vs. transferred calls. I'd start with "transfer_source" var and go from there. -MC On Wed, Feb 20, 2013 at 3:56 PM, Steven Schoch < schoch+freeswitch.org at xwin32.com> wrote: > This must be a FAQ, but I can't find it. > > In the default dialplan, how can I tell if the call was from an internal > phone, or from an external call that was transferred? > > One page said there was a variable ${internal_call}, but I can't find that > anywhere in the source. Should I just look to see if a directory variable, > like account_code or user_context is set? > > -- > Steve > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130221/877c40fe/attachment.html From dvl36.ripe.nick at gmail.com Fri Feb 22 04:42:56 2013 From: dvl36.ripe.nick at gmail.com (Dmitry Lysenko) Date: Fri, 22 Feb 2013 03:42:56 +0200 Subject: [Freeswitch-users] Changing codec during calls In-Reply-To: References: <6EF8B8E9-4CC5-480C-B1C1-09CBC533470C@kavun.ch> <35AF9E79-771D-440A-BEBB-A856D831161C@kavun.ch> Message-ID: Now this cool feature is fully working! Today Anthony committed the bugfix! Tested with 2 different SIP UA<->FS<->Callcentric. Both legs. Now possible to write application that will switch codecs on the fly, regarding on any event, such as switching to low bandwidth backup internet line or too high cpu load, etc. BTW, is there way to get API access to statistic of jitter buffer? Thanks. P.S. As I know, asterisk can't do such thing. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130222/010e6bb6/attachment.html From anthony.minessale at gmail.com Fri Feb 22 06:47:19 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 21 Feb 2013 21:47:19 -0600 Subject: [Freeswitch-users] Outbound mod event bug In-Reply-To: <68FC21E54C0D40E986A880956375D5FF@PBPC> References: <68FC21E54C0D40E986A880956375D5FF@PBPC> Message-ID: For the record all crashes are considered bugs. Sometimes its a bad code path and sometimes its just an improper build but either way we encourage bug reports on crashes. First try a full rebuild and update: UNIX: -- this is all-in-one process that updates to latest and builds everything. make current WINDOWS update to latest with your git client open project file clean solution build solution If you can still reproduce, get a backtrace per the wiki mentioned by Steven and post it to jira. On Thu, Feb 21, 2013 at 10:49 AM, Stuart Mills < stuart.mills3 at btopenworld.com> wrote: > Hi All, > > I know some of you may not consider this as a bug, especially if things > are done in the correct way this would never happen, but please read on and > I'll explain the issue as I was a little surprised at how badly FreeSwitch > handled this user error, seen as though it?s such a good piece of software. > > Basically, a call arrives into Freeswitch and a connection is established > to a remote server, then the following relay occurs (in it?s simplest > terms) - > > connect sent into Freeswitch > FreeSwitch replies OK > myevents sent into FreeSwitch > FreeSwitch replies OK > answer sent into FreeSwitch > FreeSwitch replies OK > bridge with arguments sent into FreeSwitch > FreeSwitch replies OK > > Now, everything is perfect up until here and I'd say that as a call > switching platform it?s doing exactly what is being asked of it ? however > here?s the bit most of you would agree is wrong and isn?t a ?real? bug - if > you reply again with another bridge, be it by mistake or a deliberate > attempt to fool FreeSwitch, the system crashes and causes a core dump. > > I can get this to happen every time, no warning, no nothing, just a crash > dump and FreeSwitch dies in front of my eyes. > > Answers on a postcard please ? try not to be too harsh. I am particularly > new to FreeSwitch and find it very very good, I was just really surprised > to fin a bug that could crash it. > > Kind Regards, > > Stuart > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130221/a7da3667/attachment.html From drk at drkngs.net Fri Feb 22 08:10:58 2013 From: drk at drkngs.net (Dave R. Kompel) Date: Thu, 21 Feb 2013 21:10:58 -0800 Subject: [Freeswitch-users] Outbound mod event bug In-Reply-To: Message-ID: <20130222051058.6c92dbc1@mail.tritonwest.net> One little update, Clean solution doesn't work in the MSBUILD on windows. It is safe to rebuild the FreeSwitchCoreLib, and then just build the whole solutoin. This has never been fixed since the first build for windows back in the VS2005 days. This is because the download and other projects don't have clean methods. If you need to do a "Totally Fresh" build on windows then do a "git clean -dfx" first. --Dave _____ From: Anthony Minessale [mailto:anthony.minessale at gmail.com] To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] Sent: Thu, 21 Feb 2013 19:47:19 -0800 Subject: Re: [Freeswitch-users] Outbound mod event bug For the record all crashes are considered bugs. Sometimes its a bad code path and sometimes its just an improper build but either way we encourage bug reports on crashes. First try a full rebuild and update: UNIX: -- this is all-in-one process that updates to latest and builds everything. make current WINDOWS update to latest with your git client open project file clean solution build solution If you can still reproduce, get a backtrace per the wiki mentioned by Steven and post it to jira. On Thu, Feb 21, 2013 at 10:49 AM, Stuart Mills wrote: Hi All, I know some of you may not consider this as a bug, especially if things are done in the correct way this would never happen, but please read on and I'll explain the issue as I was a little surprised at how badly FreeSwitch handled this user error, seen as though it?s such a good piece of software. Basically, a call arrives into Freeswitch and a connection is established to a remote server, then the following relay occurs (in it?s simplest terms) - connect sent into Freeswitch FreeSwitch replies OK myevents sent into FreeSwitch FreeSwitch replies OK answer sent into FreeSwitch FreeSwitch replies OK bridge with arguments sent into FreeSwitch FreeSwitch replies OK Now, everything is perfect up until here and I'd say that as a call switching platform it?s doing exactly what is being asked of it ? however here?s the bit most of you would agree is wrong and isn?t a ?real? bug - if you reply again with another bridge, be it by mistake or a deliberate attempt to fool FreeSwitch, the system crashes and causes a core dump. I can get this to happen every time, no warning, no nothing, just a crash dump and FreeSwitch dies in front of my eyes. Answers on a postcard please ? try not to be too harsh. I am particularly new to FreeSwitch and find it very very good, I was just really surprised to fin a bug that could crash it. Kind Regards, Stuart _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130221/e0cf4636/attachment-0001.html From lists at kavun.ch Fri Feb 22 08:46:21 2013 From: lists at kavun.ch (Emrah) Date: Fri, 22 Feb 2013 00:46:21 -0500 Subject: [Freeswitch-users] Changing codec during calls In-Reply-To: References: <6EF8B8E9-4CC5-480C-B1C1-09CBC533470C@kavun.ch> <35AF9E79-771D-440A-BEBB-A856D831161C@kavun.ch> Message-ID: <564748CC-A8D1-44ED-9AC4-55910DBB7F2A@kavun.ch> Hi guys, Sorry was disconnected from the list for a while. Indeed it looks like the latest commit fixed it. The issue I see now is that the channel info are not updated and a show channels will only list the initial codecs. Great job and thanks a bunch, Emrah On Feb 21, 2013, at 8:42 PM, Dmitry Lysenko wrote: > Now this cool feature is fully working! Today Anthony committed the bugfix! > Tested with 2 different SIP UA<->FS<->Callcentric. Both legs. > Now possible to write application that will switch codecs on the fly, regarding on any event, such as switching to low bandwidth backup internet line or too high cpu load, etc. > BTW, is there way to get API access to statistic of jitter buffer? > Thanks. > > P.S. As I know, asterisk can't do such thing. > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 mario at ims.bg Fri Feb 22 09:26:51 2013 From: mario at ims.bg (Mario Karakanovski) Date: Fri, 22 Feb 2013 08:26:51 +0200 Subject: [Freeswitch-users] freeswitch hack - solved References: <958ABE902C1F47B09FFAAE7EDEE074B1@imsdeveloper> Message-ID: <389F3985F98E4C69BC09E6361347ADCF@imsdeveloper> Definitely my fault! Somebody was open internet to test environment where external profile does not require authentication. Mario _____ From: Mario Karakanovski [mailto:mario at ims.bg] Sent: Thursday, February 21, 2013 8:36 AM To: 'FreeSWITCH Users Help' Subject: RE: [Freeswitch-users] freeswitch hack In my situation all calls are rejected, but I think it is because they are authenticated with invalid username. My concern is how ones can authenticate in freeswitch with user that not exists and never was configured. I was not able to reproduce that. What I found so far: they use a couple of IPs. They send OPTIONS (only one time) during the day and start try at the night. They tried a maximum of 100 calls. I am still waiting to log some packet Mario _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Wednesday, February 20, 2013 10:41 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] freeswitch hack Aren't they supposed to be rejected? On Wed, Feb 20, 2013 at 11:19 AM, Blake Priddy wrote: I have also had the situation that they are calls getting rejected. On Wed, Feb 20, 2013 at 11:08 AM, Michael Collins wrote: On Wed, Feb 20, 2013 at 1:53 AM, Mario Karakanovski wrote: Thanks Ken, It is helpful, but I still think there is some security issue. I've double check configuration. I've try to reproduce the issue trying to do direct call (TCP and UDP) or authenticate with invalid user, but everything works as expected - calls/authentication was rejected. I've decide to log the traffic - maybe I will be able to see where is the problem. What "security issue"? You said that they cannot make calls with the passwords that they've guessed, correct? About the only thing left to do is set up fail2ban and just shut the door on them when they fail too many times. -Michael _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -- Blakelund Priddy Network Systems Engineer Bryant Public School District Bryant, Arkansas 72022 http://www.bryantschools.org p 501-653-5038 f 501-847-5656 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130222/ea69537f/attachment.html From jh.zhou at outlook.com Fri Feb 22 05:44:54 2013 From: jh.zhou at outlook.com (ZhouJianhua) Date: Fri, 22 Feb 2013 02:44:54 +0000 Subject: [Freeswitch-users] Does Freeswitch support CRTP? Message-ID: RTP's head is too heavy for mobile network. There is little about CRTP on google, anyone use it? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130222/0e817311/attachment.html From andrew at cassidywebservices.co.uk Fri Feb 22 14:00:39 2013 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Fri, 22 Feb 2013 11:00:39 +0000 Subject: [Freeswitch-users] Jira does not work any more? In-Reply-To: <5478EF41-5827-4538-BF3A-A214B7A9B98E@freeswitch.org> References: <51269FC1.7030305@gmail.com> <5478EF41-5827-4538-BF3A-A214B7A9B98E@freeswitch.org> Message-ID: Can we T.38 directly to a sip address for that number though? I have some PDF faxing stuff set up :) On 21 February 2013 23:26, Brian West wrote: > Well that number is REAL, its T.38 capable from the provider. So it would > have worked thats kinda the FUNNY joke if it all LOL. > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH_Wire > T: +1.918.420.9266 | F: +1.918.420.9267 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9266 > ISN: 410*543 > > > > > > > On Feb 21, 2013, at 4:44 PM, Dmitry Lysenko > wrote: > > > My first thought: It's time to think about T38. ) > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130222/f713c804/attachment-0001.html From alex at digitalmail.com Fri Feb 22 14:04:54 2013 From: alex at digitalmail.com (Alex Lake) Date: Fri, 22 Feb 2013 11:04:54 +0000 Subject: [Freeswitch-users] Short word drop-outs Message-ID: <512750D6.5060401@digitalmail.com> Got a customer who's complaining that short words are being lost in conversations over VOIP/Freeswitch. I presume this is over-enthusiastic silence suppression which could be tweaked in the freeswitch config. I wondered if this might be a handset config issue, but they claim that they can use their handsets with other VOIP providers that don't show the same problem. Any tips? From kkgp20 at gmail.com Fri Feb 22 15:13:59 2013 From: kkgp20 at gmail.com (K K) Date: Fri, 22 Feb 2013 13:13:59 +0100 Subject: [Freeswitch-users] [mod_fsv] How the video recording really looks like? Message-ID: Hi, I would like to record the both sides video on FreeSWITCH during the video call between two UAC. I would like to record each video call separately. I have tried the mod_fsv by adding following configuration line: under the The problem is that after adding that now when I make a call to any extension I am automatically connected and the *.fsv file is being created. The party which I have called did not receive any incoming call. It seems that I am getting automatically the connection to FreeSWITCH which is recording my video and not moving my call to the destination. So i guess that is the idea of "video recording" whit mod_fsv? I am not able to achieve my goal? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130222/f4399f0e/attachment.html From frederick at targointernet.com Fri Feb 22 16:26:59 2013 From: frederick at targointernet.com (Frederick Pruneau) Date: Fri, 22 Feb 2013 08:26:59 -0500 Subject: [Freeswitch-users] Outgoing calls from unknown users Message-ID: <51277223.1050902@targointernet.com> Hi everyone! I have found in the log files some international calls from unknown extensions. These extensions don't exist in my configuration. I tried to block them in my firewall (iptables on my freeswitch server) but they always use random IP adresses. Here is a short part of my Master.csv: "1001","1001","0015972595646444","2013-02-22 02:05:27","","2013-02-22 02:05:27","0","NORMAL_CLEARING","3c876eae-7cbe-11e2-877f-b791adff5763","","","","" "1001","1001","9011972595646444","2013-02-22 02:05:28","","2013-02-22 02:05:28","0","NORMAL_CLEARING","3d0d058c-7cbe-11e2-8783-b791adff5763","","","","" "1001","1001","2011972595646444","2013-02-22 02:05:29","","2013-02-22 02:05:29","0","NORMAL_CLEARING","3da55576-7cbe-11e2-8787-b791adff5763","","","","" "1001","1001","3011972595646444","2013-02-22 02:05:30","","2013-02-22 02:05:30","0","NORMAL_CLEARING","3e4727ca-7cbe-11e2-878b-b791adff5763","","","","" "1001","1001","4011972595646444","2013-02-22 02:05:31","","2013-02-22 02:05:31","0","NORMAL_CLEARING","3eecc2e8-7cbe-11e2-878f-b791adff5763","","","","" "1001","1001","5011972595646444","2013-02-22 02:05:32","","2013-02-22 02:05:32","0","NORMAL_CLEARING","3f633b94-7cbe-11e2-8793-b791adff5763","","","","" "1001","1001","6011972595646444","2013-02-22 02:05:33","","2013-02-22 02:05:33","0","NORMAL_CLEARING","3fc49902-7cbe-11e2-8797-b791adff5763","","","","" "1001","1001","7011972595646444","2013-02-22 02:05:33","","2013-02-22 02:05:33","0","NORMAL_CLEARING","403c0622-7cbe-11e2-879b-b791adff5763","","","","" "1001","1001","8011972595646444","2013-02-22 02:05:34","","2013-02-22 02:05:34","0","NORMAL_CLEARING","40e61ef0-7cbe-11e2-879f-b791adff5763","","","","" With my configuration, I need to be registered to make a call. I tried to call with an unregistered phone and I was not able to make a call. I don't know how they are able to do this but I need to block them. Is there something that I am missing in my configuration to block unwanted extensions to make calls? Thanks in advance! Fred From krice at freeswitch.org Fri Feb 22 17:00:34 2013 From: krice at freeswitch.org (Ken Rice) Date: Fri, 22 Feb 2013 08:00:34 -0600 Subject: [Freeswitch-users] Outgoing calls from unknown users In-Reply-To: <51277223.1050902@targointernet.com> Message-ID: This is someone trying to hack your system... This is VERY common... You will notice that each of these calls are to the same number, but they are trying different prefixes... How to avoid this, A) make sure you are not using the default username and passwords for registered sip users B) don't allow unauthenticated calls to go back out to the PSTN C) Use appropriate firewall rules to only allow places you should be getting calls from D) use something like Fail2Ban to block people attempting to make repeated failed calls/registration attempts in a short period of time... fail2ban will cu the crap out pretty quick as the attempts are from automatted scripts... On 2/22/13 7:26 AM, "Frederick Pruneau" wrote: > Hi everyone! > > I have found in the log files some international calls from unknown > extensions. These extensions don't exist in my configuration. I tried to > block them in my firewall (iptables on my freeswitch server) but they > always use random IP adresses. Here is a short part of my Master.csv: > > "1001","1001","0015972595646444","2013-02-22 02:05:27","","2013-02-22 > 02:05:27","0","NORMAL_CLEARING","3c876eae-7cbe-11e2-877f-b791adff5763","",""," > ","" > "1001","1001","9011972595646444","2013-02-22 02:05:28","","2013-02-22 > 02:05:28","0","NORMAL_CLEARING","3d0d058c-7cbe-11e2-8783-b791adff5763","",""," > ","" > "1001","1001","2011972595646444","2013-02-22 02:05:29","","2013-02-22 > 02:05:29","0","NORMAL_CLEARING","3da55576-7cbe-11e2-8787-b791adff5763","",""," > ","" > "1001","1001","3011972595646444","2013-02-22 02:05:30","","2013-02-22 > 02:05:30","0","NORMAL_CLEARING","3e4727ca-7cbe-11e2-878b-b791adff5763","",""," > ","" > "1001","1001","4011972595646444","2013-02-22 02:05:31","","2013-02-22 > 02:05:31","0","NORMAL_CLEARING","3eecc2e8-7cbe-11e2-878f-b791adff5763","",""," > ","" > "1001","1001","5011972595646444","2013-02-22 02:05:32","","2013-02-22 > 02:05:32","0","NORMAL_CLEARING","3f633b94-7cbe-11e2-8793-b791adff5763","",""," > ","" > "1001","1001","6011972595646444","2013-02-22 02:05:33","","2013-02-22 > 02:05:33","0","NORMAL_CLEARING","3fc49902-7cbe-11e2-8797-b791adff5763","",""," > ","" > "1001","1001","7011972595646444","2013-02-22 02:05:33","","2013-02-22 > 02:05:33","0","NORMAL_CLEARING","403c0622-7cbe-11e2-879b-b791adff5763","",""," > ","" > "1001","1001","8011972595646444","2013-02-22 02:05:34","","2013-02-22 > 02:05:34","0","NORMAL_CLEARING","40e61ef0-7cbe-11e2-879f-b791adff5763","",""," > ","" > > With my configuration, I need to be registered to make a call. I tried > to call with an unregistered phone and I was not able to make a call. I > don't know how they are able to do this but I need to block them. Is > there something that I am missing in my configuration to block unwanted > extensions to make calls? > > Thanks in advance! > > > Fred > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch From benkokakao at gmail.com Fri Feb 22 17:01:53 2013 From: benkokakao at gmail.com (Christian Benke) Date: Fri, 22 Feb 2013 15:01:53 +0100 Subject: [Freeswitch-users] Outgoing calls from unknown users In-Reply-To: <51277223.1050902@targointernet.com> References: <51277223.1050902@targointernet.com> Message-ID: Do you have a logfile of these calls? Can you please paste it to http://pastebin.freeswitch.org/. Otherwise, please paste your dialplans to pastebin so we can figure out what's really happening with your calls, the csv has too little information. Do all of these calls have 0 billsec? If possible, you should turn off FreeSWITCH till you know the reason for this calls, it looks very much like your system is not safe. Best regards, Christian -- Central Asia by bike, starting May 2013 - http://poab.org On 22 February 2013 14:26, Frederick Pruneau wrote: > Hi everyone! > > I have found in the log files some international calls from unknown > extensions. These extensions don't exist in my configuration. I tried to > block them in my firewall (iptables on my freeswitch server) but they > always use random IP adresses. Here is a short part of my Master.csv: > > "1001","1001","0015972595646444","2013-02-22 02:05:27","","2013-02-22 > 02:05:27","0","NORMAL_CLEARING","3c876eae-7cbe-11e2-877f-b791adff5763","","","","" > "1001","1001","9011972595646444","2013-02-22 02:05:28","","2013-02-22 > 02:05:28","0","NORMAL_CLEARING","3d0d058c-7cbe-11e2-8783-b791adff5763","","","","" > "1001","1001","2011972595646444","2013-02-22 02:05:29","","2013-02-22 > 02:05:29","0","NORMAL_CLEARING","3da55576-7cbe-11e2-8787-b791adff5763","","","","" > "1001","1001","3011972595646444","2013-02-22 02:05:30","","2013-02-22 > 02:05:30","0","NORMAL_CLEARING","3e4727ca-7cbe-11e2-878b-b791adff5763","","","","" > "1001","1001","4011972595646444","2013-02-22 02:05:31","","2013-02-22 > 02:05:31","0","NORMAL_CLEARING","3eecc2e8-7cbe-11e2-878f-b791adff5763","","","","" > "1001","1001","5011972595646444","2013-02-22 02:05:32","","2013-02-22 > 02:05:32","0","NORMAL_CLEARING","3f633b94-7cbe-11e2-8793-b791adff5763","","","","" > "1001","1001","6011972595646444","2013-02-22 02:05:33","","2013-02-22 > 02:05:33","0","NORMAL_CLEARING","3fc49902-7cbe-11e2-8797-b791adff5763","","","","" > "1001","1001","7011972595646444","2013-02-22 02:05:33","","2013-02-22 > 02:05:33","0","NORMAL_CLEARING","403c0622-7cbe-11e2-879b-b791adff5763","","","","" > "1001","1001","8011972595646444","2013-02-22 02:05:34","","2013-02-22 > 02:05:34","0","NORMAL_CLEARING","40e61ef0-7cbe-11e2-879f-b791adff5763","","","","" > > With my configuration, I need to be registered to make a call. I tried > to call with an unregistered phone and I was not able to make a call. I > don't know how they are able to do this but I need to block them. Is > there something that I am missing in my configuration to block unwanted > extensions to make calls? > > Thanks in advance! > > > Fred > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 vitaliy.davudov at vts24.ru Fri Feb 22 17:12:39 2013 From: vitaliy.davudov at vts24.ru (=?UTF-8?B?0JLQuNGC0LDQu9C40Lkg0JTQsNCy0YPQtNC+0LI=?=) Date: Fri, 22 Feb 2013 18:12:39 +0400 Subject: [Freeswitch-users] Memory growth In-Reply-To: <5125AD25.6020005@vts24.ru> References: <7D4D7EF3-9889-4C06-A1E4-26225C86E791@mgtech.com> <5125AD25.6020005@vts24.ru> Message-ID: <51277CD7.20008@vts24.ru> Need a little help in the following question: can I apply the patch http://fisheye.freeswitch.org/rdiff/freeswitch.git?csid=45d849ab746e6fc0833820851dced1dc61e0fbe1&u&N to the file libs/sofia-sip/libsofia-sip-ua/su/su_time0.c? Would it be necessary to change anything else? Then I would be able to immediately begin testing... 21.02.2013 9:14, ??????? ??????? ?????: > I would be glad to do it, but my FS in git branch v1.2.stable (it is > in production environment). I will have to wait patches to the 1.2 series. > > > 21.02.2013 3:42, Mario G ?????: >> FYI, Anthony nailed my memory leak yesterday (see the jira below). 24 >> hours of testing aok so far. The leak varied between 300k-2M per >> hour, not sure if it was only OSX related so good idea for anyone >> with a leak to test with updated HEAD. >> Mario G >> >> On Feb 18, 2013, at 11:22 AM, Ken Rice wrote: >> >>> Re: [Freeswitch-users] Memory growth >>> We?re trying to figure out where this last bit of leaking is coming >>> from... Looks like it might be related to Registrations some how... >>> >>> Once we squash that there will be some more info on the 1.2 branch >>> coming >>> >>> >>> On 2/18/13 12:19 PM, "Mario G" >> > wrote: >>> >>>> FYI/in case it's related: I had a substantial memory leak on the >>>> Oct 25 though Jan 29 Head versions and 1.2.3. After Feb 11 head, >>>> now a much smaller leak I can't pinpoint, see >>>> http://jira.freeswitch.org/browse/FS-5095#comment-39379. Updating >>>> to HEAD will probably fix it, I hope someone else sees the slow >>>> leak since I am getting nowhere real fast on it..... >>>> Mario G >>>> >>>> On Feb 18, 2013, at 3:16 AM, Steven Ayre wrote: >>>> >>>>> Are you able to reproduce it on a newer version? Try current head >>>>> of the master branch. >>>>> >>>>> If you can, then try running FS through valgrind which may reveal >>>>> the source of any memory leaks. >>>>> >>>>> -Steve >>>>> >>>>> >>>>> >>>>> On 18 February 2013 11:00, ??????? ??????? >>>>> > wrote: >>>>>> Hi, list! >>>>>> I've installed FreeSWITCH Version >>>>>> 1.2.3+git~20120920T220849Z~f718a5e8e6 >>>>>> (1.2.3; git at commit f718a5e8e6 on Thu, 20 Sep 2012 22:08:49 Z) >>>>>> >>>>>> I noticed that FreeSwtich uses a lot of memory of my system, and >>>>>> percent >>>>>> of used memory is gradually growing regardless of the number of the >>>>>> simultaneous calls: >>>>>> >>>>>> # uname -a >>>>>> Linux freeswitch1 2.6.18-308.13.1.el5 #1 SMP Tue Aug 21 17:10:18 EDT >>>>>> 2012 x86_64 x86_64 x86_64 GNU/Linux >>>>>> >>>>>> # top >>>>>> top - 14:21:39 up 140 days, 29 min, 2 users, load average: >>>>>> 0.11, 0.14, >>>>>> 0.10 >>>>>> Tasks: 99 total, 1 running, 96 sleeping, 2 stopped, 0 zombie >>>>>> Cpu(s): 1.2%us, 0.8%sy, 0.0%ni, 96.9%id, 0.1%wa, 0.2%hi, 0.7%si, >>>>>> 0.0%st >>>>>> Mem: 3781680k total, 3748360k used, 33320k free, 120708k >>>>>> buffers >>>>>> Swap: 2096472k total, 17224k used, 2079248k free, 1838356k >>>>>> cached >>>>>> >>>>>> PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND >>>>>> 9584 root -2 -10 1792m 1.3g 8252 S 2.7 36.8 935:07.85 >>>>>> freeswitch >>>>>> 3095 mysql 15 0 329m 28m 4588 S 0.7 0.8 713:07.03 mysqld >>>>>> >>>>>> At this moment FS handles no more 15 simultaneous calls, and has >>>>>> about >>>>>> 117 registered users. >>>>>> Is there a tool to monitor, which of processes or modules in FS >>>>>> is using >>>>>> memory and how much? >>>>>> >>>>>> -- >>>>>> Best regards, >>>>>> Vitaly. >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> >>>>>> http://wiki.freeswitch.org >>>>>> >>>>>> http://www.cluecon.com >>>>>> >>>>>> >>>>>> 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 >>>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> 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 >>>> >>>> >>>> ------------------------------------------------------------------------ >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>> >>> -- >>> Ken >>> _http://www.FreeSWITCH.org >>> http://www.ClueCon.com >>> http://www.OSTAG.org >>> _irc.freenode.net #freeswitch >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 > > -- > ? ?????????? ???????????, > ??????? ??????? ????????? > ??? "???-???????-??????" > (?????? ???????? "ETERIA") > http://www.vts24.ru > ???: (495) 989-47-00 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -- ? ?????????? ???????????, ??????? ??????? ????????? ??? "???-???????-??????" (?????? ???????? "ETERIA") http://www.vts24.ru ???: (495) 989-47-00 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130222/6358fc90/attachment-0001.html From pm_zefman_r at mail.ru Fri Feb 22 17:17:27 2013 From: pm_zefman_r at mail.ru (=?UTF-8?B?RG1pdHJpeSBTaHVtYWV2?=) Date: Fri, 22 Feb 2013 18:17:27 +0400 Subject: [Freeswitch-users] =?utf-8?q?Repeated_extensions?= Message-ID: <1361542646.743454162@f319.mail.ru> Hello world) I have session recording extension: ? ? ?? ?? ? ? ? ? ? ? ? ?- ? ??? ? ? ? ?? ? in the very beginnin of my dialplan. But there are some (IVRs with transfer)?and (Group number with loopback) in my dialplan. So recording extension is hited multiple times and as a result I have several files with the "same" recorded session (IVR, Group number, Answered subscriber). What should I do to prohibit repeated hits of "record_3..16_digits" extension? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130222/f36e4355/attachment.html From levend.sayar at karel.com.tr Fri Feb 22 18:36:55 2013 From: levend.sayar at karel.com.tr (Levend Sayar) Date: Fri, 22 Feb 2013 15:36:55 +0000 Subject: [Freeswitch-users] optional SRTP Message-ID: Hi again. On my previous discussion on this list, i learned that RTP/AVP with a a=crypto attribute means optional SRTP. I suppose FS supports optional SRTP, but when i look at the sofia code, afaiu fs does not support it. Right ? A code snippet from sofia_glue.c if (m->m_proto != sdp_proto_srtp) { switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_ERROR, "a=crypto in RTP/AVP, refer to rfc3711\n"); match = 0; goto done; } -- _lvnd_ {^_^} -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130222/6c85c6ae/attachment.html From frederick at targointernet.com Fri Feb 22 18:48:26 2013 From: frederick at targointernet.com (Frederick Pruneau) Date: Fri, 22 Feb 2013 10:48:26 -0500 Subject: [Freeswitch-users] Outgoing calls from unknown users In-Reply-To: References: <51277223.1050902@targointernet.com> Message-ID: <5127934A.5080703@targointernet.com> PB 20618 Everytime, it is a 0 billsec. For now, international calls are not authorized. But in a near future, I want to authorized them. I verified what Ken wrote: A) make sure you are not using the default username and passwords for registered sip users *(not using default username and password. All default users/extensions have been removed)* B) don't allow unauthenticated calls to go back out to the PSTN *(I don't have PSTN lines)* C) Use appropriate firewall rules to only allow places you should be getting calls from *(Already done)* D) use something like Fail2Ban to block people attempting to make repeated failed calls/registration attempts in a short period of time... Actually, I have fail2ban that blocks registration attemps. I don't know how to block failed calls. Can you guide me to a web site or help me to ban failed calls? Thank you for your quick replies! Fred Le 2013-02-22 09:01, Christian Benke a ?crit : > Do you have a logfile of these calls? Can you please paste it to > http://pastebin.freeswitch.org/. Otherwise, please paste your > dialplans to pastebin so we can figure out what's really happening > with your calls, the csv has too little information. Do all of these > calls have 0 billsec? > > If possible, you should turn off FreeSWITCH till you know the reason > for this calls, it looks very much like your system is not safe. > > Best regards, > Christian > > -- > Central Asia by bike, starting May 2013 - http://poab.org > > > On 22 February 2013 14:26, Frederick Pruneau > wrote: >> Hi everyone! >> >> I have found in the log files some international calls from unknown >> extensions. These extensions don't exist in my configuration. I tried to >> block them in my firewall (iptables on my freeswitch server) but they >> always use random IP adresses. Here is a short part of my Master.csv: >> >> "1001","1001","0015972595646444","2013-02-22 02:05:27","","2013-02-22 >> 02:05:27","0","NORMAL_CLEARING","3c876eae-7cbe-11e2-877f-b791adff5763","","","","" >> "1001","1001","9011972595646444","2013-02-22 02:05:28","","2013-02-22 >> 02:05:28","0","NORMAL_CLEARING","3d0d058c-7cbe-11e2-8783-b791adff5763","","","","" >> "1001","1001","2011972595646444","2013-02-22 02:05:29","","2013-02-22 >> 02:05:29","0","NORMAL_CLEARING","3da55576-7cbe-11e2-8787-b791adff5763","","","","" >> "1001","1001","3011972595646444","2013-02-22 02:05:30","","2013-02-22 >> 02:05:30","0","NORMAL_CLEARING","3e4727ca-7cbe-11e2-878b-b791adff5763","","","","" >> "1001","1001","4011972595646444","2013-02-22 02:05:31","","2013-02-22 >> 02:05:31","0","NORMAL_CLEARING","3eecc2e8-7cbe-11e2-878f-b791adff5763","","","","" >> "1001","1001","5011972595646444","2013-02-22 02:05:32","","2013-02-22 >> 02:05:32","0","NORMAL_CLEARING","3f633b94-7cbe-11e2-8793-b791adff5763","","","","" >> "1001","1001","6011972595646444","2013-02-22 02:05:33","","2013-02-22 >> 02:05:33","0","NORMAL_CLEARING","3fc49902-7cbe-11e2-8797-b791adff5763","","","","" >> "1001","1001","7011972595646444","2013-02-22 02:05:33","","2013-02-22 >> 02:05:33","0","NORMAL_CLEARING","403c0622-7cbe-11e2-879b-b791adff5763","","","","" >> "1001","1001","8011972595646444","2013-02-22 02:05:34","","2013-02-22 >> 02:05:34","0","NORMAL_CLEARING","40e61ef0-7cbe-11e2-879f-b791adff5763","","","","" >> >> With my configuration, I need to be registered to make a call. I tried >> to call with an unregistered phone and I was not able to make a call. I >> don't know how they are able to do this but I need to block them. Is >> there something that I am missing in my configuration to block unwanted >> extensions to make calls? >> >> Thanks in advance! >> >> >> Fred >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -- Fr?d?rick Pruneau Administrateur r?seau | Network administrator Targo Communications Ste-Clotilde : (450) 826-0031 Montr?al : (514) 448-0773 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130222/22908711/attachment.html From steveayre at gmail.com Fri Feb 22 18:52:27 2013 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 22 Feb 2013 16:52:27 +0100 Subject: [Freeswitch-users] Does Freeswitch support CRTP? In-Reply-To: References: Message-ID: <3FCBC4B6-A8C0-4B88-8008-6BF79C7071CA@gmail.com> CRTP works by compressing the IP and UDP headers. That means it cannot be used over mobile networks. The audio is already compressed so cannot be compressed further - RTP is as light as you can get while still being routable via the Internet. You should compare the different codec options. There is a large difference in the bandwidth use of different codecs. Eg compare G711 vs GSM, CELT, Speex etc. Other advice: Increasing the ptime will fit a longer period of time into a single packet so reduce the IP/UDP/RTP header overhead. Also increasing the jitter buffer length may help compensate for the heavy jitter you may see on a mobile connection. Steve On 22 Feb 2013, at 03:44, ZhouJianhua wrote: > RTP's head is too heavy for mobile network. There is little about CRTP on google, anyone use it? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130222/9d29943d/attachment-0001.html From krice at freeswitch.org Fri Feb 22 19:03:29 2013 From: krice at freeswitch.org (Ken Rice) Date: Fri, 22 Feb 2013 10:03:29 -0600 Subject: [Freeswitch-users] Outgoing calls from unknown users In-Reply-To: <5127934A.5080703@targointernet.com> Message-ID: If you were seeing a billtime of 0 that means the calls were blocked... Some one was probably hitting the ?external? interface then hitting the public context... This is allowed in the default example configs for freeswitch, but calls coming in that way are only allowed to actually do something if a) they hit one of the pre-defined local extensions or you have modified it to allow other calling. Why is it like this? This is so you can define your local extensions and DIDs in the public context then say hey you can just call me via sip to SIP:mynumber_or_extension at hostname.or.ip.of.my.FS.box.com Bots will scan the internet (the entire internet) and find your public profile is not doing sip challenge response, then they will try to place calls. In your logs they were trying to call a number in what appears to be Palestine, this is very comon, as they are probing to see if you let the calls pass by trying various prefixes... As long as you are configuring the unauthenticated interfaces to only allow calls for your local extensions this is not a big dead, they will give up and go away. On 2/22/13 9:48 AM, "Frederick Pruneau" wrote: > > PB 20618 > > Everytime, it is a 0 billsec. For now, international calls are not > authorized. But in a near future, I want to authorized them. > > I verified what Ken wrote: > > A) make sure you are not using the default username and passwords for > registered sip users (not using default username and password. All default > users/extensions have been removed) > B) don't allow unauthenticated calls to go back out to the PSTN (I don't have > PSTN lines) > C) Use appropriate firewall rules to only allow places you should be getting > calls from (Already done) > D) use something like Fail2Ban to block people attempting to make repeated > failed calls/registration attempts in a short period of time... > > Actually, I have fail2ban that blocks registration attemps. I don't know how > to block failed calls. Can you guide me to a web site or help me to ban failed > calls? > > Thank you for your quick replies! > > Fred > > Le 2013-02-22 09:01, Christian Benke a ?crit : > > >> >> Do you have a logfile of these calls? Can you please paste it to >> http://pastebin.freeswitch.org/. Otherwise, please paste your >> dialplans to pastebin so we can figure out what's really happening >> with your calls, the csv has too little information. Do all of these >> calls have 0 billsec? >> >> If possible, you should turn off FreeSWITCH till you know the reason >> for this calls, it looks very much like your system is not safe. >> >> Best regards, >> Christian >> >> -- >> Central Asia by bike, starting May 2013 - http://poab.org >> >> >> On 22 February 2013 14:26, Frederick Pruneau >> wrote: >> >>> >>> Hi everyone! >>> >>> I have found in the log files some international calls from unknown >>> extensions. These extensions don't exist in my configuration. I tried to >>> block them in my firewall (iptables on my freeswitch server) but they >>> always use random IP adresses. Here is a short part of my Master.csv: >>> >>> "1001","1001","0015972595646444","2013-02-22 02:05:27","","2013-02-22 >>> 02:05:27","0","NORMAL_CLEARING","3c876eae-7cbe-11e2-877f-b791adff5763","","" >>> ,"","" >>> "1001","1001","9011972595646444","2013-02-22 02:05:28","","2013-02-22 >>> 02:05:28","0","NORMAL_CLEARING","3d0d058c-7cbe-11e2-8783-b791adff5763","","" >>> ,"","" >>> "1001","1001","2011972595646444","2013-02-22 02:05:29","","2013-02-22 >>> 02:05:29","0","NORMAL_CLEARING","3da55576-7cbe-11e2-8787-b791adff5763","","" >>> ,"","" >>> "1001","1001","3011972595646444","2013-02-22 02:05:30","","2013-02-22 >>> 02:05:30","0","NORMAL_CLEARING","3e4727ca-7cbe-11e2-878b-b791adff5763","","" >>> ,"","" >>> "1001","1001","4011972595646444","2013-02-22 02:05:31","","2013-02-22 >>> 02:05:31","0","NORMAL_CLEARING","3eecc2e8-7cbe-11e2-878f-b791adff5763","","" >>> ,"","" >>> "1001","1001","5011972595646444","2013-02-22 02:05:32","","2013-02-22 >>> 02:05:32","0","NORMAL_CLEARING","3f633b94-7cbe-11e2-8793-b791adff5763","","" >>> ,"","" >>> "1001","1001","6011972595646444","2013-02-22 02:05:33","","2013-02-22 >>> 02:05:33","0","NORMAL_CLEARING","3fc49902-7cbe-11e2-8797-b791adff5763","","" >>> ,"","" >>> "1001","1001","7011972595646444","2013-02-22 02:05:33","","2013-02-22 >>> 02:05:33","0","NORMAL_CLEARING","403c0622-7cbe-11e2-879b-b791adff5763","","" >>> ,"","" >>> "1001","1001","8011972595646444","2013-02-22 02:05:34","","2013-02-22 >>> 02:05:34","0","NORMAL_CLEARING","40e61ef0-7cbe-11e2-879f-b791adff5763","","" >>> ,"","" >>> >>> With my configuration, I need to be registered to make a call. I tried >>> to call with an unregistered phone and I was not able to make a call. I >>> don't know how they are able to do this but I need to block them. Is >>> there something that I am missing in my configuration to block unwanted >>> extensions to make calls? >>> >>> Thanks in advance! >>> >>> >>> Fred >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> > > > -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130222/f0192204/attachment.html From krice at freeswitch.org Fri Feb 22 19:08:40 2013 From: krice at freeswitch.org (Ken Rice) Date: Fri, 22 Feb 2013 10:08:40 -0600 Subject: [Freeswitch-users] optional SRTP In-Reply-To: Message-ID: FreeSWITCH Support SIP/TLS, SRTP and ZRTP, the error message you are refering to below is where someone has improperly implemented SRTP and is sending the crypto headers in the wrong spots. See the Wiki for setting up secure calling. Also please note, if you are going to use SRTP, you _must_ use SIP/TLS as the keys for SRTP are passed in the SDP of the SIP messaging. You can use ZRTP without using SIP/TLS as the key exchange for ZRTP happens as a DH exchange in the ZRTP stream, but if you want to protect the contents of the SIP messaging you?ll still want to use SIP/TLS. K On 2/22/13 9:36 AM, "Levend Sayar" wrote: > Hi again. > > On my previous discussion on this list, i learned that RTP/AVP with a > a=crypto attribute means optional SRTP. > > I suppose FS supports optional SRTP, but when i look at the sofia code, afaiu > fs does not support it. > > Right ? > > A code snippet from sofia_glue.c > > if (m->m_proto != sdp_proto_srtp) { > switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_ERROR, > "a=crypto in RTP/AVP, refer to rfc3711\n"); > match = 0; > goto done; > } > > > > > > -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130222/cfea20c0/attachment.html From brian at freeswitch.org Fri Feb 22 19:27:20 2013 From: brian at freeswitch.org (Brian West) Date: Fri, 22 Feb 2013 10:27:20 -0600 Subject: [Freeswitch-users] optional SRTP In-Reply-To: References: Message-ID: crypto in RTP/AVP is invalid per the spec. You are required to send two rtp profiles one with RTP/AVP or RTP/SAVP To get around this in your broken device you can set the variable sip_allow_crypto_in_avp or set profile param NDLB-allow-crypto-in-avp -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire T: +1.918.420.9266 | F: +1.918.420.9267 | M: +1.918.424.WEST iNUM: +883 5100 1420 9266 ISN: 410*543 On Feb 22, 2013, at 10:08 AM, Ken Rice wrote: > FreeSWITCH Support SIP/TLS, SRTP and ZRTP, the error message you are refering to below is where someone has improperly implemented SRTP and is sending the crypto headers in the wrong spots. See the Wiki for setting up secure calling. > > Also please note, if you are going to use SRTP, you _must_ use SIP/TLS as the keys for SRTP are passed in the SDP of the SIP messaging. You can use ZRTP without using SIP/TLS as the key exchange for ZRTP happens as a DH exchange in the ZRTP stream, but if you want to protect the contents of the SIP messaging you?ll still want to use SIP/TLS. > > > K From levend.sayar at karel.com.tr Fri Feb 22 19:30:02 2013 From: levend.sayar at karel.com.tr (Levend Sayar) Date: Fri, 22 Feb 2013 16:30:02 +0000 Subject: [Freeswitch-users] optional SRTP In-Reply-To: References: Message-ID: Ken you are right. But what i want to do is not securing all the conversation. I dont need that. What i want is FS to prefer RTP when an optional SRTP offer is received. In this list, it is mentioned that m=audio RTP/AVP a=crypto ... means optional SRTP. Namely caller prefers to talk with SRTP, but if calle prefers RTP, caller accept RTP too. First of all, the SDP I show is improper ? -- ?yi ?al??malar. _lvnd_ {^_^} KAREL`UCAP On Fri, 2013-02-22 at 10:08 -0600, Ken Rice wrote: FreeSWITCH Support SIP/TLS, SRTP and ZRTP, the error message you are refering to below is where someone has improperly implemented SRTP and is sending the crypto headers in the wrong spots. See the Wiki for setting up secure calling. Also please note, if you are going to use SRTP, you _must_ use SIP/TLS as the keys for SRTP are passed in the SDP of the SIP messaging. You can use ZRTP without using SIP/TLS as the key exchange for ZRTP happens as a DH exchange in the ZRTP stream, but if you want to protect the contents of the SIP messaging you?ll still want to use SIP/TLS. K On 2/22/13 9:36 AM, "Levend Sayar" wrote: Hi again. On my previous discussion on this list, i learned that RTP/AVP with a a=crypto attribute means optional SRTP. I suppose FS supports optional SRTP, but when i look at the sofia code, afaiu fs does not support it. Right ? A code snippet from sofia_glue.c if (m->m_proto != sdp_proto_srtp) { switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_ERROR, "a=crypto in RTP/AVP, refer to rfc3711\n"); match = 0; goto done; } plain text document attachment (ATT00001) _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130222/bcfaa7eb/attachment.html From krice at freeswitch.org Fri Feb 22 19:36:34 2013 From: krice at freeswitch.org (Ken Rice) Date: Fri, 22 Feb 2013 10:36:34 -0600 Subject: [Freeswitch-users] optional SRTP In-Reply-To: Message-ID: See rfc3711 that describes how you are supposed to pass the crypto information. If you want to disable SRTP on all calls, that?s a channel variable for that you just set Now you can use the dialplan to look at the SDP, and determine if you want to set if you want to have SRTP enabled or not See http://wiki.freeswitch.org/wiki/SRTP for more information K On 2/22/13 10:30 AM, "Levend Sayar" wrote: > Ken you are right. But what i want to do is not securing all the conversation. > I dont need that. > What i want is FS to prefer RTP when an optional SRTP offer is received. > > > In this list, it is mentioned that > > m=audio RTP/AVP > a=crypto ... > > means optional SRTP. Namely caller prefers to talk with SRTP, but if calle > prefers RTP, caller accept RTP too. > > First of all, the SDP I show is improper ? > > -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130222/e12899d4/attachment.html From sertys at gmail.com Fri Feb 22 19:59:25 2013 From: sertys at gmail.com (Daniel Ivanov) Date: Fri, 22 Feb 2013 17:59:25 +0100 Subject: [Freeswitch-users] Changing codec during calls In-Reply-To: <564748CC-A8D1-44ED-9AC4-55910DBB7F2A@kavun.ch> References: <6EF8B8E9-4CC5-480C-B1C1-09CBC533470C@kavun.ch> <35AF9E79-771D-440A-BEBB-A856D831161C@kavun.ch> <564748CC-A8D1-44ED-9AC4-55910DBB7F2A@kavun.ch> Message-ID: Freeswitch is turning into a mean, mean event-driven voip panacea... Congrats on the new commit. On Feb 22, 2013 6:49 AM, "Emrah" wrote: > Hi guys, > > Sorry was disconnected from the list for a while. > Indeed it looks like the latest commit fixed it. > The issue I see now is that the channel info are not updated and a show > channels will only list the initial codecs. > > Great job and thanks a bunch, > Emrah > On Feb 21, 2013, at 8:42 PM, Dmitry Lysenko > wrote: > > > Now this cool feature is fully working! Today Anthony committed the > bugfix! > > Tested with 2 different SIP UA<->FS<->Callcentric. Both legs. > > Now possible to write application that will switch codecs on the fly, > regarding on any event, such as switching to low bandwidth backup internet > line or too high cpu load, etc. > > BTW, is there way to get API access to statistic of jitter buffer? > > Thanks. > > > > P.S. As I know, asterisk can't do such thing. > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130222/8421d9de/attachment.html From steveayre at gmail.com Fri Feb 22 20:05:03 2013 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 22 Feb 2013 18:05:03 +0100 Subject: [Freeswitch-users] optional SRTP In-Reply-To: References: Message-ID: <575B91FD-D404-4BD2-BBDE-59FDD6ABD90F@gmail.com> The confusion was mine on an earlier thread. I said that, based on something on another mailing list that I can't find now (sip-implementors?). That post might have been inspired by this which states RTP/AVP with a=crypto allows SRTP desired but not required - but only because it 'deviates from the specification in RFC4568' http://msdn.microsoft.com/en-us/library/dd948772(v=office.12).aspx I guess such devices are what the NDLB parameter is for. The previous thread was 'SRTP disabling' where Levand has a phone that allows optional SRTP but only offers RTP/SAVP in its SDP with no RTP/AVP profile offered so FS cannot/won't send plain RTP to the SRTP port. Which I'd say is correct behaviour and a broken phone. Steve On 22 Feb 2013, at 17:27, Brian West wrote: > crypto in RTP/AVP is invalid per the spec. > > You are required to send two rtp profiles one with RTP/AVP or RTP/SAVP > > To get around this in your broken device you can set the variable sip_allow_crypto_in_avp or set profile param NDLB-allow-crypto-in-avp > > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH_Wire > T: +1.918.420.9266 | F: +1.918.420.9267 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9266 > ISN: 410*543 > > > > > > > On Feb 22, 2013, at 10:08 AM, Ken Rice wrote: > >> FreeSWITCH Support SIP/TLS, SRTP and ZRTP, the error message you are refering to below is where someone has improperly implemented SRTP and is sending the crypto headers in the wrong spots. See the Wiki for setting up secure calling. >> >> Also please note, if you are going to use SRTP, you _must_ use SIP/TLS as the keys for SRTP are passed in the SDP of the SIP messaging. You can use ZRTP without using SIP/TLS as the key exchange for ZRTP happens as a DH exchange in the ZRTP stream, but if you want to protect the contents of the SIP messaging you?ll still want to use SIP/TLS. >> >> >> K > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130222/a632a11f/attachment-0001.html From david.villasmil.work at gmail.com Fri Feb 22 20:23:57 2013 From: david.villasmil.work at gmail.com (David Villasmil) Date: Fri, 22 Feb 2013 18:23:57 +0100 Subject: [Freeswitch-users] Outgoing calls from unknown users In-Reply-To: References: Message-ID: Since a long time ago i NEVER use ports 5060/5061/5080/5081... Security by obscurity also helps (in my experience), in addition to fail2ban... You must ALWAYS setup fail2ban if you don't know the IPs of your customers and so a firewall won't do much good. But firewall+fail2ban+non-standard-ports is pretty safe (of course in addition to authenticating) David On Feb 22, 2013, at 17:03, Ken Rice wrote: > If you were seeing a billtime of 0 that means the calls were blocked... Some one was probably hitting the ?external? interface then hitting the public context... This is allowed in the default example configs for freeswitch, but calls coming in that way are only allowed to actually do something if a) they hit one of the pre-defined local extensions or you have modified it to allow other calling. > > Why is it like this? This is so you can define your local extensions and DIDs in the public context then say hey you can just call me via sip to SIP:mynumber_or_extension at hostname.or.ip.of.my.FS.box.com > > Bots will scan the internet (the entire internet) and find your public profile is not doing sip challenge response, then they will try to place calls. In your logs they were trying to call a number in what appears to be Palestine, this is very comon, as they are probing to see if you let the calls pass by trying various prefixes... > > As long as you are configuring the unauthenticated interfaces to only allow calls for your local extensions this is not a big dead, they will give up and go away. > > > > On 2/22/13 9:48 AM, "Frederick Pruneau" wrote: > > > PB 20618 > > Everytime, it is a 0 billsec. For now, international calls are not authorized. But in a near future, I want to authorized them. > > I verified what Ken wrote: > > A) make sure you are not using the default username and passwords for > registered sip users (not using default username and password. All default users/extensions have been removed) > B) don't allow unauthenticated calls to go back out to the PSTN (I don't have PSTN lines) > C) Use appropriate firewall rules to only allow places you should be getting > calls from (Already done) > D) use something like Fail2Ban to block people attempting to make repeated > failed calls/registration attempts in a short period of time... > > Actually, I have fail2ban that blocks registration attemps. I don't know how to block failed calls. Can you guide me to a web site or help me to ban failed calls? > > Thank you for your quick replies! > > Fred > > Le 2013-02-22 09:01, Christian Benke a ?crit : > > > > Do you have a logfile of these calls? Can you please paste it to > http://pastebin.freeswitch.org/. Otherwise, please paste your > dialplans to pastebin so we can figure out what's really happening > with your calls, the csv has too little information. Do all of these > calls have 0 billsec? > > If possible, you should turn off FreeSWITCH till you know the reason > for this calls, it looks very much like your system is not safe. > > Best regards, > Christian > > -- > Central Asia by bike, starting May 2013 - http://poab.org > > > On 22 February 2013 14:26, Frederick Pruneau > wrote: > > > Hi everyone! > > I have found in the log files some international calls from unknown > extensions. These extensions don't exist in my configuration. I tried to > block them in my firewall (iptables on my freeswitch server) but they > always use random IP adresses. Here is a short part of my Master.csv: > > "1001","1001","0015972595646444","2013-02-22 02:05:27","","2013-02-22 > 02:05:27","0","NORMAL_CLEARING","3c876eae-7cbe-11e2-877f-b791adff5763","","","","" > "1001","1001","9011972595646444","2013-02-22 02:05:28","","2013-02-22 > 02:05:28","0","NORMAL_CLEARING","3d0d058c-7cbe-11e2-8783-b791adff5763","","","","" > "1001","1001","2011972595646444","2013-02-22 02:05:29","","2013-02-22 > 02:05:29","0","NORMAL_CLEARING","3da55576-7cbe-11e2-8787-b791adff5763","","","","" > "1001","1001","3011972595646444","2013-02-22 02:05:30","","2013-02-22 > 02:05:30","0","NORMAL_CLEARING","3e4727ca-7cbe-11e2-878b-b791adff5763","","","","" > "1001","1001","4011972595646444","2013-02-22 02:05:31","","2013-02-22 > 02:05:31","0","NORMAL_CLEARING","3eecc2e8-7cbe-11e2-878f-b791adff5763","","","","" > "1001","1001","5011972595646444","2013-02-22 02:05:32","","2013-02-22 > 02:05:32","0","NORMAL_CLEARING","3f633b94-7cbe-11e2-8793-b791adff5763","","","","" > "1001","1001","6011972595646444","2013-02-22 02:05:33","","2013-02-22 > 02:05:33","0","NORMAL_CLEARING","3fc49902-7cbe-11e2-8797-b791adff5763","","","","" > "1001","1001","7011972595646444","2013-02-22 02:05:33","","2013-02-22 > 02:05:33","0","NORMAL_CLEARING","403c0622-7cbe-11e2-879b-b791adff5763","","","","" > "1001","1001","8011972595646444","2013-02-22 02:05:34","","2013-02-22 > 02:05:34","0","NORMAL_CLEARING","40e61ef0-7cbe-11e2-879f-b791adff5763","","","","" > > With my configuration, I need to be registered to make a call. I tried > to call with an unregistered phone and I was not able to make a call. I > don't know how they are able to do this but I need to block them. Is > there something that I am missing in my configuration to block unwanted > extensions to make calls? > > Thanks in advance! > > > Fred > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > > > -- > Ken > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > irc.freenode.net #freeswitch > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130222/f1ac05a1/attachment.html From david.villasmil.work at gmail.com Fri Feb 22 20:23:57 2013 From: david.villasmil.work at gmail.com (David Villasmil) Date: Fri, 22 Feb 2013 18:23:57 +0100 Subject: [Freeswitch-users] Outgoing calls from unknown users In-Reply-To: References: Message-ID: Since a long time ago i NEVER use ports 5060/5061/5080/5081... Security by obscurity also helps (in my experience), in addition to fail2ban... You must ALWAYS setup fail2ban if you don't know the IPs of your customers and so a firewall won't do much good. But firewall+fail2ban+non-standard-ports is pretty safe (of course in addition to authenticating) David On Feb 22, 2013, at 17:03, Ken Rice wrote: > If you were seeing a billtime of 0 that means the calls were blocked... Some one was probably hitting the ?external? interface then hitting the public context... This is allowed in the default example configs for freeswitch, but calls coming in that way are only allowed to actually do something if a) they hit one of the pre-defined local extensions or you have modified it to allow other calling. > > Why is it like this? This is so you can define your local extensions and DIDs in the public context then say hey you can just call me via sip to SIP:mynumber_or_extension at hostname.or.ip.of.my.FS.box.com > > Bots will scan the internet (the entire internet) and find your public profile is not doing sip challenge response, then they will try to place calls. In your logs they were trying to call a number in what appears to be Palestine, this is very comon, as they are probing to see if you let the calls pass by trying various prefixes... > > As long as you are configuring the unauthenticated interfaces to only allow calls for your local extensions this is not a big dead, they will give up and go away. > > > > On 2/22/13 9:48 AM, "Frederick Pruneau" wrote: > > > PB 20618 > > Everytime, it is a 0 billsec. For now, international calls are not authorized. But in a near future, I want to authorized them. > > I verified what Ken wrote: > > A) make sure you are not using the default username and passwords for > registered sip users (not using default username and password. All default users/extensions have been removed) > B) don't allow unauthenticated calls to go back out to the PSTN (I don't have PSTN lines) > C) Use appropriate firewall rules to only allow places you should be getting > calls from (Already done) > D) use something like Fail2Ban to block people attempting to make repeated > failed calls/registration attempts in a short period of time... > > Actually, I have fail2ban that blocks registration attemps. I don't know how to block failed calls. Can you guide me to a web site or help me to ban failed calls? > > Thank you for your quick replies! > > Fred > > Le 2013-02-22 09:01, Christian Benke a ?crit : > > > > Do you have a logfile of these calls? Can you please paste it to > http://pastebin.freeswitch.org/. Otherwise, please paste your > dialplans to pastebin so we can figure out what's really happening > with your calls, the csv has too little information. Do all of these > calls have 0 billsec? > > If possible, you should turn off FreeSWITCH till you know the reason > for this calls, it looks very much like your system is not safe. > > Best regards, > Christian > > -- > Central Asia by bike, starting May 2013 - http://poab.org > > > On 22 February 2013 14:26, Frederick Pruneau > wrote: > > > Hi everyone! > > I have found in the log files some international calls from unknown > extensions. These extensions don't exist in my configuration. I tried to > block them in my firewall (iptables on my freeswitch server) but they > always use random IP adresses. Here is a short part of my Master.csv: > > "1001","1001","0015972595646444","2013-02-22 02:05:27","","2013-02-22 > 02:05:27","0","NORMAL_CLEARING","3c876eae-7cbe-11e2-877f-b791adff5763","","","","" > "1001","1001","9011972595646444","2013-02-22 02:05:28","","2013-02-22 > 02:05:28","0","NORMAL_CLEARING","3d0d058c-7cbe-11e2-8783-b791adff5763","","","","" > "1001","1001","2011972595646444","2013-02-22 02:05:29","","2013-02-22 > 02:05:29","0","NORMAL_CLEARING","3da55576-7cbe-11e2-8787-b791adff5763","","","","" > "1001","1001","3011972595646444","2013-02-22 02:05:30","","2013-02-22 > 02:05:30","0","NORMAL_CLEARING","3e4727ca-7cbe-11e2-878b-b791adff5763","","","","" > "1001","1001","4011972595646444","2013-02-22 02:05:31","","2013-02-22 > 02:05:31","0","NORMAL_CLEARING","3eecc2e8-7cbe-11e2-878f-b791adff5763","","","","" > "1001","1001","5011972595646444","2013-02-22 02:05:32","","2013-02-22 > 02:05:32","0","NORMAL_CLEARING","3f633b94-7cbe-11e2-8793-b791adff5763","","","","" > "1001","1001","6011972595646444","2013-02-22 02:05:33","","2013-02-22 > 02:05:33","0","NORMAL_CLEARING","3fc49902-7cbe-11e2-8797-b791adff5763","","","","" > "1001","1001","7011972595646444","2013-02-22 02:05:33","","2013-02-22 > 02:05:33","0","NORMAL_CLEARING","403c0622-7cbe-11e2-879b-b791adff5763","","","","" > "1001","1001","8011972595646444","2013-02-22 02:05:34","","2013-02-22 > 02:05:34","0","NORMAL_CLEARING","40e61ef0-7cbe-11e2-879f-b791adff5763","","","","" > > With my configuration, I need to be registered to make a call. I tried > to call with an unregistered phone and I was not able to make a call. I > don't know how they are able to do this but I need to block them. Is > there something that I am missing in my configuration to block unwanted > extensions to make calls? > > Thanks in advance! > > > Fred > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > > > -- > Ken > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > irc.freenode.net #freeswitch > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130222/f1ac05a1/attachment-0003.html From david.villasmil.work at gmail.com Fri Feb 22 20:28:55 2013 From: david.villasmil.work at gmail.com (David Villasmil) Date: Fri, 22 Feb 2013 18:28:55 +0100 Subject: [Freeswitch-users] Outgoing calls from unknown users In-Reply-To: References: Message-ID: Also, I always completely remove ALL sample routing in the default dialplans, always. When i need an extension, I add it. There's another thread a about a user who left the external dialplan and got hacked... So always remove ALL routing! David On Feb 22, 2013, at 18:23, David Villasmil wrote: > Since a long time ago i NEVER use ports 5060/5061/5080/5081... > > Security by obscurity also helps (in my experience), in addition to fail2ban... > You must ALWAYS setup fail2ban if you don't know the IPs of your customers and so a firewall won't do much good. > > But firewall+fail2ban+non-standard-ports is pretty safe (of course in addition to authenticating) > > > David > On Feb 22, 2013, at 17:03, Ken Rice wrote: > >> If you were seeing a billtime of 0 that means the calls were blocked... Some one was probably hitting the ?external? interface then hitting the public context... This is allowed in the default example configs for freeswitch, but calls coming in that way are only allowed to actually do something if a) they hit one of the pre-defined local extensions or you have modified it to allow other calling. >> >> Why is it like this? This is so you can define your local extensions and DIDs in the public context then say hey you can just call me via sip to SIP:mynumber_or_extension at hostname.or.ip.of.my.FS.box.com >> >> Bots will scan the internet (the entire internet) and find your public profile is not doing sip challenge response, then they will try to place calls. In your logs they were trying to call a number in what appears to be Palestine, this is very comon, as they are probing to see if you let the calls pass by trying various prefixes... >> >> As long as you are configuring the unauthenticated interfaces to only allow calls for your local extensions this is not a big dead, they will give up and go away. >> >> >> >> On 2/22/13 9:48 AM, "Frederick Pruneau" wrote: >> >> >> PB 20618 >> >> Everytime, it is a 0 billsec. For now, international calls are not authorized. But in a near future, I want to authorized them. >> >> I verified what Ken wrote: >> >> A) make sure you are not using the default username and passwords for >> registered sip users (not using default username and password. All default users/extensions have been removed) >> B) don't allow unauthenticated calls to go back out to the PSTN (I don't have PSTN lines) >> C) Use appropriate firewall rules to only allow places you should be getting >> calls from (Already done) >> D) use something like Fail2Ban to block people attempting to make repeated >> failed calls/registration attempts in a short period of time... >> >> Actually, I have fail2ban that blocks registration attemps. I don't know how to block failed calls. Can you guide me to a web site or help me to ban failed calls? >> >> Thank you for your quick replies! >> >> Fred >> >> Le 2013-02-22 09:01, Christian Benke a ?crit : >> >> >> >> Do you have a logfile of these calls? Can you please paste it to >> http://pastebin.freeswitch.org/. Otherwise, please paste your >> dialplans to pastebin so we can figure out what's really happening >> with your calls, the csv has too little information. Do all of these >> calls have 0 billsec? >> >> If possible, you should turn off FreeSWITCH till you know the reason >> for this calls, it looks very much like your system is not safe. >> >> Best regards, >> Christian >> >> -- >> Central Asia by bike, starting May 2013 - http://poab.org >> >> >> On 22 February 2013 14:26, Frederick Pruneau >> wrote: >> >> >> Hi everyone! >> >> I have found in the log files some international calls from unknown >> extensions. These extensions don't exist in my configuration. I tried to >> block them in my firewall (iptables on my freeswitch server) but they >> always use random IP adresses. Here is a short part of my Master.csv: >> >> "1001","1001","0015972595646444","2013-02-22 02:05:27","","2013-02-22 >> 02:05:27","0","NORMAL_CLEARING","3c876eae-7cbe-11e2-877f-b791adff5763","","","","" >> "1001","1001","9011972595646444","2013-02-22 02:05:28","","2013-02-22 >> 02:05:28","0","NORMAL_CLEARING","3d0d058c-7cbe-11e2-8783-b791adff5763","","","","" >> "1001","1001","2011972595646444","2013-02-22 02:05:29","","2013-02-22 >> 02:05:29","0","NORMAL_CLEARING","3da55576-7cbe-11e2-8787-b791adff5763","","","","" >> "1001","1001","3011972595646444","2013-02-22 02:05:30","","2013-02-22 >> 02:05:30","0","NORMAL_CLEARING","3e4727ca-7cbe-11e2-878b-b791adff5763","","","","" >> "1001","1001","4011972595646444","2013-02-22 02:05:31","","2013-02-22 >> 02:05:31","0","NORMAL_CLEARING","3eecc2e8-7cbe-11e2-878f-b791adff5763","","","","" >> "1001","1001","5011972595646444","2013-02-22 02:05:32","","2013-02-22 >> 02:05:32","0","NORMAL_CLEARING","3f633b94-7cbe-11e2-8793-b791adff5763","","","","" >> "1001","1001","6011972595646444","2013-02-22 02:05:33","","2013-02-22 >> 02:05:33","0","NORMAL_CLEARING","3fc49902-7cbe-11e2-8797-b791adff5763","","","","" >> "1001","1001","7011972595646444","2013-02-22 02:05:33","","2013-02-22 >> 02:05:33","0","NORMAL_CLEARING","403c0622-7cbe-11e2-879b-b791adff5763","","","","" >> "1001","1001","8011972595646444","2013-02-22 02:05:34","","2013-02-22 >> 02:05:34","0","NORMAL_CLEARING","40e61ef0-7cbe-11e2-879f-b791adff5763","","","","" >> >> With my configuration, I need to be registered to make a call. I tried >> to call with an unregistered phone and I was not able to make a call. I >> don't know how they are able to do this but I need to block them. Is >> there something that I am missing in my configuration to block unwanted >> extensions to make calls? >> >> Thanks in advance! >> >> >> Fred >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> >> >> >> >> -- >> Ken >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> irc.freenode.net #freeswitch >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130222/35e7d54d/attachment.html From schoch+freeswitch.org at xwin32.com Fri Feb 22 20:37:38 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Fri, 22 Feb 2013 09:37:38 -0800 Subject: [Freeswitch-users] "Outgoing Call" on phone display In-Reply-To: <2F45CFD8-1F72-44AB-93D4-0C59BED92AD2@freeswitch.org> References: <094868EE-812C-4E21-8551-0EC80F553D3D@freeswitch.org> <2F45CFD8-1F72-44AB-93D4-0C59BED92AD2@freeswitch.org> Message-ID: Yes, that's what I'm going to do today. I enabled mod_cidlookup. Is that the best method, or should I use a lua script so I can do the lookup in the background? -- Steve On Thu, Feb 21, 2013 at 3:27 PM, Brian West wrote: > You know you can do CNAM lookups while dialing and set those values and > update the phone during the ringing phase. It puts Outgoing Call in there > when it has no other information. > -- > Brian West > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130222/ce5289e9/attachment.html From mehroz.ashraf85 at gmail.com Fri Feb 22 10:59:30 2013 From: mehroz.ashraf85 at gmail.com (mehroz) Date: Thu, 21 Feb 2013 23:59:30 -0800 (PST) Subject: [Freeswitch-users] FS with SSL/TLS issues! In-Reply-To: <1361513236022-7587844.post@n2.nabble.com> References: <1361367187947-7587736.post@n2.nabble.com> <1361513236022-7587844.post@n2.nabble.com> Message-ID: <1361519970790-7587849.post@n2.nabble.com> Moreover, i am unable to import "cafile.pem" in my 3CX phone. Could it be the reason if these errors? 1.Do i need client certificate to establish SSL connection with every softphone? 2.I have read that we can use FScomm phone to test but where does it get the certificate file ? 3.How can i check that FS has really seen the agent.pem file and OK for SSL/TLS SIP? Meanwhile i'll map domain name on FS server and will try to generate certificates using domain name. ...and ill be waiting for anyone to put his thoughts in , where i am! Thanks -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FS-with-SSL-TLS-issues-tp7587736p7587849.html Sent from the freeswitch-users mailing list archive at Nabble.com. From GB at cm.nl Fri Feb 22 15:41:10 2013 From: GB at cm.nl (Grant Bagdasarian) Date: Fri, 22 Feb 2013 13:41:10 +0100 Subject: [Freeswitch-users] Generate SIP INFO inside dialplan Message-ID: Hello, Is there a way to generate a SIP INFO message inside the dial plan? If so, how? If not, are there any other ways to accomplish this during a call? Regards, Grant -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130222/40e3ac32/attachment-0001.html From frederick at targointernet.com Fri Feb 22 21:01:28 2013 From: frederick at targointernet.com (Frederick Pruneau) Date: Fri, 22 Feb 2013 13:01:28 -0500 Subject: [Freeswitch-users] Outgoing calls from unknown users In-Reply-To: References: Message-ID: <5127B278.3010505@targointernet.com> Thanks guys for all your help! Le 2013-02-22 12:28, David Villasmil a ?crit : > Also, > > I always completely remove ALL sample routing in the default > dialplans, always. > > When i need an extension, I add it. > > There's another thread a about a user who left the external dialplan > and got hacked... > > So always remove ALL routing! > > David > > On Feb 22, 2013, at 18:23, David Villasmil > > wrote: > >> Since a long time ago i NEVER use ports 5060/5061/5080/5081... >> >> Security by obscurity also helps (in my experience), in addition to >> fail2ban... >> You must ALWAYS setup fail2ban if you don't know the IPs of your >> customers and so a firewall won't do much good. >> >> But firewall+fail2ban+non-standard-ports is pretty safe (of course in >> addition to authenticating) >> >> >> David >> On Feb 22, 2013, at 17:03, Ken Rice > > wrote: >> >>> Re: [Freeswitch-users] Outgoing calls from unknown users If you were >>> seeing a billtime of 0 that means the calls were blocked... Some one >>> was probably hitting the "external" interface then hitting the >>> public context... This is allowed in the default example configs for >>> freeswitch, but calls coming in that way are only allowed to >>> actually do something if a) they hit one of the pre-defined local >>> extensions or you have modified it to allow other calling. >>> >>> Why is it like this? This is so you can define your local extensions >>> and DIDs in the public context then say hey you can just call me via >>> sip to SIP:mynumber_or_extension at hostname.or.ip.of.my.FS.box.com >>> >>> Bots will scan the internet (the entire internet) and find your >>> public profile is not doing sip challenge response, then they will >>> try to place calls. In your logs they were trying to call a number >>> in what appears to be Palestine, this is very comon, as they are >>> probing to see if you let the calls pass by trying various prefixes... >>> >>> As long as you are configuring the unauthenticated interfaces to >>> only allow calls for your local extensions this is not a big dead, >>> they will give up and go away. >>> >>> >>> >>> On 2/22/13 9:48 AM, "Frederick Pruneau" >>> wrote: >>> >>> >>> PB 20618 >>> >>> Everytime, it is a 0 billsec. For now, international calls are >>> not authorized. But in a near future, I want to authorized them. >>> >>> I verified what Ken wrote: >>> >>> A) make sure you are not using the default username and >>> passwords for >>> registered sip users *(not using default username and password. >>> All default users/extensions have been removed) >>> * B) don't allow unauthenticated calls to go back out to the >>> PSTN *(I don't have PSTN lines) >>> * C) Use appropriate firewall rules to only allow places you >>> should be getting >>> calls from *(Already done) >>> * D) use something like Fail2Ban to block people attempting to >>> make repeated >>> failed calls/registration attempts in a short period of time... >>> >>> Actually, I have fail2ban that blocks registration attemps. I >>> don't know how to block failed calls. Can you guide me to a web >>> site or help me to ban failed calls? >>> >>> Thank you for your quick replies! >>> >>> Fred >>> >>> Le 2013-02-22 09:01, Christian Benke a ?crit : >>> >>> >>> >>> Do you have a logfile of these calls? Can you please paste it to >>> http://pastebin.freeswitch.org/. Otherwise, please paste your >>> dialplans to pastebin so we can figure out what's really >>> happening >>> with your calls, the csv has too little information. Do all >>> of these >>> calls have 0 billsec? >>> >>> If possible, you should turn off FreeSWITCH till you know >>> the reason >>> for this calls, it looks very much like your system is not safe. >>> >>> Best regards, >>> Christian >>> >>> -- >>> Central Asia by bike, starting May 2013 - http://poab.org >>> >>> >>> On 22 February 2013 14:26, Frederick Pruneau >>> >>> wrote: >>> >>> >>> Hi everyone! >>> >>> I have found in the log files some international calls >>> from unknown >>> extensions. These extensions don't exist in my >>> configuration. I tried to >>> block them in my firewall (iptables on my freeswitch >>> server) but they >>> always use random IP adresses. Here is a short part of >>> my Master.csv: >>> >>> "1001","1001","0015972595646444","2013-02-22 >>> 02:05:27","","2013-02-22 >>> 02:05:27","0","NORMAL_CLEARING","3c876eae-7cbe-11e2-877f-b791adff5763","","","","" >>> "1001","1001","9011972595646444","2013-02-22 >>> 02:05:28","","2013-02-22 >>> 02:05:28","0","NORMAL_CLEARING","3d0d058c-7cbe-11e2-8783-b791adff5763","","","","" >>> "1001","1001","2011972595646444","2013-02-22 >>> 02:05:29","","2013-02-22 >>> 02:05:29","0","NORMAL_CLEARING","3da55576-7cbe-11e2-8787-b791adff5763","","","","" >>> "1001","1001","3011972595646444","2013-02-22 >>> 02:05:30","","2013-02-22 >>> 02:05:30","0","NORMAL_CLEARING","3e4727ca-7cbe-11e2-878b-b791adff5763","","","","" >>> "1001","1001","4011972595646444","2013-02-22 >>> 02:05:31","","2013-02-22 >>> 02:05:31","0","NORMAL_CLEARING","3eecc2e8-7cbe-11e2-878f-b791adff5763","","","","" >>> "1001","1001","5011972595646444","2013-02-22 >>> 02:05:32","","2013-02-22 >>> 02:05:32","0","NORMAL_CLEARING","3f633b94-7cbe-11e2-8793-b791adff5763","","","","" >>> "1001","1001","6011972595646444","2013-02-22 >>> 02:05:33","","2013-02-22 >>> 02:05:33","0","NORMAL_CLEARING","3fc49902-7cbe-11e2-8797-b791adff5763","","","","" >>> "1001","1001","7011972595646444","2013-02-22 >>> 02:05:33","","2013-02-22 >>> 02:05:33","0","NORMAL_CLEARING","403c0622-7cbe-11e2-879b-b791adff5763","","","","" >>> "1001","1001","8011972595646444","2013-02-22 >>> 02:05:34","","2013-02-22 >>> 02:05:34","0","NORMAL_CLEARING","40e61ef0-7cbe-11e2-879f-b791adff5763","","","","" >>> >>> With my configuration, I need to be registered to make a >>> call. I tried >>> to call with an unregistered phone and I was not able to >>> make a call. I >>> don't know how they are able to do this but I need to >>> block them. Is >>> there something that I am missing in my configuration to >>> block unwanted >>> extensions to make calls? >>> >>> Thanks in advance! >>> >>> >>> Fred >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >>> >>> >>> >>> -- >>> Ken >>> _http://www.FreeSWITCH.org >>> http://www.ClueCon.com >>> http://www.OSTAG.org >>> _irc.freenode.net #freeswitch >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -- Fr?d?rick Pruneau Administrateur r?seau | Network administrator Targo Communications Ste-Clotilde : (450) 826-0031 Montr?al : (514) 448-0773 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130222/1d1167da/attachment-0001.html From msc at freeswitch.org Fri Feb 22 21:10:01 2013 From: msc at freeswitch.org (Michael Collins) Date: Fri, 22 Feb 2013 10:10:01 -0800 Subject: [Freeswitch-users] make mod_voicemail In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF23ACA6D@Mail-Kilo.squay.com> References: <1FFF97C269757C458224B7C895F35F1522A507@cantor.std.visionutv.se> <592A9CF93E12394E8472A6CC66E66BF23ACA44@Mail-Kilo.squay.com> <592A9CF93E12394E8472A6CC66E66BF23ACA6D@Mail-Kilo.squay.com> Message-ID: This means that there is something different happening on your production server. One thing you can do is take the binary that is built on your test server and manually copy it over to your production server - assuming they are the same OS and architecture. Be absolutely certain to backup the current mod_voicemail.so/.la files before you copy over the binaries - that way if you have a problem you can always go back. Just an FYI - 1.0.6 is more than 3 years old. You should really be getting your application and configs up and running on the latest git head which is about to be released as 1.2.6. Getting your app work on latest git HEAD is probably less work and more valuable than trying to backport changes into selected 1.0.6 files. -MC On Thu, Feb 21, 2013 at 2:03 AM, Archana Venugopan wrote: > Hi,**** > > ** ** > > I haven?t updated GIT recently. Freeswitch version I am using is 1.0.6. > The same version am using in my test environment as well and there if I > give make mod_voicemail-install, only mod_voicemail is being compiled. I > face this issue in my production server only.**** > > ** ** > > ** ** > > ** ** > > Regards,**** > > Archana**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Avi Marcus > *Sent:* 21 February 2013 09:54 > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] make mod_voicemail**** > > ** ** > > If you updated GIT since your last compile, then all the dependencies have > changed...**** > > > **** > > -Avi**** > > ** ** > > On Thu, Feb 21, 2013 at 11:46 AM, Archana Venugopan < > a.venugopan at mundio.com> wrote:**** > > Ya, this is being carried out in our production server. And it was built > years before I suppose. So am afraid to proceed as well with make > mod_voicemail commands, if it compiles everything then our production set > up will change completely.**** > > But I made few changes in voicemail and I need to compile mod_voicemail > part alone. Please do help me in this.**** > > **** > > Regards,**** > > Archana**** > > **** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Peter Olsson > *Sent:* 21 February 2013 09:38**** > > > *To:* 'FreeSWITCH Users Help' > *Subject:* Re: [Freeswitch-users] make mod_voicemail**** > > **** > > So, was everything else built already? I believe FS will build the > dependencies, if they were not built already ? I?m not totally sure about > that though.**** > > **** > > /Peter**** > > **** > > *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *F?r *Archana Venugopan > *Skickat:* den 21 februari 2013 10:30 > *Till:* FreeSWITCH Users Help > *?mne:* Re: [Freeswitch-users] make mod_voicemail**** > > **** > > I am afraid if that is as well installing everything so I just stopped it. > **** > > **** > > [root at sme-xswitch freeswitch]# make mod_voicemail-install**** > > quiet_libtool: compile: gcc -DHAVE_CONFIG_H -I. -I./src/include > -I./libs/xmlrpc-c -I/usr/local/src/freeswitch/libs/curl/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden > -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -pthread -DLINUX=2 > -D_REENTRANT -D_GNU_SOURCE -D_LARGEFILE64_SOURCE > -I/usr/local/src/freeswitch/libs/apr/include > -I/usr/local/src/freeswitch/libs/apr-util/include > -I/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib > -I/usr/local/src/freeswitch/libs/libtpl-1.5/src > -I/usr/local/src/freeswitch/libs/stfu > -I/usr/local/src/freeswitch/libs/sqlite > -I/usr/local/src/freeswitch/libs/pcre > -I/usr/local/src/freeswitch/libs/speex/include -Ilibs/speex/include > -I/usr/local/src/freeswitch/libs/srtp/include > -I/usr/local/src/freeswitch/libs/srtp/crypto/include > -Ilibs/srtp/crypto/include -I/usr/local/src/freeswitch/libs/spandsp/src > -I/usr/local/src/freeswitch/libs/tiff-4.0.2/libtiff -DENABLE_SRTP > -DSWITCH_HAVE_ODBC -I/usr/include > -I/usr/local/src/freeswitch/libs/libedit/src -DSWITCH_HAVE_LIBEDIT > -I/usr/local/src/freeswitch/libs/curl/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -Werror > -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb > -DHAVE_OPENSSL -Wall -std=c99 -pedantic -Wdeclaration-after-statement -g > -O2 -MT libfreeswitch_la-switch_ivr.lo -MD -MP -MF > .deps/libfreeswitch_la-switch_ivr.Tpo -c src/switch_ivr.c -fPIC -DPIC -o > .libs/libfreeswitch_la-switch_ivr.o**** > > quiet_libtool: compile: gcc -DHAVE_CONFIG_H -I. -I./src/include > -I./libs/xmlrpc-c -I/usr/local/src/freeswitch/libs/curl/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden > -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -pthread -DLINUX=2 > -D_REENTRANT -D_GNU_SOURCE -D_LARGEFILE64_SOURCE > -I/usr/local/src/freeswitch/libs/apr/include > -I/usr/local/src/freeswitch/libs/apr-util/include > -I/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib > -I/usr/local/src/freeswitch/libs/libtpl-1.5/src > -I/usr/local/src/freeswitch/libs/stfu > -I/usr/local/src/freeswitch/libs/sqlite > -I/usr/local/src/freeswitch/libs/pcre > -I/usr/local/src/freeswitch/libs/speex/include -Ilibs/speex/include > -I/usr/local/src/freeswitch/libs/srtp/include > -I/usr/local/src/freeswitch/libs/srtp/crypto/include > -Ilibs/srtp/crypto/include -I/usr/local/src/freeswitch/libs/spandsp/src > -I/usr/local/src/freeswitch/libs/tiff-4.0.2/libtiff -DENABLE_SRTP > -DSWITCH_HAVE_ODBC -I/usr/include > -I/usr/local/src/freeswitch/libs/libedit/src -DSWITCH_HAVE_LIBEDIT > -I/usr/local/src/freeswitch/libs/curl/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -Werror > -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb > -DHAVE_OPENSSL -Wall -std=c99 -pedantic -Wdeclaration-after-statement -g > -O2 -MT libfreeswitch_la-switch_ivr.lo -MD -MP -MF > .deps/libfreeswitch_la-switch_ivr.Tpo -c src/switch_ivr.c -o > libfreeswitch_la-switch_ivr.o >/dev/null 2>&1**** > > ^Cmake[1]: *** [libfreeswitch_la-switch_ivr.lo] Error 1**** > > make: *** [mod_voicemail-install] Interrupt**** > > **** > > Regards,**** > > Archana**** > > **** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [ > mailto:freeswitch-users-bounces at lists.freeswitch.org] > *On Behalf Of *Avi Marcus > *Sent:* 21 February 2013 09:24 > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] make mod_voicemail**** > > **** > > Try:**** > > > > **** > > > > **** > > > > **** > > make mod_voicemail-install**** > > > **** > > -Avi Marcus**** > > **** > > On Thu, Feb 21, 2013 at 11:10 AM, Archana Venugopan < > a.venugopan at mundio.com> wrote:**** > > Hi,**** > > **** > > When I try to give ?make mod_voicemail? command, I see its compiling > everything as like make command instead of just doing mod_voicemail. **** > > Can someone suggest why is this happening and what can be done to just > compile mod_voicemail. Many thanks.**** > > **** > > Regards,**** > > Archana**** > > **** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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**** > > **** > > !DSPAM:5125e59032761278114654! **** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130222/7f09e5eb/attachment-0001.html From msc at freeswitch.org Fri Feb 22 21:11:22 2013 From: msc at freeswitch.org (Michael Collins) Date: Fri, 22 Feb 2013 10:11:22 -0800 Subject: [Freeswitch-users] FW: FW: Event pick In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF23ACB84@Mail-Kilo.squay.com> References: <592A9CF93E12394E8472A6CC66E66BF23ACB84@Mail-Kilo.squay.com> Message-ID: You'll need a 3rd party program listening on the event socket to do this. -MC On Thu, Feb 21, 2013 at 4:13 AM, Archana Venugopan wrote: > Hi,**** > > ** ** > > Ya bridge is in dialplan XML only. In that case how am I to trace the > termination event and run a DB query? Is there a way to do?**** > > ** ** > > Thanks.**** > > ** ** > > Regards,**** > > Archana**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael > Collins > *Sent:* 20 February 2013 18:07 > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] FW: Event pick**** > > ** ** > > If you bridge in the dialplan then your Lua script won't run until the > bridge app has terminated. > -MC**** > > On Wed, Feb 20, 2013 at 9:33 AM, Archana Venugopan > wrote:**** > > Hi,**** > > **** > > I should define a obsession before going to obcause. But my outbound call > is already initiated from my dialplan XML [ data="sofia/external/0${msc_prefix}$1$2 at 10.30.3.27"/>].**** > > Am not sure how to give obsession in lua script as my outbound call is > initiated from dialplan XML.**** > > Can you please guide me. Many thanks**** > > **** > > **** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael > Collins > *Sent:* 20 February 2013 17:12 > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Event pick**** > > **** > > What value is in obCause when the originator cancels the call? Print the > value in the else block and find out. > -MC**** > > On Wed, Feb 20, 2013 at 4:37 AM, Archana Venugopan > wrote:**** > > Hi,**** > > I need to capture this ORIGINATOR _CANCEL event in my lua script. >From > google I found the below code. But we establish our calls from dialplan XML > as like this, data="sofia/external/0${msc_prefix}$1$2 at 10.30.3.27"/>.**** > > Since am not that well versed in lua script, I am not sure in the place of > obsession below what should I pass. As far I understood the = > sofia/external/0${msc_prefix}$1$2 at 10.30.3.27, this is what I should put > there but we already have this in our XML.**** > > Please suggest me on how should I proceed. Many thanks.**** > > **** > > -- Initiate an outbound call**** > > **** > > obSession = freeswitch.Session("sofia/192.168.0.4/1002")**** > > **** > > -- Check to see if the call was answered**** > > **** > > if obSession:ready() then**** > > -- Do something good here**** > > **** > > else -- This means the call was not answered ... Check for the reason** > ** > > **** > > local obCause = obSession:hangupCause()**** > > **** > > freeswitch.consoleLog("info", "obSession:hangupCause() = " .. obCause ) > **** > > **** > > if ( obCause == "USER_BUSY" ) then -- SIP 486**** > > -- For BUSY you may reschedule the call for later**** > > elseif ( obCause == "NO_ANSWER" ) then**** > > -- Call them back in an hour**** > > elseif ( obCause == "ORIGINATOR_CANCEL" ) then -- SIP 487**** > > -- May need to check for network congestion or problems**** > > else**** > > -- Log these issues**** > > end**** > > end**** > > **** > > **** > > **** > > **** > > 2013-02-19 17:56:02.277173 [NOTICE] sofia.c:7082 Hangup sofia/internal/ > 206 at fsfailover.uk01.com [CS_EXECUTE] [ORIGINATOR_CANCEL]**** > > **** > > Regards,**** > > Archana**** > > **** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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**** > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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**** > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130222/49cbd5e5/attachment.html From msc at freeswitch.org Fri Feb 22 21:24:54 2013 From: msc at freeswitch.org (Michael Collins) Date: Fri, 22 Feb 2013 10:24:54 -0800 Subject: [Freeswitch-users] Repeated extensions In-Reply-To: <1361542646.743454162@f319.mail.ru> References: <1361542646.743454162@f319.mail.ru> Message-ID: This is not valid: I recommend this: Try that in place of what you have and re-test. I have a suspicion that it will solve your problem of multiple recordings. -MC On Fri, Feb 22, 2013 at 6:17 AM, Dmitriy Shumaev wrote: > Hello world) > > I have session recording extension: > > > > > > > > > > > > - > > > > > > > > in the very beginnin of my dialplan. But there are some (IVRs with > transfer) and (Group number with loopback) in my dialplan. So recording > extension is hited multiple times and as a result I have several files with > the "same" recorded session (IVR, Group number, Answered subscriber). > > What should I do to prohibit repeated hits of "record_3..16_digits" > extension? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130222/32a1ae0c/attachment-0001.html From msc at freeswitch.org Fri Feb 22 21:29:09 2013 From: msc at freeswitch.org (Michael Collins) Date: Fri, 22 Feb 2013 10:29:09 -0800 Subject: [Freeswitch-users] Short word drop-outs In-Reply-To: <512750D6.5060401@digitalmail.com> References: <512750D6.5060401@digitalmail.com> Message-ID: Start with pcapsipdump. Get pcaps of all these calls. Then have the customer report to you the date, time, caller and callee of such an occurrence. Locate the pcap and listen to it in Wireshark. See if the "dropout" occurs in the audio stream or if there is a possible loss of packets, etc. That should yield some clues as to where to go next. If you are really suspicious of the silence suppression stuff then you might want to turn it off and see what happens. -MC On Fri, Feb 22, 2013 at 3:04 AM, Alex Lake wrote: > Got a customer who's complaining that short words are being lost in > conversations over VOIP/Freeswitch. > > I presume this is over-enthusiastic silence suppression which could be > tweaked in the freeswitch config. > > I wondered if this might be a handset config issue, but they claim that > they can use their handsets with other VOIP providers that don't show > the same problem. > > Any tips? > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130222/a09de898/attachment.html From mike at jerris.com Fri Feb 22 22:55:08 2013 From: mike at jerris.com (Michael Jerris) Date: Fri, 22 Feb 2013 14:55:08 -0500 Subject: [Freeswitch-users] Memory growth In-Reply-To: <51277CD7.20008@vts24.ru> References: <7D4D7EF3-9889-4C06-A1E4-26225C86E791@mgtech.com> <5125AD25.6020005@vts24.ru> <51277CD7.20008@vts24.ru> Message-ID: <3AA53F32-9F17-4D8C-B4EE-FCD6FE411C7E@jerris.com> apply patch cd libs/sofia-sip make touch .update cd ../.. make mod_sofia-install On Feb 22, 2013, at 9:12 AM, ??????? ??????? wrote: > Need a little help in the following question: can I apply the patch > http://fisheye.freeswitch.org/rdiff/freeswitch.git?csid=45d849ab746e6fc0833820851dced1dc61e0fbe1&u&N > to the file > libs/sofia-sip/libsofia-sip-ua/su/su_time0.c? > Would it be necessary to change anything else? > > Then I would be able to immediately begin testing... > > 21.02.2013 9:14, ??????? ??????? ?????: >> I would be glad to do it, but my FS in git branch v1.2.stable (it is in production environment). I will have to wait patches to the 1.2 series. >> >> >> 21.02.2013 3:42, Mario G ?????: >>> FYI, Anthony nailed my memory leak yesterday (see the jira below). 24 hours of testing aok so far. The leak varied between 300k-2M per hour, not sure if it was only OSX related so good idea for anyone with a leak to test with updated HEAD. >>> Mario G >>> >>> On Feb 18, 2013, at 11:22 AM, Ken Rice wrote: >>> >>>> We?re trying to figure out where this last bit of leaking is coming from... Looks like it might be related to Registrations some how... >>>> >>>> Once we squash that there will be some more info on the 1.2 branch coming -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130222/f6bf07b5/attachment.html From lloyd.aloysius at gmail.com Fri Feb 22 23:26:54 2013 From: lloyd.aloysius at gmail.com (Lloyd Aloysius) Date: Fri, 22 Feb 2013 15:26:54 -0500 Subject: [Freeswitch-users] mod_conference - moderator privileges Message-ID: Hi All How to setup a moderator for a conference? Using moderator flags ? There are two pin numbers in the conference profile. 1. pin - regular user 2. moderator-pin - I assume this pin is for moderator. If I send a call to conference - how the pin authentication work for the conference ? Any help is appreciated. Thanks Lloyd -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130222/00eef200/attachment.html From krice at freeswitch.org Fri Feb 22 23:40:22 2013 From: krice at freeswitch.org (Ken Rice) Date: Fri, 22 Feb 2013 14:40:22 -0600 Subject: [Freeswitch-users] Friday Free For All at the top of the hour Message-ID: Hey guys I?m be on there in about 20 minutes, but start the Friday FreeForAll sip:888 at conference.freeswitch.org GO! -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130222/5afe6f0c/attachment.html From levend.sayar at karel.com.tr Sat Feb 23 00:12:42 2013 From: levend.sayar at karel.com.tr (Levend Sayar) Date: Fri, 22 Feb 2013 21:12:42 +0000 Subject: [Freeswitch-users] optional SRTP In-Reply-To: <575B91FD-D404-4BD2-BBDE-59FDD6ABD90F@gmail.com> References: , <575B91FD-D404-4BD2-BBDE-59FDD6ABD90F@gmail.com> Message-ID: Thanks for your responses. So afaiu m=audio RTP/AVP a=crypto Is invalid due to spec. What Steve pointed is a Microsoft deviation. So for optional RTP, SDP must look like something like this ? m=audio RTP/AVP a=blabla m=audio RTP/SAVP a=crypto Namely contain both RTP/AVP and RTP/SAVP profiles. And callee can choose whatever it prefers. RTP or SRTP. Brian, I will test the parameters you gave ASAP and say the result. I really appreciate your help guys. _lvnd_ {^_^} On 22 ?ub 2013, at 19:09, "Steven Ayre" > wrote: The confusion was mine on an earlier thread. I said that, based on something on another mailing list that I can't find now (sip-implementors?). That post might have been inspired by this which states RTP/AVP with a=crypto allows SRTP desired but not required - but only because it 'deviates from the specification in RFC4568' http://msdn.microsoft.com/en-us/library/dd948772(v=office.12).aspx I guess such devices are what the NDLB parameter is for. The previous thread was 'SRTP disabling' where Levand has a phone that allows optional SRTP but only offers RTP/SAVP in its SDP with no RTP/AVP profile offered so FS cannot/won't send plain RTP to the SRTP port. Which I'd say is correct behaviour and a broken phone. Steve On 22 Feb 2013, at 17:27, Brian West > wrote: crypto in RTP/AVP is invalid per the spec. You are required to send two rtp profiles one with RTP/AVP or RTP/SAVP To get around this in your broken device you can set the variable sip_allow_crypto_in_avp or set profile param NDLB-allow-crypto-in-avp -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire T: +1.918.420.9266 | F: +1.918.420.9267 | M: +1.918.424.WEST iNUM: +883 5100 1420 9266 ISN: 410*543 On Feb 22, 2013, at 10:08 AM, Ken Rice > wrote: FreeSWITCH Support SIP/TLS, SRTP and ZRTP, the error message you are refering to below is where someone has improperly implemented SRTP and is sending the crypto headers in the wrong spots. See the Wiki for setting up secure calling. Also please note, if you are going to use SRTP, you _must_ use SIP/TLS as the keys for SRTP are passed in the SDP of the SIP messaging. You can use ZRTP without using SIP/TLS as the key exchange for ZRTP happens as a DH exchange in the ZRTP stream, but if you want to protect the contents of the SIP messaging you?ll still want to use SIP/TLS. K _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130222/388a8562/attachment-0001.html From msc at freeswitch.org Sat Feb 23 00:16:32 2013 From: msc at freeswitch.org (Michael Collins) Date: Fri, 22 Feb 2013 13:16:32 -0800 Subject: [Freeswitch-users] mod_conference - moderator privileges In-Reply-To: References: Message-ID: I'm pretty sure this is all covered on the mod_conference wiki page: http://wiki.freeswitch.org/wiki/Mod_conference#Examples I believe you want this example: confname++flags{endconf|moderator} IIRC you can specify the exact PIN you want the moderator to use by putting in: confname+123456+flags{endconf|moderator} Let us know how that goes. -MC On Fri, Feb 22, 2013 at 12:26 PM, Lloyd Aloysius wrote: > Hi All > > How to setup a moderator for a conference? Using moderator flags ? > > There are two pin numbers in the conference profile. > > 1. pin - regular user > > 2. moderator-pin - I assume this pin is for moderator. > > If I send a call to conference - how the pin authentication work for the > conference ? > > > Any help is appreciated. > > > Thanks > Lloyd > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130222/95ffa63c/attachment.html From msc at freeswitch.org Sat Feb 23 00:19:10 2013 From: msc at freeswitch.org (Michael Collins) Date: Fri, 22 Feb 2013 13:19:10 -0800 Subject: [Freeswitch-users] freeswitch hack In-Reply-To: References: <958ABE902C1F47B09FFAAE7EDEE074B1@imsdeveloper> Message-ID: They can't auth with a username that does not exist. The best they can do is try to guess user names and passwords. That's where Ken's best practices are quite useful. -MC On Wed, Feb 20, 2013 at 10:35 PM, Mario Karakanovski wrote: > In my situation all calls are rejected, but I think it is because they > are authenticated with invalid username.**** > > My concern is how ones can authenticate in freeswitch with user that not > exists and never was configured. I was not able to reproduce that.**** > > What I found so far: they use a couple of IPs. They send OPTIONS (only one > time) during the day and start try at the night. They tried a maximum of > 100 calls.**** > > I am still waiting to log some packet**** > > ** ** > > Mario **** > > ** ** > ------------------------------ > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael > Collins > *Sent:* Wednesday, February 20, 2013 10:41 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] freeswitch hack**** > > ** ** > > Aren't they supposed to be rejected?**** > > On Wed, Feb 20, 2013 at 11:19 AM, Blake Priddy > wrote:**** > > I have also had the situation that they are calls getting rejected.**** > > ** ** > > On Wed, Feb 20, 2013 at 11:08 AM, Michael Collins > wrote:**** > > ** ** > > On Wed, Feb 20, 2013 at 1:53 AM, Mario Karakanovski wrote:* > *** > > Thanks Ken,**** > > **** > > It is helpful, but I still think there is some security issue. I?ve double > check configuration. I?ve try to reproduce the issue trying to do direct > call (TCP and UDP) or authenticate with invalid user, but everything works > as expected ? calls/authentication was rejected. I?ve decide to log the > traffic ? maybe I will be able to see where is the problem.**** > > ** ** > > What "security issue"? You said that they cannot make calls with the > passwords that they've guessed, correct? About the only thing left to do is > set up fail2ban and just shut > the door on them when they fail too many times. > > -Michael > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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**** > > > > **** > > ** ** > > -- **** > > > *Blakelund Priddy***** > > Network Systems Engineer > Bryant Public School District > Bryant, Arkansas 72022 > http://www.bryantschools.org**** > > p 501-653-5038 > f 501-847-5656**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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**** > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130222/132a54de/attachment-0001.html From bpriddy at bryantschools.org Sat Feb 23 00:20:17 2013 From: bpriddy at bryantschools.org (Blake Priddy) Date: Fri, 22 Feb 2013 15:20:17 -0600 Subject: [Freeswitch-users] Friday Free For All at the top of the hour In-Reply-To: References: Message-ID: Anyone? http://www.youtube.com/watch?feature=player_detailpage&v=f4zyjLyBp64 On Fri, Feb 22, 2013 at 2:40 PM, Ken Rice wrote: > Hey guys I?m be on there in about 20 minutes, but start the Friday > FreeForAll > > > sip:888 at conference.freeswitch.org GO! > -- > Ken > *http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > *irc.freenode.net #freeswitch > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- *Blakelund Priddy* Network Systems Engineer Bryant Public School District Bryant, Arkansas 72022 http://www.bryantschools.org p 501-653-5038 f 501-847-5656 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130222/e9b89324/attachment.html From royce3 at gmail.com Sat Feb 23 00:28:36 2013 From: royce3 at gmail.com (Royce Mitchell III) Date: Fri, 22 Feb 2013 15:28:36 -0600 Subject: [Freeswitch-users] Adtran Total Access 916e 4242916L1 In-Reply-To: References: Message-ID: rename the zipper attachment to a zip extension. I have attached good and bad isdn traces. The "good" traces are call that have no problems The "bad" traces lose early media whenever the 180 is sent ( after the 183+media ) which happens at this point: 12:19:54.507 ISDN.EP PRI 1 Call to 832XXXXXXX alerting, no PI. On Wed, Feb 20, 2013 at 11:48 AM, Michael Collins wrote: > Do you have d-channel traces of working vs. problematic calls? I'd love to > see those. > -MC > > > On Wed, Feb 20, 2013 at 7:45 AM, Royce Mitchell III > wrote: >> >> when placing an outbound call through the 908e, via PRI hosted by >> TWTelecom and dialing a destination on the AT&T wireless network, a >> "different" series of ISDN codes comes back than from most other >> destinations. This series of ISDN codes causes the Adtran to send a >> 183+early media followed by a 180 w/o media, which causes the caller >> to hear nothing during most of the ringing process. All work-arounds I >> have tried to employ so far ( like ignore early media ) have had >> disastrous side-effects, we've almost lost clients over them. I put >> out a request to pay for a FreeSWITCH expert to help fix the issue, >> but I've been too busy with other projects to follow up. >> >> On Wed, Feb 20, 2013 at 9:27 AM, Ken Rice wrote: >> > What kind of incompatibility? As this does not make a lot of sense for a >> > TDM/SIP device to have problems with ATT Wireless a Cellular network, >> > they >> > cant talk directly to each other >> > >> > On 2/20/13 9:20 AM, "Royce Mitchell III" wrote: >> > >> >> I have a few 908e's, and I'm experiencing a serious incompatibility >> >> between FreeSWITCH, the 908e, and the AT&T wireless network. I'm >> >> confident I will get it fixed eventually, but just sharing what I know >> >> about the device. >> >> >> >> On Wed, Feb 20, 2013 at 9:12 AM, Ken Rice wrote: >> >>> The Adtran TA9XX series are pretty good units. Check out their website >> >>> at >> >>> adtran.com to find tech manuals for these things... >> >>> >> >>> I cant answer all the questions below, but these are very stable units >> >>> designed to be sold to carriers (and used by many carriers) >> >>> >> >>> How many concurrent connections depends on the specific model you are >> >>> getting. >> >>> >> >>> >> >>> On 2/20/13 9:04 AM, "Mimiko" wrote: >> >>> >> >>>> Hi. >> >>>> >> >>>> Who knows about this device: Adtran Total Access 916e 4242916L1 >> >>>> http://www.newegg.com/Product/Product.aspx?Item=N82E16833706027 >> >>>> >> >>>> How stable is it? Does it use LAN ports in failover mode? Does it use >> >>>> T1/FT1 WAN ports in failover manner connected to a PBX with two PRI >> >>>> ports? How good it is working with FS? How much concurrent connection >> >>>> does it support? >> >>>> >> >>>> I asked to the company via email a week ago, and no response till >> >>>> now. >> >>> >> >>> -- >> >>> Ken >> >>> http://www.FreeSWITCH.org >> >>> http://www.ClueCon.com >> >>> http://www.OSTAG.org >> >>> irc.freenode.net #freeswitch >> >>> >> >>> >> >>> >> >>> >> >>> _________________________________________________________________________ >> >>> Professional FreeSWITCH Consulting Services: >> >>> consulting at freeswitch.org >> >>> http://www.freeswitchsolutions.com >> >>> >> >>> >> >>> >> >>> >> >>> Official FreeSWITCH Sites >> >>> http://www.freeswitch.org >> >>> http://wiki.freeswitch.org >> >>> http://www.cluecon.com >> >>> >> >>> 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 >> >> >> >> >> > >> > -- >> > Ken >> > http://www.FreeSWITCH.org >> > http://www.ClueCon.com >> > http://www.OSTAG.org >> > irc.freenode.net #freeswitch >> > >> > >> > >> > >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > 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 >> >> >> >> -- >> Royce Mitchell >> IT Consultant >> ITAS Solutions >> royce3 at itas-solutions.com >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 > > > > > -- > Michael S Collins > Twitter: @mercutioviz > > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -- Royce Mitchell IT Consultant ITAS Solutions royce3 at itas-solutions.com -------------- next part -------------- A non-text attachment was scrubbed... Name: good_bad.zipper Type: application/octet-stream Size: 4767 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130222/f88efb82/attachment.obj From steveayre at gmail.com Sat Feb 23 00:33:04 2013 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 22 Feb 2013 22:33:04 +0100 Subject: [Freeswitch-users] optional SRTP In-Reply-To: References: <575B91FD-D404-4BD2-BBDE-59FDD6ABD90F@gmail.com> Message-ID: <9BCB7CE9-384E-4F4E-9498-6FBB7E75946D@gmail.com> Probably not only MS, but yes non-standard. > So for optional RTP, SDP must look like something like this ? > > m=audio RTP/AVP > a=blabla > m=audio RTP/SAVP > a=crypto Correct. The callee can then pick which of the profiles to use. Steve On 22 Feb 2013, at 22:12, Levend Sayar wrote: > Thanks for your responses. > > So afaiu > > m=audio RTP/AVP > a=crypto > > Is invalid due to spec. What Steve pointed is a Microsoft deviation. > > So for optional RTP, SDP must look like something like this ? > > m=audio RTP/AVP > a=blabla > m=audio RTP/SAVP > a=crypto > > Namely contain both RTP/AVP and RTP/SAVP profiles. And callee can choose whatever it prefers. RTP or SRTP. > > Brian, I will test the parameters you gave ASAP and say the result. > > I really appreciate your help guys. > > > > _lvnd_ > {^_^} > > > > On 22 ?ub 2013, at 19:09, "Steven Ayre" wrote: > >> The confusion was mine on an earlier thread. I said that, based on something on another mailing list that I can't find now (sip-implementors?). >> >> That post might have been inspired by this which states RTP/AVP with a=crypto allows SRTP desired but not required - but only because it 'deviates from the specification in RFC4568' >> >> http://msdn.microsoft.com/en-us/library/dd948772(v=office.12).aspx >> >> I guess such devices are what the NDLB parameter is for. >> >> >> >> The previous thread was 'SRTP disabling' where Levand has a phone that allows optional SRTP but only offers RTP/SAVP in its SDP with no RTP/AVP profile offered so FS cannot/won't send plain RTP to the SRTP port. Which I'd say is correct behaviour and a broken phone. >> >> >> >> Steve >> >> On 22 Feb 2013, at 17:27, Brian West wrote: >> >>> crypto in RTP/AVP is invalid per the spec. >>> >>> You are required to send two rtp profiles one with RTP/AVP or RTP/SAVP >>> >>> To get around this in your broken device you can set the variable sip_allow_crypto_in_avp or set profile param NDLB-allow-crypto-in-avp >>> >>> -- >>> Brian West >>> brian at freeswitch.org >>> FreeSWITCH Solutions, LLC >>> PO BOX PO BOX 2531 >>> Brookfield, WI 53008-2531 >>> Twitter: @FreeSWITCH_Wire >>> T: +1.918.420.9266 | F: +1.918.420.9267 | M: +1.918.424.WEST >>> iNUM: +883 5100 1420 9266 >>> ISN: 410*543 >>> >>> >>> >>> >>> >>> >>> On Feb 22, 2013, at 10:08 AM, Ken Rice wrote: >>> >>>> FreeSWITCH Support SIP/TLS, SRTP and ZRTP, the error message you are refering to below is where someone has improperly implemented SRTP and is sending the crypto headers in the wrong spots. See the Wiki for setting up secure calling. >>>> Also please note, if you are going to use SRTP, you _must_ use SIP/TLS as the keys for SRTP are passed in the SDP of the SIP messaging. You can use ZRTP without using SIP/TLS as the key exchange for ZRTP happens as a DH exchange in the ZRTP stream, but if you want to protect the contents of the SIP messaging you?ll still want to use SIP/TLS. >>>> K >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130222/a7a370c3/attachment-0001.html From jpyle at fidelityvoice.com Sat Feb 23 00:42:30 2013 From: jpyle at fidelityvoice.com (Jeff Pyle) Date: Fri, 22 Feb 2013 16:42:30 -0500 Subject: [Freeswitch-users] Adtran Total Access 916e 4242916L1 In-Reply-To: References: Message-ID: What version AOS on the 908e? On Fri, Feb 22, 2013 at 4:28 PM, Royce Mitchell III wrote: > rename the zipper attachment to a zip extension. I have attached good > and bad isdn traces. > > The "good" traces are call that have no problems > > The "bad" traces lose early media whenever the 180 is sent ( after the > 183+media ) which happens at this point: > > 12:19:54.507 ISDN.EP PRI 1 Call to 832XXXXXXX alerting, no PI. > > On Wed, Feb 20, 2013 at 11:48 AM, Michael Collins > wrote: > > Do you have d-channel traces of working vs. problematic calls? I'd love > to > > see those. > > -MC > > > > > > On Wed, Feb 20, 2013 at 7:45 AM, Royce Mitchell III > > wrote: > >> > >> when placing an outbound call through the 908e, via PRI hosted by > >> TWTelecom and dialing a destination on the AT&T wireless network, a > >> "different" series of ISDN codes comes back than from most other > >> destinations. This series of ISDN codes causes the Adtran to send a > >> 183+early media followed by a 180 w/o media, which causes the caller > >> to hear nothing during most of the ringing process. All work-arounds I > >> have tried to employ so far ( like ignore early media ) have had > >> disastrous side-effects, we've almost lost clients over them. I put > >> out a request to pay for a FreeSWITCH expert to help fix the issue, > >> but I've been too busy with other projects to follow up. > >> > >> On Wed, Feb 20, 2013 at 9:27 AM, Ken Rice wrote: > >> > What kind of incompatibility? As this does not make a lot of sense > for a > >> > TDM/SIP device to have problems with ATT Wireless a Cellular network, > >> > they > >> > cant talk directly to each other > >> > > >> > On 2/20/13 9:20 AM, "Royce Mitchell III" wrote: > >> > > >> >> I have a few 908e's, and I'm experiencing a serious incompatibility > >> >> between FreeSWITCH, the 908e, and the AT&T wireless network. I'm > >> >> confident I will get it fixed eventually, but just sharing what I > know > >> >> about the device. > >> >> > >> >> On Wed, Feb 20, 2013 at 9:12 AM, Ken Rice > wrote: > >> >>> The Adtran TA9XX series are pretty good units. Check out their > website > >> >>> at > >> >>> adtran.com to find tech manuals for these things... > >> >>> > >> >>> I cant answer all the questions below, but these are very stable > units > >> >>> designed to be sold to carriers (and used by many carriers) > >> >>> > >> >>> How many concurrent connections depends on the specific model you > are > >> >>> getting. > >> >>> > >> >>> > >> >>> On 2/20/13 9:04 AM, "Mimiko" wrote: > >> >>> > >> >>>> Hi. > >> >>>> > >> >>>> Who knows about this device: Adtran Total Access 916e 4242916L1 > >> >>>> http://www.newegg.com/Product/Product.aspx?Item=N82E16833706027 > >> >>>> > >> >>>> How stable is it? Does it use LAN ports in failover mode? Does it > use > >> >>>> T1/FT1 WAN ports in failover manner connected to a PBX with two PRI > >> >>>> ports? How good it is working with FS? How much concurrent > connection > >> >>>> does it support? > >> >>>> > >> >>>> I asked to the company via email a week ago, and no response till > >> >>>> now. > >> >>> > >> >>> -- > >> >>> Ken > >> >>> http://www.FreeSWITCH.org > >> >>> http://www.ClueCon.com > >> >>> http://www.OSTAG.org > >> >>> irc.freenode.net #freeswitch > >> >>> > >> >>> > >> >>> > >> >>> > >> >>> > _________________________________________________________________________ > >> >>> Professional FreeSWITCH Consulting Services: > >> >>> consulting at freeswitch.org > >> >>> http://www.freeswitchsolutions.com > >> >>> > >> >>> > >> >>> > >> >>> > >> >>> Official FreeSWITCH Sites > >> >>> http://www.freeswitch.org > >> >>> http://wiki.freeswitch.org > >> >>> http://www.cluecon.com > >> >>> > >> >>> 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 > >> >> > >> >> > >> > > >> > -- > >> > Ken > >> > http://www.FreeSWITCH.org > >> > http://www.ClueCon.com > >> > http://www.OSTAG.org > >> > irc.freenode.net #freeswitch > >> > > >> > > >> > > >> > > >> > > _________________________________________________________________________ > >> > Professional FreeSWITCH Consulting Services: > >> > consulting at freeswitch.org > >> > http://www.freeswitchsolutions.com > >> > > >> > > >> > > >> > > >> > Official FreeSWITCH Sites > >> > http://www.freeswitch.org > >> > http://wiki.freeswitch.org > >> > http://www.cluecon.com > >> > > >> > 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 > >> > >> > >> > >> -- > >> Royce Mitchell > >> IT Consultant > >> ITAS Solutions > >> royce3 at itas-solutions.com > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> 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 > > > > > > > > > > -- > > Michael S Collins > > Twitter: @mercutioviz > > > > http://www.FreeSWITCH.org > > http://www.ClueCon.com > > http://www.OSTAG.org > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > 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 > > > > > > -- > Royce Mitchell > IT Consultant > ITAS Solutions > royce3 at itas-solutions.com > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130222/713fa2bb/attachment.html From royce3 at gmail.com Sat Feb 23 00:49:20 2013 From: royce3 at gmail.com (Royce Mitchell III) Date: Fri, 22 Feb 2013 15:49:20 -0600 Subject: [Freeswitch-users] Adtran Total Access 916e 4242916L1 In-Reply-To: References: Message-ID: R10.5.0.E On Fri, Feb 22, 2013 at 3:42 PM, Jeff Pyle wrote: > What version AOS on the 908e? > > > On Fri, Feb 22, 2013 at 4:28 PM, Royce Mitchell III > wrote: >> >> rename the zipper attachment to a zip extension. I have attached good >> and bad isdn traces. >> >> The "good" traces are call that have no problems >> >> The "bad" traces lose early media whenever the 180 is sent ( after the >> 183+media ) which happens at this point: >> >> 12:19:54.507 ISDN.EP PRI 1 Call to 832XXXXXXX alerting, no PI. >> >> On Wed, Feb 20, 2013 at 11:48 AM, Michael Collins >> wrote: >> > Do you have d-channel traces of working vs. problematic calls? I'd love >> > to >> > see those. >> > -MC >> > >> > >> > On Wed, Feb 20, 2013 at 7:45 AM, Royce Mitchell III >> > wrote: >> >> >> >> when placing an outbound call through the 908e, via PRI hosted by >> >> TWTelecom and dialing a destination on the AT&T wireless network, a >> >> "different" series of ISDN codes comes back than from most other >> >> destinations. This series of ISDN codes causes the Adtran to send a >> >> 183+early media followed by a 180 w/o media, which causes the caller >> >> to hear nothing during most of the ringing process. All work-arounds I >> >> have tried to employ so far ( like ignore early media ) have had >> >> disastrous side-effects, we've almost lost clients over them. I put >> >> out a request to pay for a FreeSWITCH expert to help fix the issue, >> >> but I've been too busy with other projects to follow up. >> >> >> >> On Wed, Feb 20, 2013 at 9:27 AM, Ken Rice wrote: >> >> > What kind of incompatibility? As this does not make a lot of sense >> >> > for a >> >> > TDM/SIP device to have problems with ATT Wireless a Cellular network, >> >> > they >> >> > cant talk directly to each other >> >> > >> >> > On 2/20/13 9:20 AM, "Royce Mitchell III" wrote: >> >> > >> >> >> I have a few 908e's, and I'm experiencing a serious incompatibility >> >> >> between FreeSWITCH, the 908e, and the AT&T wireless network. I'm >> >> >> confident I will get it fixed eventually, but just sharing what I >> >> >> know >> >> >> about the device. >> >> >> >> >> >> On Wed, Feb 20, 2013 at 9:12 AM, Ken Rice >> >> >> wrote: >> >> >>> The Adtran TA9XX series are pretty good units. Check out their >> >> >>> website >> >> >>> at >> >> >>> adtran.com to find tech manuals for these things... >> >> >>> >> >> >>> I cant answer all the questions below, but these are very stable >> >> >>> units >> >> >>> designed to be sold to carriers (and used by many carriers) >> >> >>> >> >> >>> How many concurrent connections depends on the specific model you >> >> >>> are >> >> >>> getting. >> >> >>> >> >> >>> >> >> >>> On 2/20/13 9:04 AM, "Mimiko" wrote: >> >> >>> >> >> >>>> Hi. >> >> >>>> >> >> >>>> Who knows about this device: Adtran Total Access 916e 4242916L1 >> >> >>>> http://www.newegg.com/Product/Product.aspx?Item=N82E16833706027 >> >> >>>> >> >> >>>> How stable is it? Does it use LAN ports in failover mode? Does it >> >> >>>> use >> >> >>>> T1/FT1 WAN ports in failover manner connected to a PBX with two >> >> >>>> PRI >> >> >>>> ports? How good it is working with FS? How much concurrent >> >> >>>> connection >> >> >>>> does it support? >> >> >>>> >> >> >>>> I asked to the company via email a week ago, and no response till >> >> >>>> now. >> >> >>> >> >> >>> -- >> >> >>> Ken >> >> >>> http://www.FreeSWITCH.org >> >> >>> http://www.ClueCon.com >> >> >>> http://www.OSTAG.org >> >> >>> irc.freenode.net #freeswitch >> >> >>> >> >> >>> >> >> >>> >> >> >>> >> >> >>> >> >> >>> _________________________________________________________________________ >> >> >>> Professional FreeSWITCH Consulting Services: >> >> >>> consulting at freeswitch.org >> >> >>> http://www.freeswitchsolutions.com >> >> >>> >> >> >>> >> >> >>> >> >> >>> >> >> >>> Official FreeSWITCH Sites >> >> >>> http://www.freeswitch.org >> >> >>> http://wiki.freeswitch.org >> >> >>> http://www.cluecon.com >> >> >>> >> >> >>> 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 >> >> >> >> >> >> >> >> > >> >> > -- >> >> > Ken >> >> > http://www.FreeSWITCH.org >> >> > http://www.ClueCon.com >> >> > http://www.OSTAG.org >> >> > irc.freenode.net #freeswitch >> >> > >> >> > >> >> > >> >> > >> >> > >> >> > _________________________________________________________________________ >> >> > Professional FreeSWITCH Consulting Services: >> >> > consulting at freeswitch.org >> >> > http://www.freeswitchsolutions.com >> >> > >> >> > >> >> > >> >> > >> >> > Official FreeSWITCH Sites >> >> > http://www.freeswitch.org >> >> > http://wiki.freeswitch.org >> >> > http://www.cluecon.com >> >> > >> >> > 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 >> >> >> >> >> >> >> >> -- >> >> Royce Mitchell >> >> IT Consultant >> >> ITAS Solutions >> >> royce3 at itas-solutions.com >> >> >> >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://wiki.freeswitch.org >> >> http://www.cluecon.com >> >> >> >> 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 >> > >> > >> > >> > >> > -- >> > Michael S Collins >> > Twitter: @mercutioviz >> > >> > http://www.FreeSWITCH.org >> > http://www.ClueCon.com >> > http://www.OSTAG.org >> > >> > >> > >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > 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 >> > >> >> >> >> -- >> Royce Mitchell >> IT Consultant >> ITAS Solutions >> royce3 at itas-solutions.com >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -- Royce Mitchell IT Consultant ITAS Solutions royce3 at itas-solutions.com From lloyd.aloysius at gmail.com Sat Feb 23 00:56:05 2013 From: lloyd.aloysius at gmail.com (Lloyd Aloysius) Date: Fri, 22 Feb 2013 16:56:05 -0500 Subject: [Freeswitch-users] mod_conference - moderator privileges In-Reply-To: References: Message-ID: Michael, Thank you for the info. I saw the wiki samples. Can you tell me when we use the parameters - *pin *and *moderator-pin* in the conference profile. There is no info in the wiki. Thank you Lloyd * * On Fri, Feb 22, 2013 at 4:16 PM, Michael Collins wrote: > I'm pretty sure this is all covered on the mod_conference wiki page: > http://wiki.freeswitch.org/wiki/Mod_conference#Examples > > I believe you want this example: > confname++flags{endconf|moderator} > > IIRC you can specify the exact PIN you want the moderator to use by > putting in: > confname+123456+flags{endconf|moderator} > > Let us know how that goes. > -MC > > On Fri, Feb 22, 2013 at 12:26 PM, Lloyd Aloysius > wrote: > >> Hi All >> >> How to setup a moderator for a conference? Using moderator flags ? >> >> There are two pin numbers in the conference profile. >> >> 1. pin - regular user >> >> 2. moderator-pin - I assume this pin is for moderator. >> >> If I send a call to conference - how the pin authentication work for the >> conference ? >> >> >> Any help is appreciated. >> >> >> Thanks >> Lloyd >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130222/28c7a7cd/attachment.html From jpyle at fidelityvoice.com Sat Feb 23 01:02:02 2013 From: jpyle at fidelityvoice.com (Jeff Pyle) Date: Fri, 22 Feb 2013 17:02:02 -0500 Subject: [Freeswitch-users] Adtran Total Access 916e 4242916L1 In-Reply-To: References: Message-ID: Ok. I asked because I encountered an issue similar to this on early versions of A2 and A4 code. I run R10.3.2 in most places now and don't encounter it. In my case it's connected to a Level(3) PRIs. I haven't tried anything newer than R10.3.2. Sorry I couldn't be of more help. - Jeff On Fri, Feb 22, 2013 at 4:49 PM, Royce Mitchell III wrote: > R10.5.0.E > > On Fri, Feb 22, 2013 at 3:42 PM, Jeff Pyle > wrote: > > What version AOS on the 908e? > > > > > > On Fri, Feb 22, 2013 at 4:28 PM, Royce Mitchell III > > wrote: > >> > >> rename the zipper attachment to a zip extension. I have attached good > >> and bad isdn traces. > >> > >> The "good" traces are call that have no problems > >> > >> The "bad" traces lose early media whenever the 180 is sent ( after the > >> 183+media ) which happens at this point: > >> > >> 12:19:54.507 ISDN.EP PRI 1 Call to 832XXXXXXX alerting, no PI. > >> > >> On Wed, Feb 20, 2013 at 11:48 AM, Michael Collins > >> wrote: > >> > Do you have d-channel traces of working vs. problematic calls? I'd > love > >> > to > >> > see those. > >> > -MC > >> > > >> > > >> > On Wed, Feb 20, 2013 at 7:45 AM, Royce Mitchell III > > >> > wrote: > >> >> > >> >> when placing an outbound call through the 908e, via PRI hosted by > >> >> TWTelecom and dialing a destination on the AT&T wireless network, a > >> >> "different" series of ISDN codes comes back than from most other > >> >> destinations. This series of ISDN codes causes the Adtran to send a > >> >> 183+early media followed by a 180 w/o media, which causes the caller > >> >> to hear nothing during most of the ringing process. All work-arounds > I > >> >> have tried to employ so far ( like ignore early media ) have had > >> >> disastrous side-effects, we've almost lost clients over them. I put > >> >> out a request to pay for a FreeSWITCH expert to help fix the issue, > >> >> but I've been too busy with other projects to follow up. > >> >> > >> >> On Wed, Feb 20, 2013 at 9:27 AM, Ken Rice > wrote: > >> >> > What kind of incompatibility? As this does not make a lot of sense > >> >> > for a > >> >> > TDM/SIP device to have problems with ATT Wireless a Cellular > network, > >> >> > they > >> >> > cant talk directly to each other > >> >> > > >> >> > On 2/20/13 9:20 AM, "Royce Mitchell III" wrote: > >> >> > > >> >> >> I have a few 908e's, and I'm experiencing a serious > incompatibility > >> >> >> between FreeSWITCH, the 908e, and the AT&T wireless network. I'm > >> >> >> confident I will get it fixed eventually, but just sharing what I > >> >> >> know > >> >> >> about the device. > >> >> >> > >> >> >> On Wed, Feb 20, 2013 at 9:12 AM, Ken Rice > >> >> >> wrote: > >> >> >>> The Adtran TA9XX series are pretty good units. Check out their > >> >> >>> website > >> >> >>> at > >> >> >>> adtran.com to find tech manuals for these things... > >> >> >>> > >> >> >>> I cant answer all the questions below, but these are very stable > >> >> >>> units > >> >> >>> designed to be sold to carriers (and used by many carriers) > >> >> >>> > >> >> >>> How many concurrent connections depends on the specific model you > >> >> >>> are > >> >> >>> getting. > >> >> >>> > >> >> >>> > >> >> >>> On 2/20/13 9:04 AM, "Mimiko" wrote: > >> >> >>> > >> >> >>>> Hi. > >> >> >>>> > >> >> >>>> Who knows about this device: Adtran Total Access 916e 4242916L1 > >> >> >>>> http://www.newegg.com/Product/Product.aspx?Item=N82E16833706027 > >> >> >>>> > >> >> >>>> How stable is it? Does it use LAN ports in failover mode? Does > it > >> >> >>>> use > >> >> >>>> T1/FT1 WAN ports in failover manner connected to a PBX with two > >> >> >>>> PRI > >> >> >>>> ports? How good it is working with FS? How much concurrent > >> >> >>>> connection > >> >> >>>> does it support? > >> >> >>>> > >> >> >>>> I asked to the company via email a week ago, and no response > till > >> >> >>>> now. > >> >> >>> > >> >> >>> -- > >> >> >>> Ken > >> >> >>> http://www.FreeSWITCH.org > >> >> >>> http://www.ClueCon.com > >> >> >>> http://www.OSTAG.org > >> >> >>> irc.freenode.net #freeswitch > >> >> >>> > >> >> >>> > >> >> >>> > >> >> >>> > >> >> >>> > >> >> >>> > _________________________________________________________________________ > >> >> >>> Professional FreeSWITCH Consulting Services: > >> >> >>> consulting at freeswitch.org > >> >> >>> http://www.freeswitchsolutions.com > >> >> >>> > >> >> >>> > >> >> >>> > >> >> >>> > >> >> >>> Official FreeSWITCH Sites > >> >> >>> http://www.freeswitch.org > >> >> >>> http://wiki.freeswitch.org > >> >> >>> http://www.cluecon.com > >> >> >>> > >> >> >>> 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 > >> >> >> > >> >> >> > >> >> > > >> >> > -- > >> >> > Ken > >> >> > http://www.FreeSWITCH.org > >> >> > http://www.ClueCon.com > >> >> > http://www.OSTAG.org > >> >> > irc.freenode.net #freeswitch > >> >> > > >> >> > > >> >> > > >> >> > > >> >> > > >> >> > > _________________________________________________________________________ > >> >> > Professional FreeSWITCH Consulting Services: > >> >> > consulting at freeswitch.org > >> >> > http://www.freeswitchsolutions.com > >> >> > > >> >> > > >> >> > > >> >> > > >> >> > Official FreeSWITCH Sites > >> >> > http://www.freeswitch.org > >> >> > http://wiki.freeswitch.org > >> >> > http://www.cluecon.com > >> >> > > >> >> > 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 > >> >> > >> >> > >> >> > >> >> -- > >> >> Royce Mitchell > >> >> IT Consultant > >> >> ITAS Solutions > >> >> royce3 at itas-solutions.com > >> >> > >> >> > >> >> > _________________________________________________________________________ > >> >> Professional FreeSWITCH Consulting Services: > >> >> consulting at freeswitch.org > >> >> http://www.freeswitchsolutions.com > >> >> > >> >> > >> >> > >> >> > >> >> Official FreeSWITCH Sites > >> >> http://www.freeswitch.org > >> >> http://wiki.freeswitch.org > >> >> http://www.cluecon.com > >> >> > >> >> 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 > >> > > >> > > >> > > >> > > >> > -- > >> > Michael S Collins > >> > Twitter: @mercutioviz > >> > > >> > http://www.FreeSWITCH.org > >> > http://www.ClueCon.com > >> > http://www.OSTAG.org > >> > > >> > > >> > > >> > > _________________________________________________________________________ > >> > Professional FreeSWITCH Consulting Services: > >> > consulting at freeswitch.org > >> > http://www.freeswitchsolutions.com > >> > > >> > > >> > > >> > > >> > Official FreeSWITCH Sites > >> > http://www.freeswitch.org > >> > http://wiki.freeswitch.org > >> > http://www.cluecon.com > >> > > >> > 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 > >> > > >> > >> > >> > >> -- > >> Royce Mitchell > >> IT Consultant > >> ITAS Solutions > >> royce3 at itas-solutions.com > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> 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 > >> > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > 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 > > > > > > -- > Royce Mitchell > IT Consultant > ITAS Solutions > royce3 at itas-solutions.com > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130222/04db0f34/attachment-0001.html From brian at freeswitch.org Sat Feb 23 01:35:05 2013 From: brian at freeswitch.org (Brian West) Date: Fri, 22 Feb 2013 16:35:05 -0600 Subject: [Freeswitch-users] optional SRTP In-Reply-To: References: , <575B91FD-D404-4BD2-BBDE-59FDD6ABD90F@gmail.com> Message-ID: <04C07BED-96ED-4CA8-8344-E09C018DA8FF@freeswitch.org> This is what Polycom and many others do. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire T: +1.918.420.9266 | F: +1.918.420.9267 | M: +1.918.424.WEST iNUM: +883 5100 1420 9266 ISN: 410*543 On Feb 22, 2013, at 3:12 PM, Levend Sayar wrote: > So for optional RTP, SDP must look like something like this ? > > m=audio RTP/AVP > a=blabla > m=audio RTP/SAVP > a=crypto From brian at freeswitch.org Sat Feb 23 01:38:26 2013 From: brian at freeswitch.org (Brian West) Date: Fri, 22 Feb 2013 16:38:26 -0600 Subject: [Freeswitch-users] FS with SSL/TLS issues! In-Reply-To: <1361519970790-7587849.post@n2.nabble.com> References: <1361367187947-7587736.post@n2.nabble.com> <1361513236022-7587844.post@n2.nabble.com> <1361519970790-7587849.post@n2.nabble.com> Message-ID: <113D484C-EA7E-4611-B958-6514F5F2B5B9@freeswitch.org> Just go get a regular signed cert like you would for mod_ssl for apache, It will work. cat the key and the cert into agent.pem and the chain cert into cafile.pem and fire it up... I did this it works great. Startssl.com is a good alternative -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire T: +1.918.420.9266 | F: +1.918.420.9267 | M: +1.918.424.WEST iNUM: +883 5100 1420 9266 ISN: 410*543 On Feb 22, 2013, at 1:59 AM, mehroz wrote: > Moreover, i am unable to import "cafile.pem" in my 3CX phone. Could it be the > reason if these errors? > > 1.Do i need client certificate to establish SSL connection with every > softphone? > 2.I have read that we can use FScomm phone to test but where does it get the > certificate file ? > 3.How can i check that FS has really seen the agent.pem file and OK for > SSL/TLS SIP? > > Meanwhile i'll map domain name on FS server and will try to generate > certificates using domain name. > ...and ill be waiting for anyone to put his thoughts in , where i am! Thanks From msc at freeswitch.org Sat Feb 23 01:57:33 2013 From: msc at freeswitch.org (Michael Collins) Date: Fri, 22 Feb 2013 14:57:33 -0800 Subject: [Freeswitch-users] mod_conference - moderator privileges In-Reply-To: References: Message-ID: I believe you can use those when you want the pin to always be the same for every caller. IIRC, if you set the PIN in the conf profile then it will always ask for a PIN when someone enters a conference. However, if you set it in the dialplan call then you can have a custom PIN for each caller. Standard disclaimerapplies, so please test and report back. -MC On Fri, Feb 22, 2013 at 1:56 PM, Lloyd Aloysius wrote: > Michael, > > Thank you for the info. > > I saw the wiki samples. > > Can you tell me when we use the parameters - *pin *and *moderator-pin* in the conference profile. There is no info in the wiki. > > Thank you > Lloyd > > * > * > > > On Fri, Feb 22, 2013 at 4:16 PM, Michael Collins wrote: > >> I'm pretty sure this is all covered on the mod_conference wiki page: >> http://wiki.freeswitch.org/wiki/Mod_conference#Examples >> >> I believe you want this example: >> confname++flags{endconf|moderator} >> >> IIRC you can specify the exact PIN you want the moderator to use by >> putting in: >> confname+123456+flags{endconf|moderator} >> >> Let us know how that goes. >> -MC >> >> On Fri, Feb 22, 2013 at 12:26 PM, Lloyd Aloysius < >> lloyd.aloysius at gmail.com> wrote: >> >>> Hi All >>> >>> How to setup a moderator for a conference? Using moderator flags ? >>> >>> There are two pin numbers in the conference profile. >>> >>> 1. pin - regular user >>> >>> 2. moderator-pin - I assume this pin is for moderator. >>> >>> If I send a call to conference - how the pin authentication work for the >>> conference ? >>> >>> >>> Any help is appreciated. >>> >>> >>> Thanks >>> Lloyd >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130222/85c2a6fd/attachment.html From dujinfang at gmail.com Sat Feb 23 03:39:00 2013 From: dujinfang at gmail.com (Seven Du) Date: Sat, 23 Feb 2013 08:39:00 +0800 Subject: [Freeswitch-users] [mod_fsv] How the video recording really looks like? In-Reply-To: References: Message-ID: <14FF31176F964AD6A1D70E66EC16FAB9@gmail.com> check http://wiki.freeswitch.org/wiki/Video-recording On Friday, February 22, 2013 at 8:13 PM, K K wrote: > Hi, I would like to record the both sides video on FreeSWITCH during the video call between two UAC. I would like to record each video call separately. > I have tried the mod_fsv by adding following configuration line: > > > > under the > > > > > The problem is that after adding that now when I make a call to any extension I am automatically connected and the *.fsv file is being created. > The party which I have called did not receive any incoming call. > It seems that I am getting automatically the connection to FreeSWITCH which is recording my video and not moving my call to the destination. > > So i guess that is the idea of "video recording" whit mod_fsv? I am not able to achieve my goal? > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto: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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130223/8eb1d735/attachment-0001.html From schoch+freeswitch.org at xwin32.com Sat Feb 23 04:41:58 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Fri, 22 Feb 2013 17:41:58 -0800 Subject: [Freeswitch-users] "Outgoing Call" on phone display In-Reply-To: <2F45CFD8-1F72-44AB-93D4-0C59BED92AD2@freeswitch.org> References: <094868EE-812C-4E21-8551-0EC80F553D3D@freeswitch.org> <2F45CFD8-1F72-44AB-93D4-0C59BED92AD2@freeswitch.org> Message-ID: On Thu, Feb 21, 2013 at 3:27 PM, Brian West wrote: > You know you can do CNAM lookups while dialing and set those values and > update the phone during the ringing phase. It puts Outgoing Call in there > when it has no other information. I tried the following: This had one intended consequence in that it left the 7-digit number in the display, rather than replacing it with the 11-digit number, like it usually does, but setting the effective_called_id_name did not behave as expected. What it did was change the name on the display to the CNAM, as expected, while it was ringing, but as soon as the remote party answered, the display changed again to "Outbound Call". -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130222/9f764cbe/attachment.html From nathandownes at hotmail.com Sat Feb 23 05:32:30 2013 From: nathandownes at hotmail.com (Mr Nathan Downes) Date: Sat, 23 Feb 2013 13:32:30 +1100 Subject: [Freeswitch-users] Door intercom/gate controller In-Reply-To: <144901ce0bbe$5ab583a0$10208ae0$@freeswitch.org> References: <144901ce0bbe$5ab583a0$10208ae0$@freeswitch.org> Message-ID: Hi Michael, Thanks for the response, it worked in the opposite direction, the gate controller (OOB) calls phones (Inband) kind of like the door thing on an apartment block. Start_dtmf worked if I called the gate controller from the phones, but not in the other direction which is how it will be used. It has been resolved as it was discovered the ATA devices inbuilt to the fibre ONT have an option for OOB DTMF, enabling this on the POTS profile allowed the device to function. I am still interested on how I could of got it to work though, is start_dtmf a one directional app? Thanks, Nathan From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Saturday, 16 February 2013 6:35 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Door intercom/gate controller Can you show us the dialplan for sending a call to the gate controller? Also, pastebin the console debug output for a call to the gate controller where you press digits but the controller doesn't respond. Thanks, MC On Wed, Feb 13, 2013 at 2:42 AM, Mr Nathan Downes wrote: Hi list, I have a gate controller that only understands RFC 2833 but it calls endpoints that can only provide inband DTMF, I can't seem to get the digits recognised by the gate controller to trigger the relay using start_dtmf_generate, as they are coming back to the a-leg? Rather than out the b-leg.. It works fine when I call a SPA502g as it will provide RFC 2833.. is there some trickery I can achieve this? Or am I just doing it wrong?? J _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130223/cd1c375a/attachment.html From msc at freeswitch.org Sat Feb 23 06:40:03 2013 From: msc at freeswitch.org (Michael Collins) Date: Fri, 22 Feb 2013 19:40:03 -0800 Subject: [Freeswitch-users] "Outgoing Call" on phone display In-Reply-To: References: <094868EE-812C-4E21-8551-0EC80F553D3D@freeswitch.org> <2F45CFD8-1F72-44AB-93D4-0C59BED92AD2@freeswitch.org> Message-ID: Have you tried setting this variable for the leg in question? http://wiki.freeswitch.org/wiki/Variable_ignore_display_updates -MC On Fri, Feb 22, 2013 at 5:41 PM, Steven Schoch < schoch+freeswitch.org at xwin32.com> wrote: > On Thu, Feb 21, 2013 at 3:27 PM, Brian West wrote: > >> You know you can do CNAM lookups while dialing and set those values and >> update the phone during the ringing phase. It puts Outgoing Call in there >> when it has no other information. > > > I tried the following: > > > > data="effective_caller_id_number=${outbound_caller_id_number}"/> > data="effective_caller_id_name=${outbound_caller_name}"/> > > data="effective_callee_id_name=${cidlookup(1${default_areacode}$1)}"/> > data="sofia/gateway/flowroute/1${default_areacode}$1"/> > > > > This had one intended consequence in that it left the 7-digit number in > the display, rather than replacing it with the 11-digit number, like it > usually does, but setting the effective_called_id_name did not behave as > expected. > > What it did was change the name on the display to the CNAM, as expected, > while it was ringing, but as soon as the remote party answered, the display > changed again to "Outbound Call". > > -- > Steve > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130222/647a0e6e/attachment.html From royce3 at gmail.com Sat Feb 23 07:37:50 2013 From: royce3 at gmail.com (Royce Mitchell III) Date: Fri, 22 Feb 2013 22:37:50 -0600 Subject: [Freeswitch-users] Adtran Total Access 916e 4242916L1 In-Reply-To: References: Message-ID: I'll try R10.3.2 when I get a chance and see if it makes a difference On Fri, Feb 22, 2013 at 4:02 PM, Jeff Pyle wrote: > Ok. I asked because I encountered an issue similar to this on early > versions of A2 and A4 code. I run R10.3.2 in most places now and don't > encounter it. In my case it's connected to a Level(3) PRIs. > > I haven't tried anything newer than R10.3.2. > > Sorry I couldn't be of more help. > > > - Jeff > > > On Fri, Feb 22, 2013 at 4:49 PM, Royce Mitchell III > wrote: >> >> R10.5.0.E >> >> On Fri, Feb 22, 2013 at 3:42 PM, Jeff Pyle >> wrote: >> > What version AOS on the 908e? >> > >> > >> > On Fri, Feb 22, 2013 at 4:28 PM, Royce Mitchell III >> > wrote: >> >> >> >> rename the zipper attachment to a zip extension. I have attached good >> >> and bad isdn traces. >> >> >> >> The "good" traces are call that have no problems >> >> >> >> The "bad" traces lose early media whenever the 180 is sent ( after the >> >> 183+media ) which happens at this point: >> >> >> >> 12:19:54.507 ISDN.EP PRI 1 Call to 832XXXXXXX alerting, no PI. >> >> >> >> On Wed, Feb 20, 2013 at 11:48 AM, Michael Collins >> >> wrote: >> >> > Do you have d-channel traces of working vs. problematic calls? I'd >> >> > love >> >> > to >> >> > see those. >> >> > -MC >> >> > >> >> > >> >> > On Wed, Feb 20, 2013 at 7:45 AM, Royce Mitchell III >> >> > >> >> > wrote: >> >> >> >> >> >> when placing an outbound call through the 908e, via PRI hosted by >> >> >> TWTelecom and dialing a destination on the AT&T wireless network, a >> >> >> "different" series of ISDN codes comes back than from most other >> >> >> destinations. This series of ISDN codes causes the Adtran to send a >> >> >> 183+early media followed by a 180 w/o media, which causes the caller >> >> >> to hear nothing during most of the ringing process. All work-arounds >> >> >> I >> >> >> have tried to employ so far ( like ignore early media ) have had >> >> >> disastrous side-effects, we've almost lost clients over them. I put >> >> >> out a request to pay for a FreeSWITCH expert to help fix the issue, >> >> >> but I've been too busy with other projects to follow up. >> >> >> >> >> >> On Wed, Feb 20, 2013 at 9:27 AM, Ken Rice >> >> >> wrote: >> >> >> > What kind of incompatibility? As this does not make a lot of sense >> >> >> > for a >> >> >> > TDM/SIP device to have problems with ATT Wireless a Cellular >> >> >> > network, >> >> >> > they >> >> >> > cant talk directly to each other >> >> >> > >> >> >> > On 2/20/13 9:20 AM, "Royce Mitchell III" wrote: >> >> >> > >> >> >> >> I have a few 908e's, and I'm experiencing a serious >> >> >> >> incompatibility >> >> >> >> between FreeSWITCH, the 908e, and the AT&T wireless network. I'm >> >> >> >> confident I will get it fixed eventually, but just sharing what I >> >> >> >> know >> >> >> >> about the device. >> >> >> >> >> >> >> >> On Wed, Feb 20, 2013 at 9:12 AM, Ken Rice >> >> >> >> wrote: >> >> >> >>> The Adtran TA9XX series are pretty good units. Check out their >> >> >> >>> website >> >> >> >>> at >> >> >> >>> adtran.com to find tech manuals for these things... >> >> >> >>> >> >> >> >>> I cant answer all the questions below, but these are very stable >> >> >> >>> units >> >> >> >>> designed to be sold to carriers (and used by many carriers) >> >> >> >>> >> >> >> >>> How many concurrent connections depends on the specific model >> >> >> >>> you >> >> >> >>> are >> >> >> >>> getting. >> >> >> >>> >> >> >> >>> >> >> >> >>> On 2/20/13 9:04 AM, "Mimiko" wrote: >> >> >> >>> >> >> >> >>>> Hi. >> >> >> >>>> >> >> >> >>>> Who knows about this device: Adtran Total Access 916e 4242916L1 >> >> >> >>>> http://www.newegg.com/Product/Product.aspx?Item=N82E16833706027 >> >> >> >>>> >> >> >> >>>> How stable is it? Does it use LAN ports in failover mode? Does >> >> >> >>>> it >> >> >> >>>> use >> >> >> >>>> T1/FT1 WAN ports in failover manner connected to a PBX with two >> >> >> >>>> PRI >> >> >> >>>> ports? How good it is working with FS? How much concurrent >> >> >> >>>> connection >> >> >> >>>> does it support? >> >> >> >>>> >> >> >> >>>> I asked to the company via email a week ago, and no response >> >> >> >>>> till >> >> >> >>>> now. >> >> >> >>> >> >> >> >>> -- >> >> >> >>> Ken >> >> >> >>> http://www.FreeSWITCH.org >> >> >> >>> http://www.ClueCon.com >> >> >> >>> http://www.OSTAG.org >> >> >> >>> irc.freenode.net #freeswitch >> >> >> >>> >> >> >> >>> >> >> >> >>> >> >> >> >>> >> >> >> >>> >> >> >> >>> >> >> >> >>> _________________________________________________________________________ >> >> >> >>> Professional FreeSWITCH Consulting Services: >> >> >> >>> consulting at freeswitch.org >> >> >> >>> http://www.freeswitchsolutions.com >> >> >> >>> >> >> >> >>> >> >> >> >>> >> >> >> >>> >> >> >> >>> Official FreeSWITCH Sites >> >> >> >>> http://www.freeswitch.org >> >> >> >>> http://wiki.freeswitch.org >> >> >> >>> http://www.cluecon.com >> >> >> >>> >> >> >> >>> 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 >> >> >> >> >> >> >> >> >> >> >> > >> >> >> > -- >> >> >> > Ken >> >> >> > http://www.FreeSWITCH.org >> >> >> > http://www.ClueCon.com >> >> >> > http://www.OSTAG.org >> >> >> > irc.freenode.net #freeswitch >> >> >> > >> >> >> > >> >> >> > >> >> >> > >> >> >> > >> >> >> > >> >> >> > _________________________________________________________________________ >> >> >> > Professional FreeSWITCH Consulting Services: >> >> >> > consulting at freeswitch.org >> >> >> > http://www.freeswitchsolutions.com >> >> >> > >> >> >> > >> >> >> > >> >> >> > >> >> >> > Official FreeSWITCH Sites >> >> >> > http://www.freeswitch.org >> >> >> > http://wiki.freeswitch.org >> >> >> > http://www.cluecon.com >> >> >> > >> >> >> > 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 >> >> >> >> >> >> >> >> >> >> >> >> -- >> >> >> Royce Mitchell >> >> >> IT Consultant >> >> >> ITAS Solutions >> >> >> royce3 at itas-solutions.com >> >> >> >> >> >> >> >> >> >> >> >> _________________________________________________________________________ >> >> >> Professional FreeSWITCH Consulting Services: >> >> >> consulting at freeswitch.org >> >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> >> http://www.freeswitch.org >> >> >> http://wiki.freeswitch.org >> >> >> http://www.cluecon.com >> >> >> >> >> >> 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 >> >> > >> >> > >> >> > >> >> > >> >> > -- >> >> > Michael S Collins >> >> > Twitter: @mercutioviz >> >> > >> >> > http://www.FreeSWITCH.org >> >> > http://www.ClueCon.com >> >> > http://www.OSTAG.org >> >> > >> >> > >> >> > >> >> > >> >> > _________________________________________________________________________ >> >> > Professional FreeSWITCH Consulting Services: >> >> > consulting at freeswitch.org >> >> > http://www.freeswitchsolutions.com >> >> > >> >> > >> >> > >> >> > >> >> > Official FreeSWITCH Sites >> >> > http://www.freeswitch.org >> >> > http://wiki.freeswitch.org >> >> > http://www.cluecon.com >> >> > >> >> > 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 >> >> > >> >> >> >> >> >> >> >> -- >> >> Royce Mitchell >> >> IT Consultant >> >> ITAS Solutions >> >> royce3 at itas-solutions.com >> >> >> >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://wiki.freeswitch.org >> >> http://www.cluecon.com >> >> >> >> 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 >> >> >> > >> > >> > >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > 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 >> > >> >> >> >> -- >> Royce Mitchell >> IT Consultant >> ITAS Solutions >> royce3 at itas-solutions.com >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -- Royce Mitchell IT Consultant ITAS Solutions royce3 at itas-solutions.com From shayne.alone at gmail.com Sat Feb 23 10:48:10 2013 From: shayne.alone at gmail.com (shayne.alone at gmail.com) Date: Sat, 23 Feb 2013 11:18:10 +0330 Subject: [Freeswitch-users] outbound calls control Message-ID: I need to control over the return value of bridge application: How should on find if the call was rejected bye Carrier_One ( before I fill 10 calls which I am permitted to ) and route then via Carrier_Two? -- Regards, Ali R. Taleghani -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130223/08be1ce2/attachment.html From brett at launch3.net Sat Feb 23 11:25:42 2013 From: brett at launch3.net (Wilsonbrett85) Date: Sat, 23 Feb 2013 00:25:42 -0800 (PST) Subject: [Freeswitch-users] DID providers - any thoughts? (jan 2013) In-Reply-To: References: Message-ID: <1361607942545-7587900.post@n2.nabble.com> +1 for flowroute. No complaints. Web interface is excellent, I believe they have an api as well. Call quality is good and tech support is great. I've dealt with aidin @ flowroute and he really went over and above to help me diagnose an issue. He did his own packet caps and examined my packet caps as well. In the end it was as simple as me needing to configure our pfsense firewall for symmetric Nat. I use them currently on 3 installations. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/DID-providers-any-thoughts-jan-2013-tp7586851p7587900.html Sent from the freeswitch-users mailing list archive at Nabble.com. From a.venugopan at mundio.com Sat Feb 23 13:16:20 2013 From: a.venugopan at mundio.com (Archana Venugopan) Date: Sat, 23 Feb 2013 10:16:20 +0000 Subject: [Freeswitch-users] make mod_voicemail In-Reply-To: References: <1FFF97C269757C458224B7C895F35F1522A507@cantor.std.visionutv.se> <592A9CF93E12394E8472A6CC66E66BF23ACA44@Mail-Kilo.squay.com> <592A9CF93E12394E8472A6CC66E66BF23ACA6D@Mail-Kilo.squay.com> Message-ID: <592A9CF93E12394E8472A6CC66E66BF23AD074@Mail-Kilo.squay.com> Thanks. In that case if I copy mod_voicemail.so/la file from my test environment to production along with mod_voicemail.c without compiling(make mod_voicemail) whether my changes will be changed? Regards, Archana From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 22 February 2013 18:10 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] make mod_voicemail This means that there is something different happening on your production server. One thing you can do is take the binary that is built on your test server and manually copy it over to your production server - assuming they are the same OS and architecture. Be absolutely certain to backup the current mod_voicemail.so/.la files before you copy over the binaries - that way if you have a problem you can always go back. Just an FYI - 1.0.6 is more than 3 years old. You should really be getting your application and configs up and running on the latest git head which is about to be released as 1.2.6. Getting your app work on latest git HEAD is probably less work and more valuable than trying to backport changes into selected 1.0.6 files. -MC On Thu, Feb 21, 2013 at 2:03 AM, Archana Venugopan > wrote: Hi, I haven't updated GIT recently. Freeswitch version I am using is 1.0.6. The same version am using in my test environment as well and there if I give make mod_voicemail-install, only mod_voicemail is being compiled. I face this issue in my production server only. Regards, Archana From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi Marcus Sent: 21 February 2013 09:54 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] make mod_voicemail If you updated GIT since your last compile, then all the dependencies have changed... -Avi On Thu, Feb 21, 2013 at 11:46 AM, Archana Venugopan > wrote: Ya, this is being carried out in our production server. And it was built years before I suppose. So am afraid to proceed as well with make mod_voicemail commands, if it compiles everything then our production set up will change completely. But I made few changes in voicemail and I need to compile mod_voicemail part alone. Please do help me in this. Regards, Archana From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peter Olsson Sent: 21 February 2013 09:38 To: 'FreeSWITCH Users Help' Subject: Re: [Freeswitch-users] make mod_voicemail So, was everything else built already? I believe FS will build the dependencies, if they were not built already - I'm not totally sure about that though. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Archana Venugopan Skickat: den 21 februari 2013 10:30 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] make mod_voicemail I am afraid if that is as well installing everything so I just stopped it. [root at sme-xswitch freeswitch]# make mod_voicemail-install quiet_libtool: compile: gcc -DHAVE_CONFIG_H -I. -I./src/include -I./libs/xmlrpc-c -I/usr/local/src/freeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -pthread -DLINUX=2 -D_REENTRANT -D_GNU_SOURCE -D_LARGEFILE64_SOURCE -I/usr/local/src/freeswitch/libs/apr/include -I/usr/local/src/freeswitch/libs/apr-util/include -I/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib -I/usr/local/src/freeswitch/libs/libtpl-1.5/src -I/usr/local/src/freeswitch/libs/stfu -I/usr/local/src/freeswitch/libs/sqlite -I/usr/local/src/freeswitch/libs/pcre -I/usr/local/src/freeswitch/libs/speex/include -Ilibs/speex/include -I/usr/local/src/freeswitch/libs/srtp/include -I/usr/local/src/freeswitch/libs/srtp/crypto/include -Ilibs/srtp/crypto/include -I/usr/local/src/freeswitch/libs/spandsp/src -I/usr/local/src/freeswitch/libs/tiff-4.0.2/libtiff -DENABLE_SRTP -DSWITCH_HAVE_ODBC -I/usr/include -I/usr/local/src/freeswitch/libs/libedit/src -DSWITCH_HAVE_LIBEDIT -I/usr/local/src/freeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -Wall -std=c99 -pedantic -Wdeclaration-after-statement -g -O2 -MT libfreeswitch_la-switch_ivr.lo -MD -MP -MF .deps/libfreeswitch_la-switch_ivr.Tpo -c src/switch_ivr.c -fPIC -DPIC -o .libs/libfreeswitch_la-switch_ivr.o quiet_libtool: compile: gcc -DHAVE_CONFIG_H -I. -I./src/include -I./libs/xmlrpc-c -I/usr/local/src/freeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -pthread -DLINUX=2 -D_REENTRANT -D_GNU_SOURCE -D_LARGEFILE64_SOURCE -I/usr/local/src/freeswitch/libs/apr/include -I/usr/local/src/freeswitch/libs/apr-util/include -I/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib -I/usr/local/src/freeswitch/libs/libtpl-1.5/src -I/usr/local/src/freeswitch/libs/stfu -I/usr/local/src/freeswitch/libs/sqlite -I/usr/local/src/freeswitch/libs/pcre -I/usr/local/src/freeswitch/libs/speex/include -Ilibs/speex/include -I/usr/local/src/freeswitch/libs/srtp/include -I/usr/local/src/freeswitch/libs/srtp/crypto/include -Ilibs/srtp/crypto/include -I/usr/local/src/freeswitch/libs/spandsp/src -I/usr/local/src/freeswitch/libs/tiff-4.0.2/libtiff -DENABLE_SRTP -DSWITCH_HAVE_ODBC -I/usr/include -I/usr/local/src/freeswitch/libs/libedit/src -DSWITCH_HAVE_LIBEDIT -I/usr/local/src/freeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -Wall -std=c99 -pedantic -Wdeclaration-after-statement -g -O2 -MT libfreeswitch_la-switch_ivr.lo -MD -MP -MF .deps/libfreeswitch_la-switch_ivr.Tpo -c src/switch_ivr.c -o libfreeswitch_la-switch_ivr.o >/dev/null 2>&1 ^Cmake[1]: *** [libfreeswitch_la-switch_ivr.lo] Error 1 make: *** [mod_voicemail-install] Interrupt Regards, Archana From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi Marcus Sent: 21 February 2013 09:24 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] make mod_voicemail Try: make mod_voicemail-install -Avi Marcus On Thu, Feb 21, 2013 at 11:10 AM, Archana Venugopan > wrote: Hi, When I try to give 'make mod_voicemail' command, I see its compiling everything as like make command instead of just doing mod_voicemail. Can someone suggest why is this happening and what can be done to just compile mod_voicemail. Many thanks. Regards, Archana _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 !DSPAM:5125e59032761278114654! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130223/3b672134/attachment-0001.html From brett at launch3.net Sat Feb 23 13:28:50 2013 From: brett at launch3.net (Wilsonbrett85) Date: Sat, 23 Feb 2013 02:28:50 -0800 (PST) Subject: [Freeswitch-users] lua and playAndGetDigits with tts Message-ID: <1361615330934-7587902.post@n2.nabble.com> FreeSWITCH Version 1.3.13b+git~20130208T015804Z~7989798ec4 (git 7989798 2013-02-08 01:58:04Z) Not sure if this is a bug: in lua: session:set_tts_parms("flite", "kal") acct_num = session:playAndGetDigits(1, 5, 3, 3000, "#", "say:Please enter your account number now, followed by the pound key.", "say:Your entry was incorrect.", "\\d+") I get the following in the log and the tts does not work: 2013-02-21 21:15:20.736283 [ERR] switch_ivr_play_say.c:1207 Invalid Args The only way to get it to work is to call the script from an ivr with tts-engine and tts-voice set, or call after the dialplan like so: Why does session:set_tts_parms("flite", "kal") not function as expected? Interestingly I have no issues with session:speak in the same use case. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/lua-and-playAndGetDigits-with-tts-tp7587902.html Sent from the freeswitch-users mailing list archive at Nabble.com. From brett at launch3.net Sat Feb 23 13:40:38 2013 From: brett at launch3.net (Wilsonbrett85) Date: Sat, 23 Feb 2013 02:40:38 -0800 (PST) Subject: [Freeswitch-users] lua and playAndGetDigits with tts In-Reply-To: <1361615330934-7587902.post@n2.nabble.com> References: <1361615330934-7587902.post@n2.nabble.com> Message-ID: <1361616038337-7587903.post@n2.nabble.com> Found that this will fix the issue. Not sure why the set_tts_parms does not. session:setVariable("tts_engine", "flite"); session:setVariable("tts_voice", "kal"); -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/lua-and-playAndGetDigits-with-tts-tp7587902p7587903.html Sent from the freeswitch-users mailing list archive at Nabble.com. From steveayre at gmail.com Sat Feb 23 16:01:44 2013 From: steveayre at gmail.com (Steven Ayre) Date: Sat, 23 Feb 2013 13:01:44 +0000 Subject: [Freeswitch-users] make mod_voicemail In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF23AC9F1@Mail-Kilo.squay.com> References: <592A9CF93E12394E8472A6CC66E66BF23AC9F1@Mail-Kilo.squay.com> Message-ID: mod_voicemail depends on libfreeswitch to be built (the FS core library). That is required, not optional (for the plugin functionality, core API usage etc). It is not possible to build mod_voicemail without either building the core or having it already been built. Also if you had cleaned the directory (eg make clean) that would mean it would need to be rebuilt too. -Steve On 21 February 2013 09:10, Archana Venugopan wrote: > Hi,**** > > ** ** > > When I try to give ?make mod_voicemail? command, I see its compiling > everything as like make command instead of just doing mod_voicemail. **** > > Can someone suggest why is this happening and what can be done to just > compile mod_voicemail. Many thanks.**** > > ** ** > > Regards,**** > > Archana**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130223/d1849a94/attachment.html From steveayre at gmail.com Sat Feb 23 16:06:02 2013 From: steveayre at gmail.com (Steven Ayre) Date: Sat, 23 Feb 2013 13:06:02 +0000 Subject: [Freeswitch-users] make mod_voicemail In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF23ACA1B@Mail-Kilo.squay.com> References: <592A9CF93E12394E8472A6CC66E66BF23AC9F1@Mail-Kilo.squay.com> <592A9CF93E12394E8472A6CC66E66BF23ACA1B@Mail-Kilo.squay.com> Message-ID: This shows that it's building the FS core (src/switch_ivr.c is part of that). That means that file hadn't been built or the build wasn't up-to-date. Building the module without the core is not possible - the below is expected behaviour. Once the core and module are built, the -install will only install the module not the rest. -Steve On 21 February 2013 09:30, Archana Venugopan wrote: > I am afraid if that is as well installing everything so I just stopped > it. **** > > ** ** > > [root at sme-xswitch freeswitch]# make mod_voicemail-install**** > > quiet_libtool: compile: gcc -DHAVE_CONFIG_H -I. -I./src/include > -I./libs/xmlrpc-c -I/usr/local/src/freeswitch/libs/curl/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden > -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -pthread -DLINUX=2 > -D_REENTRANT -D_GNU_SOURCE -D_LARGEFILE64_SOURCE > -I/usr/local/src/freeswitch/libs/apr/include > -I/usr/local/src/freeswitch/libs/apr-util/include > -I/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib > -I/usr/local/src/freeswitch/libs/libtpl-1.5/src > -I/usr/local/src/freeswitch/libs/stfu > -I/usr/local/src/freeswitch/libs/sqlite > -I/usr/local/src/freeswitch/libs/pcre > -I/usr/local/src/freeswitch/libs/speex/include -Ilibs/speex/include > -I/usr/local/src/freeswitch/libs/srtp/include > -I/usr/local/src/freeswitch/libs/srtp/crypto/include > -Ilibs/srtp/crypto/include -I/usr/local/src/freeswitch/libs/spandsp/src > -I/usr/local/src/freeswitch/libs/tiff-4.0.2/libtiff -DENABLE_SRTP > -DSWITCH_HAVE_ODBC -I/usr/include > -I/usr/local/src/freeswitch/libs/libedit/src -DSWITCH_HAVE_LIBEDIT > -I/usr/local/src/freeswitch/libs/curl/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -Werror > -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb > -DHAVE_OPENSSL -Wall -std=c99 -pedantic -Wdeclaration-after-statement -g > -O2 -MT libfreeswitch_la-switch_ivr.lo -MD -MP -MF > .deps/libfreeswitch_la-switch_ivr.Tpo -c src/switch_ivr.c -fPIC -DPIC -o > .libs/libfreeswitch_la-switch_ivr.o**** > > quiet_libtool: compile: gcc -DHAVE_CONFIG_H -I. -I./src/include > -I./libs/xmlrpc-c -I/usr/local/src/freeswitch/libs/curl/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden > -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -pthread -DLINUX=2 > -D_REENTRANT -D_GNU_SOURCE -D_LARGEFILE64_SOURCE > -I/usr/local/src/freeswitch/libs/apr/include > -I/usr/local/src/freeswitch/libs/apr-util/include > -I/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib > -I/usr/local/src/freeswitch/libs/libtpl-1.5/src > -I/usr/local/src/freeswitch/libs/stfu > -I/usr/local/src/freeswitch/libs/sqlite > -I/usr/local/src/freeswitch/libs/pcre > -I/usr/local/src/freeswitch/libs/speex/include -Ilibs/speex/include > -I/usr/local/src/freeswitch/libs/srtp/include > -I/usr/local/src/freeswitch/libs/srtp/crypto/include > -Ilibs/srtp/crypto/include -I/usr/local/src/freeswitch/libs/spandsp/src > -I/usr/local/src/freeswitch/libs/tiff-4.0.2/libtiff -DENABLE_SRTP > -DSWITCH_HAVE_ODBC -I/usr/include > -I/usr/local/src/freeswitch/libs/libedit/src -DSWITCH_HAVE_LIBEDIT > -I/usr/local/src/freeswitch/libs/curl/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -Werror > -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb > -DHAVE_OPENSSL -Wall -std=c99 -pedantic -Wdeclaration-after-statement -g > -O2 -MT libfreeswitch_la-switch_ivr.lo -MD -MP -MF > .deps/libfreeswitch_la-switch_ivr.Tpo -c src/switch_ivr.c -o > libfreeswitch_la-switch_ivr.o >/dev/null 2>&1**** > > ^Cmake[1]: *** [libfreeswitch_la-switch_ivr.lo] Error 1**** > > make: *** [mod_voicemail-install] Interrupt**** > > ** ** > > Regards,**** > > Archana**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Avi Marcus > *Sent:* 21 February 2013 09:24 > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] make mod_voicemail**** > > ** ** > > Try:**** > > > > make mod_voicemail-install**** > > > **** > > -Avi Marcus**** > > ** ** > > On Thu, Feb 21, 2013 at 11:10 AM, Archana Venugopan < > a.venugopan at mundio.com> wrote:**** > > Hi,**** > > **** > > When I try to give ?make mod_voicemail? command, I see its compiling > everything as like make command instead of just doing mod_voicemail. **** > > Can someone suggest why is this happening and what can be done to just > compile mod_voicemail. Many thanks.**** > > **** > > Regards,**** > > Archana**** > > **** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130223/f2d1d8a8/attachment-0001.html From garmt.noname at gmail.com Sat Feb 23 16:32:14 2013 From: garmt.noname at gmail.com (grmt) Date: Sat, 23 Feb 2013 14:32:14 +0100 Subject: [Freeswitch-users] lua and playAndGetDigits with tts In-Reply-To: <1361616038337-7587903.post@n2.nabble.com> References: <1361615330934-7587902.post@n2.nabble.com> <1361616038337-7587903.post@n2.nabble.com> Message-ID: <00f901ce11ca$31e15500$95a3ff00$@gmail.com> Try setparAms -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Wilsonbrett85 Sent: Saturday, February 23, 2013 11:41 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] lua and playAndGetDigits with tts Found that this will fix the issue. Not sure why the set_tts_parms does not. session:setVariable("tts_engine", "flite"); session:setVariable("tts_voice", "kal"); -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/lua-and-playAndGetDigits-with- tts-tp7587902p7587903.html Sent from the freeswitch-users mailing list archive at Nabble.com. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 a.venugopan at mundio.com Sat Feb 23 16:35:13 2013 From: a.venugopan at mundio.com (Archana Venugopan) Date: Sat, 23 Feb 2013 13:35:13 +0000 Subject: [Freeswitch-users] make mod_voicemail In-Reply-To: References: <592A9CF93E12394E8472A6CC66E66BF23AC9F1@Mail-Kilo.squay.com> <592A9CF93E12394E8472A6CC66E66BF23ACA1B@Mail-Kilo.squay.com> Message-ID: <592A9CF93E12394E8472A6CC66E66BF23AD13E@Mail-Kilo.squay.com> If its building the core whether it would affect the scripts inside /usr/local/freeswitch folder as well? If yes then all the scripts inside would be lost. Please let me know. Regards, Archana From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven Ayre Sent: 23 February 2013 13:06 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] make mod_voicemail This shows that it's building the FS core (src/switch_ivr.c is part of that). That means that file hadn't been built or the build wasn't up-to-date. Building the module without the core is not possible - the below is expected behaviour. Once the core and module are built, the -install will only install the module not the rest. -Steve On 21 February 2013 09:30, Archana Venugopan > wrote: I am afraid if that is as well installing everything so I just stopped it. [root at sme-xswitch freeswitch]# make mod_voicemail-install quiet_libtool: compile: gcc -DHAVE_CONFIG_H -I. -I./src/include -I./libs/xmlrpc-c -I/usr/local/src/freeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -pthread -DLINUX=2 -D_REENTRANT -D_GNU_SOURCE -D_LARGEFILE64_SOURCE -I/usr/local/src/freeswitch/libs/apr/include -I/usr/local/src/freeswitch/libs/apr-util/include -I/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib -I/usr/local/src/freeswitch/libs/libtpl-1.5/src -I/usr/local/src/freeswitch/libs/stfu -I/usr/local/src/freeswitch/libs/sqlite -I/usr/local/src/freeswitch/libs/pcre -I/usr/local/src/freeswitch/libs/speex/include -Ilibs/speex/include -I/usr/local/src/freeswitch/libs/srtp/include -I/usr/local/src/freeswitch/libs/srtp/crypto/include -Ilibs/srtp/crypto/include -I/usr/local/src/freeswitch/libs/spandsp/src -I/usr/local/src/freeswitch/libs/tiff-4.0.2/libtiff -DENABLE_SRTP -DSWITCH_HAVE_ODBC -I/usr/include -I/usr/local/src/freeswitch/libs/libedit/src -DSWITCH_HAVE_LIBEDIT -I/usr/local/src/freeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -Wall -std=c99 -pedantic -Wdeclaration-after-statement -g -O2 -MT libfreeswitch_la-switch_ivr.lo -MD -MP -MF .deps/libfreeswitch_la-switch_ivr.Tpo -c src/switch_ivr.c -fPIC -DPIC -o .libs/libfreeswitch_la-switch_ivr.o quiet_libtool: compile: gcc -DHAVE_CONFIG_H -I. -I./src/include -I./libs/xmlrpc-c -I/usr/local/src/freeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -pthread -DLINUX=2 -D_REENTRANT -D_GNU_SOURCE -D_LARGEFILE64_SOURCE -I/usr/local/src/freeswitch/libs/apr/include -I/usr/local/src/freeswitch/libs/apr-util/include -I/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib -I/usr/local/src/freeswitch/libs/libtpl-1.5/src -I/usr/local/src/freeswitch/libs/stfu -I/usr/local/src/freeswitch/libs/sqlite -I/usr/local/src/freeswitch/libs/pcre -I/usr/local/src/freeswitch/libs/speex/include -Ilibs/speex/include -I/usr/local/src/freeswitch/libs/srtp/include -I/usr/local/src/freeswitch/libs/srtp/crypto/include -Ilibs/srtp/crypto/include -I/usr/local/src/freeswitch/libs/spandsp/src -I/usr/local/src/freeswitch/libs/tiff-4.0.2/libtiff -DENABLE_SRTP -DSWITCH_HAVE_ODBC -I/usr/include -I/usr/local/src/freeswitch/libs/libedit/src -DSWITCH_HAVE_LIBEDIT -I/usr/local/src/freeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -Wall -std=c99 -pedantic -Wdeclaration-after-statement -g -O2 -MT libfreeswitch_la-switch_ivr.lo -MD -MP -MF .deps/libfreeswitch_la-switch_ivr.Tpo -c src/switch_ivr.c -o libfreeswitch_la-switch_ivr.o >/dev/null 2>&1 ^Cmake[1]: *** [libfreeswitch_la-switch_ivr.lo] Error 1 make: *** [mod_voicemail-install] Interrupt Regards, Archana From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi Marcus Sent: 21 February 2013 09:24 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] make mod_voicemail Try: make mod_voicemail-install -Avi Marcus On Thu, Feb 21, 2013 at 11:10 AM, Archana Venugopan > wrote: Hi, When I try to give 'make mod_voicemail' command, I see its compiling everything as like make command instead of just doing mod_voicemail. Can someone suggest why is this happening and what can be done to just compile mod_voicemail. Many thanks. Regards, Archana _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130223/fe4be583/attachment-0001.html From steveayre at gmail.com Sat Feb 23 17:36:46 2013 From: steveayre at gmail.com (Steven Ayre) Date: Sat, 23 Feb 2013 14:36:46 +0000 Subject: [Freeswitch-users] make mod_voicemail In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF23AD13E@Mail-Kilo.squay.com> References: <592A9CF93E12394E8472A6CC66E66BF23AC9F1@Mail-Kilo.squay.com> <592A9CF93E12394E8472A6CC66E66BF23ACA1B@Mail-Kilo.squay.com> <592A9CF93E12394E8472A6CC66E66BF23AD13E@Mail-Kilo.squay.com> Message-ID: Building will only affect files within the source directory - not your install paths (/usr/local/freeswitch). Make sure you build mod_voicemail using the same version of FS as you're running though - otherwise there may be differences in the core API between the version being used by freeswitch and by the module. -Steve On 23 February 2013 13:35, Archana Venugopan wrote: > If its building the core whether it would affect the scripts inside > /usr/local/freeswitch folder as well? If yes then all the scripts inside > would be lost. Please let me know.** > > ** ** > > Regards,**** > > Archana**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Steven Ayre > *Sent:* 23 February 2013 13:06 > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] make mod_voicemail**** > > ** ** > > This shows that it's building the FS core (src/switch_ivr.c is part of > that).**** > > ** ** > > That means that file hadn't been built or the build wasn't up-to-date. > Building the module without the core is not possible - the below is > expected behaviour. Once the core and module are built, the -install will > only install the module not the rest.**** > > ** ** > > -Steve**** > > ** ** > > ** ** > > On 21 February 2013 09:30, Archana Venugopan > wrote:**** > > I am afraid if that is as well installing everything so I just stopped it. > **** > > **** > > [root at sme-xswitch freeswitch]# make mod_voicemail-install**** > > quiet_libtool: compile: gcc -DHAVE_CONFIG_H -I. -I./src/include > -I./libs/xmlrpc-c -I/usr/local/src/freeswitch/libs/curl/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden > -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -pthread -DLINUX=2 > -D_REENTRANT -D_GNU_SOURCE -D_LARGEFILE64_SOURCE > -I/usr/local/src/freeswitch/libs/apr/include > -I/usr/local/src/freeswitch/libs/apr-util/include > -I/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib > -I/usr/local/src/freeswitch/libs/libtpl-1.5/src > -I/usr/local/src/freeswitch/libs/stfu > -I/usr/local/src/freeswitch/libs/sqlite > -I/usr/local/src/freeswitch/libs/pcre > -I/usr/local/src/freeswitch/libs/speex/include -Ilibs/speex/include > -I/usr/local/src/freeswitch/libs/srtp/include > -I/usr/local/src/freeswitch/libs/srtp/crypto/include > -Ilibs/srtp/crypto/include -I/usr/local/src/freeswitch/libs/spandsp/src > -I/usr/local/src/freeswitch/libs/tiff-4.0.2/libtiff -DENABLE_SRTP > -DSWITCH_HAVE_ODBC -I/usr/include > -I/usr/local/src/freeswitch/libs/libedit/src -DSWITCH_HAVE_LIBEDIT > -I/usr/local/src/freeswitch/libs/curl/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -Werror > -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb > -DHAVE_OPENSSL -Wall -std=c99 -pedantic -Wdeclaration-after-statement -g > -O2 -MT libfreeswitch_la-switch_ivr.lo -MD -MP -MF > .deps/libfreeswitch_la-switch_ivr.Tpo -c src/switch_ivr.c -fPIC -DPIC -o > .libs/libfreeswitch_la-switch_ivr.o**** > > quiet_libtool: compile: gcc -DHAVE_CONFIG_H -I. -I./src/include > -I./libs/xmlrpc-c -I/usr/local/src/freeswitch/libs/curl/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden > -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -pthread -DLINUX=2 > -D_REENTRANT -D_GNU_SOURCE -D_LARGEFILE64_SOURCE > -I/usr/local/src/freeswitch/libs/apr/include > -I/usr/local/src/freeswitch/libs/apr-util/include > -I/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib > -I/usr/local/src/freeswitch/libs/libtpl-1.5/src > -I/usr/local/src/freeswitch/libs/stfu > -I/usr/local/src/freeswitch/libs/sqlite > -I/usr/local/src/freeswitch/libs/pcre > -I/usr/local/src/freeswitch/libs/speex/include -Ilibs/speex/include > -I/usr/local/src/freeswitch/libs/srtp/include > -I/usr/local/src/freeswitch/libs/srtp/crypto/include > -Ilibs/srtp/crypto/include -I/usr/local/src/freeswitch/libs/spandsp/src > -I/usr/local/src/freeswitch/libs/tiff-4.0.2/libtiff -DENABLE_SRTP > -DSWITCH_HAVE_ODBC -I/usr/include > -I/usr/local/src/freeswitch/libs/libedit/src -DSWITCH_HAVE_LIBEDIT > -I/usr/local/src/freeswitch/libs/curl/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -Werror > -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb > -DHAVE_OPENSSL -Wall -std=c99 -pedantic -Wdeclaration-after-statement -g > -O2 -MT libfreeswitch_la-switch_ivr.lo -MD -MP -MF > .deps/libfreeswitch_la-switch_ivr.Tpo -c src/switch_ivr.c -o > libfreeswitch_la-switch_ivr.o >/dev/null 2>&1**** > > ^Cmake[1]: *** [libfreeswitch_la-switch_ivr.lo] Error 1**** > > make: *** [mod_voicemail-install] Interrupt**** > > **** > > Regards,**** > > Archana**** > > **** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Avi Marcus > *Sent:* 21 February 2013 09:24 > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] make mod_voicemail**** > > **** > > Try:**** > > > > **** > > make mod_voicemail-install**** > > > **** > > -Avi Marcus**** > > **** > > On Thu, Feb 21, 2013 at 11:10 AM, Archana Venugopan < > a.venugopan at mundio.com> wrote:**** > > Hi,**** > > **** > > When I try to give ?make mod_voicemail? command, I see its compiling > everything as like make command instead of just doing mod_voicemail. **** > > Can someone suggest why is this happening and what can be done to just > compile mod_voicemail. Many thanks.**** > > **** > > Regards,**** > > Archana**** > > **** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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**** > > **** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130223/0471eeed/attachment.html From hynek.cihlar at gmail.com Sat Feb 23 20:06:25 2013 From: hynek.cihlar at gmail.com (Hynek Cihlar) Date: Sat, 23 Feb 2013 18:06:25 +0100 Subject: [Freeswitch-users] CCS_RINGING too early In-Reply-To: References: Message-ID: Is there even a valid use case for Freeswitch to set a channel to RINGING before the actual ring is signaled by the far endpoint? What other evidence would be helpful to diagnose the issue? Hynek On Tue, Feb 12, 2013 at 12:03 PM, Hynek Cihlar wrote: > When originating a call the respective call channel's call state is set to > RINGING right after progress 100 is received. > > Here's the captured flow: > > |Time | | > | | | | > |2.848 | INVITE SDP (g711A g711U GSM telephone-eventRTP...e-101 > CN) |SIP From: "" | |(5080) ------------------> (5060) | > |2.848 | 407 Proxy Authentication Required |SIP Status > | |(5080) <------------------ (5060) | > |2.849 | ACK | |SIP Request > | |(5080) ------------------> (5060) | > |2.849 | INVITE SDP (g711A g711U GSM telephone-eventRTP...e-101 > CN) |SIP From: "" | |(5080) ------------------> (5060) | > |2.850 | 100 Trying| |SIP Status > | |(5080) <------------------ (5060) | > |13.444 | 180 Ringing |SIP Status > | |(5080) <------------------ (5060) | > |13.445 | 183 Session Progress SDP (g711A g711U GSM > tele...ne-eventRTPType-101) |SIP Status > | |(5080) <------------------ (5060) | > |13.445 | RTP (g711A) |RTP Num packets:230 > Duration:4.574s SSRC:0x1E777E26 > | |(26056) <------------------ (19312) | > |13.601 | RTP (g711A) |RTP Num packets:220 > Duration:4.419s SSRC:0xA530C519 > | |(26056) ------------------> (19312) | > > After the 100 Trying is received the switch executes > switch_channel_perform_set_running_state (switch_channel.c) and the channel > call state is set to RINGING and ESL event CHANNEL_CALLSTATE:RINGING is > generated. > > I would expect the channel call state to be set to RINGING only after 180 > Ringing is received from the far endpoint. > > Could anybody give me a hint what could be wrong or what steps to take to > figure out? I am already out of ideas. > > Thanks! > Hynek > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130223/9257d463/attachment-0001.html From brett at launch3.net Sat Feb 23 20:35:13 2013 From: brett at launch3.net (Wilsonbrett85) Date: Sat, 23 Feb 2013 09:35:13 -0800 (PST) Subject: [Freeswitch-users] lua and playAndGetDigits with tts In-Reply-To: <00f901ce11ca$31e15500$95a3ff00$@gmail.com> References: <1361615330934-7587902.post@n2.nabble.com> <1361616038337-7587903.post@n2.nabble.com> <00f901ce11ca$31e15500$95a3ff00$@gmail.com> Message-ID: <1361640913216-7587910.post@n2.nabble.com> Then I get this: 2013-02-22 04:35:48.536287 [ERR] mod_lua.cpp:198 /usr/local/freeswitch/scripts/shell_demo.lua:48: attempt to call method 'set_tts_params' (a nil value) stack traceback: /usr/local/freeswitch/scripts/shell_demo.lua:48: in main chunk -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/lua-and-playAndGetDigits-with-tts-tp7587902p7587910.html Sent from the freeswitch-users mailing list archive at Nabble.com. From steveayre at gmail.com Sat Feb 23 20:38:39 2013 From: steveayre at gmail.com (Steven Ayre) Date: Sat, 23 Feb 2013 17:38:39 +0000 Subject: [Freeswitch-users] CCS_RINGING too early In-Reply-To: References: Message-ID: There are some use-cases. The main one would be that you might get FS to generate a ringback tone if you're using ignore_early_media=true since otherwise the caller would never hear any ringing. -Steve On 23 February 2013 17:06, Hynek Cihlar wrote: > Is there even a valid use case for Freeswitch to set a channel to RINGING > before the actual ring is signaled by the far endpoint? What other evidence > would be helpful to diagnose the issue? > > Hynek > > > On Tue, Feb 12, 2013 at 12:03 PM, Hynek Cihlar wrote: > >> When originating a call the respective call channel's call state is set >> to RINGING right after progress 100 is received. >> >> Here's the captured flow: >> >> |Time | | >> | | | | >> |2.848 | INVITE SDP (g711A g711U GSM >> telephone-eventRTP...e-101 CN) |SIP From: "" > To:> | |(5080) ------------------> (5060) | >> |2.848 | 407 Proxy Authentication Required |SIP Status >> | |(5080) <------------------ (5060) | >> |2.849 | ACK | |SIP Request >> | |(5080) ------------------> (5060) | >> |2.849 | INVITE SDP (g711A g711U GSM >> telephone-eventRTP...e-101 CN) |SIP From: "" > To:> | |(5080) ------------------> (5060) | >> |2.850 | 100 Trying| |SIP Status >> | |(5080) <------------------ (5060) | >> |13.444 | 180 Ringing |SIP Status >> | |(5080) <------------------ (5060) | >> |13.445 | 183 Session Progress SDP (g711A g711U GSM >> tele...ne-eventRTPType-101) |SIP Status >> | |(5080) <------------------ (5060) | >> |13.445 | RTP (g711A) |RTP Num packets:230 >> Duration:4.574s SSRC:0x1E777E26 >> | |(26056) <------------------ (19312) | >> |13.601 | RTP (g711A) |RTP Num packets:220 >> Duration:4.419s SSRC:0xA530C519 >> | |(26056) ------------------> (19312) | >> >> After the 100 Trying is received the switch executes >> switch_channel_perform_set_running_state (switch_channel.c) and the channel >> call state is set to RINGING and ESL event CHANNEL_CALLSTATE:RINGING is >> generated. >> >> I would expect the channel call state to be set to RINGING only after 180 >> Ringing is received from the far endpoint. >> >> Could anybody give me a hint what could be wrong or what steps to take to >> figure out? I am already out of ideas. >> >> Thanks! >> Hynek >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130223/7001c765/attachment.html From brett at launch3.net Sat Feb 23 20:41:35 2013 From: brett at launch3.net (Wilsonbrett85) Date: Sat, 23 Feb 2013 09:41:35 -0800 (PST) Subject: [Freeswitch-users] Possible to avoid sending callcenter call to agent if they are on the line? Message-ID: <1361641295507-7587911.post@n2.nabble.com> When a call comes in through the callcenter queue and an agent answers, naturally another call coming in will NOT ring their phone. But what if I want to have this same behavior if an agent calls a local extension or calls out the gateway? So if an agent is on the phone at all, they are skipped over for incoming queue calls. Is it possible? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Possible-to-avoid-sending-callcenter-call-to-agent-if-they-are-on-the-line-tp7587911.html Sent from the freeswitch-users mailing list archive at Nabble.com. From vbvbrj at gmail.com Sat Feb 23 20:58:03 2013 From: vbvbrj at gmail.com (Mimiko) Date: Sat, 23 Feb 2013 19:58:03 +0200 Subject: [Freeswitch-users] Possible to avoid sending callcenter call to agent if they are on the line? In-Reply-To: <1361641295507-7587911.post@n2.nabble.com> References: <1361641295507-7587911.post@n2.nabble.com> Message-ID: <5129032B.7080500@gmail.com> On 23.02.2013 19:41, Wilsonbrett85 wrote: > When a call comes in through the callcenter queue and an agent answers, > naturally another call coming in will NOT ring their phone. > > But what if I want to have this same behavior if an agent calls a local > extension or calls out the gateway? So if an agent is on the phone at all, > they are skipped over for incoming queue calls. Is it possible? There are two situations: 1) The phone can handle second line and call waiting. In this case agent will see/hear that another call is incoming and will answer or not. If agent will not answer, this will be considered that agent is not near phone and the status will change "On break" after a certain number of unanswered calls. 2) The phone does not support second line. Then callcenter will try to establish a connection, but will get "leg B is busy" and will move to next agent, generating for this agent "bridge failed with busy error" event. -- Mimiko desu. From brett at launch3.net Sat Feb 23 21:37:14 2013 From: brett at launch3.net (Wilsonbrett85) Date: Sat, 23 Feb 2013 10:37:14 -0800 (PST) Subject: [Freeswitch-users] How to enable dtmf prompt while in callcenter queue? Message-ID: <1361644634372-7587914.post@n2.nabble.com> So I've got my callcenter setup working, using the following queue position announcement script: http://pastebin.com/sP3Gdq2g My dialplan looks like this: I'm wondering if it's possible to allow the caller to press a key during the announcements to exit the callcenter queue and go straight to voicemail? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/How-to-enable-dtmf-prompt-while-in-callcenter-queue-tp7587914.html Sent from the freeswitch-users mailing list archive at Nabble.com. From anthony.minessale at gmail.com Sat Feb 23 22:17:18 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 23 Feb 2013 13:17:18 -0600 Subject: [Freeswitch-users] CCS_RINGING too early In-Reply-To: References: Message-ID: The data provided is not really relevant to FS. If you want a diagnosis of what is happening in FS you need to provide FS logs from the console. I am not even 100% sure which direction this is but I think you mean outbound from FS to somewhere else. console loglevel debug sofia global siptrace on On Sat, Feb 23, 2013 at 11:38 AM, Steven Ayre wrote: > There are some use-cases. > > The main one would be that you might get FS to generate a ringback tone if > you're using ignore_early_media=true since otherwise the caller would never > hear any ringing. > > -Steve > > > > On 23 February 2013 17:06, Hynek Cihlar wrote: > >> Is there even a valid use case for Freeswitch to set a channel to RINGING >> before the actual ring is signaled by the far endpoint? What other evidence >> would be helpful to diagnose the issue? >> >> Hynek >> >> >> On Tue, Feb 12, 2013 at 12:03 PM, Hynek Cihlar wrote: >> >>> When originating a call the respective call channel's call state is set >>> to RINGING right after progress 100 is received. >>> >>> Here's the captured flow: >>> >>> |Time | | >>> | | | | >>> |2.848 | INVITE SDP (g711A g711U GSM >>> telephone-eventRTP...e-101 CN) |SIP From: "" >> To:>> | |(5080) ------------------> (5060) | >>> |2.848 | 407 Proxy Authentication Required |SIP >>> Status >>> | |(5080) <------------------ (5060) | >>> |2.849 | ACK | |SIP Request >>> | |(5080) ------------------> (5060) | >>> |2.849 | INVITE SDP (g711A g711U GSM >>> telephone-eventRTP...e-101 CN) |SIP From: "" >> To:>> | |(5080) ------------------> (5060) | >>> |2.850 | 100 Trying| |SIP Status >>> | |(5080) <------------------ (5060) | >>> |13.444 | 180 Ringing |SIP Status >>> | |(5080) <------------------ (5060) | >>> |13.445 | 183 Session Progress SDP (g711A g711U GSM >>> tele...ne-eventRTPType-101) |SIP Status >>> | |(5080) <------------------ (5060) | >>> |13.445 | RTP (g711A) |RTP Num packets:230 >>> Duration:4.574s SSRC:0x1E777E26 >>> | |(26056) <------------------ (19312) | >>> |13.601 | RTP (g711A) |RTP Num packets:220 >>> Duration:4.419s SSRC:0xA530C519 >>> | |(26056) ------------------> (19312) | >>> >>> After the 100 Trying is received the switch executes >>> switch_channel_perform_set_running_state (switch_channel.c) and the channel >>> call state is set to RINGING and ESL event CHANNEL_CALLSTATE:RINGING is >>> generated. >>> >>> I would expect the channel call state to be set to RINGING only after >>> 180 Ringing is received from the far endpoint. >>> >>> Could anybody give me a hint what could be wrong or what steps to take >>> to figure out? I am already out of ideas. >>> >>> Thanks! >>> Hynek >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130223/a83002b4/attachment-0001.html From anthony.minessale at gmail.com Sat Feb 23 22:40:26 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 23 Feb 2013 13:40:26 -0600 Subject: [Freeswitch-users] Short word drop-outs In-Reply-To: References: <512750D6.5060401@digitalmail.com> Message-ID: There is nothing of the sort enabled by default anywhere except conference calls. Regular calls are passed through unaltered unless you maybe have a transcoding path so if its happening in that setting its not FS doing it and more likely the TDM gateway or cell phone. Get someone to reproduce it and take a pcap of the call, you should be able to extract the audio and see if any is missing. On Fri, Feb 22, 2013 at 12:29 PM, Michael Collins wrote: > Start with pcapsipdump. Get pcaps of all these calls. Then have the > customer report to you the date, time, caller and callee of such an > occurrence. Locate the pcap and listen to it in Wireshark. See if the > "dropout" occurs in the audio stream or if there is a possible loss of > packets, etc. That should yield some clues as to where to go next. > > If you are really suspicious of the silence suppression stuff then you > might want to turn it off and see what happens. > > -MC > > > On Fri, Feb 22, 2013 at 3:04 AM, Alex Lake wrote: > >> Got a customer who's complaining that short words are being lost in >> conversations over VOIP/Freeswitch. >> >> I presume this is over-enthusiastic silence suppression which could be >> tweaked in the freeswitch config. >> >> I wondered if this might be a handset config issue, but they claim that >> they can use their handsets with other VOIP providers that don't show >> the same problem. >> >> Any tips? >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130223/3f40dd17/attachment.html From hynek.cihlar at gmail.com Sat Feb 23 23:02:02 2013 From: hynek.cihlar at gmail.com (Hynek Cihlar) Date: Sat, 23 Feb 2013 21:02:02 +0100 Subject: [Freeswitch-users] CCS_RINGING too early In-Reply-To: References: Message-ID: Yes, it is outboubd from FS to somewhere else. Here's the console output http://pastebin.com/Awme44SZ. Hynek On Sat, Feb 23, 2013 at 8:17 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > The data provided is not really relevant to FS. > If you want a diagnosis of what is happening in FS you need to provide FS > logs from the console. > > I am not even 100% sure which direction this is but I think you mean > outbound from FS to somewhere else. > > console loglevel debug > sofia global siptrace on > > > > On Sat, Feb 23, 2013 at 11:38 AM, Steven Ayre wrote: > >> There are some use-cases. >> >> The main one would be that you might get FS to generate a ringback tone >> if you're using ignore_early_media=true since otherwise the caller would >> never hear any ringing. >> >> -Steve >> >> >> >> On 23 February 2013 17:06, Hynek Cihlar wrote: >> >>> Is there even a valid use case for Freeswitch to set a channel to >>> RINGING before the actual ring is signaled by the far endpoint? What other >>> evidence would be helpful to diagnose the issue? >>> >>> Hynek >>> >>> >>> On Tue, Feb 12, 2013 at 12:03 PM, Hynek Cihlar wrote: >>> >>>> When originating a call the respective call channel's call state is set >>>> to RINGING right after progress 100 is received. >>>> >>>> Here's the captured flow: >>>> >>>> |Time | | >>>> | | | | >>>> |2.848 | INVITE SDP (g711A g711U GSM >>>> telephone-eventRTP...e-101 CN) |SIP From: "" >>> To:>>> | |(5080) ------------------> (5060) | >>>> |2.848 | 407 Proxy Authentication Required |SIP >>>> Status >>>> | |(5080) <------------------ (5060) | >>>> |2.849 | ACK | |SIP Request >>>> | |(5080) ------------------> (5060) | >>>> |2.849 | INVITE SDP (g711A g711U GSM >>>> telephone-eventRTP...e-101 CN) |SIP From: "" >>> To:>>> | |(5080) ------------------> (5060) | >>>> |2.850 | 100 Trying| |SIP Status >>>> | |(5080) <------------------ (5060) | >>>> |13.444 | 180 Ringing |SIP Status >>>> | |(5080) <------------------ (5060) | >>>> |13.445 | 183 Session Progress SDP (g711A g711U GSM >>>> tele...ne-eventRTPType-101) |SIP Status >>>> | |(5080) <------------------ (5060) | >>>> |13.445 | RTP (g711A) |RTP Num packets:230 >>>> Duration:4.574s SSRC:0x1E777E26 >>>> | |(26056) <------------------ (19312) | >>>> |13.601 | RTP (g711A) |RTP Num packets:220 >>>> Duration:4.419s SSRC:0xA530C519 >>>> | |(26056) ------------------> (19312) | >>>> >>>> After the 100 Trying is received the switch executes >>>> switch_channel_perform_set_running_state (switch_channel.c) and the channel >>>> call state is set to RINGING and ESL event CHANNEL_CALLSTATE:RINGING is >>>> generated. >>>> >>>> I would expect the channel call state to be set to RINGING only after >>>> 180 Ringing is received from the far endpoint. >>>> >>>> Could anybody give me a hint what could be wrong or what steps to take >>>> to figure out? I am already out of ideas. >>>> >>>> Thanks! >>>> Hynek >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130223/fbdf3ad4/attachment-0001.html From anthony.minessale at gmail.com Sat Feb 23 23:16:18 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 23 Feb 2013 14:16:18 -0600 Subject: [Freeswitch-users] CCS_RINGING too early In-Reply-To: References: Message-ID: I think I see why, nobody seems to use those callstate events enough to notice anything wrong with them, Try latest HEAD. On Sat, Feb 23, 2013 at 2:02 PM, Hynek Cihlar wrote: > Yes, it is outboubd from FS to somewhere else. Here's the console output > http://pastebin.com/Awme44SZ. > > Hynek > > > On Sat, Feb 23, 2013 at 8:17 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> The data provided is not really relevant to FS. >> If you want a diagnosis of what is happening in FS you need to provide FS >> logs from the console. >> >> I am not even 100% sure which direction this is but I think you mean >> outbound from FS to somewhere else. >> >> console loglevel debug >> sofia global siptrace on >> >> >> >> On Sat, Feb 23, 2013 at 11:38 AM, Steven Ayre wrote: >> >>> There are some use-cases. >>> >>> The main one would be that you might get FS to generate a ringback tone >>> if you're using ignore_early_media=true since otherwise the caller would >>> never hear any ringing. >>> >>> -Steve >>> >>> >>> >>> On 23 February 2013 17:06, Hynek Cihlar wrote: >>> >>>> Is there even a valid use case for Freeswitch to set a channel to >>>> RINGING before the actual ring is signaled by the far endpoint? What other >>>> evidence would be helpful to diagnose the issue? >>>> >>>> Hynek >>>> >>>> >>>> On Tue, Feb 12, 2013 at 12:03 PM, Hynek Cihlar wrote: >>>> >>>>> When originating a call the respective call channel's call state is >>>>> set to RINGING right after progress 100 is received. >>>>> >>>>> Here's the captured flow: >>>>> >>>>> |Time | | >>>>> | | | | >>>>> |2.848 | INVITE SDP (g711A g711U GSM >>>>> telephone-eventRTP...e-101 CN) |SIP From: "" >>>> To:>>>> | |(5080) ------------------> (5060) | >>>>> |2.848 | 407 Proxy Authentication Required |SIP >>>>> Status >>>>> | |(5080) <------------------ (5060) | >>>>> |2.849 | ACK | |SIP Request >>>>> | |(5080) ------------------> (5060) | >>>>> |2.849 | INVITE SDP (g711A g711U GSM >>>>> telephone-eventRTP...e-101 CN) |SIP From: "" >>>> To:>>>> | |(5080) ------------------> (5060) | >>>>> |2.850 | 100 Trying| |SIP Status >>>>> | |(5080) <------------------ (5060) | >>>>> |13.444 | 180 Ringing |SIP Status >>>>> | |(5080) <------------------ (5060) | >>>>> |13.445 | 183 Session Progress SDP (g711A g711U GSM >>>>> tele...ne-eventRTPType-101) |SIP Status >>>>> | |(5080) <------------------ (5060) | >>>>> |13.445 | RTP (g711A) |RTP Num packets:230 >>>>> Duration:4.574s SSRC:0x1E777E26 >>>>> | |(26056) <------------------ (19312) | >>>>> |13.601 | RTP (g711A) |RTP Num packets:220 >>>>> Duration:4.419s SSRC:0xA530C519 >>>>> | |(26056) ------------------> (19312) | >>>>> >>>>> After the 100 Trying is received the switch executes >>>>> switch_channel_perform_set_running_state (switch_channel.c) and the channel >>>>> call state is set to RINGING and ESL event CHANNEL_CALLSTATE:RINGING is >>>>> generated. >>>>> >>>>> I would expect the channel call state to be set to RINGING only after >>>>> 180 Ringing is received from the far endpoint. >>>>> >>>>> Could anybody give me a hint what could be wrong or what steps to take >>>>> to figure out? I am already out of ideas. >>>>> >>>>> Thanks! >>>>> Hynek >>>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130223/b0cfe8b9/attachment.html From hynek.cihlar at gmail.com Sun Feb 24 00:47:27 2013 From: hynek.cihlar at gmail.com (Hynek Cihlar) Date: Sat, 23 Feb 2013 22:47:27 +0100 Subject: [Freeswitch-users] CCS_RINGING too early In-Reply-To: References: Message-ID: With the HEAD it now works as expected, RINGING comes only after 180/183 is received from the originated endpoint. Could you also merge the change into more stable branch? Do you want me to file a bug for that? Hynek On Sat, Feb 23, 2013 at 9:16 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > I think I see why, nobody seems to use those callstate events enough to > notice anything wrong with them, > Try latest HEAD. > > > > On Sat, Feb 23, 2013 at 2:02 PM, Hynek Cihlar wrote: > >> Yes, it is outboubd from FS to somewhere else. Here's the console output >> http://pastebin.com/Awme44SZ. >> >> Hynek >> >> >> On Sat, Feb 23, 2013 at 8:17 PM, Anthony Minessale < >> anthony.minessale at gmail.com> wrote: >> >>> The data provided is not really relevant to FS. >>> If you want a diagnosis of what is happening in FS you need to provide >>> FS logs from the console. >>> >>> I am not even 100% sure which direction this is but I think you mean >>> outbound from FS to somewhere else. >>> >>> console loglevel debug >>> sofia global siptrace on >>> >>> >>> >>> On Sat, Feb 23, 2013 at 11:38 AM, Steven Ayre wrote: >>> >>>> There are some use-cases. >>>> >>>> The main one would be that you might get FS to generate a ringback tone >>>> if you're using ignore_early_media=true since otherwise the caller would >>>> never hear any ringing. >>>> >>>> -Steve >>>> >>>> >>>> >>>> On 23 February 2013 17:06, Hynek Cihlar wrote: >>>> >>>>> Is there even a valid use case for Freeswitch to set a channel to >>>>> RINGING before the actual ring is signaled by the far endpoint? What other >>>>> evidence would be helpful to diagnose the issue? >>>>> >>>>> Hynek >>>>> >>>>> >>>>> On Tue, Feb 12, 2013 at 12:03 PM, Hynek Cihlar >>>> > wrote: >>>>> >>>>>> When originating a call the respective call channel's call state is >>>>>> set to RINGING right after progress 100 is received. >>>>>> >>>>>> Here's the captured flow: >>>>>> >>>>>> |Time | | >>>>>> | | | | >>>>>> |2.848 | INVITE SDP (g711A g711U GSM >>>>>> telephone-eventRTP...e-101 CN) |SIP From: "" >>>>> To:>>>>> | |(5080) ------------------> (5060) | >>>>>> |2.848 | 407 Proxy Authentication Required |SIP >>>>>> Status >>>>>> | |(5080) <------------------ (5060) | >>>>>> |2.849 | ACK | |SIP Request >>>>>> | |(5080) ------------------> (5060) | >>>>>> |2.849 | INVITE SDP (g711A g711U GSM >>>>>> telephone-eventRTP...e-101 CN) |SIP From: "" >>>>> To:>>>>> | |(5080) ------------------> (5060) | >>>>>> |2.850 | 100 Trying| |SIP Status >>>>>> | |(5080) <------------------ (5060) | >>>>>> |13.444 | 180 Ringing |SIP Status >>>>>> | |(5080) <------------------ (5060) | >>>>>> |13.445 | 183 Session Progress SDP (g711A g711U GSM >>>>>> tele...ne-eventRTPType-101) |SIP Status >>>>>> | |(5080) <------------------ (5060) | >>>>>> |13.445 | RTP (g711A) |RTP Num >>>>>> packets:230 Duration:4.574s SSRC:0x1E777E26 >>>>>> | |(26056) <------------------ (19312) | >>>>>> |13.601 | RTP (g711A) |RTP Num >>>>>> packets:220 Duration:4.419s SSRC:0xA530C519 >>>>>> | |(26056) ------------------> (19312) | >>>>>> >>>>>> After the 100 Trying is received the switch executes >>>>>> switch_channel_perform_set_running_state (switch_channel.c) and the channel >>>>>> call state is set to RINGING and ESL event CHANNEL_CALLSTATE:RINGING is >>>>>> generated. >>>>>> >>>>>> I would expect the channel call state to be set to RINGING only after >>>>>> 180 Ringing is received from the far endpoint. >>>>>> >>>>>> Could anybody give me a hint what could be wrong or what steps to >>>>>> take to figure out? I am already out of ideas. >>>>>> >>>>>> Thanks! >>>>>> Hynek >>>>>> >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> 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 >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>>> >>>> >>> >>> >>> -- >>> Anthony Minessale II >>> >>> FreeSWITCH http://www.freeswitch.org/ >>> ClueCon http://www.cluecon.com/ >>> Twitter: http://twitter.com/FreeSWITCH_wire >>> >>> AIM: anthm >>> MSN:anthony_minessale at hotmail.com >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>> IRC: irc.freenode.net #freeswitch >>> >>> FreeSWITCH Developer Conference >>> sip:888 at conference.freeswitch.org >>> googletalk:conf+888 at conference.freeswitch.org >>> pstn:+19193869900 >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130223/6a798eb0/attachment-0001.html From anthony.minessale at gmail.com Sun Feb 24 01:26:08 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 23 Feb 2013 16:26:08 -0600 Subject: [Freeswitch-users] CCS_RINGING too early In-Reply-To: References: Message-ID: Head is the most stable branch currently, its in final stage. We will not merge things until 1.2 is branched and 1.5 is tagged on head. On Feb 23, 2013 3:51 PM, "Hynek Cihlar" wrote: > With the HEAD it now works as expected, RINGING comes only after 180/183 > is received from the originated endpoint. > > Could you also merge the change into more stable branch? Do you want me to > file a bug for that? > > Hynek > > > On Sat, Feb 23, 2013 at 9:16 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> I think I see why, nobody seems to use those callstate events enough to >> notice anything wrong with them, >> Try latest HEAD. >> >> >> >> On Sat, Feb 23, 2013 at 2:02 PM, Hynek Cihlar wrote: >> >>> Yes, it is outboubd from FS to somewhere else. Here's the console output >>> http://pastebin.com/Awme44SZ. >>> >>> Hynek >>> >>> >>> On Sat, Feb 23, 2013 at 8:17 PM, Anthony Minessale < >>> anthony.minessale at gmail.com> wrote: >>> >>>> The data provided is not really relevant to FS. >>>> If you want a diagnosis of what is happening in FS you need to provide >>>> FS logs from the console. >>>> >>>> I am not even 100% sure which direction this is but I think you mean >>>> outbound from FS to somewhere else. >>>> >>>> console loglevel debug >>>> sofia global siptrace on >>>> >>>> >>>> >>>> On Sat, Feb 23, 2013 at 11:38 AM, Steven Ayre wrote: >>>> >>>>> There are some use-cases. >>>>> >>>>> The main one would be that you might get FS to generate a ringback >>>>> tone if you're using ignore_early_media=true since otherwise the caller >>>>> would never hear any ringing. >>>>> >>>>> -Steve >>>>> >>>>> >>>>> >>>>> On 23 February 2013 17:06, Hynek Cihlar wrote: >>>>> >>>>>> Is there even a valid use case for Freeswitch to set a channel to >>>>>> RINGING before the actual ring is signaled by the far endpoint? What other >>>>>> evidence would be helpful to diagnose the issue? >>>>>> >>>>>> Hynek >>>>>> >>>>>> >>>>>> On Tue, Feb 12, 2013 at 12:03 PM, Hynek Cihlar < >>>>>> hynek.cihlar at gmail.com> wrote: >>>>>> >>>>>>> When originating a call the respective call channel's call state is >>>>>>> set to RINGING right after progress 100 is received. >>>>>>> >>>>>>> Here's the captured flow: >>>>>>> >>>>>>> |Time | | >>>>>>> | | | | >>>>>>> >>>>>>> |2.848 | INVITE SDP (g711A g711U GSM >>>>>>> telephone-eventRTP...e-101 CN) |SIP From: "" >>>>>> To:>>>>>> | |(5080) ------------------> (5060) | >>>>>>> |2.848 | 407 Proxy Authentication Required |SIP >>>>>>> Status >>>>>>> | |(5080) <------------------ (5060) | >>>>>>> |2.849 | ACK | |SIP Request >>>>>>> | |(5080) ------------------> (5060) | >>>>>>> |2.849 | INVITE SDP (g711A g711U GSM >>>>>>> telephone-eventRTP...e-101 CN) |SIP From: "" >>>>>> To:>>>>>> | |(5080) ------------------> (5060) | >>>>>>> |2.850 | 100 Trying| |SIP Status >>>>>>> | |(5080) <------------------ (5060) | >>>>>>> |13.444 | 180 Ringing |SIP Status >>>>>>> | |(5080) <------------------ (5060) | >>>>>>> |13.445 | 183 Session Progress SDP (g711A g711U GSM >>>>>>> tele...ne-eventRTPType-101) |SIP Status >>>>>>> | |(5080) <------------------ (5060) | >>>>>>> |13.445 | RTP (g711A) |RTP Num >>>>>>> packets:230 Duration:4.574s SSRC:0x1E777E26 >>>>>>> | |(26056) <------------------ (19312) | >>>>>>> |13.601 | RTP (g711A) |RTP Num >>>>>>> packets:220 Duration:4.419s SSRC:0xA530C519 >>>>>>> | |(26056) ------------------> (19312) | >>>>>>> >>>>>>> After the 100 Trying is received the switch executes >>>>>>> switch_channel_perform_set_running_state (switch_channel.c) and the channel >>>>>>> call state is set to RINGING and ESL event CHANNEL_CALLSTATE:RINGING is >>>>>>> generated. >>>>>>> >>>>>>> I would expect the channel call state to be set to RINGING only >>>>>>> after 180 Ringing is received from the far endpoint. >>>>>>> >>>>>>> Could anybody give me a hint what could be wrong or what steps to >>>>>>> take to figure out? I am already out of ideas. >>>>>>> >>>>>>> Thanks! >>>>>>> Hynek >>>>>>> >>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> 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 >>>>>> >>>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> 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 >>>>> >>>>> >>>> >>>> >>>> -- >>>> Anthony Minessale II >>>> >>>> FreeSWITCH http://www.freeswitch.org/ >>>> ClueCon http://www.cluecon.com/ >>>> Twitter: http://twitter.com/FreeSWITCH_wire >>>> >>>> AIM: anthm >>>> MSN:anthony_minessale at hotmail.com >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>> IRC: irc.freenode.net #freeswitch >>>> >>>> FreeSWITCH Developer Conference >>>> sip:888 at conference.freeswitch.org >>>> googletalk:conf+888 at conference.freeswitch.org >>>> pstn:+19193869900 >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130223/99b89a82/attachment-0001.html From gabe at gundy.org Sun Feb 24 05:09:02 2013 From: gabe at gundy.org (Gabriel Gunderson) Date: Sat, 23 Feb 2013 19:09:02 -0700 Subject: [Freeswitch-users] CCS_RINGING too early In-Reply-To: References: Message-ID: On Sat, Feb 23, 2013 at 1:16 PM, Anthony Minessale wrote: > I think I see why, nobody seems to use those callstate events enough to > notice anything wrong with them, I use the call state events; I'll keep my eyes open :) Gabe From david.villasmil.work at gmail.com Sun Feb 24 05:27:27 2013 From: david.villasmil.work at gmail.com (David Villasmil) Date: Sun, 24 Feb 2013 03:27:27 +0100 Subject: [Freeswitch-users] outbound calls control In-Reply-To: References: Message-ID: <3BE08C3D-447A-4657-9F5B-717542363B5A@gmail.com> Maybe with: Failover for your outbound gateway is easy to implement at bridge time using the | separator: Or By checking ${bridge_hangup_cause}? --- David Villasmil On Feb 23, 2013, at 8:48, "shayne.alone at gmail.com" wrote: > I need to control over the return value of bridge application: > > How should on find if the call was rejected bye Carrier_One ( before I fill 10 calls which I am permitted to ) and route then via Carrier_Two? > > > > > > > > > > > > > > > > > > > > > > > > > > > -- > Regards, > Ali R. Taleghani > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130224/974db51c/attachment.html From hynek.cihlar at gmail.com Sun Feb 24 14:31:23 2013 From: hynek.cihlar at gmail.com (Hynek Cihlar) Date: Sun, 24 Feb 2013 12:31:23 +0100 Subject: [Freeswitch-users] CCS_RINGING too early In-Reply-To: References: Message-ID: Good enough, thanks! Hynek On Sat, Feb 23, 2013 at 11:26 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > Head is the most stable branch currently, its in final stage. We will not > merge things until 1.2 is branched and 1.5 is tagged on head. > On Feb 23, 2013 3:51 PM, "Hynek Cihlar" wrote: > >> With the HEAD it now works as expected, RINGING comes only after 180/183 >> is received from the originated endpoint. >> >> Could you also merge the change into more stable branch? Do you want me >> to file a bug for that? >> >> Hynek >> >> >> On Sat, Feb 23, 2013 at 9:16 PM, Anthony Minessale < >> anthony.minessale at gmail.com> wrote: >> >>> I think I see why, nobody seems to use those callstate events enough to >>> notice anything wrong with them, >>> Try latest HEAD. >>> >>> >>> >>> On Sat, Feb 23, 2013 at 2:02 PM, Hynek Cihlar wrote: >>> >>>> Yes, it is outboubd from FS to somewhere else. Here's the console >>>> output http://pastebin.com/Awme44SZ. >>>> >>>> Hynek >>>> >>>> >>>> On Sat, Feb 23, 2013 at 8:17 PM, Anthony Minessale < >>>> anthony.minessale at gmail.com> wrote: >>>> >>>>> The data provided is not really relevant to FS. >>>>> If you want a diagnosis of what is happening in FS you need to provide >>>>> FS logs from the console. >>>>> >>>>> I am not even 100% sure which direction this is but I think you mean >>>>> outbound from FS to somewhere else. >>>>> >>>>> console loglevel debug >>>>> sofia global siptrace on >>>>> >>>>> >>>>> >>>>> On Sat, Feb 23, 2013 at 11:38 AM, Steven Ayre wrote: >>>>> >>>>>> There are some use-cases. >>>>>> >>>>>> The main one would be that you might get FS to generate a ringback >>>>>> tone if you're using ignore_early_media=true since otherwise the caller >>>>>> would never hear any ringing. >>>>>> >>>>>> -Steve >>>>>> >>>>>> >>>>>> >>>>>> On 23 February 2013 17:06, Hynek Cihlar wrote: >>>>>> >>>>>>> Is there even a valid use case for Freeswitch to set a channel to >>>>>>> RINGING before the actual ring is signaled by the far endpoint? What other >>>>>>> evidence would be helpful to diagnose the issue? >>>>>>> >>>>>>> Hynek >>>>>>> >>>>>>> >>>>>>> On Tue, Feb 12, 2013 at 12:03 PM, Hynek Cihlar < >>>>>>> hynek.cihlar at gmail.com> wrote: >>>>>>> >>>>>>>> When originating a call the respective call channel's call state is >>>>>>>> set to RINGING right after progress 100 is received. >>>>>>>> >>>>>>>> Here's the captured flow: >>>>>>>> >>>>>>>> |Time | | >>>>>>>> | | | | >>>>>>>> >>>>>>>> |2.848 | INVITE SDP (g711A g711U GSM >>>>>>>> telephone-eventRTP...e-101 CN) |SIP From: "" >>>>>>> To:>>>>>>> | |(5080) ------------------> (5060) | >>>>>>>> |2.848 | 407 Proxy Authentication Required |SIP >>>>>>>> Status >>>>>>>> | |(5080) <------------------ (5060) | >>>>>>>> |2.849 | ACK | |SIP Request >>>>>>>> | |(5080) ------------------> (5060) | >>>>>>>> |2.849 | INVITE SDP (g711A g711U GSM >>>>>>>> telephone-eventRTP...e-101 CN) |SIP From: "" >>>>>>> To:>>>>>>> | |(5080) ------------------> (5060) | >>>>>>>> |2.850 | 100 Trying| |SIP Status >>>>>>>> | |(5080) <------------------ (5060) | >>>>>>>> |13.444 | 180 Ringing |SIP Status >>>>>>>> | |(5080) <------------------ (5060) | >>>>>>>> |13.445 | 183 Session Progress SDP (g711A g711U GSM >>>>>>>> tele...ne-eventRTPType-101) |SIP Status >>>>>>>> | |(5080) <------------------ (5060) | >>>>>>>> |13.445 | RTP (g711A) |RTP Num >>>>>>>> packets:230 Duration:4.574s SSRC:0x1E777E26 >>>>>>>> | |(26056) <------------------ (19312) | >>>>>>>> |13.601 | RTP (g711A) |RTP Num >>>>>>>> packets:220 Duration:4.419s SSRC:0xA530C519 >>>>>>>> | |(26056) ------------------> (19312) | >>>>>>>> >>>>>>>> After the 100 Trying is received the switch executes >>>>>>>> switch_channel_perform_set_running_state (switch_channel.c) and the channel >>>>>>>> call state is set to RINGING and ESL event CHANNEL_CALLSTATE:RINGING is >>>>>>>> generated. >>>>>>>> >>>>>>>> I would expect the channel call state to be set to RINGING only >>>>>>>> after 180 Ringing is received from the far endpoint. >>>>>>>> >>>>>>>> Could anybody give me a hint what could be wrong or what steps to >>>>>>>> take to figure out? I am already out of ideas. >>>>>>>> >>>>>>>> Thanks! >>>>>>>> Hynek >>>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://wiki.freeswitch.org >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> 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 >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> 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 >>>>>> >>>>>> >>>>> >>>>> >>>>> -- >>>>> Anthony Minessale II >>>>> >>>>> FreeSWITCH http://www.freeswitch.org/ >>>>> ClueCon http://www.cluecon.com/ >>>>> Twitter: http://twitter.com/FreeSWITCH_wire >>>>> >>>>> AIM: anthm >>>>> MSN:anthony_minessale at hotmail.com >>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>>> IRC: irc.freenode.net #freeswitch >>>>> >>>>> FreeSWITCH Developer Conference >>>>> sip:888 at conference.freeswitch.org >>>>> googletalk:conf+888 at conference.freeswitch.org >>>>> pstn:+19193869900 >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> 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 >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>>> >>>> >>> >>> >>> -- >>> Anthony Minessale II >>> >>> FreeSWITCH http://www.freeswitch.org/ >>> ClueCon http://www.cluecon.com/ >>> Twitter: http://twitter.com/FreeSWITCH_wire >>> >>> AIM: anthm >>> MSN:anthony_minessale at hotmail.com >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>> IRC: irc.freenode.net #freeswitch >>> >>> FreeSWITCH Developer Conference >>> sip:888 at conference.freeswitch.org >>> googletalk:conf+888 at conference.freeswitch.org >>> pstn:+19193869900 >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130224/0cc2ac99/attachment-0001.html From vittorio.guglielmo at gmail.com Mon Feb 25 00:58:11 2013 From: vittorio.guglielmo at gmail.com (Vittorio Guglielmo) Date: Sun, 24 Feb 2013 22:58:11 +0100 Subject: [Freeswitch-users] FreeSWITCH suddenly core dump Message-ID: <512A8CF3.30107@gmail.com> Hello everybody, I'm new to freeswitch (I recently switched form Asterisk). This evening, after one week of work (I have only 3 accounts enabled), freeswitch suddenly core-dumped. Now, every time I try to start it, it pushes a core dump. I read the wiki about procedures to gain back traces, and used the fscore_pb from the support-d of the source-dir, and here is the URL it gave to me for you: http://pastebin.freeswitch.org/20623 Thanks for your help Vic -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130224/e7f64b1f/attachment.html From krice at freeswitch.org Mon Feb 25 01:54:46 2013 From: krice at freeswitch.org (Ken Rice) Date: Sun, 24 Feb 2013 16:54:46 -0600 Subject: [Freeswitch-users] FreeSWITCH suddenly core dump In-Reply-To: <512A8CF3.30107@gmail.com> Message-ID: According to your coredump, the crash is not in FreeSWITCH, but in libmysql. Make sure you are using the threadsafe version of libmysql check there wiki there is info on this On 2/24/13 3:58 PM, "Vittorio Guglielmo" wrote: > Hello everybody, > > I'm new to freeswitch (I recently switched form Asterisk). > > This evening, after one week of work (I have only 3 accounts enabled), > freeswitch suddenly core-dumped. > > Now, every time I try to start it, it pushes a core dump. > > I read the wiki about procedures to gain back traces, and used the fscore_pb > from the support-d of the source-dir, > > and here is the URL it gave to me for you: > http://pastebin.freeswitch.org/20623 > > > Thanks for your help > > Vic > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130224/6b9ee71f/attachment.html From krice at freeswitch.org Mon Feb 25 02:12:56 2013 From: krice at freeswitch.org (Ken Rice) Date: Sun, 24 Feb 2013 17:12:56 -0600 Subject: [Freeswitch-users] SwitchPi - FreeSWITCH Meets the Raspberry Pi with some small Open Hardware Addons Message-ID: Hey Guys, May of you know I have been working to get FreeSWITCH crammed on to the RaspberryPi. And a few of you have noticed where I have been blogging about it at http://www.switchpi.org/ well its come time to ask for a little help... If you have an EE background and have recently sent boards sent off to a short run proto fab and want to assist in a little refinement of the PCB, I think its time to start heading down that road. Please contact me off list with how you can help. Where am I currently? A) FreeSWITCH Building cleaning on Raspbian (I?ll be making an image with FS precompiled and installed shortly), B) breadboard hardware working with Python and limited C sources. Eagle Files and Source code is on Github (Linked off the blog for easy reference) K -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org http://www.switchpi.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130224/c592af0b/attachment.html From anthony.minessale at gmail.com Mon Feb 25 02:32:59 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 24 Feb 2013 17:32:59 -0600 Subject: [Freeswitch-users] FreeSWITCH suddenly core dump In-Reply-To: References: <512A8CF3.30107@gmail.com> Message-ID: Can't even tell which version this is: Build from fresh GIT HEAD (make current) and report to JIRA if its still there. On Sun, Feb 24, 2013 at 4:54 PM, Ken Rice wrote: > According to your coredump, the crash is not in FreeSWITCH, but in > libmysql. Make sure you are using the threadsafe version of libmysql check > there wiki there is info on this > > > > On 2/24/13 3:58 PM, "Vittorio Guglielmo" > wrote: > > Hello everybody, > > I'm new to freeswitch (I recently switched form Asterisk). > > This evening, after one week of work (I have only 3 accounts enabled), > freeswitch suddenly core-dumped. > > Now, every time I try to start it, it pushes a core dump. > > I read the wiki about procedures to gain back traces, and used the > fscore_pb from the support-d of the source-dir, > > and here is the URL it gave to me for you: > http://pastebin.freeswitch.org/20623 < > http://pastebin.freeswitch.org/20623> > > > Thanks for your help > > Vic > > > ------------------------------ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > -- > Ken > *http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > *irc.freenode.net #freeswitch > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130224/97594f5c/attachment.html From lists at kavun.ch Mon Feb 25 07:26:56 2013 From: lists at kavun.ch (Emrah) Date: Sun, 24 Feb 2013 23:26:56 -0500 Subject: [Freeswitch-users] "Outgoing Call" on phone display In-Reply-To: References: <094868EE-812C-4E21-8551-0EC80F553D3D@freeswitch.org> <2F45CFD8-1F72-44AB-93D4-0C59BED92AD2@freeswitch.org> Message-ID: Curious to know how we can fix this as well. Cheers On Feb 22, 2013, at 8:41 PM, Steven Schoch wrote: > On Thu, Feb 21, 2013 at 3:27 PM, Brian West wrote: > You know you can do CNAM lookups while dialing and set those values and update the phone during the ringing phase. It puts Outgoing Call in there when it has no other information. > > I tried the following: > > > > > > > > > > > > This had one intended consequence in that it left the 7-digit number in the display, rather than replacing it with the 11-digit number, like it usually does, but setting the effective_called_id_name did not behave as expected. > > What it did was change the name on the display to the CNAM, as expected, while it was ringing, but as soon as the remote party answered, the display changed again to "Outbound Call". > > -- > Steve > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 lists at kavun.ch Mon Feb 25 07:33:14 2013 From: lists at kavun.ch (Emrah) Date: Sun, 24 Feb 2013 23:33:14 -0500 Subject: [Freeswitch-users] Changing codec during calls In-Reply-To: References: <6EF8B8E9-4CC5-480C-B1C1-09CBC533470C@kavun.ch> <35AF9E79-771D-440A-BEBB-A856D831161C@kavun.ch> <564748CC-A8D1-44ED-9AC4-55910DBB7F2A@kavun.ch> Message-ID: <5EAECFBC-CB73-4633-9F18-56B7818F6DE6@kavun.ch> Is it necessary to file a bug on the channel info issue? The channel data does not get updated when changing codecs and a "show channel" always shows the first negotiated codec. This is an incredible feature and ever so useful, thanks a bunch. E On Feb 22, 2013, at 11:59 AM, Daniel Ivanov wrote: > Freeswitch is turning into a mean, mean event-driven voip panacea... > Congrats on the new commit. > > On Feb 22, 2013 6:49 AM, "Emrah" wrote: > Hi guys, > > Sorry was disconnected from the list for a while. > Indeed it looks like the latest commit fixed it. > The issue I see now is that the channel info are not updated and a show channels will only list the initial codecs. > > Great job and thanks a bunch, > Emrah > On Feb 21, 2013, at 8:42 PM, Dmitry Lysenko wrote: > > > Now this cool feature is fully working! Today Anthony committed the bugfix! > > Tested with 2 different SIP UA<->FS<->Callcentric. Both legs. > > Now possible to write application that will switch codecs on the fly, regarding on any event, such as switching to low bandwidth backup internet line or too high cpu load, etc. > > BTW, is there way to get API access to statistic of jitter buffer? > > Thanks. > > > > P.S. As I know, asterisk can't do such thing. > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 lists at kavun.ch Mon Feb 25 10:32:38 2013 From: lists at kavun.ch (Emrah) Date: Mon, 25 Feb 2013 02:32:38 -0500 Subject: [Freeswitch-users] conference-flags In-Reply-To: References: <68031E41-1E74-49C5-AC80-2ABFCBB1DB52@kavun.ch> <07101B0D-F9A9-4845-A9B8-5F591A9F8949@kavun.ch> Message-ID: <9F5AD8DC-733D-49EC-AC91-000697A9A304@kavun.ch> Anybody up for some clarifications on this? Cheers On Jan 16, 2013, at 8:27 PM, Michael Collins wrote: > I believe that these have to do with which video stream gets displayed. We have a user (sorry, can't recall name off top of my head) who does work with deaf/hearing-impaired and he needs to do video conferences where the translator does audio between the speaker and video to the deaf person. I think these affect that. I someone can shed some light on this that would be helpful. It's the third Monday of the week for me and I'm a little fried... > > Thanks, > MC > > On Wed, Jan 16, 2013 at 7:41 AM, Emrah wrote: > Hi guys, > > If any of you know the answer to this I would be very grateful if you could take a minute and post it on the Wiki or reply to my email. > > Thanks! > On Jan 11, 2013, at 2:28 PM, Emrah wrote: > > > Hi all, > > > > Where can I find some explanations on how to use conference-flags? > > > > How do you use video-bridge and video-floor and what do they do? > > > > Thanks for any pointers. > > > > E > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 dvl36.ripe.nick at gmail.com Mon Feb 25 10:53:34 2013 From: dvl36.ripe.nick at gmail.com (Dmitry Lysenko) Date: Mon, 25 Feb 2013 09:53:34 +0200 Subject: [Freeswitch-users] Changing codec during calls In-Reply-To: <5EAECFBC-CB73-4633-9F18-56B7818F6DE6@kavun.ch> References: <6EF8B8E9-4CC5-480C-B1C1-09CBC533470C@kavun.ch> <35AF9E79-771D-440A-BEBB-A856D831161C@kavun.ch> <564748CC-A8D1-44ED-9AC4-55910DBB7F2A@kavun.ch> <5EAECFBC-CB73-4633-9F18-56B7818F6DE6@kavun.ch> Message-ID: Emrah, please file a bug to jira. Btw, are you checked changing of codec ptime during call? Thanks. 2013/2/25 Emrah > Is it necessary to file a bug on the channel info issue? > The channel data does not get updated when changing codecs and a "show > channel" always shows the first negotiated codec. > > This is an incredible feature and ever so useful, thanks a bunch. > > E > On Feb 22, 2013, at 11:59 AM, Daniel Ivanov wrote: > > > Freeswitch is turning into a mean, mean event-driven voip panacea... > > Congrats on the new commit. > > > > On Feb 22, 2013 6:49 AM, "Emrah" wrote: > > Hi guys, > > > > Sorry was disconnected from the list for a while. > > Indeed it looks like the latest commit fixed it. > > The issue I see now is that the channel info are not updated and a show > channels will only list the initial codecs. > > > > Great job and thanks a bunch, > > Emrah > > On Feb 21, 2013, at 8:42 PM, Dmitry Lysenko > wrote: > > > > > Now this cool feature is fully working! Today Anthony committed the > bugfix! > > > Tested with 2 different SIP UA<->FS<->Callcentric. Both legs. > > > Now possible to write application that will switch codecs on the fly, > regarding on any event, such as switching to low bandwidth backup internet > line or too high cpu load, etc. > > > BTW, is there way to get API access to statistic of jitter buffer? > > > Thanks. > > > > > > P.S. As I know, asterisk can't do such thing. > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > > > > > 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 > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > 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 > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130225/43805c32/attachment.html From emamirazavi at gmail.com Mon Feb 25 11:32:26 2013 From: emamirazavi at gmail.com (Sayyed Mohammad Emami Razavi) Date: Mon, 25 Feb 2013 12:02:26 +0330 Subject: [Freeswitch-users] mod_cdr and MySQL Message-ID: Can i use some modules like mod_cdr_mongodb for MySQL? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130225/608f2184/attachment.html From shaheryarkh at gmail.com Mon Feb 25 12:45:59 2013 From: shaheryarkh at gmail.com (Muhammad Shahzad) Date: Mon, 25 Feb 2013 10:45:59 +0100 Subject: [Freeswitch-users] mod_cdr and MySQL In-Reply-To: References: Message-ID: Use FS XML CDR module to push cdrs to webserver where you can do whatever you want including storing them in mysql db. Thank you. On 25 Feb 2013 09:36, "Sayyed Mohammad Emami Razavi" wrote: > > Can i use some modules like mod_cdr_mongodb for MySQL? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130225/0ab07ac7/attachment-0001.html From avi at avimarcus.net Mon Feb 25 13:03:25 2013 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 25 Feb 2013 12:03:25 +0200 Subject: [Freeswitch-users] mod_cdr and MySQL In-Reply-To: References: Message-ID: You can have xml_cdr or json_cdr post to a web server to save to MySQL... or safer, to save to disk and then process right after. Or you can use http://wiki.freeswitch.org/wiki/Mod_cdr_csv and create an INSERT template, then run a cron and actually do those inserts when you rotate the logfile. -Avi On Mon, Feb 25, 2013 at 10:32 AM, Sayyed Mohammad Emami Razavi < emamirazavi at gmail.com> wrote: > Can i use some modules like mod_cdr_mongodb for MySQL? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130225/ea45b28a/attachment.html From kkgp20 at gmail.com Mon Feb 25 13:04:17 2013 From: kkgp20 at gmail.com (K K) Date: Mon, 25 Feb 2013 11:04:17 +0100 Subject: [Freeswitch-users] [mod_fsv] How the video recording really looks like? In-Reply-To: <14FF31176F964AD6A1D70E66EC16FAB9@gmail.com> References: <14FF31176F964AD6A1D70E66EC16FAB9@gmail.com> Message-ID: Yes, I have checked, I used the: but without any effect, no error, no recorded file. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130225/3b12ca78/attachment.html From kkgp20 at gmail.com Mon Feb 25 14:14:19 2013 From: kkgp20 at gmail.com (K K) Date: Mon, 25 Feb 2013 12:14:19 +0100 Subject: [Freeswitch-users] [mod_fsv] How the video recording really looks like? In-Reply-To: References: <14FF31176F964AD6A1D70E66EC16FAB9@gmail.com> Message-ID: OK now I have tried: But I receive errors: Error finding the folder path section in /recordings/firstTest.fsv Error invalid file format /recordings/firstTest.fsv Error opening file /recordings/firstTest.fsv -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130225/1207f2c2/attachment.html From regis.freeswitch.org at tornad.net Mon Feb 25 15:20:07 2013 From: regis.freeswitch.org at tornad.net (Regis M) Date: Mon, 25 Feb 2013 13:20:07 +0100 Subject: [Freeswitch-users] How to enable dtmf prompt while in callcenter queue? In-Reply-To: <1361644634372-7587914.post@n2.nabble.com> References: <1361644634372-7587914.post@n2.nabble.com> Message-ID: Hi, It's possible by binding digit action before entering in callcenter queue. Try that : [..] If you user hit DTMF 1 during waiting agent, it will be transferred to extension 1000 You can also unbind DTMF if the call is answered Regards, 2013/2/23 Wilsonbrett85 > So I've got my callcenter setup working, using the following queue position > announcement script: http://pastebin.com/sP3Gdq2g > > My dialplan looks like this: > > > data="result=${luarun(callcenter-announce-position.lua ${uuid} > support@***.net 15000)}"/> > > > I'm wondering if it's possible to allow the caller to press a key during > the > announcements to exit the callcenter queue and go straight to voicemail? > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/How-to-enable-dtmf-prompt-while-in-callcenter-queue-tp7587914.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130225/548ce34e/attachment.html From shayne.alone at gmail.com Mon Feb 25 15:28:23 2013 From: shayne.alone at gmail.com (shayne.alone at gmail.com) Date: Mon, 25 Feb 2013 15:58:23 +0330 Subject: [Freeswitch-users] inbound concurrent calls per provider Message-ID: Hi all; where should i define the maximum number of concurrent calls which is allowed from each of my providers? * I had about 30 providers which are defined under directory each of witch have more that one origination source address. example: -- Regards, Ali R. Taleghani -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130225/e8989457/attachment.html From shayne.alone at gmail.com Mon Feb 25 15:50:14 2013 From: shayne.alone at gmail.com (shayne.alone at gmail.com) Date: Mon, 25 Feb 2013 16:20:14 +0330 Subject: [Freeswitch-users] inbound concurrent calls per provider In-Reply-To: References: Message-ID: I have no authentication and use ACL on providers... On Mon, Feb 25, 2013 at 3:58 PM, shayne.alone at gmail.com < shayne.alone at gmail.com> wrote: > Hi all; > > where should i define the maximum number of concurrent calls which is > allowed from each of my providers? > > * I had about 30 providers which are defined under directory each of witch > have more that one origination source address. > example: > > > > > > > > > > > -- > Regards, > Ali R. Taleghani > -- Regards, Ali R. Taleghani -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130225/719da257/attachment.html From david.villasmil.work at gmail.com Mon Feb 25 15:59:27 2013 From: david.villasmil.work at gmail.com (David Villasmil) Date: Mon, 25 Feb 2013 13:59:27 +0100 Subject: [Freeswitch-users] inbound concurrent calls per provider In-Reply-To: References: Message-ID: Hello, Take a look at limit: http://wiki.freeswitch.org/wiki/Limit David On Mon, Feb 25, 2013 at 1:50 PM, shayne.alone at gmail.com < shayne.alone at gmail.com> wrote: > I have no authentication and use ACL on providers... > > > On Mon, Feb 25, 2013 at 3:58 PM, shayne.alone at gmail.com < > shayne.alone at gmail.com> wrote: > >> Hi all; >> >> where should i define the maximum number of concurrent calls which is >> allowed from each of my providers? >> >> * I had about 30 providers which are defined under directory each of >> witch have more that one origination source address. >> example: >> >> >> >> >> >> >> >> >> >> >> -- >> Regards, >> Ali R. Taleghani >> > > > > -- > Regards, > Ali R. Taleghani > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130225/4fa7e3f3/attachment-0001.html From vittorio.guglielmo at gmail.com Mon Feb 25 13:07:01 2013 From: vittorio.guglielmo at gmail.com (Vittorio Guglielmo) Date: Mon, 25 Feb 2013 11:07:01 +0100 Subject: [Freeswitch-users] FreeSWITCH suddenly core dump In-Reply-To: References: <512A8CF3.30107@gmail.com> Message-ID: <512B37C5.6010707@gmail.com> Hi Krice, I use fusionpbx with postgresql, don't have mysql installed. Anthony, here is freeswitch version: FreeSWITCH version: 1.2.6+git~20130104T154559Z~a4247651ca (git a424765 2013-01-04 15:45:59Z) Thank you all for your help. Vic On 25/02/13 00:32, Anthony Minessale wrote: > Can't even tell which version this is: > > Build from fresh GIT HEAD (make current) and report to JIRA if its > still there. > > > > On Sun, Feb 24, 2013 at 4:54 PM, Ken Rice > wrote: > > According to your coredump, the crash is not in FreeSWITCH, but in > libmysql. Make sure you are using the threadsafe version of > libmysql check there wiki there is info on this > > > > On 2/24/13 3:58 PM, "Vittorio Guglielmo" > > wrote: > > Hello everybody, > > I'm new to freeswitch (I recently switched form Asterisk). > > This evening, after one week of work (I have only 3 accounts > enabled), freeswitch suddenly core-dumped. > > Now, every time I try to start it, it pushes a core dump. > > I read the wiki about procedures to gain back traces, and > used the fscore_pb from the support-d of the source-dir, > > and here is the URL it gave to me for you: > http://pastebin.freeswitch.org/20623 > > > > Thanks for your help > > Vic > > > ------------------------------------------------------------------------ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > -- > Ken > _http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > _irc.freenode.net #freeswitch > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > googletalk:conf+888 at conference.freeswitch.org > > pstn:+19193869900 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130225/88b6d5a4/attachment-0001.html From vittorio.guglielmo at gmail.com Mon Feb 25 15:49:46 2013 From: vittorio.guglielmo at gmail.com (Vittorio Guglielmo) Date: Mon, 25 Feb 2013 13:49:46 +0100 Subject: [Freeswitch-users] FreeSWITCH suddenly core dump In-Reply-To: References: <512A8CF3.30107@gmail.com> Message-ID: <512B5DEA.30702@gmail.com> Hi Anthony, tried to build from fresh GIT HEAD but the issue is still there. So I reported to Jira : http://jira.freeswitch.org/browse/FS-5123. Thanks. Vic On 25/02/13 00:32, Anthony Minessale wrote: > Can't even tell which version this is: > > Build from fresh GIT HEAD (make current) and report to JIRA if its > still there. > > > > On Sun, Feb 24, 2013 at 4:54 PM, Ken Rice > wrote: > > According to your coredump, the crash is not in FreeSWITCH, but in > libmysql. Make sure you are using the threadsafe version of > libmysql check there wiki there is info on this > > > > On 2/24/13 3:58 PM, "Vittorio Guglielmo" > > wrote: > > Hello everybody, > > I'm new to freeswitch (I recently switched form Asterisk). > > This evening, after one week of work (I have only 3 accounts > enabled), freeswitch suddenly core-dumped. > > Now, every time I try to start it, it pushes a core dump. > > I read the wiki about procedures to gain back traces, and > used the fscore_pb from the support-d of the source-dir, > > and here is the URL it gave to me for you: > http://pastebin.freeswitch.org/20623 > > > > Thanks for your help > > Vic > > > ------------------------------------------------------------------------ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > -- > Ken > _http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > _irc.freenode.net #freeswitch > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > googletalk:conf+888 at conference.freeswitch.org > > pstn:+19193869900 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130225/1a94de64/attachment-0001.html From venkateshwaran54 at gmail.com Mon Feb 25 12:49:23 2013 From: venkateshwaran54 at gmail.com (Venkateshwaran Thirugnanam) Date: Mon, 25 Feb 2013 15:19:23 +0530 Subject: [Freeswitch-users] Freeswitch Installation Message-ID: Hi All, 1.I need to install freeswitch on centos 6.3 please guild me with procedure..I can't find a procedure on internet...I need to install with Autoboot freeswitch from startup... 2.Why we need to install epel for Centos 6?whether its mandatory or optional? Guild me Regards, Kumaran T -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130225/2d1f9d6d/attachment.html From venkateshwaran54 at gmail.com Mon Feb 25 16:07:41 2013 From: venkateshwaran54 at gmail.com (Venkateshwaran Thirugnanam) Date: Mon, 25 Feb 2013 18:37:41 +0530 Subject: [Freeswitch-users] Configure Message-ID: Hi All, How to configure 3 freeswitch server to Centralized mysql DB? Please guild Regards, Kumaran T -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130225/89771880/attachment.html From kkgp20 at gmail.com Mon Feb 25 16:36:44 2013 From: kkgp20 at gmail.com (K K) Date: Mon, 25 Feb 2013 14:36:44 +0100 Subject: [Freeswitch-users] [mod_fsv] How the video recording really looks like? In-Reply-To: References: <14FF31176F964AD6A1D70E66EC16FAB9@gmail.com> Message-ID: When I use the: I have no errors, but also no recorded file in above path. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130225/7bf2010e/attachment.html From flavio at sippulse.com Mon Feb 25 17:08:49 2013 From: flavio at sippulse.com (Flavio Goncalves) Date: Mon, 25 Feb 2013 11:08:49 -0300 Subject: [Freeswitch-users] Possible BUG in G729 negotiation Message-ID: Hi, I have searched the list, but I couldn't find a final answer for this case. I'm receiving calls from a provider using G729 annex B. INVITE a=rtpmap:18 G729/8000. a=fmtp:18 annexb=yes. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=rtpmap:19 CN/8000. and FS is answering 200OK m=audio 31388 RTP/AVP 18 101 19. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=yes. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=rtpmap:19 CN/8000. a=ptime:20. If freeswitch supports only G729A, shouldn't FS refuse the call with 488 or suppress the line annexb=yes? The audio quality is really bad when I receive calls in this way. After removing the annexb on the gateway the audio is fine, but unfortunately I can't remove from it from all VoIP providers terminating calls. For outbound calls I simply removed the annexb, but for inbound there is no way to do it. Flavio -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130225/6b92f78b/attachment.html From dujinfang at gmail.com Mon Feb 25 17:28:58 2013 From: dujinfang at gmail.com (Seven Du) Date: Mon, 25 Feb 2013 22:28:58 +0800 Subject: [Freeswitch-users] [mod_fsv] How the video recording really looks like? In-Reply-To: References: <14FF31176F964AD6A1D70E66EC16FAB9@gmail.com> Message-ID: <922DB54402EF453C96DB20C17697B2C0@gmail.com> 1) Are you sure you are on the video-media-bug branch as required on the wiki? 2) Did you tried all the commands listed on the wiki before trying your stuff? On Monday, February 25, 2013 at 9:36 PM, K K wrote: > When I use the: > > > > I have no errors, but also no recorded file in above path. > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto: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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130225/6a956413/attachment.html From lloyd.aloysius at gmail.com Mon Feb 25 17:40:51 2013 From: lloyd.aloysius at gmail.com (Lloyd Aloysius) Date: Mon, 25 Feb 2013 09:40:51 -0500 Subject: [Freeswitch-users] Configure In-Reply-To: References: Message-ID: 1. setup the odbc in each freeswitch server point to the shared db 2. conf/autoload_configs - switch.xml - parameter core-odbc-name = your odbc name http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core Lloyd On Mon, Feb 25, 2013 at 8:07 AM, Venkateshwaran Thirugnanam < venkateshwaran54 at gmail.com> wrote: > Hi All, > How to configure 3 freeswitch server to Centralized mysql DB? Please > guild > > Regards, > Kumaran T > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130225/57a2482d/attachment.html From lloyd.aloysius at gmail.com Mon Feb 25 17:45:56 2013 From: lloyd.aloysius at gmail.com (Lloyd Aloysius) Date: Mon, 25 Feb 2013 09:45:56 -0500 Subject: [Freeswitch-users] Freeswitch Installation In-Reply-To: References: Message-ID: here is how I install 1. Install CentOS 6.3 basic 2. Disable SELinux - /etc/selinux/config SELINUX=disabled 3. Some of the following not necessary yum -y groupinstall base yum -y install perl yum -y install crontabs yum -y install sudo yum -y install wget yum -y install vim-enhanced yum -y install emacs yum -y install git yum -y install bind-utils yum -y install sendmail sendmail-cf system-switch-mail mailx yum -y install rsync chkconfig sendmail on chkconfig crond on 4.FreeSWITCH Required Packages yum install autoconf automake gcc-c++ git-core libjpeg-devel libtool make ncurses-devel unixODBC-devel openssl-devel gnutls-devel libogg-devel libvorbis-devel curl-devel libtiff-devel libjpeg-devel python-devel expat-devel zlib zlib-devel bzip2 which patch 5.cd /usr/src git clone git://git.freeswitch.org/freeswitch.git cd freeswitch/ ./bootstrap.sh ./configure make make install make cd-sounds-install make cd-moh-install adduser freeswitch chown -R freeswitch:freeswitch /usr/local/freeswitch Lloyd On Mon, Feb 25, 2013 at 4:49 AM, Venkateshwaran Thirugnanam < venkateshwaran54 at gmail.com> wrote: > Hi All, > 1.I need to install freeswitch on centos 6.3 please guild me with > procedure..I can't find a procedure on internet...I need to install with > Autoboot freeswitch from startup... > 2.Why we need to install epel for Centos 6?whether its mandatory or > optional? > Guild me > > Regards, > Kumaran T > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130225/cdf0337d/attachment-0001.html From david.villasmil.work at gmail.com Mon Feb 25 17:48:20 2013 From: david.villasmil.work at gmail.com (David Villasmil) Date: Mon, 25 Feb 2013 15:48:20 +0100 Subject: [Freeswitch-users] Configure In-Reply-To: References: Message-ID: Hello, What do you mean exactly by "3 freeswitch server to centralized db"? If you mean the core, just set your unixodbc dsn to pont to the db server... http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core David --- David Villasmil On Feb 25, 2013, at 14:07, Venkateshwaran Thirugnanam wrote: > Hi All, > How to configure 3 freeswitch server to Centralized mysql DB? Please guild > > Regards, > Kumaran T > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130225/b37df708/attachment.html From lloyd.aloysius at gmail.com Mon Feb 25 17:47:52 2013 From: lloyd.aloysius at gmail.com (Lloyd Aloysius) Date: Mon, 25 Feb 2013 09:47:52 -0500 Subject: [Freeswitch-users] Freeswitch Installation In-Reply-To: References: Message-ID: For Auto start at boot cp /usr/src/freeswitch.git/build/freeswitch.init.redhat /etc/init.d/freeswitch Edit /etc/init.d/freeswitch like below PROG_NAME=freeswitch PID_FILE=${PID_FILE-/usr/local/freeswitch/log/freeswitch.pid} FS_USER=${FS_USER-freeswitch} FS_FILE=${FS_FILE-/usr/local/freeswitch/bin/freeswitch} FS_HOME=${FS_HOME-/usr/local/freeswitch} LOCK_FILE=/var/lock/subsys/freeswitch FREESWITCH_ARGS="-nc" RETVAL=0 chmod 755 /etc/init.d/freeswitch chkconfig --add freeswitch chkconfig --level 2345 freeswitch on * * On Mon, Feb 25, 2013 at 9:45 AM, Lloyd Aloysius wrote: > here is how I install > > 1. Install CentOS 6.3 basic > > 2. Disable SELinux - /etc/selinux/config > > SELINUX=disabled > > 3. Some of the following not necessary > > yum -y groupinstall base > yum -y install perl > yum -y install crontabs > yum -y install sudo > yum -y install wget > yum -y install vim-enhanced > yum -y install emacs > yum -y install git > yum -y install bind-utils > yum -y install sendmail sendmail-cf system-switch-mail mailx > yum -y install rsync > chkconfig sendmail on > chkconfig crond on > > > 4.FreeSWITCH Required Packages > > yum install autoconf automake gcc-c++ git-core libjpeg-devel libtool make > ncurses-devel unixODBC-devel openssl-devel gnutls-devel libogg-devel > libvorbis-devel curl-devel libtiff-devel libjpeg-devel python-devel > expat-devel zlib zlib-devel bzip2 which patch > > > 5.cd /usr/src > > git clone git://git.freeswitch.org/freeswitch.git > > cd freeswitch/ > > ./bootstrap.sh > > ./configure > > make > > make install > > make cd-sounds-install > > make cd-moh-install > > adduser freeswitch > > chown -R freeswitch:freeswitch /usr/local/freeswitch > > Lloyd > > > On Mon, Feb 25, 2013 at 4:49 AM, Venkateshwaran Thirugnanam < > venkateshwaran54 at gmail.com> wrote: > >> Hi All, >> 1.I need to install freeswitch on centos 6.3 please guild me with >> procedure..I can't find a procedure on internet...I need to install with >> Autoboot freeswitch from startup... >> 2.Why we need to install epel for Centos 6?whether its mandatory or >> optional? >> Guild me >> >> Regards, >> Kumaran T >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130225/54a1a8b0/attachment.html From kkgp20 at gmail.com Mon Feb 25 18:04:59 2013 From: kkgp20 at gmail.com (K K) Date: Mon, 25 Feb 2013 16:04:59 +0100 Subject: [Freeswitch-users] [mod_fsv] How the video recording really looks like? In-Reply-To: <922DB54402EF453C96DB20C17697B2C0@gmail.com> References: <14FF31176F964AD6A1D70E66EC16FAB9@gmail.com> <922DB54402EF453C96DB20C17697B2C0@gmail.com> Message-ID: AD.1 OK so I see now that I need to compile the video-media-bug branch to achieve my goal. I am sorry, but in my opinion this Wiki page is very confusing, because of the fact that the mod_fsv is delivered with the current FreeSWITCH installer, and you do not need to be in the video-media-bug to use this mod as it is described in the "Record a single legged call" section. It should be clarified which features are and which are not available. Thanks to such one Wiki sentence you can get less mails on yours mailing list. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130225/fff85d96/attachment.html From alex at digitalmail.com Mon Feb 25 18:29:07 2013 From: alex at digitalmail.com (Alex Lake) Date: Mon, 25 Feb 2013 15:29:07 +0000 Subject: [Freeswitch-users] Getting Freeswitch 1.2.0 (git-771819c 2012-05-25 04-46-28 +0000) Message-ID: <512B8343.2040906@digitalmail.com> How can I get this version of Freeswitch? I tried "git clone -b v1.2.0 git://git.freeswitch.org/freeswitch.git" but that reported that the branch wasn't found. From vermeulen.deon at gmail.com Mon Feb 25 18:19:01 2013 From: vermeulen.deon at gmail.com (Deon Vermeulen) Date: Mon, 25 Feb 2013 17:19:01 +0200 Subject: [Freeswitch-users] INVITE Via: header parameters Message-ID: <5CEB828A-8DDB-4E08-AD10-A292CB7CD359@gmail.com> Hi I've been all over forums, the wiki, mailing lists but I just can't find the answer to my problem. Perhaps the reason why I don't get the answer is that I don't properly understand what to search for? ;-) Any way. The problem I'm facing is that I have to present the Via: string in the INVITE in a specific manner to successfully setup my SIP trunk to a SIP enabled Siemens MSC. It should be presented as follows: INVITE sip:xxxxxxxxxxxx@:5060 SIP/2.0 Via: SIP/2.0/UDP ;rport;branch=z9hG4bKQ96vgD8Z1c2rN How can I force the :5060 in the Via: uri? Before I forget. The server is all running on Public IP's except to this specific trunk. In this scenario FS is a Full-ON SBC handling media in Full Proxy. Thank you very much Kind Regards Sent from my iPhone -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130225/b23206dd/attachment.html From krice at freeswitch.org Mon Feb 25 18:57:07 2013 From: krice at freeswitch.org (Ken Rice) Date: Mon, 25 Feb 2013 09:57:07 -0600 Subject: [Freeswitch-users] Getting Freeswitch 1.2.0 (git-771819c 2012-05-25 04-46-28 +0000) In-Reply-To: <512B8343.2040906@digitalmail.com> Message-ID: Thast because v1.2.0 is not a branch its a tag on the v1.2.stable branch... Just git clone freeswitch as normal, then git checkout TAG_NAME Not sure why you would want 1.2.0 tho... There are major issues in that release and it is not recommended for production due to security issues On 2/25/13 9:29 AM, "Alex Lake" wrote: > How can I get this version of Freeswitch? > > I tried "git clone -b v1.2.0 git://git.freeswitch.org/freeswitch.git" > but that reported that the branch wasn't found. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch From msc at freeswitch.org Mon Feb 25 19:34:20 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 25 Feb 2013 08:34:20 -0800 Subject: [Freeswitch-users] Door intercom/gate controller In-Reply-To: References: <144901ce0bbe$5ab583a0$10208ae0$@freeswitch.org> Message-ID: Hi Nathan, Yes, start_dtmf is one-directional as far as I am aware, that is, it only works on the sending stream of the channel on which it is set. BTW, I'm glad you got it figured out! -MC On Fri, Feb 22, 2013 at 6:32 PM, Mr Nathan Downes wrote: > Hi Michael,**** > > ** ** > > Thanks for the response, it worked in the opposite direction, the gate > controller (OOB) calls phones (Inband) kind of like the door thing on an > apartment block. Start_dtmf worked if I called the gate controller from > the phones, but not in the other direction which is how it will be used.** > ** > > ** ** > > It has been resolved as it was discovered the ATA devices inbuilt to the > fibre ONT have an option for OOB DTMF, enabling this on the POTS profile > allowed the device to function.**** > > ** ** > > I am still interested on how I could of got it to work though, is > start_dtmf a one directional app?**** > > ** ** > > Thanks,**** > > ** ** > > Nathan**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael > Collins > *Sent:* Saturday, 16 February 2013 6:35 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Door intercom/gate controller**** > > ** ** > > Can you show us the dialplan for sending a call to the gate controller? > Also, pastebin the console debug output for a call to the gate controller > where you press digits but the controller doesn't respond.**** > > ** ** > > Thanks,**** > > MC**** > > On Wed, Feb 13, 2013 at 2:42 AM, Mr Nathan Downes < > nathandownes at hotmail.com> wrote:**** > > Hi list,**** > > **** > > I have a gate controller that only understands RFC 2833 but it calls > endpoints that can only provide inband DTMF, I can?t seem to get the digits > recognised by the gate controller to trigger the relay using > start_dtmf_generate, as they are coming back to the a-leg? Rather than out > the b-leg.. It works fine when I call a SPA502g as it will provide RFC > 2833.. is there some trickery I can achieve this? Or am I just doing it > wrong?? J**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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**** > > > > **** > > ** ** > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130225/07fbf86b/attachment.html From msc at freeswitch.org Mon Feb 25 19:41:25 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 25 Feb 2013 08:41:25 -0800 Subject: [Freeswitch-users] Possible BUG in G729 negotiation In-Reply-To: References: Message-ID: Hey Flavio! Hope all is well with you (aside from this little SDP issue). Looking forward to seeing you guys at ClueCon 2013. :) Are you able to reproduce this issue on latest git HEAD? If so, could you open a ticket at jira.freeswitch.org please? That will be the quickest way to get it resolved. Thanks! -Michael On Mon, Feb 25, 2013 at 6:08 AM, Flavio Goncalves wrote: > Hi, > > I have searched the list, but I couldn't find a final answer for this > case. I'm receiving calls from a provider using G729 annex B. > > INVITE > a=rtpmap:18 G729/8000. > a=fmtp:18 annexb=yes. > a=rtpmap:0 PCMU/8000. > a=rtpmap:8 PCMA/8000. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=rtpmap:19 CN/8000. > > and FS is answering > > 200OK > m=audio 31388 RTP/AVP 18 101 19. > a=rtpmap:18 G729/8000. > a=fmtp:18 annexb=yes. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=rtpmap:19 CN/8000. > a=ptime:20. > > If freeswitch supports only G729A, shouldn't FS refuse the call with 488 > or suppress the line annexb=yes? > > The audio quality is really bad when I receive calls in this way. After > removing the annexb on the gateway the audio is fine, but unfortunately I > can't remove from it from all VoIP providers terminating calls. For > outbound calls I simply removed the annexb, but for inbound there is no way > to do it. > > Flavio > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130225/b6e36758/attachment.html From freeswitch-list at puzzled.xs4all.nl Mon Feb 25 20:03:30 2013 From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists) Date: Mon, 25 Feb 2013 18:03:30 +0100 Subject: [Freeswitch-users] Freeswitch Installation In-Reply-To: References: Message-ID: <512B9962.9030609@puzzled.xs4all.nl> On 02/25/2013 03:45 PM, Lloyd Aloysius wrote: > here is how I install > > 1. Install CentOS 6.3 basic > > 2. Disable SELinux - /etc/selinux/config > > SELINUX=disabled Disabling is a bad idea. Regards, Patrick From sdevoy at bizfocused.com Mon Feb 25 20:03:43 2013 From: sdevoy at bizfocused.com (Sean Devoy) Date: Mon, 25 Feb 2013 12:03:43 -0500 Subject: [Freeswitch-users] Polycom specific problem Message-ID: <124301ce137a$11342150$339c63f0$@bizfocused.com> This was initially reported to me as a call being dropped issue. I have just had an update that says if the user stays on the line for "a few seconds or so" the call returned to its previous state. Now I am skeptical if it is a FS issue at all. Alas, you all are my best source of information so I appeal to you with my question and undying gratitude. My first question for you Polycom users is "Does Polycom have a different call waiting tone when the users is dialing out and not connected yet?" The current issue, reported by several users, is intermittent. If the Polycom user is dialing out (and the call is not answered yet) and a new call comes in (incoming calls ring all 6 extensions) the Polycom user gets a "busy signal" for 1 to 2 seconds and then it returns to their ringing outgoing call. They have been absolutely adamant that it is not the "nice little call waiting tone" they get when a call is connected and another call comes in. It is only in this "pre-connected state" that they get the "busy signal" and they swear it is intermittent. I will add that the phone configuration is Line ringing on both extensions on the phone. I have turned on global siptrace to capture everything I can the next time it happens. Any ideas? Sean -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130225/b366f95b/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 70 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130225/b366f95b/attachment-0001.gif From krice at freeswitch.org Mon Feb 25 20:17:14 2013 From: krice at freeswitch.org (Ken Rice) Date: Mon, 25 Feb 2013 11:17:14 -0600 Subject: [Freeswitch-users] Freeswitch Installation In-Reply-To: <512B9962.9030609@puzzled.xs4all.nl> Message-ID: Disabling selinux is required to run freeswitch unless you are going to take the time and set up a proper SELinux profile for it... On 2/25/13 11:03 AM, "Patrick Lists" wrote: > On 02/25/2013 03:45 PM, Lloyd Aloysius wrote: >> here is how I install >> >> 1. Install CentOS 6.3 basic >> >> 2. Disable SELinux - /etc/selinux/config >> >> SELINUX=disabled > > Disabling is a bad idea. > > Regards, > Patrick > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch From venkateshwaran54 at gmail.com Mon Feb 25 20:30:35 2013 From: venkateshwaran54 at gmail.com (Venkateshwaran Thirugnanam) Date: Mon, 25 Feb 2013 23:00:35 +0530 Subject: [Freeswitch-users] Configure In-Reply-To: References: Message-ID: Hello David, I need to install 3 freeswitch server in different location and all 3 servers should have centralized DB(Mysql)...I don't know how to point these servers to centralized DB Regards, Kumaran T On Mon, Feb 25, 2013 at 8:18 PM, David Villasmil < david.villasmil.work at gmail.com> wrote: > Hello, > > What do you mean exactly by "3 freeswitch server to centralized db"? > > If you mean the core, just set your unixodbc dsn to pont to the db > server... > > http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core > > David > --- > > David Villasmil > > > On Feb 25, 2013, at 14:07, Venkateshwaran Thirugnanam < > venkateshwaran54 at gmail.com> wrote: > > Hi All, > How to configure 3 freeswitch server to Centralized mysql DB? Please > guild > > Regards, > Kumaran T > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130225/59e3d9f4/attachment.html From victor.chukalovskiy at gmail.com Mon Feb 25 20:37:12 2013 From: victor.chukalovskiy at gmail.com (Victor Chukalovskiy) Date: Mon, 25 Feb 2013 12:37:12 -0500 Subject: [Freeswitch-users] Bounty for FS-5118 - Tel URI parameters (npdi, rn) appearing in SIP URI on outbound calls Message-ID: <512BA148.5040703@gmail.com> Hello, I'm offering bounty to have npdi and rn= issue sorted out. I'll be also posting this in the bounty section of the WiKi shortly. Amount depends on the scope of work. If I get it right, FS-5118 needs to be first reviewed by developers to move any further with this. I'm ready to offer my feedback and detailed description of what is needed. At this point, I see two different path: (1) Attack FS-5118 directly. That is, provide a way to pass tel URI parameters correctly when bridging call to a SIP gateway OR (2) Come-up with channel variables for npdi or npdi= and for rn= that are: -Automatically inserted into tel URI in the INVITE on outbound SIP calls -Automatically populated with parsed values on incoming SIP call (to avoid doing this in the dial-plan http://wiki.freeswitch.org/wiki/Regular_Expression#LNP_.28Local_Number_Portability.29) I believe FS project will benefit from the ability to easily send and receive npdi or npdi= and rn= (ideally, by means of dedicated channel variables). -Victor From krice at freeswitch.org Mon Feb 25 20:44:17 2013 From: krice at freeswitch.org (Ken Rice) Date: Mon, 25 Feb 2013 11:44:17 -0600 Subject: [Freeswitch-users] Bounty for FS-5118 - Tel URI parameters (npdi, rn) appearing in SIP URI on outbound calls In-Reply-To: <512BA148.5040703@gmail.com> Message-ID: Just to clarify, FreeSWITCH can already send NDPI and RN as per RFC4694. You are specifically referring to adding support for this to the gateways mechanism that is part of mod_sofia in FreeSWITCH. On 2/25/13 11:37 AM, "Victor Chukalovskiy" wrote: > Hello, > > I'm offering bounty to have npdi and rn= issue sorted out. I'll be also > posting this in the bounty section of the WiKi shortly. Amount depends > on the scope of work. > > If I get it right, FS-5118 needs to be first reviewed by developers to > move any further with this. I'm ready to offer my feedback and detailed > description of what is needed. > > At this point, I see two different path: > > (1) Attack FS-5118 directly. That is, provide a way to pass tel URI > parameters correctly when bridging call to a SIP gateway > > OR > > (2) Come-up with channel variables for npdi or npdi= and for rn= that are: > -Automatically inserted into tel URI in the INVITE on outbound SIP > calls > -Automatically populated with parsed values on incoming SIP call > (to avoid doing this in the dial-plan > http://wiki.freeswitch.org/wiki/Regular_Expression#LNP_.28Local_Number_Portabi > lity.29) > > I believe FS project will benefit from the ability to easily send and > receive npdi or npdi= and rn= (ideally, by means of dedicated channel > variables). > > -Victor > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch From venkateshwaran54 at gmail.com Mon Feb 25 20:45:35 2013 From: venkateshwaran54 at gmail.com (Venkateshwaran Thirugnanam) Date: Mon, 25 Feb 2013 23:15:35 +0530 Subject: [Freeswitch-users] freeswitch--Automatic Startup At Boot Message-ID: Hi All, How to make freeswitch automatic startup on centos 6.3...I followed doc but I think I'm missing so its not starting after reboot....Pls guide me.. Regards, Kumaran T -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130225/d043ea2b/attachment.html From mario_fs at mgtech.com Mon Feb 25 21:02:22 2013 From: mario_fs at mgtech.com (Mario G) Date: Mon, 25 Feb 2013 10:02:22 -0800 Subject: [Freeswitch-users] [Solved] Can local_stream MOH have a time limit/max duration? In-Reply-To: <006916BC-5985-4489-9515-7E13BDF0FDFC@mgtech.com> References: <219BCB63-3A5A-419F-B374-BDD45FDE3FAB@mgtech.com> <006916BC-5985-4489-9515-7E13BDF0FDFC@mgtech.com> Message-ID: Update, this also works with file_string: Needless to say this only works with the latest HEAD for now. Thanks to Anthony! Mario G I updated the wiki with a new Execution Parameters section: http://wiki.freeswitch.org/wiki/Mod_local_stream#Execution_Parameters . I am using this now: I updated the wiki with a new Execution Parameters section: http://wiki.freeswitch.org/wiki/File_string#Execution_Parameters. This works great: > On Feb 12, 2013, at 10:43 AM, Mario G wrote: > >> I tried a few things with transfer_ringaback but no dice. The wiki also states transfer_ringback is only for after the call is answered, use ringback for early media, which is what I need it for. i looked at source code but did not find anything that would allow limiting moh duration, bummer. Thanks. >> Mario G >> >> On Feb 11, 2013, at 12:57 PM, Michael Collins wrote: >> >>> For #2 use transfer_ringback channel variable and set it to MOH and it should work the same way, i.e. instead of ringing the caller will hear music. >>> >>> -MC >>> >>> On Fri, Feb 8, 2013 at 8:55 AM, Mario G wrote: >>> I looked into chime-ins, but it only chimes in at fixed intervals, and it was not flexible enough for true customization. I could not find anything else on the wiki that would play moh (or any other file) for a specific time. See my previous post for a details of how it works and what I wanted to do. Thanks. >>> Mario G >>> >>> On Feb 8, 2013, at 5:48 AM, Vik Killa wrote: >>> >>> > I don't quite understand what you are trying to do here. >>> > mod_local_stream plays audio in a loop constantly with chime-ins >>> > (announcements) set to play periodically. If you need something to >>> > play from the beginning for a certain amount of time, you need to use >>> > something other than mod_local_stream. >>> > >>> > >>> >> On Wed, Feb 6, 2013 at 9:04 AM, Mario G wrote: >>> >>> >>> >>> Could not find anything in the wiki and this may not be possible but... I >>> >>> have early media with instructions, is has voice, ring, voice, music. I want >>> >>> to change the ring to music but I could not find a way to limit the time the >>> >>> music plays. See the second example, I would like to limit the first moh to >>> >>> a maximum time. If not possible, is this a "bounty" enhancement? Thanks for >>> >>> any help. >>> >>> Mario G >>> >>> >>> >>> >>> >>> Currently: >>> >>> >> >>> data="ringback=file_string://${lua_ring1}!tone_stream://${us-ring};loops=3!file_string://${lua_ring2}!local_stream://moh"/> >>> >>> >>> >>> Would Like (add something to limit the first moh): >>> >>> >> >>> data="ringback=file_string://${lua_ring1}!local_stream://moh!file_string://${lua_ring2}!local_stream://moh"/> >>> >>> >>> >>> >>> >>> ______________________________________________________________________ >>> > >>> > _________________________________________________________________________ > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130225/cfccf5fc/attachment-0001.html From jdiaz at coinfru.com Mon Feb 25 21:10:01 2013 From: jdiaz at coinfru.com (Josue Diaz Cruz) Date: Mon, 25 Feb 2013 19:10:01 +0100 Subject: [Freeswitch-users] Best Installation MOD_GSMOPEN Message-ID: Good Afternoon. I beg your pardon if my question is really from a newbee. I am trying to install gsmopen in a system. I did it with CentOS 6 X64, Debian 6, Ubuntu 12.10. Can some one provide me the best way to do it. Cause i allways have problems in one point of the installation. Depending the distro could it be one or other. I was thinking that could be nice to give a complete requierements (libs) to install before do anything. If some one can provide me a easy, simple and not troubles, could be nice: * System and version * List of requierements. * Better steps to avoid issues with libraries and compile. What i tried: Centos 6 x86_64 --- Huawey E169 unlocked and voice active. (Issues when try to load mod_gsmopen.so) Debian 6 x86_64 --- Huawey E169 unlocked and voice active. (Issues with the ttyUSB port) Ubuntu 12.10 --- Huawey E169 unlocked and voice active. (Issues with the ttyUSB port) E169 works perfectly. I have a mobile partner with voicecall and i test it calling my land line without issues. Thank you very much for your consideration. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130225/a763aba9/attachment.html From victor.chukalovskiy at gmail.com Mon Feb 25 21:13:10 2013 From: victor.chukalovskiy at gmail.com (Victor Chukalovskiy) Date: Mon, 25 Feb 2013 13:13:10 -0500 Subject: [Freeswitch-users] Bounty for FS-5118 - Tel URI parameters (npdi, rn) appearing in SIP URI on outbound calls In-Reply-To: References: Message-ID: <512BA9B6.2000506@gmail.com> Thanks, that's right. FS-5118 is to add support to send it over gateways. Any idea on how much work is involved and what's a good bounty for this? On 13-02-25 12:44 PM, Ken Rice wrote: > Just to clarify, FreeSWITCH can already send NDPI and RN as per RFC4694. > You are specifically referring to adding support for this to the gateways > mechanism that is part of mod_sofia in FreeSWITCH. > > > > > On 2/25/13 11:37 AM, "Victor Chukalovskiy" > wrote: > >> Hello, >> >> I'm offering bounty to have npdi and rn= issue sorted out. I'll be also >> posting this in the bounty section of the WiKi shortly. Amount depends >> on the scope of work. >> >> If I get it right, FS-5118 needs to be first reviewed by developers to >> move any further with this. I'm ready to offer my feedback and detailed >> description of what is needed. >> >> At this point, I see two different path: >> >> (1) Attack FS-5118 directly. That is, provide a way to pass tel URI >> parameters correctly when bridging call to a SIP gateway >> >> OR >> >> (2) Come-up with channel variables for npdi or npdi= and for rn= that are: >> -Automatically inserted into tel URI in the INVITE on outbound SIP >> calls >> -Automatically populated with parsed values on incoming SIP call >> (to avoid doing this in the dial-plan >> http://wiki.freeswitch.org/wiki/Regular_Expression#LNP_.28Local_Number_Portabi >> lity.29) >> >> I believe FS project will benefit from the ability to easily send and >> receive npdi or npdi= and rn= (ideally, by means of dedicated channel >> variables). >> >> -Victor >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 freeswitch-list at puzzled.xs4all.nl Mon Feb 25 21:14:05 2013 From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists) Date: Mon, 25 Feb 2013 19:14:05 +0100 Subject: [Freeswitch-users] Freeswitch Installation In-Reply-To: References: Message-ID: <512BA9ED.8000708@puzzled.xs4all.nl> On 02/25/2013 06:17 PM, Ken Rice wrote: > Disabling selinux is required to run freeswitch unless you are going to take > the time and set up a proper SELinux profile for it... Yes but IMHO making it work is definitely worth it. And the Red Hat SELinux folks are very cooperative when you file a bug with all SELinux AVCs. If there is room in the budget one could also hire a Red Hat Certified Security Specialist (RHCSS) to write a policy. Regards, Patrick From dgarcia at anew.com.ve Mon Feb 25 21:20:12 2013 From: dgarcia at anew.com.ve (Saugort Dario Garcia Tovar) Date: Mon, 25 Feb 2013 13:50:12 -0430 Subject: [Freeswitch-users] freeswitch--Automatic Startup At Boot In-Reply-To: References: Message-ID: <512BAB5C.7090104@anew.com.ve> For Auto start at boot cp /usr/src/freeswitch.git/build/freeswitch.init.redhat /etc/init.d/freeswitch Edit /etc/init.d/freeswitch like below PROG_NAME=freeswitch PID_FILE=${PID_FILE-/usr/local/freeswitch/log/freeswitch.pid} FS_USER=${FS_USER-freeswitch} FS_FILE=${FS_FILE-/usr/local/freeswitch/bin/freeswitch} FS_HOME=${FS_HOME-/usr/local/freeswitch} LOCK_FILE=/var/lock/subsys/freeswitch FREESWITCH_ARGS="-nc" RETVAL=0 chmod 755 /etc/init.d/freeswitch chkconfig --add freeswitch chkconfig --level 2345 freeswitch on * * On 2/25/2013 1:15 PM, Venkateshwaran Thirugnanam wrote: > Hi All, > How to make freeswitch automatic startup on centos 6.3...I followed > doc but I think I'm missing so its not starting after reboot....Pls > guide me.. > > Regards, > Kumaran T > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.2238 / Virus Database: 2641/5632 - Release Date: 02/25/13 > -- Atentamente, *Dario Garc?a* Consultor. CCCT, Nivel C2, Sector Yarey, Mz, Ofc. MZ03a. Caracas-Venezuela. Tel?fono: +58 212 9081842 Cel: +58 412 2221515 dgarcia at anew.com.ve http://www.anew.com.ve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130225/16781d66/attachment.html From marketing at cluecon.com Mon Feb 25 21:29:26 2013 From: marketing at cluecon.com (Michael Collins) Date: Mon, 25 Feb 2013 10:29:26 -0800 Subject: [Freeswitch-users] FreeSWITCH Weekly News and Notes Message-ID: Greetings! First item in the news today is a happy report from long-time FreeSWITCH user Henry Gavin. Henry runs a company in the U.K. called SureVoIP. He is pleased to report that "thanks to FreeSWITCH and FusionPBX" his company is once again a finalist for the annual ITSPA awards. Congrats to Henry for leveraging FreeSWITCH in a successful business endeavor. Another annual event is the Google Summer of Code(GSoC). FreeSWITCH will once again apply as a mentoring organization. Please start thinking of project ideas that we can include in our organization's application. Applications will be submitted starting March 18 and no later than March 29. Ken Rice and I will be coordinating this process. Stay tuned for more details. On last week's conference callwe did a nice tour of the CudaTel <> Communication Server. In the coming weeks we will have more presentations for GUI front-ends that community members have built. On this week's conference callwe will have Ken Rice give us an update on his new project: SwitchPi . If you like DIY projects then you'll appreciate what Ken has done with integrating the Raspberry Pi with FreeSWITCH and some other items to create something new. We look forward to seeing it in action. Don't forget about the FreeSWITCH HA discussion on Tuesday evening at 8PM EST. Last week's discussion was very fruitful. Eliot Gable gave us all a lot of information about the different approaches that he can take for building mod_ha_cluster . We look forward to his report on the potential of using OpenMPI. For those who can't make it to the HA discussion please join weekly conference call on Wednesday where we will have a brief recap of the HA call. In ClueCon news we have uploaded two new videos: * What's new in sipXecs 4.6 - Douglas Hubler * Challenges and Opportunities in Open Source VoIP- Travis Cross Stay tuned for more ClueCon 2012 videos and ClueCon 2013 announcements. Have a great week and we look forward to talking to you on Wednesday. -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130225/7279814a/attachment-0001.html From msc at freeswitch.org Mon Feb 25 22:05:58 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 25 Feb 2013 11:05:58 -0800 Subject: [Freeswitch-users] Freeswitch Installation In-Reply-To: <512BA9ED.8000708@puzzled.xs4all.nl> References: <512BA9ED.8000708@puzzled.xs4all.nl> Message-ID: I would love to see an SELinux policy created and donated to the community. If anyone wants to put money toward this then send an email to us at OSTAG: ostag at ostag.org. We'll coordinate everything there. Thanks, MC On Mon, Feb 25, 2013 at 10:14 AM, Patrick Lists < freeswitch-list at puzzled.xs4all.nl> wrote: > On 02/25/2013 06:17 PM, Ken Rice wrote: > > Disabling selinux is required to run freeswitch unless you are going to > take > > the time and set up a proper SELinux profile for it... > > Yes but IMHO making it work is definitely worth it. And the Red Hat > SELinux folks are very cooperative when you file a bug with all SELinux > AVCs. If there is room in the budget one could also hire a Red Hat > Certified Security Specialist (RHCSS) to write a policy. > > Regards, > Patrick > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130225/747ac189/attachment.html From schoch+freeswitch.org at xwin32.com Mon Feb 25 22:22:05 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Mon, 25 Feb 2013 11:22:05 -0800 Subject: [Freeswitch-users] "Outgoing Call" on phone display In-Reply-To: References: <094868EE-812C-4E21-8551-0EC80F553D3D@freeswitch.org> <2F45CFD8-1F72-44AB-93D4-0C59BED92AD2@freeswitch.org> Message-ID: On Fri, Feb 22, 2013 at 7:40 PM, Michael Collins wrote: > Have you tried setting this variable for the leg in question? > http://wiki.freeswitch.org/wiki/Variable_ignore_display_updates Thank you! Problem solved! I added: after and it worked as required. -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130225/4b74c9d6/attachment.html From talk2ram at gmail.com Mon Feb 25 22:30:48 2013 From: talk2ram at gmail.com (ram) Date: Tue, 26 Feb 2013 01:00:48 +0530 Subject: [Freeswitch-users] Configure In-Reply-To: References: Message-ID: Hi this is the best place to start http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core On Mon, Feb 25, 2013 at 11:00 PM, Venkateshwaran Thirugnanam < venkateshwaran54 at gmail.com> wrote: > Hello David, > I need to install 3 freeswitch server in different location and all 3 > servers should have centralized DB(Mysql)...I don't know how to point these > servers to centralized DB > > Regards, > Kumaran T > > > On Mon, Feb 25, 2013 at 8:18 PM, David Villasmil < > david.villasmil.work at gmail.com> wrote: > >> Hello, >> >> What do you mean exactly by "3 freeswitch server to centralized db"? >> >> If you mean the core, just set your unixodbc dsn to pont to the db >> server... >> >> http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core >> >> David >> --- >> >> David Villasmil >> >> >> On Feb 25, 2013, at 14:07, Venkateshwaran Thirugnanam < >> venkateshwaran54 at gmail.com> wrote: >> >> Hi All, >> How to configure 3 freeswitch server to Centralized mysql DB? Please >> guild >> >> Regards, >> Kumaran T >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130226/b54791be/attachment.html From jmoran at secureachsystems.com Mon Feb 25 22:32:49 2013 From: jmoran at secureachsystems.com (Jason Moran) Date: Mon, 25 Feb 2013 14:32:49 -0500 Subject: [Freeswitch-users] Nuance Speech Server / Vocalizer MRCPv2 mod_unimrcp simultaneous licenses not releasing quickly enough. Message-ID: <361E98F99D3CC3439EED59BC1924ED6971B08A@SERVER2003.SecuReachSystems.local> In connecting to Nuance Speech Server's MRCPv2 for text-to-speech (Nuance Vocalizer) using mod_unimrcp, a single phone call which has several different TTS phrases should only use up 1 single simultaneous license. However, I am seeing a call use up 2,3,4 (etc) like it is not releasing the connection/license. I looked this up and some forums have noted that Nuance "does things wrong" and uniMRCP needs to send Nuance the "RTCP BYE" to tell Nuance that it is done with the synthesizer speech. But I tried that, and it still eats up the licenses. Within a couple of calls I'm out of TTS ports and the calls fail (but they clear again after several more seconds). Ideas? Experience? -Jason -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130225/8b3e5479/attachment-0001.html From schoch+freeswitch.org at xwin32.com Mon Feb 25 22:42:53 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Mon, 25 Feb 2013 11:42:53 -0800 Subject: [Freeswitch-users] Favorite CNAM provider Message-ID: Now that I got Caller-ID-Name working, I need to choose a provider. I have it setup now to use opencnam.com, but many numbers are "UNKNOWN" (which I'm not complaining about, because that service is free). There is also VoIPCNAM.com, which costs $2 a month, no big deal and well worth it, but I want to make sure it has good data. Feedback anyone? -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130225/0537342d/attachment.html From msc at freeswitch.org Mon Feb 25 22:45:10 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 25 Feb 2013 11:45:10 -0800 Subject: [Freeswitch-users] "Outgoing Call" on phone display In-Reply-To: References: <094868EE-812C-4E21-8551-0EC80F553D3D@freeswitch.org> <2F45CFD8-1F72-44AB-93D4-0C59BED92AD2@freeswitch.org> Message-ID: Thanks for checking in. -MC On Mon, Feb 25, 2013 at 11:22 AM, Steven Schoch < schoch+freeswitch.org at xwin32.com> wrote: > On Fri, Feb 22, 2013 at 7:40 PM, Michael Collins wrote: > >> Have you tried setting this variable for the leg in question? >> http://wiki.freeswitch.org/wiki/Variable_ignore_display_updates > > > Thank you! Problem solved! > > I added: > > > > after > > data="effective_callee_id_name=${cidlookup(1${default_areacode}$1)}"/> > > and it worked as required. > > -- > Steve > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130225/7e87938e/attachment.html From msc at freeswitch.org Mon Feb 25 22:47:20 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 25 Feb 2013 11:47:20 -0800 Subject: [Freeswitch-users] Nuance Speech Server / Vocalizer MRCPv2 mod_unimrcp simultaneous licenses not releasing quickly enough. In-Reply-To: <361E98F99D3CC3439EED59BC1924ED6971B08A@SERVER2003.SecuReachSystems.local> References: <361E98F99D3CC3439EED59BC1924ED6971B08A@SERVER2003.SecuReachSystems.local> Message-ID: Do we know why Nuance holds the port open? Is the port actually open after the RTCP BYE? Just curious. I think Chris Rienzo is the best person to ask, so let's wait for him to chime in. -MC On Mon, Feb 25, 2013 at 11:32 AM, Jason Moran wrote: > In connecting to Nuance Speech Server?s MRCPv2 for text-to-speech > (Nuance Vocalizer) using mod_unimrcp, a single phone call which has several > different TTS phrases should only use up 1 single simultaneous license. > However, I am seeing a call use up 2,3,4 (etc) like it is not releasing the > connection/license.**** > > ** ** > > I looked this up and some forums have noted that Nuance ?does things > wrong? and uniMRCP needs to send Nuance the ?RTCP BYE? to tell Nuance that > it is done with the synthesizer speech. But I tried that, and it still > eats up the licenses. Within a couple of calls I?m out of TTS ports and the > calls fail (but they clear again after several more seconds).**** > > ** ** > > **** > > **** > > **** > > **** > > **** > > ** ** > > Ideas? Experience?**** > > -Jason**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130225/9b6e7b6a/attachment.html From ga at steadfasttelecom.com Mon Feb 25 22:51:26 2013 From: ga at steadfasttelecom.com (Gilad Abada) Date: Mon, 25 Feb 2013 14:51:26 -0500 Subject: [Freeswitch-users] Favorite CNAM provider In-Reply-To: References: Message-ID: We use bulkcnam.com and havent had any complaints. On Mon, Feb 25, 2013 at 2:42 PM, Steven Schoch < schoch+freeswitch.org at xwin32.com> wrote: > Now that I got Caller-ID-Name working, I need to choose a provider. I > have it setup now to use opencnam.com, but many numbers are "UNKNOWN" > (which I'm not complaining about, because that service is free). There is > also VoIPCNAM.com, which costs $2 a month, no big deal and well worth it, > but I want to make sure it has good data. Feedback anyone? > > -- > Steve > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Gilad Abada SteadFast Telecommunications, Inc. Call us to find out how much you can save with VoIP! V: 212.589.1001 For over 35 years, Steadfast Telecommunications has been providing state-of-the-art communications technology to businesses and government agencies - large and small. Steadfast Telecommunications tailors Unified Communications and Voice-Over IP Solutions to single-site offices or multi-site and worldwide enterprises. Make your virtual office a reality. Enjoy the freedom to travel while remaining connected to your office. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130225/24eab8e2/attachment.html From justlikeef at gmail.com Mon Feb 25 23:21:33 2013 From: justlikeef at gmail.com (Rob Hutton) Date: Mon, 25 Feb 2013 15:21:33 -0500 Subject: [Freeswitch-users] Nuance Speech Server / Vocalizer MRCPv2 mod_unimrcp simultaneous licenses not releasing quickly enough. In-Reply-To: <361E98F99D3CC3439EED59BC1924ED6971B08A@SERVER2003.SecuReachSystems.local> References: <361E98F99D3CC3439EED59BC1924ED6971B08A@SERVER2003.SecuReachSystems.local> Message-ID: We have a similar situation on a Cisco UCCX platform that we ended up having to write a script to bounce the services every couple of hours to clear. We haven't seen a fix from Nuance yet. On Mon, Feb 25, 2013 at 2:32 PM, Jason Moran wrote: > In connecting to Nuance Speech Server?s MRCPv2 for text-to-speech > (Nuance Vocalizer) using mod_unimrcp, a single phone call which has several > different TTS phrases should only use up 1 single simultaneous license. > However, I am seeing a call use up 2,3,4 (etc) like it is not releasing the > connection/license.**** > > ** ** > > I looked this up and some forums have noted that Nuance ?does things > wrong? and uniMRCP needs to send Nuance the ?RTCP BYE? to tell Nuance that > it is done with the synthesizer speech. But I tried that, and it still > eats up the licenses. Within a couple of calls I?m out of TTS ports and the > calls fail (but they clear again after several more seconds).**** > > ** ** > > **** > > **** > > **** > > **** > > **** > > ** ** > > Ideas? Experience?**** > > -Jason**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130225/23afbe6d/attachment-0001.html From avi at avimarcus.net Mon Feb 25 23:21:55 2013 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 25 Feb 2013 22:21:55 +0200 Subject: [Freeswitch-users] Favorite CNAM provider In-Reply-To: References: Message-ID: Opencnam's paid service for 0.004 per query seems the way to go. bulkcnam is 0.009... callwithus is 0.006. Eek, http://www.voipcnam.com/ is quite expensive. $2 = 25 free queries for that month. Additional are 2cents each. If you use exactly 25 queries, that's 8 cents each - 20x more expensive than opencnam. -Avi On Mon, Feb 25, 2013 at 9:51 PM, Gilad Abada wrote: > We use bulkcnam.com and havent had any complaints. > > > On Mon, Feb 25, 2013 at 2:42 PM, Steven Schoch < > schoch+freeswitch.org at xwin32.com> wrote: > >> Now that I got Caller-ID-Name working, I need to choose a provider. I >> have it setup now to use opencnam.com, but many numbers are "UNKNOWN" >> (which I'm not complaining about, because that service is free). There is >> also VoIPCNAM.com, which costs $2 a month, no big deal and well worth it, >> but I want to make sure it has good data. Feedback anyone? >> >> -- >> Steve >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > > -- > Gilad Abada > > > > SteadFast Telecommunications, Inc. > > Call us to find out how much you can save with VoIP! > > V: 212.589.1001 > > > For over 35 years, Steadfast Telecommunications has been providing > state-of-the-art communications technology to businesses and government > agencies - large and small. Steadfast Telecommunications tailors Unified > Communications and Voice-Over IP Solutions to single-site offices or > multi-site and worldwide enterprises. Make your virtual office a reality. > Enjoy the freedom to travel while remaining connected to your office. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130225/aabed2b8/attachment.html From julian.pawlowski at gmail.com Tue Feb 26 00:26:02 2013 From: julian.pawlowski at gmail.com (Julian Pawlowski) Date: Mon, 25 Feb 2013 22:26:02 +0100 Subject: [Freeswitch-users] German voice prompts In-Reply-To: References: Message-ID: Hi Michael, On Mon, Feb 18, 2013 at 8:41 PM, Michael Collins wrote: > Thanks for taking the lead on this. Once you get a set of sound files > ready for production please let me know. I will get them house in our > freeswitch-sounds git repo so that we can keep track of them and properly > roll tarballs to put out on files.freeswitch.org. > I wanted to give you an update on this. It actually became a lot more than just only creating german voice prompts now. The repo currently hosted on https://github.com/jpawlowski/freeswitch-sounds-tts provides capability to create voice prompts for any language via Google TTS and Bing TTS services. Although we have lovely Callie for the english people already I found this useful as there may be people out there wanting to create their own prompts but wish to mix it with existing ones (which is doable with Callie but it mixes the voices...). You may download pre-compiled prompts for german and english here: http://repo.profhost.eu/static/freeswitch/ Although I gave them version 1.0.1 already I'm not sure if they are ready to be published on files.freeswitch.org yet, maybe wait for some feedback first. Oh and I did some improvements to the english part also, e.g. - typos in the filename - adjust long texts with a line break for separate processing (e.g. Google wants shorter texts) - identify tones and music files - remove duplicates (mostly related to the tones) - (can't remember right now, might be more) For the tones part: When I was looking through the files it seemed there was an intention to include tone files for e.g. joining/leaving a conference. I moved them to a dedicated folder and tried to find some first files: https://github.com/jpawlowski/freeswitch-sounds-tts/tree/master/tone https://github.com/jpawlowski/freeswitch-sounds-tts/tree/master/music Those files will be included into the normal voice path during packaging (see files mentioned above). I didn't check if tones are used anywhere in the FS source code already to adapt the file names yet. There are also a bunch of administrative scripts included to import/export files from XML or existing Callie voice. I also have plans to create a new XML file to update phrase_en.xml and phrase_de.xml in the FS Git repo but it's not finished yet. The scripts are not really pretty and mix different kind of programming languages. However they are working for now, may be consolidation can be done later :-) For now I would like to keep the repo for the TTS stuff at my Github account but surely we can talk about a transfer to the FS servers later. Cheers, Julian -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130225/45ddb43f/attachment.html From msc at freeswitch.org Tue Feb 26 00:54:19 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 25 Feb 2013 13:54:19 -0800 Subject: [Freeswitch-users] German voice prompts In-Reply-To: References: Message-ID: Thanks for checking in and showing us this tool. If you have a minute to join us on the Wednesday conference call maybe you could call in and talk about it. :) Thanks, MC On Mon, Feb 25, 2013 at 1:26 PM, Julian Pawlowski < julian.pawlowski at gmail.com> wrote: > Hi Michael, > > On Mon, Feb 18, 2013 at 8:41 PM, Michael Collins > wrote: > >> Thanks for taking the lead on this. Once you get a set of sound files >> ready for production please let me know. I will get them house in our >> freeswitch-sounds git repo so that we can keep track of them and properly >> roll tarballs to put out on files.freeswitch.org. >> > > I wanted to give you an update on this. > > It actually became a lot more than just only creating german voice prompts > now. > The repo currently hosted on > https://github.com/jpawlowski/freeswitch-sounds-tts provides capability > to create voice prompts for any language via Google TTS and Bing TTS > services. > > Although we have lovely Callie for the english people already I found this > useful as there may be people out there wanting to create their own > prompts but wish to mix it with existing ones (which is doable with > Callie but it mixes the voices...). > > You may download pre-compiled prompts for german and english here: > http://repo.profhost.eu/static/freeswitch/ > > Although I gave them version 1.0.1 already I'm not sure if they are ready > to be published on files.freeswitch.org yet, maybe wait for some feedback > first. > > Oh and I did some improvements to the english part also, e.g. > > - typos in the filename > - adjust long texts with a line break for separate processing (e.g. Google > wants shorter texts) > - identify tones and music files > - remove duplicates (mostly related to the tones) > - (can't remember right now, might be more) > > For the tones part: When I was looking through the files it seemed there > was an intention to include tone files for e.g. joining/leaving a > conference. I moved them to a dedicated folder and tried to find some first > files: > https://github.com/jpawlowski/freeswitch-sounds-tts/tree/master/tone > https://github.com/jpawlowski/freeswitch-sounds-tts/tree/master/music > > Those files will be included into the normal voice path during packaging > (see files mentioned above). > I didn't check if tones are used anywhere in the FS source code already to > adapt the file names yet. > > There are also a bunch of administrative scripts included to import/export > files from XML or existing Callie voice. I also have plans to create a > new XML file to update phrase_en.xml and phrase_de.xml in the FS Git repo > but it's not finished yet. > > The scripts are not really pretty and mix different kind of programming > languages. However they are working for now, may be consolidation can be > done later :-) > For now I would like to keep the repo for the TTS stuff at my Github > account but surely we can talk about a transfer to the FS servers later. > > > Cheers, > Julian > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130225/da844744/attachment-0001.html From jaykris at gmail.com Tue Feb 26 00:57:10 2013 From: jaykris at gmail.com (JP) Date: Mon, 25 Feb 2013 13:57:10 -0800 Subject: [Freeswitch-users] How do I execute shell command from Lua? Message-ID: I entered this into my Lua script local f = io.popen("pwd") -- runs command local o = f:read("*a") -- read output of command and executed the script from fs_cli. This only caused the fs_cli to become unresponsive. Is there something I am doing wrong? Thanks JP -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130225/ce72f3bb/attachment.html From msc at freeswitch.org Tue Feb 26 01:01:45 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 25 Feb 2013 14:01:45 -0800 Subject: [Freeswitch-users] How do I execute shell command from Lua? In-Reply-To: References: Message-ID: Is it supposed to be f:read() or f.read() ? -MC On Mon, Feb 25, 2013 at 1:57 PM, JP wrote: > I entered this into my Lua script > > local f = io.popen("pwd") -- runs command > local o = f:read("*a") -- read output of command > > and executed the script from fs_cli. This only caused the fs_cli to become > unresponsive. > > Is there something I am doing wrong? > > Thanks > JP > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130225/88adf5d4/attachment.html From jaykris at gmail.com Tue Feb 26 01:06:06 2013 From: jaykris at gmail.com (JP) Date: Mon, 25 Feb 2013 14:06:06 -0800 Subject: [Freeswitch-users] How do I execute shell command from Lua? In-Reply-To: References: Message-ID: I tried both with the same result. -JP On Mon, Feb 25, 2013 at 2:01 PM, Michael Collins wrote: > Is it supposed to be f:read() or f.read() ? > -MC > > On Mon, Feb 25, 2013 at 1:57 PM, JP wrote: > >> I entered this into my Lua script >> >> local f = io.popen("pwd") -- runs command >> local o = f:read("*a") -- read output of command >> >> and executed the script from fs_cli. This only caused the fs_cli to >> become unresponsive. >> >> Is there something I am doing wrong? >> >> Thanks >> JP >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130225/19ba2306/attachment.html From cmrienzo at gmail.com Tue Feb 26 01:21:21 2013 From: cmrienzo at gmail.com (cmrienzo at gmail.com) Date: Mon, 25 Feb 2013 17:21:21 -0500 Subject: [Freeswitch-users] Nuance Speech Server / Vocalizer MRCPv2 mod_unimrcp simultaneous licenses not releasing quickly enough. In-Reply-To: References: <361E98F99D3CC3439EED59BC1924ED6971B08A@SERVER2003.SecuReachSystems.local> Message-ID: <95382631-D364-43A5-B93F-E553F68248F9@gmail.com> We also had an issue with nuance eating licenses. We had enough room to tolerate this but did have to occasionally reboot servers when too many were consumed. Chris On Feb 25, 2013, at 3:21 PM, Rob Hutton wrote: > We have a similar situation on a Cisco UCCX platform that we ended up having to write a script to bounce the services every couple of hours to clear. We haven't seen a fix from Nuance yet. > > On Mon, Feb 25, 2013 at 2:32 PM, Jason Moran wrote: >> In connecting to Nuance Speech Server?s MRCPv2 for text-to-speech (Nuance Vocalizer) using mod_unimrcp, a single phone call which has several different TTS phrases should only use up 1 single simultaneous license. However, I am seeing a call use up 2,3,4 (etc) like it is not releasing the connection/license. >> >> >> >> I looked this up and some forums have noted that Nuance ?does things wrong? and uniMRCP needs to send Nuance the ?RTCP BYE? to tell Nuance that it is done with the synthesizer speech. But I tried that, and it still eats up the licenses. Within a couple of calls I?m out of TTS ports and the calls fail (but they clear again after several more seconds). >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> Ideas? Experience? >> >> -Jason >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130225/da48e405/attachment-0001.html From msc at freeswitch.org Tue Feb 26 02:20:37 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 25 Feb 2013 15:20:37 -0800 Subject: [Freeswitch-users] How do I execute shell command from Lua? In-Reply-To: References: Message-ID: Did you use "lua" or "luarun" to launch the script from fs_cli? -MC On Mon, Feb 25, 2013 at 2:06 PM, JP wrote: > I tried both with the same result. > > -JP > > > On Mon, Feb 25, 2013 at 2:01 PM, Michael Collins wrote: > >> Is it supposed to be f:read() or f.read() ? >> -MC >> >> On Mon, Feb 25, 2013 at 1:57 PM, JP wrote: >> >>> I entered this into my Lua script >>> >>> local f = io.popen("pwd") -- runs command >>> local o = f:read("*a") -- read output of command >>> >>> and executed the script from fs_cli. This only caused the fs_cli to >>> become unresponsive. >>> >>> Is there something I am doing wrong? >>> >>> Thanks >>> JP >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130225/ae91dcfe/attachment.html From anthony.minessale at gmail.com Tue Feb 26 02:40:03 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 25 Feb 2013 17:40:03 -0600 Subject: [Freeswitch-users] Door intercom/gate controller In-Reply-To: References: <144901ce0bbe$5ab583a0$10208ae0$@freeswitch.org> Message-ID: you put start_dtmf in the execute_on_answer variable to run it on the other leg. On Mon, Feb 25, 2013 at 10:34 AM, Michael Collins wrote: > Hi Nathan, > > Yes, start_dtmf is one-directional as far as I am aware, that is, it only > works on the sending stream of the channel on which it is set. BTW, I'm > glad you got it figured out! > > -MC > > > On Fri, Feb 22, 2013 at 6:32 PM, Mr Nathan Downes < > nathandownes at hotmail.com> wrote: > >> Hi Michael,**** >> >> ** ** >> >> Thanks for the response, it worked in the opposite direction, the gate >> controller (OOB) calls phones (Inband) kind of like the door thing on an >> apartment block. Start_dtmf worked if I called the gate controller from >> the phones, but not in the other direction which is how it will be used.* >> *** >> >> ** ** >> >> It has been resolved as it was discovered the ATA devices inbuilt to the >> fibre ONT have an option for OOB DTMF, enabling this on the POTS profile >> allowed the device to function.**** >> >> ** ** >> >> I am still interested on how I could of got it to work though, is >> start_dtmf a one directional app?**** >> >> ** ** >> >> Thanks,**** >> >> ** ** >> >> Nathan**** >> >> ** ** >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael >> Collins >> *Sent:* Saturday, 16 February 2013 6:35 AM >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] Door intercom/gate controller**** >> >> ** ** >> >> Can you show us the dialplan for sending a call to the gate controller? >> Also, pastebin the console debug output for a call to the gate controller >> where you press digits but the controller doesn't respond.**** >> >> ** ** >> >> Thanks,**** >> >> MC**** >> >> On Wed, Feb 13, 2013 at 2:42 AM, Mr Nathan Downes < >> nathandownes at hotmail.com> wrote:**** >> >> Hi list,**** >> >> **** >> >> I have a gate controller that only understands RFC 2833 but it calls >> endpoints that can only provide inband DTMF, I can?t seem to get the digits >> recognised by the gate controller to trigger the relay using >> start_dtmf_generate, as they are coming back to the a-leg? Rather than out >> the b-leg.. It works fine when I call a SPA502g as it will provide RFC >> 2833.. is there some trickery I can achieve this? Or am I just doing it >> wrong?? J**** >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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**** >> >> >> >> **** >> >> ** ** >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org**** >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130225/b5d70c1e/attachment-0001.html From msc at freeswitch.org Tue Feb 26 02:42:45 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 25 Feb 2013 15:42:45 -0800 Subject: [Freeswitch-users] Nuance Speech Server / Vocalizer MRCPv2 mod_unimrcp simultaneous licenses not releasing quickly enough. In-Reply-To: <95382631-D364-43A5-B93F-E553F68248F9@gmail.com> References: <361E98F99D3CC3439EED59BC1924ED6971B08A@SERVER2003.SecuReachSystems.local> <95382631-D364-43A5-B93F-E553F68248F9@gmail.com> Message-ID: Another case of DRM hurting only those who've paid for it... *sigh* I recommend calling your VAR or Nuance and gently but firmly insisting on a resolution. -MC On Mon, Feb 25, 2013 at 2:21 PM, wrote: > We also had an issue with nuance eating licenses. We had enough room to > tolerate this but did have to occasionally reboot servers when too many > were consumed. > > Chris > > > > > On Feb 25, 2013, at 3:21 PM, Rob Hutton wrote: > > We have a similar situation on a Cisco UCCX platform that we ended up > having to write a script to bounce the services every couple of hours to > clear. We haven't seen a fix from Nuance yet. > > On Mon, Feb 25, 2013 at 2:32 PM, Jason Moran wrote: > >> In connecting to Nuance Speech Server?s MRCPv2 for text-to-speech >> (Nuance Vocalizer) using mod_unimrcp, a single phone call which has several >> different TTS phrases should only use up 1 single simultaneous license. >> However, I am seeing a call use up 2,3,4 (etc) like it is not releasing the >> connection/license.**** >> >> ** ** >> >> I looked this up and some forums have noted that Nuance ?does things >> wrong? and uniMRCP needs to send Nuance the ?RTCP BYE? to tell Nuance that >> it is done with the synthesizer speech. But I tried that, and it still >> eats up the licenses. Within a couple of calls I?m out of TTS ports and the >> calls fail (but they clear again after several more seconds).**** >> >> ** ** >> >> **** >> >> **** >> >> **** >> >> **** >> >> **** >> >> ** ** >> >> Ideas? Experience?**** >> >> -Jason**** >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130225/838e1ed5/attachment.html From jaykris at gmail.com Tue Feb 26 02:53:46 2013 From: jaykris at gmail.com (JP) Date: Mon, 25 Feb 2013 15:53:46 -0800 Subject: [Freeswitch-users] How do I execute shell command from Lua? In-Reply-To: References: Message-ID: Thanks MC. I was using "lua" and not "luarun". With "luarun" it is working :) -JP On Mon, Feb 25, 2013 at 3:20 PM, Michael Collins wrote: > Did you use "lua" or "luarun" to launch the script from fs_cli? > -MC > > > On Mon, Feb 25, 2013 at 2:06 PM, JP wrote: > >> I tried both with the same result. >> >> -JP >> >> >> On Mon, Feb 25, 2013 at 2:01 PM, Michael Collins wrote: >> >>> Is it supposed to be f:read() or f.read() ? >>> -MC >>> >>> On Mon, Feb 25, 2013 at 1:57 PM, JP wrote: >>> >>>> I entered this into my Lua script >>>> >>>> local f = io.popen("pwd") -- runs command >>>> local o = f:read("*a") -- read output of command >>>> >>>> and executed the script from fs_cli. This only caused the fs_cli to >>>> become unresponsive. >>>> >>>> Is there something I am doing wrong? >>>> >>>> Thanks >>>> JP >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>>> >>>> >>> >>> >>> -- >>> Michael S Collins >>> Twitter: @mercutioviz >>> http://www.FreeSWITCH.org >>> http://www.ClueCon.com >>> http://www.OSTAG.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130225/40b2c104/attachment-0001.html From brian at freeswitch.org Tue Feb 26 02:58:54 2013 From: brian at freeswitch.org (Brian West) Date: Mon, 25 Feb 2013 17:58:54 -0600 Subject: [Freeswitch-users] Favorite CNAM provider In-Reply-To: References: Message-ID: Open CNAM will usually say UNKNOWN on first query for that number. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire T: +1.918.420.9266 | F: +1.918.420.9267 | M: +1.918.424.WEST iNUM: +883 5100 1420 9266 ISN: 410*543 On Feb 25, 2013, at 1:42 PM, Steven Schoch wrote: > Now that I got Caller-ID-Name working, I need to choose a provider. I have it setup now to use opencnam.com, but many numbers are "UNKNOWN" (which I'm not complaining about, because that service is free). There is also VoIPCNAM.com, which costs $2 a month, no big deal and well worth it, but I want to make sure it has good data. Feedback anyone? > > -- > Steve > ______ From schoch+freeswitch.org at xwin32.com Tue Feb 26 03:03:57 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Mon, 25 Feb 2013 16:03:57 -0800 Subject: [Freeswitch-users] Best current ATA and Setup for Faxing In-Reply-To: <0B7A549A-767A-4DA6-B732-9999DC6B1B48@gmail.com> References: <0B7A549A-767A-4DA6-B732-9999DC6B1B48@gmail.com> Message-ID: On Wed, Feb 13, 2013 at 8:14 AM, Nick Vines wrote: > I havent used the cisco atas, but I've had good success with the grand > stream ht502s and ht702s. My preferred faxing carrier is also flowroute. I just installed an HT502, and it worked! -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130225/f29f32f9/attachment.html From brian at freeswitch.org Tue Feb 26 03:04:54 2013 From: brian at freeswitch.org (Brian West) Date: Mon, 25 Feb 2013 18:04:54 -0600 Subject: [Freeswitch-users] Possible BUG in G729 negotiation In-Reply-To: References: Message-ID: Try setting this variable: sip_append_audio_sdp=a=fmtp:18 annexb=no Can you verify that your provider is actually sending CN on 19 and NOT 13. There are some Sonus switches that say 19 but mean 13 and send on 13. 19=13 in hex. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire T: +1.918.420.9266 | F: +1.918.420.9267 | M: +1.918.424.WEST iNUM: +883 5100 1420 9266 ISN: 410*543 On Feb 25, 2013, at 8:08 AM, Flavio Goncalves wrote: > Hi, > > I have searched the list, but I couldn't find a final answer for this case. I'm receiving calls from a provider using G729 annex B. > > INVITE > a=rtpmap:18 G729/8000. > a=fmtp:18 annexb=yes. > a=rtpmap:0 PCMU/8000. > a=rtpmap:8 PCMA/8000. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=rtpmap:19 CN/8000. > > and FS is answering > > 200OK > m=audio 31388 RTP/AVP 18 101 19. > a=rtpmap:18 G729/8000. > a=fmtp:18 annexb=yes. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=rtpmap:19 CN/8000. > a=ptime:20. > > If freeswitch supports only G729A, shouldn't FS refuse the call with 488 or suppress the line annexb=yes? > > The audio quality is really bad when I receive calls in this way. After removing the annexb on the gateway the audio is fine, but unfortunately I can't remove from it from all VoIP providers terminating calls. For outbound calls I simply removed the annexb, but for inbound there is no way to do it. > > Flavio > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 brian at freeswitch.org Tue Feb 26 03:10:35 2013 From: brian at freeswitch.org (Brian West) Date: Mon, 25 Feb 2013 18:10:35 -0600 Subject: [Freeswitch-users] INVITE Via: header parameters In-Reply-To: <5CEB828A-8DDB-4E08-AD10-A292CB7CD359@gmail.com> References: <5CEB828A-8DDB-4E08-AD10-A292CB7CD359@gmail.com> Message-ID: <6798653F-8C10-44B2-9E6D-0B9C527D74F0@freeswitch.org> You can't, we specifically do not as you shouldn't have to when its on 5060. If your device requires you to say 5060, look in sofia_glue.c for sofia_glue_create_via, Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire T: +1.918.420.9266 | F: +1.918.420.9267 | M: +1.918.424.WEST iNUM: +883 5100 1420 9266 ISN: 410*543 On Feb 25, 2013, at 9:19 AM, Deon Vermeulen wrote: > How can I force the :5060 in the Via: uri? From dujinfang at gmail.com Tue Feb 26 03:11:46 2013 From: dujinfang at gmail.com (Seven Du) Date: Tue, 26 Feb 2013 08:11:46 +0800 Subject: [Freeswitch-users] [mod_fsv] How the video recording really looks like? In-Reply-To: References: <14FF31176F964AD6A1D70E66EC16FAB9@gmail.com> <922DB54402EF453C96DB20C17697B2C0@gmail.com> Message-ID: <384B4B29-021D-4307-AC5D-C4AD4F099BF2@gmail.com> Thanks, but You can also register an account and update the wiki page. ???? iPad ? 2013-2-25???11:04?K K ??? > AD.1 > > OK so I see now that I need to compile the video-media-bug branch to achieve my goal. I am sorry, but in my opinion this Wiki page is very confusing, because of the fact that the mod_fsv is delivered with the current FreeSWITCH installer, and you do not need to be in the video-media-bug to use this mod as it is described in the "Record a single legged call" section. It should be clarified which features are and which are not available. Thanks to such one Wiki sentence you can get less mails on yours mailing list. > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 jnankin at gmail.com Tue Feb 26 03:49:01 2013 From: jnankin at gmail.com (Josh Nankin) Date: Mon, 25 Feb 2013 18:49:01 -0600 Subject: [Freeswitch-users] event socket serialize(format) - what is format? Message-ID: In the documentation here: http://wiki.freeswitch.org/wiki/Esl#serialize There is a 'format' argument for serialize. What does that do? Can't easily tell from the documentation, or the client library code. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130225/68b170b2/attachment.html From cmrienzo at gmail.com Tue Feb 26 04:33:31 2013 From: cmrienzo at gmail.com (Christopher Rienzo) Date: Mon, 25 Feb 2013 20:33:31 -0500 Subject: [Freeswitch-users] Nuance Speech Server / Vocalizer MRCPv2 mod_unimrcp simultaneous licenses not releasing quickly enough. In-Reply-To: References: <361E98F99D3CC3439EED59BC1924ED6971B08A@SERVER2003.SecuReachSystems.local> <95382631-D364-43A5-B93F-E553F68248F9@gmail.com> Message-ID: You can also try MRCPv1 (RTSP) and see if there is better performance. Using SIP to set up an MRCP session is silly. Chris On Mon, Feb 25, 2013 at 6:42 PM, Michael Collins wrote: > Another case of DRM hurting only those who've paid for it... *sigh* > > I recommend calling your VAR or Nuance and gently but firmly insisting on > a resolution. > > -MC > > > On Mon, Feb 25, 2013 at 2:21 PM, wrote: > >> We also had an issue with nuance eating licenses. We had enough room to >> tolerate this but did have to occasionally reboot servers when too many >> were consumed. >> >> Chris >> >> >> >> >> On Feb 25, 2013, at 3:21 PM, Rob Hutton wrote: >> >> We have a similar situation on a Cisco UCCX platform that we ended up >> having to write a script to bounce the services every couple of hours to >> clear. We haven't seen a fix from Nuance yet. >> >> On Mon, Feb 25, 2013 at 2:32 PM, Jason Moran > > wrote: >> >>> In connecting to Nuance Speech Server?s MRCPv2 for text-to-speech >>> (Nuance Vocalizer) using mod_unimrcp, a single phone call which has several >>> different TTS phrases should only use up 1 single simultaneous license. >>> However, I am seeing a call use up 2,3,4 (etc) like it is not releasing the >>> connection/license.**** >>> >>> ** ** >>> >>> I looked this up and some forums have noted that Nuance ?does things >>> wrong? and uniMRCP needs to send Nuance the ?RTCP BYE? to tell Nuance that >>> it is done with the synthesizer speech. But I tried that, and it still >>> eats up the licenses. Within a couple of calls I?m out of TTS ports and the >>> calls fail (but they clear again after several more seconds).**** >>> >>> ** ** >>> >>> **** >>> >>> **** >>> >>> **** >>> >>> **** >>> >>> **** >>> >>> ** ** >>> >>> Ideas? Experience?**** >>> >>> -Jason**** >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130225/000a449e/attachment-0001.html From schoch+freeswitch.org at xwin32.com Tue Feb 26 04:51:19 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Mon, 25 Feb 2013 17:51:19 -0800 Subject: [Freeswitch-users] Retrieving voicemail from home Message-ID: My users are used to voicemail systems where you press a key (like #) during your greeting to be prompted for a password to retrieve voicemail. Is such a method available in FreeSWITCH? Also, is there a way to deny access to voicemail if their vm-password is set to "user-choose" and they haven't set one yet? -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130225/f08f82b2/attachment.html From anthony.minessale at gmail.com Tue Feb 26 04:52:02 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 25 Feb 2013 19:52:02 -0600 Subject: [Freeswitch-users] event socket serialize(format) - what is format? In-Reply-To: References: Message-ID: "xml" "json" or "plain" (default) On Mon, Feb 25, 2013 at 6:49 PM, Josh Nankin wrote: > In the documentation here: > http://wiki.freeswitch.org/wiki/Esl#serialize > > There is a 'format' argument for serialize. What does that do? Can't > easily tell from the documentation, or the client library code. > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130225/27915482/attachment.html From msc at freeswitch.org Tue Feb 26 06:41:01 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 25 Feb 2013 19:41:01 -0800 Subject: [Freeswitch-users] event socket serialize(format) - what is format? In-Reply-To: References: Message-ID: On Mon, Feb 25, 2013 at 5:52 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > "xml" "json" or "plain" (default) > Thanks Tony. I added this to the wiki: http://wiki.freeswitch.org/wiki/Event_Socket_Library#serialize -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130225/cf2736fd/attachment.html From msc at freeswitch.org Tue Feb 26 06:50:43 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 25 Feb 2013 19:50:43 -0800 Subject: [Freeswitch-users] Retrieving voicemail from home In-Reply-To: References: Message-ID: On Mon, Feb 25, 2013 at 5:51 PM, Steven Schoch < schoch+freeswitch.org at xwin32.com> wrote: > My users are used to voicemail systems where you press a key (like #) > during your greeting to be prompted for a password to retrieve voicemail. > Is such a method available in FreeSWITCH? > Yes. I believe the default value is "0" when they hear their own outgoing message. The value can be adjusted in conf/autoload_configs/voicemail.conf.xml. Look for this line: What I don't know is if you can set it to more than one keystroke, like "##" or somesuch. In any case, have fun tinkering with that value. Be sure to reload mod_voicemail after making changes to that file. > > Also, is there a way to deny access to voicemail if their vm-password is > set to "user-choose" and they haven't set one yet? > Yes. I personally just put a value into the database for the user(s) in question. I use something like the following (keep in mind that you need to define ${domain} or insert the proper value): for I in `seq 1000 1019`; do sqlite3 /usr/local/freeswitch/db/voicemail_default.db "insert into voicemail_prefs (username,domain,password) values ('$I','${domain}','12345');" done That sets the default password on all user VMs to "12345" for users 1000 - 1019. Use passwords and user ID's appropriate for your installation. -MC > > -- > Steve > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130225/1540341f/attachment.html From lists at kavun.ch Tue Feb 26 08:36:01 2013 From: lists at kavun.ch (Emrah) Date: Tue, 26 Feb 2013 00:36:01 -0500 Subject: [Freeswitch-users] Concatenated multiple playback types In-Reply-To: References: <1FFF97C269757C458224B7C895F35F1514B3F0@cantor.std.visionutv.se> <55F547E7-F986-4268-9BBD-C4210D997D12@kavun.ch> Message-ID: Hi there, Just wanted to inquire about this again. FS is ever so flexible in so many ways that I cannot understand why this wouldn't work. I would like to apply the following to my alone sound in conference.conf.xml. silence_stream://6000!file_string:///path-to-alone-sound.wav I am imagining other scenarios where I would want to use some silence before generating a ringback. What is your take on this? Cheers On Aug 22, 2012, at 12:40 PM, Michael Collins wrote: > One thing I like about the sox method is that it requires very little in the way of processing/resources. > -MC > > On Wed, Aug 22, 2012 at 9:27 AM, Emrah wrote: > Hey Michael, > > That was my work around before I posted, but I thought since you guys do allow some flexibility with silence, tone generation and file playback all usable with the same app, I thought it would make more sense to take advantage of it and combine some things there. > > All the best, thanks for your response. > Emrah > On Aug 22, 2012, at 12:07 PM, Michael Collins wrote: > > > You could always take the sneaky approach and use sox to add some silence to the beginning of conf-alone.wav. > > -MC > > > > On Wed, Aug 22, 2012 at 8:56 AM, Emrah wrote: > > Thanks for your response. > > I don't think it would be something bad to implement. > > > > I also wanted to correct a little issueI experience both with FS and Asterisk. > > If you generate a tone, even if it's played in a file, and you execute a record action right after it is played, the last 30 ms (or so) of the tone is distorded and kind of faded out. > > In some sircompstances, you might even hear the hint of the tone when you execute the next playback. > > I wanted to replace my record tones with the tone followed by 30 MS silence. > > > > Best, > > Emrah > > > > On Aug 22, 2012, at 10:36 AM, Peter Olsson wrote: > > > > > I don't believe this is possible. If you record 5 seconds of silence into a file you might be able to use file_string:// to play them both. > > > > > > /Peter > > > > > > > > > -----Ursprungligt meddelande----- > > > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Emrah > > > Skickat: den 22 augusti 2012 16:24 > > > Till: FreeSWITCH Users Help > > > ?mne: [Freeswitch-users] Concatenated multiple playback types > > > > > > Hello there, > > > > > > I am trying to add some silence before the alone sound file in my conference.cnf.xml and can't get it to work. > > > > > > Here is what I tried: > > > > > > > > > > > > > > > > > > Any idea on how to do this? > > > > > > Thanks! > > > Emrah > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > > > > > 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 > > > > > > !DSPAM:5034e9a632761303016055! > > > > > > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > > > > > 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 > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > 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 > > > > > > > > -- > > Michael S Collins > > Twitter: @mercutioviz > > http://www.FreeSWITCH.org > > http://www.ClueCon.com > > http://www.OSTAG.org > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 vermeulen.deon at gmail.com Tue Feb 26 10:47:34 2013 From: vermeulen.deon at gmail.com (Deon Vermeulen) Date: Tue, 26 Feb 2013 09:47:34 +0200 Subject: [Freeswitch-users] INVITE Via: header parameters In-Reply-To: <6798653F-8C10-44B2-9E6D-0B9C527D74F0@freeswitch.org> References: <5CEB828A-8DDB-4E08-AD10-A292CB7CD359@gmail.com> <6798653F-8C10-44B2-9E6D-0B9C527D74F0@freeswitch.org> Message-ID: <512C6896.3050504@gmail.com> Thanks for the reply Brian The only reason we're trying to do this is because of a misconfigured MSC that only accepts and sends SIP on 5060 based fromthe Via header. I don't like messing with internal system code, so the provider just have to get their act together. 5060 is not a secure port any way. I'm busy with the Administrator of the MSC to allocate at least another non-default port on which one can setup the trunk. Kind Regards On 2/26/2013 2:10 AM, Brian West wrote: > You can't, we specifically do not as you shouldn't have to when its on 5060. If your device requires you to say 5060, look in sofia_glue.c for sofia_glue_create_via, > > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH_Wire > T: +1.918.420.9266 | F: +1.918.420.9267 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9266 > ISN: 410*543 > > > > > > > On Feb 25, 2013, at 9:19 AM, Deon Vermeulen wrote: > >> How can I force the :5060 in the Via: uri? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130226/3377833a/attachment.html From pm_zefman_r at mail.ru Tue Feb 26 11:01:23 2013 From: pm_zefman_r at mail.ru (=?UTF-8?B?RG1pdHJpeSBTaHVtYWV2?=) Date: Tue, 26 Feb 2013 12:01:23 +0400 Subject: [Freeswitch-users] =?utf-8?q?Logical_AND_+_NOT?= Message-ID: <1361865683.776896435@f250.mail.ru> Hi I want to record all sessions, except calls to group numbers and to IVR. So I need something like: ? ? ?? ?? ?? ?? ?? ?? ? ? ? ? ? ? ?? ? . But the syntax of condition does not allow the operator " not equal" or "!=". What should I do? With best regards, Shumaev DA, KBR Ltd. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130226/cd9558c5/attachment.html From avi at avimarcus.net Tue Feb 26 11:53:03 2013 From: avi at avimarcus.net (Avi Marcus) Date: Tue, 26 Feb 2013 10:53:03 +0200 Subject: [Freeswitch-users] Logical AND + NOT In-Reply-To: <1361865683.776896435@f250.mail.ru> References: <1361865683.776896435@f250.mail.ru> Message-ID: Not necessarily the best answer, but you can use negative lookahead, e.g.: (?!399|400)^(\d{3,16})$ See it here: http://www.rubular.com/r/XVTB40DPZZ Alternatives... 1) A different extension that sets a variable record=false for 400, 399, and an anti-action sets it to true. (not sure that's exactly the same though) 2) a short lua script that sets the variable or actually starts the recording. 3) I'm sure there are other options... -Avi Marcus On Tue, Feb 26, 2013 at 10:01 AM, Dmitriy Shumaev wrote: > Hi > > I want to record all sessions, except calls to group numbers and to IVR. > So I need something like: > > > > > > > > > > > data="$${base_dir}/recordings/archive/${strftime(%Y-%m-%d-%H-%M-%S)}_${caller_id_name}_${caller_id_number}_2_${destination_number}.mp3"/> > > > > . But the syntax of condition does not allow the operator "not equal" or > "!=". What should I do? > > > With best regards, Shumaev DA, KBR Ltd. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130226/0a8967e0/attachment.html From a.venugopan at mundio.com Tue Feb 26 12:12:47 2013 From: a.venugopan at mundio.com (Archana Venugopan) Date: Tue, 26 Feb 2013 09:12:47 +0000 Subject: [Freeswitch-users] make mod_voicemail In-Reply-To: References: <592A9CF93E12394E8472A6CC66E66BF23AC9F1@Mail-Kilo.squay.com> <592A9CF93E12394E8472A6CC66E66BF23ACA1B@Mail-Kilo.squay.com> <592A9CF93E12394E8472A6CC66E66BF23AD13E@Mail-Kilo.squay.com> Message-ID: <592A9CF93E12394E8472A6CC66E66BF23AD7D8@Mail-Kilo.squay.com> Hi, Thanks. I tried running make mod_voicemail command. But after running this command we re-started freeswitch. And it throwed few errors as like below and freeswitch is not coming up. Please let me know what changed after running make mod_voicemail command. Should I need to re-build my FS completely now? 2013-02-26 08:58:52.872222 [CRIT] switch_loadable_module.c:1310 Error Loading module /usr/local/freeswitch/mod/mod_cdr_csv.so **/usr/local/freeswitch/mod/mod_cdr_csv.so: undefined symbol: switch_channel_expand_variables** 2013-02-26 08:58:52.872607 [CRIT] switch_loadable_module.c:1310 Error Loading module /usr/local/freeswitch/mod/mod_event_socket.so **/usr/local/freeswitch/mod/mod_event_socket.so: undefined symbol: switch_core_session_locate** 2013-02-26 08:58:52.873116 [CRIT] switch_loadable_module.c:1310 Error Loading module /usr/local/freeswitch/mod/mod_lua.so **/usr/local/freeswitch/mod/mod_lua.so: undefined symbol: _ZN13EventConsumerC1EPKcS1_** 2013-02-26 08:58:52.873661 [CRIT] switch_loadable_module.c:1310 Error Loading module /usr/local/freeswitch/mod/mod_sofia.so **/usr/local/freeswitch/mod/mod_sofia.so: undefined symbol: switch_core_session_get_partner** 2013-02-26 08:58:52.874002 [CRIT] switch_loadable_module.c:1310 Error Loading module /usr/local/freeswitch/mod/mod_loopback.so **/usr/local/freeswitch/mod/mod_loopback.so: undefined symbol: switch_core_session_get_partner** 2013-02-26 08:58:52.874354 [CRIT] switch_loadable_module.c:1310 Error Loading module /usr/local/freeswitch/mod/mod_commands.so **/usr/local/freeswitch/mod/mod_commands.so: undefined symbol: switch_core_session_get_partner** 2013-02-26 08:58:52.874731 [CRIT] switch_loadable_module.c:1310 Error Loading module /usr/local/freeswitch/mod/mod_conference.so **/usr/local/freeswitch/mod/mod_conference.so: undefined symbol: switch_channel_expand_variables** Regards, Archana From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven Ayre Sent: 23 February 2013 14:37 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] make mod_voicemail Building will only affect files within the source directory - not your install paths (/usr/local/freeswitch). Make sure you build mod_voicemail using the same version of FS as you're running though - otherwise there may be differences in the core API between the version being used by freeswitch and by the module. -Steve On 23 February 2013 13:35, Archana Venugopan > wrote: If its building the core whether it would affect the scripts inside /usr/local/freeswitch folder as well? If yes then all the scripts inside would be lost. Please let me know. Regards, Archana From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven Ayre Sent: 23 February 2013 13:06 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] make mod_voicemail This shows that it's building the FS core (src/switch_ivr.c is part of that). That means that file hadn't been built or the build wasn't up-to-date. Building the module without the core is not possible - the below is expected behaviour. Once the core and module are built, the -install will only install the module not the rest. -Steve On 21 February 2013 09:30, Archana Venugopan > wrote: I am afraid if that is as well installing everything so I just stopped it. [root at sme-xswitch freeswitch]# make mod_voicemail-install quiet_libtool: compile: gcc -DHAVE_CONFIG_H -I. -I./src/include -I./libs/xmlrpc-c -I/usr/local/src/freeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -pthread -DLINUX=2 -D_REENTRANT -D_GNU_SOURCE -D_LARGEFILE64_SOURCE -I/usr/local/src/freeswitch/libs/apr/include -I/usr/local/src/freeswitch/libs/apr-util/include -I/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib -I/usr/local/src/freeswitch/libs/libtpl-1.5/src -I/usr/local/src/freeswitch/libs/stfu -I/usr/local/src/freeswitch/libs/sqlite -I/usr/local/src/freeswitch/libs/pcre -I/usr/local/src/freeswitch/libs/speex/include -Ilibs/speex/include -I/usr/local/src/freeswitch/libs/srtp/include -I/usr/local/src/freeswitch/libs/srtp/crypto/include -Ilibs/srtp/crypto/include -I/usr/local/src/freeswitch/libs/spandsp/src -I/usr/local/src/freeswitch/libs/tiff-4.0.2/libtiff -DENABLE_SRTP -DSWITCH_HAVE_ODBC -I/usr/include -I/usr/local/src/freeswitch/libs/libedit/src -DSWITCH_HAVE_LIBEDIT -I/usr/local/src/freeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -Wall -std=c99 -pedantic -Wdeclaration-after-statement -g -O2 -MT libfreeswitch_la-switch_ivr.lo -MD -MP -MF .deps/libfreeswitch_la-switch_ivr.Tpo -c src/switch_ivr.c -fPIC -DPIC -o .libs/libfreeswitch_la-switch_ivr.o quiet_libtool: compile: gcc -DHAVE_CONFIG_H -I. -I./src/include -I./libs/xmlrpc-c -I/usr/local/src/freeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -pthread -DLINUX=2 -D_REENTRANT -D_GNU_SOURCE -D_LARGEFILE64_SOURCE -I/usr/local/src/freeswitch/libs/apr/include -I/usr/local/src/freeswitch/libs/apr-util/include -I/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib -I/usr/local/src/freeswitch/libs/libtpl-1.5/src -I/usr/local/src/freeswitch/libs/stfu -I/usr/local/src/freeswitch/libs/sqlite -I/usr/local/src/freeswitch/libs/pcre -I/usr/local/src/freeswitch/libs/speex/include -Ilibs/speex/include -I/usr/local/src/freeswitch/libs/srtp/include -I/usr/local/src/freeswitch/libs/srtp/crypto/include -Ilibs/srtp/crypto/include -I/usr/local/src/freeswitch/libs/spandsp/src -I/usr/local/src/freeswitch/libs/tiff-4.0.2/libtiff -DENABLE_SRTP -DSWITCH_HAVE_ODBC -I/usr/include -I/usr/local/src/freeswitch/libs/libedit/src -DSWITCH_HAVE_LIBEDIT -I/usr/local/src/freeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -Wall -std=c99 -pedantic -Wdeclaration-after-statement -g -O2 -MT libfreeswitch_la-switch_ivr.lo -MD -MP -MF .deps/libfreeswitch_la-switch_ivr.Tpo -c src/switch_ivr.c -o libfreeswitch_la-switch_ivr.o >/dev/null 2>&1 ^Cmake[1]: *** [libfreeswitch_la-switch_ivr.lo] Error 1 make: *** [mod_voicemail-install] Interrupt Regards, Archana From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi Marcus Sent: 21 February 2013 09:24 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] make mod_voicemail Try: make mod_voicemail-install -Avi Marcus On Thu, Feb 21, 2013 at 11:10 AM, Archana Venugopan > wrote: Hi, When I try to give 'make mod_voicemail' command, I see its compiling everything as like make command instead of just doing mod_voicemail. Can someone suggest why is this happening and what can be done to just compile mod_voicemail. Many thanks. Regards, Archana _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130226/0c8493ec/attachment-0001.html From mail at loredo.me Tue Feb 26 12:21:03 2013 From: mail at loredo.me (Julian Pawlowski) Date: Tue, 26 Feb 2013 10:21:03 +0100 Subject: [Freeswitch-users] German voice prompts In-Reply-To: References: Message-ID: Hi, On Mon, Feb 25, 2013 at 10:54 PM, Michael Collins wrote: > Thanks for checking in and showing us this tool. You're welcome. > If you have a minute to join us on the Wednesday conference call maybe you > could call in and talk about it. :) > Let me see if I can make it, not sure if I'm available for this at my client's office. Br, Julian -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130226/50848a50/attachment.html From steveayre at gmail.com Tue Feb 26 12:25:00 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 26 Feb 2013 09:25:00 +0000 Subject: [Freeswitch-users] make mod_voicemail In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF23AD7D8@Mail-Kilo.squay.com> References: <592A9CF93E12394E8472A6CC66E66BF23AC9F1@Mail-Kilo.squay.com> <592A9CF93E12394E8472A6CC66E66BF23ACA1B@Mail-Kilo.squay.com> <592A9CF93E12394E8472A6CC66E66BF23AD13E@Mail-Kilo.squay.com> <592A9CF93E12394E8472A6CC66E66BF23AD7D8@Mail-Kilo.squay.com> Message-ID: Yes, there's something up with the linking. Those modules are looking for an API call in libfreeswitch that doesn't appear to be there. Return to your backup copies to get FS to start if you need to (you did make one didn't you?), build all of FS & the modules you want from scratch, then install that. -Steve On 26 February 2013 09:12, Archana Venugopan wrote: > Hi,**** > > Thanks. I tried running make mod_voicemail command. But after running this > command we re-started freeswitch. And it throwed few errors as like below > and freeswitch is not coming up. Please let me know what changed after > running make mod_voicemail command.**** > > Should I need to re-build my FS completely now?**** > > ** ** > > 2013-02-26 08:58:52.872222 [CRIT] switch_loadable_module.c:1310 Error > Loading module /usr/local/freeswitch/mod/mod_cdr_csv.so**** > > **/usr/local/freeswitch/mod/mod_cdr_csv.so: undefined symbol: > switch_channel_expand_variables****** > > 2013-02-26 08:58:52.872607 [CRIT] switch_loadable_module.c:1310 Error > Loading module /usr/local/freeswitch/mod/mod_event_socket.so**** > > **/usr/local/freeswitch/mod/mod_event_socket.so: undefined symbol: > switch_core_session_locate****** > > 2013-02-26 08:58:52.873116 [CRIT] switch_loadable_module.c:1310 Error > Loading module /usr/local/freeswitch/mod/mod_lua.so**** > > **/usr/local/freeswitch/mod/mod_lua.so: undefined symbol: > _ZN13EventConsumerC1EPKcS1_****** > > 2013-02-26 08:58:52.873661 [CRIT] switch_loadable_module.c:1310 Error > Loading module /usr/local/freeswitch/mod/mod_sofia.so**** > > **/usr/local/freeswitch/mod/mod_sofia.so: undefined symbol: > switch_core_session_get_partner****** > > 2013-02-26 08:58:52.874002 [CRIT] switch_loadable_module.c:1310 Error > Loading module /usr/local/freeswitch/mod/mod_loopback.so**** > > **/usr/local/freeswitch/mod/mod_loopback.so: undefined symbol: > switch_core_session_get_partner****** > > 2013-02-26 08:58:52.874354 [CRIT] switch_loadable_module.c:1310 Error > Loading module /usr/local/freeswitch/mod/mod_commands.so**** > > **/usr/local/freeswitch/mod/mod_commands.so: undefined symbol: > switch_core_session_get_partner****** > > 2013-02-26 08:58:52.874731 [CRIT] switch_loadable_module.c:1310 Error > Loading module /usr/local/freeswitch/mod/mod_conference.so**** > > **/usr/local/freeswitch/mod/mod_conference.so: undefined symbol: > switch_channel_expand_variables****** > > ** ** > > Regards,**** > > Archana**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Steven Ayre > *Sent:* 23 February 2013 14:37 > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] make mod_voicemail**** > > ** ** > > Building will only affect files within the source directory - not your > install paths (/usr/local/freeswitch).**** > > ** ** > > Make sure you build mod_voicemail using the same version of FS as you're > running though - otherwise there may be differences in the core API between > the version being used by freeswitch and by the module.**** > > ** ** > > -Steve**** > > ** ** > > ** ** > > On 23 February 2013 13:35, Archana Venugopan > wrote:**** > > If its building the core whether it would affect the scripts inside > /usr/local/freeswitch folder as well? If yes then all the scripts inside > would be lost. Please let me know.**** > > **** > > Regards,**** > > Archana**** > > **** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Steven Ayre > *Sent:* 23 February 2013 13:06**** > > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] make mod_voicemail**** > > **** > > This shows that it's building the FS core (src/switch_ivr.c is part of > that).**** > > **** > > That means that file hadn't been built or the build wasn't up-to-date. > Building the module without the core is not possible - the below is > expected behaviour. Once the core and module are built, the -install will > only install the module not the rest.**** > > **** > > -Steve**** > > **** > > **** > > On 21 February 2013 09:30, Archana Venugopan > wrote:**** > > I am afraid if that is as well installing everything so I just stopped it. > **** > > **** > > [root at sme-xswitch freeswitch]# make mod_voicemail-install**** > > quiet_libtool: compile: gcc -DHAVE_CONFIG_H -I. -I./src/include > -I./libs/xmlrpc-c -I/usr/local/src/freeswitch/libs/curl/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden > -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -pthread -DLINUX=2 > -D_REENTRANT -D_GNU_SOURCE -D_LARGEFILE64_SOURCE > -I/usr/local/src/freeswitch/libs/apr/include > -I/usr/local/src/freeswitch/libs/apr-util/include > -I/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib > -I/usr/local/src/freeswitch/libs/libtpl-1.5/src > -I/usr/local/src/freeswitch/libs/stfu > -I/usr/local/src/freeswitch/libs/sqlite > -I/usr/local/src/freeswitch/libs/pcre > -I/usr/local/src/freeswitch/libs/speex/include -Ilibs/speex/include > -I/usr/local/src/freeswitch/libs/srtp/include > -I/usr/local/src/freeswitch/libs/srtp/crypto/include > -Ilibs/srtp/crypto/include -I/usr/local/src/freeswitch/libs/spandsp/src > -I/usr/local/src/freeswitch/libs/tiff-4.0.2/libtiff -DENABLE_SRTP > -DSWITCH_HAVE_ODBC -I/usr/include > -I/usr/local/src/freeswitch/libs/libedit/src -DSWITCH_HAVE_LIBEDIT > -I/usr/local/src/freeswitch/libs/curl/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -Werror > -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb > -DHAVE_OPENSSL -Wall -std=c99 -pedantic -Wdeclaration-after-statement -g > -O2 -MT libfreeswitch_la-switch_ivr.lo -MD -MP -MF > .deps/libfreeswitch_la-switch_ivr.Tpo -c src/switch_ivr.c -fPIC -DPIC -o > .libs/libfreeswitch_la-switch_ivr.o**** > > quiet_libtool: compile: gcc -DHAVE_CONFIG_H -I. -I./src/include > -I./libs/xmlrpc-c -I/usr/local/src/freeswitch/libs/curl/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden > -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -pthread -DLINUX=2 > -D_REENTRANT -D_GNU_SOURCE -D_LARGEFILE64_SOURCE > -I/usr/local/src/freeswitch/libs/apr/include > -I/usr/local/src/freeswitch/libs/apr-util/include > -I/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib > -I/usr/local/src/freeswitch/libs/libtpl-1.5/src > -I/usr/local/src/freeswitch/libs/stfu > -I/usr/local/src/freeswitch/libs/sqlite > -I/usr/local/src/freeswitch/libs/pcre > -I/usr/local/src/freeswitch/libs/speex/include -Ilibs/speex/include > -I/usr/local/src/freeswitch/libs/srtp/include > -I/usr/local/src/freeswitch/libs/srtp/crypto/include > -Ilibs/srtp/crypto/include -I/usr/local/src/freeswitch/libs/spandsp/src > -I/usr/local/src/freeswitch/libs/tiff-4.0.2/libtiff -DENABLE_SRTP > -DSWITCH_HAVE_ODBC -I/usr/include > -I/usr/local/src/freeswitch/libs/libedit/src -DSWITCH_HAVE_LIBEDIT > -I/usr/local/src/freeswitch/libs/curl/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -Werror > -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb > -DHAVE_OPENSSL -Wall -std=c99 -pedantic -Wdeclaration-after-statement -g > -O2 -MT libfreeswitch_la-switch_ivr.lo -MD -MP -MF > .deps/libfreeswitch_la-switch_ivr.Tpo -c src/switch_ivr.c -o > libfreeswitch_la-switch_ivr.o >/dev/null 2>&1**** > > ^Cmake[1]: *** [libfreeswitch_la-switch_ivr.lo] Error 1**** > > make: *** [mod_voicemail-install] Interrupt**** > > **** > > Regards,**** > > Archana**** > > **** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Avi Marcus > *Sent:* 21 February 2013 09:24 > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] make mod_voicemail**** > > **** > > Try:**** > > > > **** > > > > **** > > make mod_voicemail-install**** > > > **** > > -Avi Marcus**** > > **** > > On Thu, Feb 21, 2013 at 11:10 AM, Archana Venugopan < > a.venugopan at mundio.com> wrote:**** > > Hi,**** > > **** > > When I try to give ?make mod_voicemail? command, I see its compiling > everything as like make command instead of just doing mod_voicemail. **** > > Can someone suggest why is this happening and what can be done to just > compile mod_voicemail. Many thanks.**** > > **** > > Regards,**** > > Archana**** > > **** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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**** > > **** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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**** > > **** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130226/7774785d/attachment-0001.html From huw.selley at netdev.co.uk Tue Feb 26 12:28:06 2013 From: huw.selley at netdev.co.uk (Huw Selley) Date: Tue, 26 Feb 2013 09:28:06 +0000 Subject: [Freeswitch-users] [Freeswitch-dev] FreeSWITCH Weekly News and Notes In-Reply-To: References: Message-ID: On 25 Feb 2013, at 18:29, Michael Collins wrote: > Greetings! > > First item in the news today is a happy report from long-time FreeSWITCH user Henry Gavin. Henry runs a company in the U.K. called SureVoIP. He is pleased to report that "thanks to FreeSWITCH and FusionPBX" his company is once again a finalist for the annual ITSPA awards. Congrats to Henry for leveraging FreeSWITCH in a successful business endeavor. Don't you mean Gavin Henry? ;) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130226/431ed525/attachment.html From venkateshwaran54 at gmail.com Tue Feb 26 12:51:57 2013 From: venkateshwaran54 at gmail.com (Venkateshwaran Thirugnanam) Date: Tue, 26 Feb 2013 15:21:57 +0530 Subject: [Freeswitch-users] freeswitch--Automatic Startup At Boot In-Reply-To: <512BAB5C.7090104@anew.com.ve> References: <512BAB5C.7090104@anew.com.ve> Message-ID: Hello, Thanks for the update Regards, Kumaran T On Mon, Feb 25, 2013 at 11:50 PM, Saugort Dario Garcia Tovar < dgarcia at anew.com.ve> wrote: > For Auto start at boot > > cp /usr/src/freeswitch.git/build/freeswitch.init.redhat > /etc/init.d/freeswitch > > Edit /etc/init.d/freeswitch like below > > PROG_NAME=freeswitch > PID_FILE=${PID_FILE-/usr/local/freeswitch/log/freeswitch.pid} > FS_USER=${FS_USER-freeswitch} > FS_FILE=${FS_FILE-/usr/local/freeswitch/bin/freeswitch} > FS_HOME=${FS_HOME-/usr/local/freeswitch} > LOCK_FILE=/var/lock/subsys/freeswitch > FREESWITCH_ARGS="-nc" > RETVAL=0 > > chmod 755 /etc/init.d/freeswitch > > chkconfig --add freeswitch > > chkconfig --level 2345 freeswitch on > > * > * > > > On 2/25/2013 1:15 PM, Venkateshwaran Thirugnanam wrote: > > Hi All, > How to make freeswitch automatic startup on centos 6.3...I followed doc > but I think I'm missing so its not starting after reboot....Pls guide me.. > > Regards, > Kumaran T > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.2238 / Virus Database: 2641/5632 - Release Date: 02/25/13 > > > > -- > Atentamente, > *Dario Garc?a* > Consultor. > > CCCT, Nivel C2, Sector Yarey, Mz, > Ofc. MZ03a. > Caracas-Venezuela. > Tel?fono: +58 212 9081842 > Cel: +58 412 2221515 > dgarcia at anew.com.ve > http://www.anew.com.ve > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130226/06148328/attachment.html From alex at digitalmail.com Tue Feb 26 12:59:06 2013 From: alex at digitalmail.com (Alex Lake) Date: Tue, 26 Feb 2013 09:59:06 +0000 Subject: [Freeswitch-users] Getting Freeswitch 1.2.0 (git-771819c 2012-05-25 04-46-28 +0000) In-Reply-To: References: Message-ID: <512C876A.2070702@digitalmail.com> Ah, OK, well we've got a couple of those in production and I just wanted another of the same. I get the impression there have been significant changes (maybe even technically bug-fixes) since then which might break our service. Not quite sure what our policy should be with respect to upgrades. For OS, we generally use LTS versions only. I wonder if there's an equivalent with FreeSwitch? Thanks for the tip, though! From mehroz.ashraf85 at gmail.com Tue Feb 26 13:39:17 2013 From: mehroz.ashraf85 at gmail.com (mehroz) Date: Tue, 26 Feb 2013 02:39:17 -0800 (PST) Subject: [Freeswitch-users] FS with SSL/TLS issues! In-Reply-To: <113D484C-EA7E-4611-B958-6514F5F2B5B9@freeswitch.org> References: <1361367187947-7587736.post@n2.nabble.com> <1361513236022-7587844.post@n2.nabble.com> <1361519970790-7587849.post@n2.nabble.com> <113D484C-EA7E-4611-B958-6514F5F2B5B9@freeswitch.org> Message-ID: <1361875157474-7588012.post@n2.nabble.com> Alright, Thanks for inputs! I can successfully Register with FS on TLS now. The issue was not the CA root certificate nor domains requirement or Signed certificate, but it was because client certificate were not properly installed. (For those , trying on 3CX , you have to rename "cafile.pem" to "root_cert_3CXPHONE.pem"). Now, the problem is internal calling! and i guess it is same as Limitations of the Freeswitch TLS/SSLv23 Implementation (FS-3877) . I have been looking for the solution but coudnt get that yet! I can make call from A but the called party B does not rang, but then i can call B to A and it rings (strange, as both are in same network , same NAT). Can one can guide me step wise what to do now ! ? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FS-with-SSL-TLS-issues-tp7587736p7588012.html Sent from the freeswitch-users mailing list archive at Nabble.com. From steveayre at gmail.com Tue Feb 26 14:33:28 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 26 Feb 2013 11:33:28 +0000 Subject: [Freeswitch-users] Getting Freeswitch 1.2.0 (git-771819c 2012-05-25 04-46-28 +0000) In-Reply-To: <512C876A.2070702@digitalmail.com> References: <512C876A.2070702@digitalmail.com> Message-ID: Obviously test before using in production, but there shouldn't be anything that causes you any significant problems. AFAIK the only ones that're likely to cause you issues are: - if you're using odbc-dsn from 1.2.4 you should prefix the DSN with odbc:// - mod_nibblebill's db-dsn param is now called odbc-dsn The v1.2.stable branch is the closest to LTS at the moment... there're some patches pending in master that'll get merged into 1.2 soon, at that time I believe that master is going to become the development branch for 1.4 and 1.2 will become even more static. -Steve On 26 February 2013 09:59, Alex Lake wrote: > Ah, OK, well we've got a couple of those in production and I just > wanted another of the same. I get the impression there have been > significant changes (maybe even technically bug-fixes) since then which > might break our service. > > Not quite sure what our policy should be with respect to upgrades. For > OS, we generally use LTS versions only. I wonder if there's an > equivalent with FreeSwitch? > > Thanks for the tip, though! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130226/e4c716eb/attachment.html From alex at digitalmail.com Tue Feb 26 14:44:45 2013 From: alex at digitalmail.com (Alex Lake) Date: Tue, 26 Feb 2013 11:44:45 +0000 Subject: [Freeswitch-users] Getting Freeswitch 1.2.0 (git-771819c 2012-05-25 04-46-28 +0000) In-Reply-To: References: <512C876A.2070702@digitalmail.com> Message-ID: <512CA02D.5020101@digitalmail.com> Thanks for that. This new nested conditions thing doesn't change the behaviour of classic stacked conditions, does it? > Obviously test before using in production, but there shouldn't be > anything that causes you any significant problems. > > AFAIK the only ones that're likely to cause you issues are: > - if you're using odbc-dsn from 1.2.4 you should prefix the DSN with > odbc:// > - mod_nibblebill's db-dsn param is now called odbc-dsn > > The v1.2.stable branch is the closest to LTS at the moment... there're > some patches pending in master that'll get merged into 1.2 soon, at > that time I believe that master is going to become the development > branch for 1.4 and 1.2 will become even more static. > > -Steve > > > > On 26 February 2013 09:59, Alex Lake > wrote: > > Ah, OK, well we've got a couple of those in production and I just > wanted another of the same. I get the impression there have been > significant changes (maybe even technically bug-fixes) since then > which > might break our service. > > Not quite sure what our policy should be with respect to upgrades. For > OS, we generally use LTS versions only. I wonder if there's an > equivalent with FreeSwitch? > > Thanks for the tip, though! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.2238 / Virus Database: 2641/5633 - Release Date: 02/25/13 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130226/a326f9ce/attachment-0001.html From avi at avimarcus.net Tue Feb 26 15:33:59 2013 From: avi at avimarcus.net (Avi Marcus) Date: Tue, 26 Feb 2013 14:33:59 +0200 Subject: [Freeswitch-users] Getting Freeswitch 1.2.0 (git-771819c 2012-05-25 04-46-28 +0000) In-Reply-To: <512CA02D.5020101@digitalmail.com> References: <512C876A.2070702@digitalmail.com> <512CA02D.5020101@digitalmail.com> Message-ID: No it doesn't. We are trying to keep a list of breaking changes from a link on the wiki main page, called release notes. http://wiki.freeswitch.org/wiki/Release_Notes -Avi On Tue, Feb 26, 2013 at 1:44 PM, Alex Lake wrote: > Thanks for that. > This new nested conditions thing doesn't change the behaviour of classic > stacked conditions, does it? > > Obviously test before using in production, but there shouldn't be anything > that causes you any significant problems. > > AFAIK the only ones that're likely to cause you issues are: > - if you're using odbc-dsn from 1.2.4 you should prefix the DSN with > odbc:// > - mod_nibblebill's db-dsn param is now called odbc-dsn > > The v1.2.stable branch is the closest to LTS at the moment... there're > some patches pending in master that'll get merged into 1.2 soon, at that > time I believe that master is going to become the development branch for > 1.4 and 1.2 will become even more static. > > -Steve > > > > On 26 February 2013 09:59, Alex Lake wrote: > >> Ah, OK, well we've got a couple of those in production and I just >> wanted another of the same. I get the impression there have been >> significant changes (maybe even technically bug-fixes) since then which >> might break our service. >> >> Not quite sure what our policy should be with respect to upgrades. For >> OS, we generally use LTS versions only. I wonder if there's an >> equivalent with FreeSwitch? >> >> Thanks for the tip, though! >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.2238 / Virus Database: 2641/5633 - Release Date: 02/25/13 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130226/d60abee9/attachment.html From flavio at sippulse.com Tue Feb 26 13:02:31 2013 From: flavio at sippulse.com (Flavio Goncalves) Date: Tue, 26 Feb 2013 07:02:31 -0300 Subject: [Freeswitch-users] Possible BUG in G729 negotiation In-Reply-To: References: Message-ID: Hi Brian, Thanks for the reply, I will try the solution, The Cisco gateway sends the CN (Comfort Noise) in payload 19, it is usual. Best regards, Flavio E. Goncalves CTO - SipPulse Routing and Billing Solutions for SIP 2013/2/25 Brian West > Try setting this variable: > > sip_append_audio_sdp=a=fmtp:18 annexb=no > > Can you verify that your provider is actually sending CN on 19 and NOT 13. > There are some Sonus switches that say 19 but mean 13 and send on 13. > 19=13 in hex. > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH_Wire > T: +1.918.420.9266 | F: +1.918.420.9267 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9266 > ISN: 410*543 > > > > > > > On Feb 25, 2013, at 8:08 AM, Flavio Goncalves wrote: > > > Hi, > > > > I have searched the list, but I couldn't find a final answer for this > case. I'm receiving calls from a provider using G729 annex B. > > > > INVITE > > a=rtpmap:18 G729/8000. > > a=fmtp:18 annexb=yes. > > a=rtpmap:0 PCMU/8000. > > a=rtpmap:8 PCMA/8000. > > a=rtpmap:101 telephone-event/8000. > > a=fmtp:101 0-16. > > a=rtpmap:19 CN/8000. > > > > and FS is answering > > > > 200OK > > m=audio 31388 RTP/AVP 18 101 19. > > a=rtpmap:18 G729/8000. > > a=fmtp:18 annexb=yes. > > a=rtpmap:101 telephone-event/8000. > > a=fmtp:101 0-16. > > a=rtpmap:19 CN/8000. > > a=ptime:20. > > > > If freeswitch supports only G729A, shouldn't FS refuse the call with > 488 or suppress the line annexb=yes? > > > > The audio quality is really bad when I receive calls in this way. After > removing the annexb on the gateway the audio is fine, but unfortunately I > can't remove from it from all VoIP providers terminating calls. For > outbound calls I simply removed the annexb, but for inbound there is no way > to do it. > > > > Flavio > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130226/e2aec24b/attachment-0001.html From Joep.Vrienten at damovo.com Tue Feb 26 15:25:15 2013 From: Joep.Vrienten at damovo.com (Joep Vrienten) Date: Tue, 26 Feb 2013 13:25:15 +0100 Subject: [Freeswitch-users] hangup a call if 181 Call is being forwarded is received Message-ID: Hi, I make an outbound call to a provider (actually a PBX). Is it possible to hang up the call if I receive "181 Call is being forwarded" from the provider? I would like to prevent that the call is forwarded to a voicemail system in the PBX. Thanks for your help Joep -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130226/dd67a618/attachment-0001.html From venkateshwaran54 at gmail.com Tue Feb 26 17:18:57 2013 From: venkateshwaran54 at gmail.com (Venkateshwaran Thirugnanam) Date: Tue, 26 Feb 2013 19:48:57 +0530 Subject: [Freeswitch-users] user and extension creation Message-ID: Hi All, I had installed freeswitch,I need to create a user and extension to test the connection,Please guild me to create users and extension for them to make a call to their extension... Regards, Kumaran T -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130226/c6428458/attachment.html From talk2ram at gmail.com Tue Feb 26 17:30:59 2013 From: talk2ram at gmail.com (ram) Date: Tue, 26 Feb 2013 20:00:59 +0530 Subject: [Freeswitch-users] user and extension creation In-Reply-To: References: Message-ID: how about this http://wiki.freeswitch.org/wiki/Getting_Started_Guide On Tue, Feb 26, 2013 at 7:48 PM, Venkateshwaran Thirugnanam < venkateshwaran54 at gmail.com> wrote: > Hi All, > I had installed freeswitch,I need to create a user and extension to test > the connection,Please guild me to create users and extension for them to > make a call to their extension... > > Regards, > Kumaran T > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130226/eae574cc/attachment.html From royj at yandex.ru Tue Feb 26 18:26:14 2013 From: royj at yandex.ru (royj) Date: Tue, 26 Feb 2013 19:26:14 +0400 Subject: [Freeswitch-users] t38 200OK timeout Message-ID: <20130226192614.c8767f514b9168395c611121@yandex.ru> I wonder if anyone faced the issue that posted on jira - http://jira.freeswitch.org/browse/FS-5120 I understand employment of developers and hope Anthony or somebody else will give the time soon, but may be here would be any thoughts. -- regards royj From willbelair at yahoo.com Tue Feb 26 18:33:28 2013 From: willbelair at yahoo.com (Will Smith) Date: Tue, 26 Feb 2013 07:33:28 -0800 (PST) Subject: [Freeswitch-users] Will Smith Message-ID: <1361892808.13628.YahooMailNeo@web122101.mail.ne1.yahoo.com>    http://www.locksmithhounslow.com/ifggk/dgt.89lodcr6za?id7z       Will Smith2/26/2013 4:33:26 PM -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130226/b47c37b6/attachment.html From lloyd.aloysius at gmail.com Tue Feb 26 18:33:50 2013 From: lloyd.aloysius at gmail.com (Lloyd Aloysius) Date: Tue, 26 Feb 2013 10:33:50 -0500 Subject: [Freeswitch-users] mod_callcenter - xml curl Message-ID: Hi All Can I provide the configuration for mod_ callcenter through xml_curl? Thanks Lloyd -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130226/74641280/attachment.html From tculjaga at gmail.com Tue Feb 26 18:42:01 2013 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Tue, 26 Feb 2013 16:42:01 +0100 Subject: [Freeswitch-users] Fax document number of pages Message-ID: hello, i know its stupid, but please, is there a way to somehow print the number of pages on a document in a / format instead of p. ? i cannot find this in mod_spandsp so i assume its somewhere in spandsp library... any idea where ? regards, Tihomir. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130226/1afa191c/attachment.html From regis.freeswitch.org at tornad.net Tue Feb 26 18:43:48 2013 From: regis.freeswitch.org at tornad.net (Regis M) Date: Tue, 26 Feb 2013 16:43:48 +0100 Subject: [Freeswitch-users] mod_callcenter - xml curl In-Reply-To: References: Message-ID: yes 2013/2/26 Lloyd Aloysius > Hi All > > Can I provide the configuration for mod_ callcenter through xml_curl? > > Thanks > Lloyd > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130226/319492b0/attachment.html From vipkilla at gmail.com Tue Feb 26 18:54:02 2013 From: vipkilla at gmail.com (Vik Killa) Date: Tue, 26 Feb 2013 10:54:02 -0500 Subject: [Freeswitch-users] mod_callcenter - xml curl In-Reply-To: References: Message-ID: You can provide the configuration for almost everything with XML_CURL. I think freeswitch.xml is the only REQUIRED static XML file. In addition, there are certain parameters required in freeswitch.xml (such as the contents of switch.conf.xml) On Tue, Feb 26, 2013 at 10:43 AM, Regis M wrote: > yes > > From andrew at cassidywebservices.co.uk Tue Feb 26 18:56:49 2013 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Tue, 26 Feb 2013 15:56:49 +0000 Subject: [Freeswitch-users] mod_callcenter - xml curl In-Reply-To: References: Message-ID: Yes but as it only loads it on module load (or when explicitly requested) I only store queue configurations in there and modify tiers and agents directly in the database. On 26 February 2013 15:43, Regis M wrote: > yes > > > 2013/2/26 Lloyd Aloysius > >> Hi All >> >> Can I provide the configuration for mod_ callcenter through xml_curl? >> >> Thanks >> Lloyd >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130226/99e6f573/attachment-0001.html From bdfoster at endigotech.com Tue Feb 26 19:06:39 2013 From: bdfoster at endigotech.com (Brian Foster) Date: Tue, 26 Feb 2013 11:06:39 -0500 Subject: [Freeswitch-users] Fax document number of pages In-Reply-To: References: Message-ID: <7529AF3C-01EB-42D8-A245-E373BC8B577D@endigotech.com> http://wiki.freeswitch.org/wiki/Mod_spandsp#Checking_the_results Specifically: fax_document_transferred_pages fax_document_total_pages I don't know the difference between the two, I personally use the former. Welcome to the FreeSWITCH Community! -BDF Sent from my iPhone On Feb 26, 2013, at 10:42 AM, Tihomir Culjaga wrote: > hello, > i know its stupid, but please, is there a way to somehow print the number of pages on a document in a / format instead of p. ? > > i cannot find this in mod_spandsp so i assume its somewhere in spandsp library... any idea where ? > > > regards, > Tihomir. > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130226/c2b1cbdf/attachment.html From vipkilla at gmail.com Tue Feb 26 19:08:45 2013 From: vipkilla at gmail.com (Vik Killa) Date: Tue, 26 Feb 2013 11:08:45 -0500 Subject: [Freeswitch-users] mod_callcenter - xml curl In-Reply-To: References: Message-ID: It's probably not good practice to modify the database directly. The API is there to interface with the database. It's probably best to have the XML the same as the configuration so when the module loads it's in sync, and if the XML changes, use the API to change the configuration (instead of reloading the module). On Tue, Feb 26, 2013 at 10:56 AM, Andrew Cassidy < andrew at cassidywebservices.co.uk> wrote: > Yes but as it only loads it on module load (or when explicitly requested) > I only store queue configurations in there and modify tiers and agents > directly in the database. > > > On 26 February 2013 15:43, Regis M wrote: > >> yes >> >> >> 2013/2/26 Lloyd Aloysius >> >>> Hi All >>> >>> Can I provide the configuration for mod_ callcenter through xml_curl? >>> >>> Thanks >>> Lloyd >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > > -- > *Andrew Cassidy BSc (Hons) MBCS SSCA* > Managing Director > > > *T *03300 100 960 *F > *03300 100 961 > *E *andrew at cassidywebservices.co.uk > *W *www.cassidywebservices.co.uk > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130226/a5f92e6d/attachment.html From bdfoster at endigotech.com Tue Feb 26 19:10:23 2013 From: bdfoster at endigotech.com (Brian Foster) Date: Tue, 26 Feb 2013 11:10:23 -0500 Subject: [Freeswitch-users] Fax document number of pages In-Reply-To: References: Message-ID: <196B5F02-02C4-4A5A-8795-492412B95E72@endigotech.com> Just realized I didn't answer your question, disregard my answer! Sent from my iPhone On Feb 26, 2013, at 10:42 AM, Tihomir Culjaga wrote: > hello, > i know its stupid, but please, is there a way to somehow print the number of pages on a document in a / format instead of p. ? > > i cannot find this in mod_spandsp so i assume its somewhere in spandsp library... any idea where ? > > > regards, > Tihomir. > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 msc at freeswitch.org Tue Feb 26 19:28:48 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 26 Feb 2013 08:28:48 -0800 Subject: [Freeswitch-users] [Freeswitch-dev] FreeSWITCH Weekly News and Notes In-Reply-To: References: Message-ID: Yes, that's exactly what I meant! See what happens when you get your endianness wrong? ;) -MC On Tue, Feb 26, 2013 at 1:28 AM, Huw Selley wrote: > > On 25 Feb 2013, at 18:29, Michael Collins wrote: > > Greetings! > > First item in the news today is a happy report from long-time FreeSWITCH > user Henry Gavin. Henry runs a company in the U.K. called SureVoIP. > He is pleased to report that "thanks to FreeSWITCH and FusionPBX" his > company is once again a finalist for the annual ITSPA awards. > Congrats to Henry for leveraging FreeSWITCH in a successful business > endeavor. > > > Don't you mean Gavin Henry? ;) > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130226/8ee7be50/attachment-0001.html From igor.cappello at edistar.com Tue Feb 26 18:13:45 2013 From: igor.cappello at edistar.com (Igor Cappello) Date: Tue, 26 Feb 2013 16:13:45 +0100 (CET) Subject: [Freeswitch-users] Problem setting up bind_digit_action Message-ID: <1506773689.16712.1361891625264.JavaMail.root@mailserver.edistar.com> We are experiencing some problems with bind_digit_action. We are trying to get a notification of the digits pressed by the bleg of a bridged call. Our simple test dialplan is the following: --> We can use only inband DTMF. Here you can find the log of a call (IPs are obscured for security) http://pastebin.com/j8pBFVDL Any hint on why the extension MYLOG is not executed? Maybe we are missing something obvious.. Thanks Igor Cappello From flavio at sippulse.com Tue Feb 26 19:52:48 2013 From: flavio at sippulse.com (Flavio Goncalves) Date: Tue, 26 Feb 2013 13:52:48 -0300 Subject: [Freeswitch-users] Possible BUG in G729 negotiation In-Reply-To: References: Message-ID: Hi Brian, This setting was already set. For outbound calls it works just fine. For inbound calls, I can't disable the annexb. I need to reply (200 Ok) with annexb=no. Best regards, Flavio E. Goncalves CTO - SipPulse Routing and Billing Solutions for SIP 2013/2/25 Brian West > Try setting this variable: > > sip_append_audio_sdp=a=fmtp:18 annexb=no > > Can you verify that your provider is actually sending CN on 19 and NOT 13. > There are some Sonus switches that say 19 but mean 13 and send on 13. > 19=13 in hex. > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH_Wire > T: +1.918.420.9266 | F: +1.918.420.9267 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9266 > ISN: 410*543 > > > > > > > On Feb 25, 2013, at 8:08 AM, Flavio Goncalves wrote: > > > Hi, > > > > I have searched the list, but I couldn't find a final answer for this > case. I'm receiving calls from a provider using G729 annex B. > > > > INVITE > > a=rtpmap:18 G729/8000. > > a=fmtp:18 annexb=yes. > > a=rtpmap:0 PCMU/8000. > > a=rtpmap:8 PCMA/8000. > > a=rtpmap:101 telephone-event/8000. > > a=fmtp:101 0-16. > > a=rtpmap:19 CN/8000. > > > > and FS is answering > > > > 200OK > > m=audio 31388 RTP/AVP 18 101 19. > > a=rtpmap:18 G729/8000. > > a=fmtp:18 annexb=yes. > > a=rtpmap:101 telephone-event/8000. > > a=fmtp:101 0-16. > > a=rtpmap:19 CN/8000. > > a=ptime:20. > > > > If freeswitch supports only G729A, shouldn't FS refuse the call with > 488 or suppress the line annexb=yes? > > > > The audio quality is really bad when I receive calls in this way. After > removing the annexb on the gateway the audio is fine, but unfortunately I > can't remove from it from all VoIP providers terminating calls. For > outbound calls I simply removed the annexb, but for inbound there is no way > to do it. > > > > Flavio > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130226/e0f33d6b/attachment.html From venkateshwaran54 at gmail.com Tue Feb 26 20:17:23 2013 From: venkateshwaran54 at gmail.com (Venkateshwaran Thirugnanam) Date: Tue, 26 Feb 2013 22:47:23 +0530 Subject: [Freeswitch-users] user and extension creation In-Reply-To: References: Message-ID: Hi Ram, I checked it, but I'm very new to it,so I can't add a user and extension to it ... Thanks Kumaran T On Tue, Feb 26, 2013 at 8:00 PM, ram wrote: > how about this > > http://wiki.freeswitch.org/wiki/Getting_Started_Guide > > > > > On Tue, Feb 26, 2013 at 7:48 PM, Venkateshwaran Thirugnanam < > venkateshwaran54 at gmail.com> wrote: > >> Hi All, >> I had installed freeswitch,I need to create a user and extension to >> test the connection,Please guild me to create users and extension for them >> to make a call to their extension... >> >> Regards, >> Kumaran T >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130226/d2a218dc/attachment.html From msc at freeswitch.org Tue Feb 26 20:28:26 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 26 Feb 2013 09:28:26 -0800 Subject: [Freeswitch-users] Logical AND + NOT In-Reply-To: <1361865683.776896435@f250.mail.ru> References: <1361865683.776896435@f250.mail.ru> Message-ID: try that and report back... -MC On Tue, Feb 26, 2013 at 12:01 AM, Dmitriy Shumaev wrote: > Hi > > I want to record all sessions, except calls to group numbers and to IVR. > So I need something like: > > > > > > > > > > > data="$${base_dir}/recordings/archive/${strftime(%Y-%m-%d-%H-%M-%S)}_${caller_id_name}_${caller_id_number}_2_${destination_number}.mp3"/> > > > > . But the syntax of condition does not allow the operator "not equal" or > "!=". What should I do? > > > With best regards, Shumaev DA, KBR Ltd. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130226/39ee3b01/attachment-0001.html From msc at freeswitch.org Tue Feb 26 20:42:37 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 26 Feb 2013 09:42:37 -0800 Subject: [Freeswitch-users] Problem setting up bind_digit_action In-Reply-To: <1506773689.16712.1361891625264.JavaMail.root@mailserver.edistar.com> References: <1506773689.16712.1361891625264.JavaMail.root@mailserver.edistar.com> Message-ID: You need to set the digit action realm: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_digit_action_set_realm -MC On Tue, Feb 26, 2013 at 7:13 AM, Igor Cappello wrote: > We are experiencing some problems with bind_digit_action. We are trying to > get a notification of the digits pressed by the bleg of a bridged call. > > Our simple test dialplan is the following: > > > > > > > > > data="myrealm,*12,exec:execute_extension,MYLOG"/> > > > > > > > > --> > data="execute_on_answer_2=execute_extension MYSETUP"/> > > > > > > > > We can use only inband DTMF. > > Here you can find the log of a call (IPs are obscured for security) > > http://pastebin.com/j8pBFVDL > > Any hint on why the extension MYLOG is not executed? Maybe we are missing > something obvious.. > > Thanks > Igor Cappello > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130226/553cc229/attachment.html From vipkilla at gmail.com Tue Feb 26 20:58:32 2013 From: vipkilla at gmail.com (Vik Killa) Date: Tue, 26 Feb 2013 12:58:32 -0500 Subject: [Freeswitch-users] Problem setting up bind_digit_action In-Reply-To: References: <1506773689.16712.1361891625264.JavaMail.root@mailserver.edistar.com> Message-ID: Please use http://pastebin.freeswitch.org/ On Tue, Feb 26, 2013 at 12:42 PM, Michael Collins wrote: > You need to set the digit action realm: > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_digit_action_set_realm > > -MC > From msc at freeswitch.org Tue Feb 26 21:02:24 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 26 Feb 2013 10:02:24 -0800 Subject: [Freeswitch-users] user and extension creation In-Reply-To: References: Message-ID: On Tue, Feb 26, 2013 at 9:17 AM, Venkateshwaran Thirugnanam < venkateshwaran54 at gmail.com> wrote: > Hi Ram, > I checked it, but I'm very new to it,so I can't add a user and extension > to it ... > http://www.linuxpromagazine.com/Issues/2009/106/FreeSWITCH -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130226/f64134b7/attachment.html From schoch+freeswitch.org at xwin32.com Tue Feb 26 21:06:27 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Tue, 26 Feb 2013 10:06:27 -0800 Subject: [Freeswitch-users] new site, cant call OUT! In-Reply-To: <08f401ce0988$21b97e90$652c7bb0$@bizfocused.com> References: <076e01ce0961$c14511d0$43cf3570$@bizfocused.com> <08f401ce0988$21b97e90$652c7bb0$@bizfocused.com> Message-ID: On Tue, Feb 12, 2013 at 5:19 PM, Sean Devoy wrote: > Then I noticed the Cisco (which is in Gateway mode) has a WAN address of > 10.1.10.10!!! I poked around and found the Comcast Business Router on the > other side of the Cisco. Yippee DOUBLE NAT. I chose to just say ?NO.? > We just got Comcast business class here. We have static addresses assigned to our account, but the Comcast router/modem does both NAT and regular routing at the same time. It's DHCP server will serve addresses from 10.1.10.10 to 10.1.10.199. That's the Comcast fail safe - if you "forget" to configure your interface to one of the static addresses they give you, and leave it on DHCP, then it will still work. (At least until you try to use VoIP!) -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130226/96239d46/attachment.html From alex at digitalmail.com Tue Feb 26 21:00:36 2013 From: alex at digitalmail.com (Alex Lake) Date: Tue, 26 Feb 2013 18:00:36 +0000 Subject: [Freeswitch-users] Silence suppression again Message-ID: <512CF844.5070509@digitalmail.com> Does Freeswitch enable any kind of silence suppression by default? If I have bridge_generate_comfort_noise=true, does that only have an effect when the audio-generating end decides to stop sending RTP? Any other tips for diagnosing silence (it's kind of like a slow noise gate effect)? From mike at jerris.com Tue Feb 26 21:17:15 2013 From: mike at jerris.com (Michael Jerris) Date: Tue, 26 Feb 2013 13:17:15 -0500 Subject: [Freeswitch-users] user and extension creation In-Reply-To: References: Message-ID: <66BDC7F8-FE91-4A90-ABAE-FEC0735E3DCB@jerris.com> http://wiki.freeswitch.org/wiki/Getting_Started_Guide#User_Configuration see Config_1000.xml for a simple (1000.xml) configuration of a user (endpoint) which will register to freeswitch. http://wiki.freeswitch.org/wiki/Config_1000.xml On Feb 26, 2013, at 12:17 PM, Venkateshwaran Thirugnanam wrote: > Hi Ram, > I checked it, but I'm very new to it,so I can't add a user and extension to it ... > > Thanks > Kumaran T > > On Tue, Feb 26, 2013 at 8:00 PM, ram wrote: > how about this > > http://wiki.freeswitch.org/wiki/Getting_Started_Guide > > > > > On Tue, Feb 26, 2013 at 7:48 PM, Venkateshwaran Thirugnanam wrote: > Hi All, > I had installed freeswitch,I need to create a user and extension to test the connection,Please guild me to create users and extension for them to make a call to their extension... > > Regards, > Kumaran T -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130226/40135997/attachment-0001.html From brian at freeswitch.org Tue Feb 26 21:43:45 2013 From: brian at freeswitch.org (Brian West) Date: Tue, 26 Feb 2013 12:43:45 -0600 Subject: [Freeswitch-users] Possible BUG in G729 negotiation In-Reply-To: References: Message-ID: <0317A212-A71A-418E-B3B2-5DED7C8F9EFE@freeswitch.org> Inbound late negotiation and set the variable prior to answer. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire T: +1.918.420.9266 | F: +1.918.420.9267 | M: +1.918.424.WEST iNUM: +883 5100 1420 9266 ISN: 410*543 On Feb 26, 2013, at 10:52 AM, Flavio Goncalves wrote: > Hi Brian, > > This setting was already set. For outbound calls it works just fine. For inbound calls, I can't disable the annexb. I need to reply (200 Ok) with annexb=no. > > Best regards, > > Flavio E. Goncalves > CTO - SipPulse Routing and Billing Solutions for SIP > > 2013/2/25 Brian West > Try setting this variable: > > sip_append_audio_sdp=a=fmtp:18 annexb=no > > Can you verify that your provider is actually sending CN on 19 and NOT 13. There are some Sonus switches that say 19 but mean 13 and send on 13. 19=13 in hex. > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH_Wire > T: +1.918.420.9266 | F: +1.918.420.9267 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9266 > ISN: 410*543 > > > > > > > On Feb 25, 2013, at 8:08 AM, Flavio Goncalves wrote: > > > Hi, > > > > I have searched the list, but I couldn't find a final answer for this case. I'm receiving calls from a provider using G729 annex B. > > > > INVITE > > a=rtpmap:18 G729/8000. > > a=fmtp:18 annexb=yes. > > a=rtpmap:0 PCMU/8000. > > a=rtpmap:8 PCMA/8000. > > a=rtpmap:101 telephone-event/8000. > > a=fmtp:101 0-16. > > a=rtpmap:19 CN/8000. > > > > and FS is answering > > > > 200OK > > m=audio 31388 RTP/AVP 18 101 19. > > a=rtpmap:18 G729/8000. > > a=fmtp:18 annexb=yes. > > a=rtpmap:101 telephone-event/8000. > > a=fmtp:101 0-16. > > a=rtpmap:19 CN/8000. > > a=ptime:20. > > > > If freeswitch supports only G729A, shouldn't FS refuse the call with 488 or suppress the line annexb=yes? > > > > The audio quality is really bad when I receive calls in this way. After removing the annexb on the gateway the audio is fine, but unfortunately I can't remove from it from all VoIP providers terminating calls. For outbound calls I simply removed the annexb, but for inbound there is no way to do it. > > > > Flavio > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 Hussein.Said at cassidiancommunications.com Tue Feb 26 21:44:25 2013 From: Hussein.Said at cassidiancommunications.com (Hussein Said) Date: Tue, 26 Feb 2013 13:44:25 -0500 Subject: [Freeswitch-users] dialplan action when sending a REFER to FreeSWITCH Message-ID: <3173206D0F5F6B418F76F5997B2AE7520739E3F9@gatex01.CMLnet.peinet.peinc.com> Hi, I have an application that routes a call to an ad-hoc conference. That application then hunts an extension to answer the call by sending a REFER to FreeSWITCH which will send an INVITE from the conference to join. I need a way to make the "hunted" extension the moderator so that it can control the conference. I did set a moderator controls profile and can see it works if I change the dial plan(public.xml) to set the "member-flag=moderator". Also, "mod-wait|rfc4579" are set in the conference.conf.xml. My challenge is that when I hunt for an extension and find one, I send a REFER to FREESwitch so that it sends an INVITE to the extension that needs to answer the call, and I could not find a variable that worked in the dialplan that I can use as a condition to set the "member-flag=moderator" for the channel that is about to join the conference. Is what I am attempting feasible? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130226/10bb40d9/attachment.html From msc at freeswitch.org Tue Feb 26 22:14:10 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 26 Feb 2013 11:14:10 -0800 Subject: [Freeswitch-users] Possible BUG in G729 negotiation In-Reply-To: <0317A212-A71A-418E-B3B2-5DED7C8F9EFE@freeswitch.org> References: <0317A212-A71A-418E-B3B2-5DED7C8F9EFE@freeswitch.org> Message-ID: Flavio, Let me know how this goes. If you need a hand with the late negotiation stuff give us a shout. It's all documented here: http://wiki.freeswitch.org/wiki/Codec_negotiation#Late_Negotiation_.28requires_param.29 We also cover it in the FS Cookbook; last recipe in chapter 2. -MC On Tue, Feb 26, 2013 at 10:43 AM, Brian West wrote: > Inbound late negotiation and set the variable prior to answer. > > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH_Wire > T: +1.918.420.9266 | F: +1.918.420.9267 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9266 > ISN: 410*543 > > > > > > > On Feb 26, 2013, at 10:52 AM, Flavio Goncalves > wrote: > > > Hi Brian, > > > > This setting was already set. For outbound calls it works just fine. For > inbound calls, I can't disable the annexb. I need to reply (200 Ok) with > annexb=no. > > > > Best regards, > > > > Flavio E. Goncalves > > CTO - SipPulse Routing and Billing Solutions for SIP > > > > 2013/2/25 Brian West > > Try setting this variable: > > > > sip_append_audio_sdp=a=fmtp:18 annexb=no > > > > Can you verify that your provider is actually sending CN on 19 and NOT > 13. There are some Sonus switches that say 19 but mean 13 and send on 13. > 19=13 in hex. > > -- > > Brian West > > brian at freeswitch.org > > FreeSWITCH Solutions, LLC > > PO BOX PO BOX 2531 > > Brookfield, WI 53008-2531 > > Twitter: @FreeSWITCH_Wire > > T: +1.918.420.9266 | F: +1.918.420.9267 | M: +1.918.424.WEST > > iNUM: +883 5100 1420 9266 > > ISN: 410*543 > > > > > > > > > > > > > > On Feb 25, 2013, at 8:08 AM, Flavio Goncalves > wrote: > > > > > Hi, > > > > > > I have searched the list, but I couldn't find a final answer for this > case. I'm receiving calls from a provider using G729 annex B. > > > > > > INVITE > > > a=rtpmap:18 G729/8000. > > > a=fmtp:18 annexb=yes. > > > a=rtpmap:0 PCMU/8000. > > > a=rtpmap:8 PCMA/8000. > > > a=rtpmap:101 telephone-event/8000. > > > a=fmtp:101 0-16. > > > a=rtpmap:19 CN/8000. > > > > > > and FS is answering > > > > > > 200OK > > > m=audio 31388 RTP/AVP 18 101 19. > > > a=rtpmap:18 G729/8000. > > > a=fmtp:18 annexb=yes. > > > a=rtpmap:101 telephone-event/8000. > > > a=fmtp:101 0-16. > > > a=rtpmap:19 CN/8000. > > > a=ptime:20. > > > > > > If freeswitch supports only G729A, shouldn't FS refuse the call with > 488 or suppress the line annexb=yes? > > > > > > The audio quality is really bad when I receive calls in this way. > After removing the annexb on the gateway the audio is fine, but > unfortunately I can't remove from it from all VoIP providers terminating > calls. For outbound calls I simply removed the annexb, but for inbound > there is no way to do it. > > > > > > Flavio > > > > > > > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > > > > > 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 > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > 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 > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130226/ced582ba/attachment-0001.html From msc at freeswitch.org Tue Feb 26 22:17:12 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 26 Feb 2013 11:17:12 -0800 Subject: [Freeswitch-users] dialplan action when sending a REFER to FreeSWITCH In-Reply-To: <3173206D0F5F6B418F76F5997B2AE7520739E3F9@gatex01.CMLnet.peinet.peinc.com> References: <3173206D0F5F6B418F76F5997B2AE7520739E3F9@gatex01.CMLnet.peinet.peinc.com> Message-ID: Would you mind putting a snippet of your dialplan into our pastebin? ( pastebin.freeswitch.org) I'm sure the folks here can help you figure it out. With FreeSWITCH there's almost always a way to do it and it's just a matter of figuring out what works for your scenario. -MC On Tue, Feb 26, 2013 at 10:44 AM, Hussein Said < Hussein.Said at cassidiancommunications.com> wrote: > Hi,**** > > I have an application that routes a call to an ad-hoc conference. That > application then hunts an extension to answer the call by sending a REFER > to FreeSWITCH which will send an INVITE from the conference to join. **** > > I need a way to make the ?hunted? extension the moderator so that it can > control the conference. **** > > I did set a moderator controls profile and can see it works if I change > the dial plan(public.xml) to set the ?member-flag=moderator?.**** > > Also, ?mod-wait|rfc4579? are set in the conference.conf.xml.**** > > My challenge is that when I hunt for an extension and find one, I send a > REFER to FREESwitch so that it sends an INVITE to the extension that needs > to answer the call, and I could not find a variable that worked in the > dialplan that I can use as a condition to set the ?member-flag=moderator? > for the channel that is about to join the conference. Is what I am > attempting feasible?**** > > ** ** > > Thanks**** > > ** ** > > ** ** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130226/e243b2df/attachment.html From jmoran at secureachsystems.com Tue Feb 26 22:33:39 2013 From: jmoran at secureachsystems.com (Jason Moran) Date: Tue, 26 Feb 2013 14:33:39 -0500 Subject: [Freeswitch-users] Nuance Speech Server / Vocalizer MRCPv2mod_unimrcp simultaneous licenses not releasing quickly enough. References: <361E98F99D3CC3439EED59BC1924ED6971B08A@SERVER2003.SecuReachSystems.local><95382631-D364-43A5-B93F-E553F68248F9@gmail.com> Message-ID: <361E98F99D3CC3439EED59BC1924ED6971B0AE@SERVER2003.SecuReachSystems.local> After trying MRCPv1 - I don't think anybody out there is using Nuance for TTS remotely / externally! The audio is always blank / silence(both Nuance and FreeSWITCH seem to think it is working ok). I can get it to work with Nuance on localhost/127.0.0.1, but I believe there is a problem with FreeSWITCH/UniMRCP determining the correct rtp-ip when in MRCPv1 mode when using a server other than localhost. As shown in earlier posts in this thread, I can get it to work with MRCPv2 on the same external server. Regardless of all of these settings (client-ext-ip , client-ip , rtp-ip , rtp-ext-ip being set to an IP address other than 127.0.0.1, RTP is still trying to use 127.0.0.1, although RTSP is using the external IP address provided. mpf_rtp_stream.c from uniMRCP returns several references to 127.0.0.1 although it should not: mpf_rtp_stream.c:495 () Close RTP Receiver 127.0.0.1:4018 <- 10.0.0.67:7914 [r:0 l:0 j:0 d:0 i:0] and mpf_rtp_stream.c:1294 () Send Compound RTCP Packet [BYE] [76 bytes] 127.0.0.1:4019 -> 10.0.0.67:7915 mpf_rtp_stream.c:1306 () Failed to Send Compound RTCP Packet [BYE] [0 bytes] 127.0.0.1:4019 -> 10.0.0.67:7915 I believe this explains why I cannot hear the audio that is generated, as it should be using a 10.0.0.x IP address to receive and sent over RTP. To be safe I also changed the external_rtp_ip in vars.xml, but that hasn't fixed the issue either. Does anybody know where mpf_rtp_stream.c is getting this RTP IP Address from? Why does it insist on 127.0.0.1 despite all settings saying otherwise? Jason From: Christopher Rienzo [mailto:cmrienzo at gmail.com] Sent: Monday, February 25, 2013 8:34 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Nuance Speech Server / Vocalizer MRCPv2mod_unimrcp simultaneous licenses not releasing quickly enough. You can also try MRCPv1 (RTSP) and see if there is better performance. Using SIP to set up an MRCP session is silly. Chris On Mon, Feb 25, 2013 at 6:42 PM, Michael Collins wrote: Another case of DRM hurting only those who've paid for it... *sigh* I recommend calling your VAR or Nuance and gently but firmly insisting on a resolution. -MC On Mon, Feb 25, 2013 at 2:21 PM, wrote: We also had an issue with nuance eating licenses. We had enough room to tolerate this but did have to occasionally reboot servers when too many were consumed. Chris On Feb 25, 2013, at 3:21 PM, Rob Hutton wrote: We have a similar situation on a Cisco UCCX platform that we ended up having to write a script to bounce the services every couple of hours to clear. We haven't seen a fix from Nuance yet. On Mon, Feb 25, 2013 at 2:32 PM, Jason Moran wrote: In connecting to Nuance Speech Server's MRCPv2 for text-to-speech (Nuance Vocalizer) using mod_unimrcp, a single phone call which has several different TTS phrases should only use up 1 single simultaneous license. However, I am seeing a call use up 2,3,4 (etc) like it is not releasing the connection/license. I looked this up and some forums have noted that Nuance "does things wrong" and uniMRCP needs to send Nuance the "RTCP BYE" to tell Nuance that it is done with the synthesizer speech. But I tried that, and it still eats up the licenses. Within a couple of calls I'm out of TTS ports and the calls fail (but they clear again after several more seconds). Ideas? Experience? -Jason ________________________________________________________________________ _ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 ________________________________________________________________________ _ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 ________________________________________________________________________ _ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org ________________________________________________________________________ _ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130226/43425752/attachment-0001.html From msc at freeswitch.org Tue Feb 26 22:53:41 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 26 Feb 2013 11:53:41 -0800 Subject: [Freeswitch-users] Concatenated multiple playback types In-Reply-To: References: <1FFF97C269757C458224B7C895F35F1514B3F0@cantor.std.visionutv.se> <55F547E7-F986-4268-9BBD-C4210D997D12@kavun.ch> Message-ID: What about this? -MC On Mon, Feb 25, 2013 at 9:36 PM, Emrah wrote: > Hi there, > > Just wanted to inquire about this again. > FS is ever so flexible in so many ways that I cannot understand why this > wouldn't work. > I would like to apply the following to my alone sound in > conference.conf.xml. > silence_stream://6000!file_string:///path-to-alone-sound.wav > > I am imagining other scenarios where I would want to use some silence > before generating a ringback. > > What is your take on this? > Cheers > On Aug 22, 2012, at 12:40 PM, Michael Collins wrote: > > > One thing I like about the sox method is that it requires very little in > the way of processing/resources. > > -MC > > > > On Wed, Aug 22, 2012 at 9:27 AM, Emrah wrote: > > Hey Michael, > > > > That was my work around before I posted, but I thought since you guys do > allow some flexibility with silence, tone generation and file playback all > usable with the same app, I thought it would make more sense to take > advantage of it and combine some things there. > > > > All the best, thanks for your response. > > Emrah > > On Aug 22, 2012, at 12:07 PM, Michael Collins > wrote: > > > > > You could always take the sneaky approach and use sox to add some > silence to the beginning of conf-alone.wav. > > > -MC > > > > > > On Wed, Aug 22, 2012 at 8:56 AM, Emrah wrote: > > > Thanks for your response. > > > I don't think it would be something bad to implement. > > > > > > I also wanted to correct a little issueI experience both with FS and > Asterisk. > > > If you generate a tone, even if it's played in a file, and you execute > a record action right after it is played, the last 30 ms (or so) of the > tone is distorded and kind of faded out. > > > In some sircompstances, you might even hear the hint of the tone when > you execute the next playback. > > > I wanted to replace my record tones with the tone followed by 30 MS > silence. > > > > > > Best, > > > Emrah > > > > > > On Aug 22, 2012, at 10:36 AM, Peter Olsson < > peter.olsson at visionutveckling.se> wrote: > > > > > > > I don't believe this is possible. If you record 5 seconds of silence > into a file you might be able to use file_string:// to play them both. > > > > > > > > /Peter > > > > > > > > > > > > -----Ursprungligt meddelande----- > > > > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] F?r Emrah > > > > Skickat: den 22 augusti 2012 16:24 > > > > Till: FreeSWITCH Users Help > > > > ?mne: [Freeswitch-users] Concatenated multiple playback types > > > > > > > > Hello there, > > > > > > > > I am trying to add some silence before the alone sound file in my > conference.cnf.xml and can't get it to work. > > > > > > > > Here is what I tried: > > > > value="silence_stream://5000!file_string://conference/conf-alone.wav"/> > > > > value="silence_stream://5000&file_string://conference/conf-alone.wav"/> > > > > value="silence_stream://5000|file_string://conference/conf-alone.wav"/> > > > > value="silence_stream://5000;file_string://conference/conf-alone.wav"/> > > > > > > > > Any idea on how to do this? > > > > > > > > Thanks! > > > > Emrah > > > > > _________________________________________________________________________ > > > > Professional FreeSWITCH Consulting Services: > > > > consulting at freeswitch.org > > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > > http://www.freeswitch.org > > > > http://wiki.freeswitch.org > > > > http://www.cluecon.com > > > > > > > > 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 > > > > > > > > !DSPAM:5034e9a632761303016055! > > > > > > > > > > > > > _________________________________________________________________________ > > > > Professional FreeSWITCH Consulting Services: > > > > consulting at freeswitch.org > > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > > http://www.freeswitch.org > > > > http://wiki.freeswitch.org > > > > http://www.cluecon.com > > > > > > > > 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 > > > > > > > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > > > > > 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 > > > > > > > > > > > > -- > > > Michael S Collins > > > Twitter: @mercutioviz > > > http://www.FreeSWITCH.org > > > http://www.ClueCon.com > > > http://www.OSTAG.org > > > > > > > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > > > > > 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 > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > 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 > > > > > > > > -- > > Michael S Collins > > Twitter: @mercutioviz > > http://www.FreeSWITCH.org > > http://www.ClueCon.com > > http://www.OSTAG.org > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130226/b2c26c16/attachment.html From tomasz.szuster at gmail.com Tue Feb 26 23:48:10 2013 From: tomasz.szuster at gmail.com (Tomasz Szuster) Date: Tue, 26 Feb 2013 21:48:10 +0100 Subject: [Freeswitch-users] nibblebill, spidermonkey_odbc issue Message-ID: Hi, I'm struggling with making nibblebill working. What I've did till now is: Installed odbc: * libmyodbc * libodbc1 * odbcinst * odbcinst1debian2 * unixodbc * unixodbc-dev compile freeswitch using ./configure --enable-core-odbc-support make; make install My spidermonkey.conf file has: My odbc.ini: [nibblebill1] Driver = /usr/lib/x86_64-linux-gnu/odbc/libmyodbc.so SERVER = callcenter PORT = 3306 DATABASE = nibblebill1 OPTION = 67108864 USER = nibblebill1 PASSWORD = XXXXXXX ldd /usr/local/freeswitch/mod/mod_spidermonkey_odbc.so linux-vdso.so.1 => (0x00007fffbd7ff000) libfreeswitch.so.1 => /usr/local/freeswitch/lib/libfreeswitch.so.1 (0x00007f2f193ee000) libjs.so.1 => /usr/local/freeswitch/lib/libjs.so.1 (0x00007f2f19120000) libnspr4.so => /usr/local/freeswitch/lib/libnspr4.so (0x00007f2f18eef000) libodbc.so.1 => /usr/lib/x86_64-linux-gnu/libodbc.so.1 (0x00007f2f18c82000) libpthread.so.0 => /lib/x86_64-linux-gnu/libpthread.so.0 (0x00007f2f18a65000) libc.so.6 => /lib/x86_64-linux-gnu/libc.so.6 (0x00007f2f186a5000) libdl.so.2 => /lib/x86_64-linux-gnu/libdl.so.2 (0x00007f2f184a1000) libcrypt.so.1 => /lib/x86_64-linux-gnu/libcrypt.so.1 (0x00007f2f18268000) librt.so.1 => /lib/x86_64-linux-gnu/librt.so.1 (0x00007f2f1805f000) libssl.so.1.0.0 => /lib/x86_64-linux-gnu/libssl.so.1.0.0 (0x00007f2f17e03000) libcrypto.so.1.0.0 => /lib/x86_64-linux-gnu/libcrypto.so.1.0.0 (0x00007f2f17a3b000) libtinfo.so.5 => /lib/x86_64-linux-gnu/libtinfo.so.5 (0x00007f2f17813000) libstdc++.so.6 => /usr/lib/x86_64-linux-gnu/libstdc++.so.6 (0x00007f2f17513000) libm.so.6 => /lib/x86_64-linux-gnu/libm.so.6 (0x00007f2f17217000) libgcc_s.so.1 => /lib/x86_64-linux-gnu/libgcc_s.so.1 (0x00007f2f17000000) libltdl.so.7 => /usr/lib/x86_64-linux-gnu/libltdl.so.7 (0x00007f2f16df6000) /lib64/ld-linux-x86-64.so.2 (0x00007f2f19a48000) libz.so.1 => /lib/x86_64-linux-gnu/libz.so.1 (0x00007f2f16bde000) FreeSWITCH (Version 1.2.6 git a424765 2013-01-04 15:45:59Z) When I try to run* load mod_spidermonkey_odbc* I get: [CRIT] switch_loadable_module.c:1330 Error Loading module /usr/local/freeswitch/mod/mod_spidermonkey_odbc.so **/usr/local/freeswitch/mod/mod_spidermonkey_odbc.so: undefined symbol: mod_spidermonkey_odbc_module_interface** *load mod_nibblebill:* 2013-02-26 21:46:40.116678 [ERR] switch_odbc.c:365 STATE: IM002 CODE 0 ERROR: [unixODBC][Driver Manager]Data source name not found, and no default driver specified 2013-02-26 21:46:40.116678 [CRIT] mod_nibblebill.c:220 Cannot connect to ODBC driver/database odbc://callcenter (user: nibblebill1 / pass XXXXX)! 2013-02-26 21:46:40.116678 [CONSOLE] switch_loadable_module.c:1348 Successfully Loaded [mod_nibblebill] Also from time to time in logs I've see: [ERR] switch_odbc.c:365 STATE: IM002 CODE 0 ERROR: [unixODBC][Driver Manager]Data source name not found, and no default driver specified Will you be able to help with this issue ? Thank you. -- Regards. Tomasz -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130226/a14eeaa5/attachment-0001.html From jason.holden at start.ca Wed Feb 27 00:22:15 2013 From: jason.holden at start.ca (Jason Holden) Date: Tue, 26 Feb 2013 16:22:15 -0500 Subject: [Freeswitch-users] sending multiple voicemail attachments to different emails Message-ID: Hi, Tried the recommendation from the wiki for setting up the information in the directory and separating entries by a , but this does not work. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130226/c8f33ff1/attachment.html From krice at freeswitch.org Wed Feb 27 00:28:09 2013 From: krice at freeswitch.org (Ken Rice) Date: Tue, 26 Feb 2013 15:28:09 -0600 Subject: [Freeswitch-users] nibblebill, spidermonkey_odbc issue In-Reply-To: Message-ID: You don?t load mod_spidermoney_odbc You uncomment it in autoload_configs/spidermonkey.conf.xml Mod_spidermonkey loads that module... Mod_spidermonkey is probably the only module that can actually load its own extension modules On 2/26/13 2:48 PM, "Tomasz Szuster" wrote: > Hi, > > I'm struggling with making nibblebill working. > What I've did till now is: > > Installed odbc: > * libmyodbc > * libodbc1 > *?odbcinst > *?odbcinst1debian2 > *?unixodbc > *?unixodbc-dev > > compile freeswitch using > > > > ./configure --enable-core-odbc-support > make; make install > > My spidermonkey.conf file has: > > > ? > ? > ? > ? > ? > ? > > > > My odbc.ini: > > [nibblebill1] > Driver ? ? ? ? ?= /usr/lib/x86_64-linux-gnu/odbc/libmyodbc.so > SERVER ? ? ? ? ?= callcenter > PORT ? ? ? ? ? ?= 3306 > DATABASE ? ? ? ?= nibblebill1 > OPTION ?= 67108864 > USER ? ? ? ? ? ?= nibblebill1 > PASSWORD ? ? ? ?= XXXXXXX > > > > > ldd /usr/local/freeswitch/mod/mod_spidermonkey_odbc.so > ? ? ? ? linux-vdso.so.1 => ?(0x00007fffbd7ff000) > ? ? ? ? libfreeswitch.so.1 => /usr/local/freeswitch/lib/libfreeswitch.so.1 > (0x00007f2f193ee000) > ? ? ? ? libjs.so.1 => /usr/local/freeswitch/lib/libjs.so.1 > (0x00007f2f19120000) > ? ? ? ? libnspr4.so => /usr/local/freeswitch/lib/libnspr4.so > (0x00007f2f18eef000) > ? ? ? ? libodbc.so.1 => /usr/lib/x86_64-linux-gnu/libodbc.so.1 > (0x00007f2f18c82000) > ? ? ? ? libpthread.so.0 => /lib/x86_64-linux-gnu/libpthread.so.0 > (0x00007f2f18a65000) > ? ? ? ? libc.so.6 => /lib/x86_64-linux-gnu/libc.so.6 (0x00007f2f186a5000) > ? ? ? ? libdl.so.2 => /lib/x86_64-linux-gnu/libdl.so.2 (0x00007f2f184a1000) > ? ? ? ? libcrypt.so.1 => /lib/x86_64-linux-gnu/libcrypt.so.1 > (0x00007f2f18268000) > ? ? ? ? librt.so.1 => /lib/x86_64-linux-gnu/librt.so.1 (0x00007f2f1805f000) > ? ? ? ? libssl.so.1.0.0 => /lib/x86_64-linux-gnu/libssl.so.1.0.0 > (0x00007f2f17e03000) > ? ? ? ? libcrypto.so.1.0.0 => /lib/x86_64-linux-gnu/libcrypto.so.1.0.0 > (0x00007f2f17a3b000) > ? ? ? ? libtinfo.so.5 => /lib/x86_64-linux-gnu/libtinfo.so.5 > (0x00007f2f17813000) > ? ? ? ? libstdc++.so.6 => /usr/lib/x86_64-linux-gnu/libstdc++.so.6 > (0x00007f2f17513000) > ? ? ? ? libm.so.6 => /lib/x86_64-linux-gnu/libm.so.6 (0x00007f2f17217000) > ? ? ? ? libgcc_s.so.1 => /lib/x86_64-linux-gnu/libgcc_s.so.1 > (0x00007f2f17000000) > ? ? ? ? libltdl.so.7 => /usr/lib/x86_64-linux-gnu/libltdl.so.7 > (0x00007f2f16df6000) > ? ? ? ? /lib64/ld-linux-x86-64.so.2 (0x00007f2f19a48000) > ? ? ? ? libz.so.1 => /lib/x86_64-linux-gnu/libz.so.1 (0x00007f2f16bde000) > > > FreeSWITCH (Version 1.2.6 git a424765 2013-01-04 15:45:59Z) > > When I try to run load mod_spidermonkey_odbc I get: > > [CRIT] switch_loadable_module.c:1330 Error Loading module > /usr/local/freeswitch/mod/mod_spidermonkey_odbc.so? > **/usr/local/freeswitch/mod/mod_spidermonkey_odbc.so: undefined symbol: > mod_spidermonkey_odbc_module_interface**? > > load mod_nibblebill: > > 2013-02-26 21:46:40.116678 [ERR] switch_odbc.c:365 STATE: IM002 CODE 0 ERROR: > [unixODBC][Driver Manager]Data source name not found, and no default driver > specified > > 2013-02-26 21:46:40.116678 [CRIT] mod_nibblebill.c:220 Cannot connect to ODBC > driver/database odbc://callcenter (user: nibblebill1 / pass XXXXX)! > 2013-02-26 21:46:40.116678 [CONSOLE] switch_loadable_module.c:1348 > Successfully Loaded [mod_nibblebill] > > > Also from time to time in logs I've see: > > [ERR] switch_odbc.c:365 STATE: IM002 CODE 0 ERROR: [unixODBC][Driver > Manager]Data source name not found, and no default driver specified? > > Will you be able to help with this issue ? > Thank you. -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130226/e4279a01/attachment.html From msc at freeswitch.org Wed Feb 27 00:53:13 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 26 Feb 2013 13:53:13 -0800 Subject: [Freeswitch-users] sending multiple voicemail attachments to different emails In-Reply-To: References: Message-ID: Malfunction! Need input! Can you pastebin some logs of it not working, plus the relevant part of the config file? -MC On Tue, Feb 26, 2013 at 1:22 PM, Jason Holden wrote: > Hi,**** > > Tried the recommendation from the wiki for setting up the information in > the directory and separating entries by a , but this does not work.**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130226/a145e8d6/attachment.html From jmoran at secureachsystems.com Wed Feb 27 01:00:43 2013 From: jmoran at secureachsystems.com (Jason Moran) Date: Tue, 26 Feb 2013 17:00:43 -0500 Subject: [Freeswitch-users] nibblebill, spidermonkey_odbc issue References: Message-ID: <361E98F99D3CC3439EED59BC1924ED6971B0B2@SERVER2003.SecuReachSystems.local> Can you make connections to your database using the ODBC connection outside of FreeSWITCH (but from the same server that FS is installed on)? I?ve often caught problems either in my firewall or a dumb typo in my ODBC configurations. From: Tomasz Szuster [mailto:tomasz.szuster at gmail.com] Sent: Tuesday, February 26, 2013 3:48 PM To: FreeSWITCH Users Help Subject: [Freeswitch-users] nibblebill, spidermonkey_odbc issue Hi, I'm struggling with making nibblebill working. What I've did till now is: Installed odbc: * libmyodbc * libodbc1 * odbcinst * odbcinst1debian2 * unixodbc * unixodbc-dev compile freeswitch using ./configure --enable-core-odbc-support make; make install My spidermonkey.conf file has: My odbc.ini: [nibblebill1] Driver = /usr/lib/x86_64-linux-gnu/odbc/libmyodbc.so SERVER = callcenter PORT = 3306 DATABASE = nibblebill1 OPTION = 67108864 USER = nibblebill1 PASSWORD = XXXXXXX ldd /usr/local/freeswitch/mod/mod_spidermonkey_odbc.so linux-vdso.so.1 => (0x00007fffbd7ff000) libfreeswitch.so.1 => /usr/local/freeswitch/lib/libfreeswitch.so.1 (0x00007f2f193ee000) libjs.so.1 => /usr/local/freeswitch/lib/libjs.so.1 (0x00007f2f19120000) libnspr4.so => /usr/local/freeswitch/lib/libnspr4.so (0x00007f2f18eef000) libodbc.so.1 => /usr/lib/x86_64-linux-gnu/libodbc.so.1 (0x00007f2f18c82000) libpthread.so.0 => /lib/x86_64-linux-gnu/libpthread.so.0 (0x00007f2f18a65000) libc.so.6 => /lib/x86_64-linux-gnu/libc.so.6 (0x00007f2f186a5000) libdl.so.2 => /lib/x86_64-linux-gnu/libdl.so.2 (0x00007f2f184a1000) libcrypt.so.1 => /lib/x86_64-linux-gnu/libcrypt.so.1 (0x00007f2f18268000) librt.so.1 => /lib/x86_64-linux-gnu/librt.so.1 (0x00007f2f1805f000) libssl.so.1.0.0 => /lib/x86_64-linux-gnu/libssl.so.1.0.0 (0x00007f2f17e03000) libcrypto.so.1.0.0 => /lib/x86_64-linux-gnu/libcrypto.so.1.0.0 (0x00007f2f17a3b000) libtinfo.so.5 => /lib/x86_64-linux-gnu/libtinfo.so.5 (0x00007f2f17813000) libstdc++.so.6 => /usr/lib/x86_64-linux-gnu/libstdc++.so.6 (0x00007f2f17513000) libm.so.6 => /lib/x86_64-linux-gnu/libm.so.6 (0x00007f2f17217000) libgcc_s.so.1 => /lib/x86_64-linux-gnu/libgcc_s.so.1 (0x00007f2f17000000) libltdl.so.7 => /usr/lib/x86_64-linux-gnu/libltdl.so.7 (0x00007f2f16df6000) /lib64/ld-linux-x86-64.so.2 (0x00007f2f19a48000) libz.so.1 => /lib/x86_64-linux-gnu/libz.so.1 (0x00007f2f16bde000) FreeSWITCH (Version 1.2.6 git a424765 2013-01-04 15:45:59Z) When I try to run load mod_spidermonkey_odbc I get: [CRIT] switch_loadable_module.c:1330 Error Loading module /usr/local/freeswitch/mod/mod_spidermonkey_odbc.so **/usr/local/freeswitch/mod/mod_spidermonkey_odbc.so: undefined symbol: mod_spidermonkey_odbc_module_interface** load mod_nibblebill: 2013-02-26 21:46:40.116678 [ERR] switch_odbc.c:365 STATE: IM002 CODE 0 ERROR: [unixODBC][Driver Manager]Data source name not found, and no default driver specified 2013-02-26 21:46:40.116678 [CRIT] mod_nibblebill.c:220 Cannot connect to ODBC driver/database odbc://callcenter (user: nibblebill1 / pass XXXXX)! 2013-02-26 21:46:40.116678 [CONSOLE] switch_loadable_module.c:1348 Successfully Loaded [mod_nibblebill] Also from time to time in logs I've see: [ERR] switch_odbc.c:365 STATE: IM002 CODE 0 ERROR: [unixODBC][Driver Manager]Data source name not found, and no default driver specified Will you be able to help with this issue ? Thank you. -- Regards. Tomasz -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130226/2fcf6198/attachment-0001.html From krice at freeswitch.org Wed Feb 27 01:14:59 2013 From: krice at freeswitch.org (Ken Rice) Date: Tue, 26 Feb 2013 16:14:59 -0600 Subject: [Freeswitch-users] nibblebill, spidermonkey_odbc issue In-Reply-To: <361E98F99D3CC3439EED59BC1924ED6971B0B2@SERVER2003.SecuReachSystems.local> Message-ID: Also, I didn?t read the whole email earlier.. Spidermonkey odbc and nibblebill odbc are 2 different config settings... You need to check the wiki for the proper odbc dsn syntax and make sure the DSN you configured in your odbc.ini works from isql K On 2/26/13 4:00 PM, "Jason Moran" wrote: > Can you make connections to your database using the ODBC connection outside of > FreeSWITCH (but from the same server that FS is installed on)? I?ve often > caught problems either in my firewall or a dumb typo in my ODBC > configurations. > > > From: Tomasz Szuster [mailto:tomasz.szuster at gmail.com] > Sent: Tuesday, February 26, 2013 3:48 PM > To: FreeSWITCH Users Help > Subject: [Freeswitch-users] nibblebill, spidermonkey_odbc issue > > Hi, > > > > I'm struggling with making nibblebill working. > > What I've did till now is: > > > > Installed odbc: > > * libmyodbc > > * libodbc1 > > * odbcinst > > * odbcinst1debian2 > > * unixodbc > > * unixodbc-dev > > > > compile freeswitch using > > > > > > ./configure --enable-core-odbc-support > make; make install > > > My spidermonkey.conf file has: > > > > > > > > > > > > > > > > > > > > > > > > My odbc.ini: > > > > [nibblebill1] > > Driver = /usr/lib/x86_64-linux-gnu/odbc/libmyodbc.so > > SERVER = callcenter > > PORT = 3306 > > DATABASE = nibblebill1 > > OPTION = 67108864 > > USER = nibblebill1 > > PASSWORD = XXXXXXX > > > > > > > > > > ldd /usr/local/freeswitch/mod/mod_spidermonkey_odbc.so > > linux-vdso.so.1 => (0x00007fffbd7ff000) > > libfreeswitch.so.1 => /usr/local/freeswitch/lib/libfreeswitch.so.1 > (0x00007f2f193ee000) > > libjs.so.1 => /usr/local/freeswitch/lib/libjs.so.1 > (0x00007f2f19120000) > > libnspr4.so => /usr/local/freeswitch/lib/libnspr4.so > (0x00007f2f18eef000) > > libodbc.so.1 => /usr/lib/x86_64-linux-gnu/libodbc.so.1 > (0x00007f2f18c82000) > > libpthread.so.0 => /lib/x86_64-linux-gnu/libpthread.so.0 > (0x00007f2f18a65000) > > libc.so.6 => /lib/x86_64-linux-gnu/libc.so.6 (0x00007f2f186a5000) > > libdl.so.2 => /lib/x86_64-linux-gnu/libdl.so.2 (0x00007f2f184a1000) > > libcrypt.so.1 => /lib/x86_64-linux-gnu/libcrypt.so.1 > (0x00007f2f18268000) > > librt.so.1 => /lib/x86_64-linux-gnu/librt.so.1 (0x00007f2f1805f000) > > libssl.so.1.0.0 => /lib/x86_64-linux-gnu/libssl.so.1.0.0 > (0x00007f2f17e03000) > > libcrypto.so.1.0.0 => /lib/x86_64-linux-gnu/libcrypto.so.1.0.0 > (0x00007f2f17a3b000) > > libtinfo.so.5 => /lib/x86_64-linux-gnu/libtinfo.so.5 > (0x00007f2f17813000) > > libstdc++.so.6 => /usr/lib/x86_64-linux-gnu/libstdc++.so.6 > (0x00007f2f17513000) > > libm.so.6 => /lib/x86_64-linux-gnu/libm.so.6 (0x00007f2f17217000) > > libgcc_s.so.1 => /lib/x86_64-linux-gnu/libgcc_s.so.1 > (0x00007f2f17000000) > > libltdl.so.7 => /usr/lib/x86_64-linux-gnu/libltdl.so.7 > (0x00007f2f16df6000) > > /lib64/ld-linux-x86-64.so.2 (0x00007f2f19a48000) > > libz.so.1 => /lib/x86_64-linux-gnu/libz.so.1 (0x00007f2f16bde000) > > > > > FreeSWITCH (Version 1.2.6 git a424765 2013-01-04 15:45:59Z) > > > > When I try to run load mod_spidermonkey_odbc I get: > > > > [CRIT] switch_loadable_module.c:1330 Error Loading module > /usr/local/freeswitch/mod/mod_spidermonkey_odbc.so > **/usr/local/freeswitch/mod/mod_spidermonkey_odbc.so: undefined symbol: > mod_spidermonkey_odbc_module_interface** > > > > load mod_nibblebill: > > > > 2013-02-26 21:46:40.116678 [ERR] switch_odbc.c:365 STATE: IM002 CODE 0 ERROR: > [unixODBC][Driver Manager]Data source name not found, and no default driver > specified > > > > 2013-02-26 21:46:40.116678 [CRIT] mod_nibblebill.c:220 Cannot connect to ODBC > driver/database odbc://callcenter (user: nibblebill1 / pass XXXXX)! > > 2013-02-26 21:46:40.116678 [CONSOLE] switch_loadable_module.c:1348 > Successfully Loaded [mod_nibblebill] > > > > > > Also from time to time in logs I've see: > > > > [ERR] switch_odbc.c:365 STATE: IM002 CODE 0 ERROR: [unixODBC][Driver > Manager]Data source name not found, and no default driver specified > > > > Will you be able to help with this issue ? > > Thank you. -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130226/66f570c9/attachment.html From cmrienzo at gmail.com Wed Feb 27 02:47:38 2013 From: cmrienzo at gmail.com (Christopher Rienzo) Date: Tue, 26 Feb 2013 18:47:38 -0500 Subject: [Freeswitch-users] Nuance Speech Server / Vocalizer MRCPv2mod_unimrcp simultaneous licenses not releasing quickly enough. In-Reply-To: <361E98F99D3CC3439EED59BC1924ED6971B0AE@SERVER2003.SecuReachSystems.local> References: <361E98F99D3CC3439EED59BC1924ED6971B08A@SERVER2003.SecuReachSystems.local> <95382631-D364-43A5-B93F-E553F68248F9@gmail.com> <361E98F99D3CC3439EED59BC1924ED6971B0AE@SERVER2003.SecuReachSystems.local> Message-ID: I used MRCPv1 with remote servers. 3 MRCP servers handling requests from 12 FS servers. Start with the example nuance MRCPv1 configuration in conf/mrcp_profiles and replace the rtp-ip, server-ip, and server-port. Use IP address, not hostnames, not "auto". If that doesn't work, reply with your configuration. Chris On Tue, Feb 26, 2013 at 2:33 PM, Jason Moran wrote: > After trying MRCPv1 ? I don?t think anybody out there is using Nuance > for TTS remotely / externally! The audio is always blank / silence(both > Nuance and FreeSWITCH seem to think it is working ok). I can get it to work > with Nuance on localhost/127.0.0.1, but I believe there is a problem with > FreeSWITCH/UniMRCP determining the correct rtp-ip when in MRCPv1 mode when > using a server other than localhost. As shown in earlier posts in this > thread, I can get it to work with MRCPv2 on the same external server.**** > > ** ** > > Regardless of all of these settings (client-ext-ip , client-ip , rtp-ip , > rtp-ext-ip being set to an IP address other than 127.0.0.1, RTP is still > trying to use 127.0.0.1, although RTSP is using the external IP address > provided.**** > > ** ** > > mpf_rtp_stream.c from uniMRCP returns several references to 127.0.0.1 > although it should not: > mpf_rtp_stream.c:495 () Close RTP Receiver 127.0.0.1:4018 <- > 10.0.0.67:7914 [r:0 l:0 j:0 d:0 i:0]**** > > and**** > > mpf_rtp_stream.c:1294 () Send Compound RTCP Packet [BYE] [76 bytes] > 127.0.0.1:4019 -> 10.0.0.67:7915**** > > mpf_rtp_stream.c:1306 () Failed to Send Compound RTCP Packet [BYE] [0 > bytes] 127.0.0.1:4019 -> 10.0.0.67:7915**** > > ** ** > > I believe this explains why I cannot hear the audio that is generated, as > it should be using a 10.0.0.x IP address to receive and sent over RTP. To > be safe I also changed the external_rtp_ip in vars.xml, but that hasn?t > fixed the issue either.**** > > ** ** > > Does anybody know where mpf_rtp_stream.c is getting this RTP IP Address > from? Why does it insist on 127.0.0.1 despite all settings saying otherwise? > **** > > ** ** > > Jason**** > > ** ** > > *From:* Christopher Rienzo [mailto:cmrienzo at gmail.com] > *Sent:* Monday, February 25, 2013 8:34 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Nuance Speech Server / Vocalizer > MRCPv2mod_unimrcp simultaneous licenses not releasing quickly enough.**** > > ** ** > > You can also try MRCPv1 (RTSP) and see if there is better performance. > Using SIP to set up an MRCP session is silly.**** > > ** ** > > Chris**** > > ** ** > > On Mon, Feb 25, 2013 at 6:42 PM, Michael Collins > wrote:**** > > Another case of DRM hurting only those who've paid for it... *sigh* > > I recommend calling your VAR or Nuance and gently but firmly insisting on > a resolution. > > -MC**** > > ** ** > > On Mon, Feb 25, 2013 at 2:21 PM, wrote:**** > > We also had an issue with nuance eating licenses. We had enough room to > tolerate this but did have to occasionally reboot servers when too many > were consumed. **** > > ** ** > > Chris**** > > ** ** > > ** ** > > ** ** > > > On Feb 25, 2013, at 3:21 PM, Rob Hutton wrote:**** > > We have a similar situation on a Cisco UCCX platform that we ended up > having to write a script to bounce the services every couple of hours to > clear. We haven't seen a fix from Nuance yet.**** > > On Mon, Feb 25, 2013 at 2:32 PM, Jason Moran > wrote:**** > > In connecting to Nuance Speech Server?s MRCPv2 for text-to-speech (Nuance > Vocalizer) using mod_unimrcp, a single phone call which has several > different TTS phrases should only use up 1 single simultaneous license. > However, I am seeing a call use up 2,3,4 (etc) like it is not releasing the > connection/license.**** > > **** > > I looked this up and some forums have noted that Nuance ?does things > wrong? and uniMRCP needs to send Nuance the ?RTCP BYE? to tell Nuance that > it is done with the synthesizer speech. But I tried that, and it still > eats up the licenses. Within a couple of calls I?m out of TTS ports and the > calls fail (but they clear again after several more seconds).**** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > Ideas? Experience?**** > > -Jason**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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**** > > > > **** > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130226/2b1ffd2f/attachment-0001.html From cmrienzo at gmail.com Wed Feb 27 02:50:47 2013 From: cmrienzo at gmail.com (Christopher Rienzo) Date: Tue, 26 Feb 2013 18:50:47 -0500 Subject: [Freeswitch-users] Concatenated multiple playback types In-Reply-To: References: <1FFF97C269757C458224B7C895F35F1514B3F0@cantor.std.visionutv.se> <55F547E7-F986-4268-9BBD-C4210D997D12@kavun.ch> Message-ID: file_string:// does the concatenation.. so you need: file_string://silence_stream://6000!conference/conf-alone.wav On Tue, Feb 26, 2013 at 2:53 PM, Michael Collins wrote: > What about this? > > value="silence_stream://6000!conference/conf-alone.wav"/> > > -MC > > > On Mon, Feb 25, 2013 at 9:36 PM, Emrah wrote: > >> Hi there, >> >> Just wanted to inquire about this again. >> FS is ever so flexible in so many ways that I cannot understand why this >> wouldn't work. >> I would like to apply the following to my alone sound in >> conference.conf.xml. >> silence_stream://6000!file_string:///path-to-alone-sound.wav >> >> I am imagining other scenarios where I would want to use some silence >> before generating a ringback. >> >> What is your take on this? >> Cheers >> On Aug 22, 2012, at 12:40 PM, Michael Collins wrote: >> >> > One thing I like about the sox method is that it requires very little >> in the way of processing/resources. >> > -MC >> > >> > On Wed, Aug 22, 2012 at 9:27 AM, Emrah wrote: >> > Hey Michael, >> > >> > That was my work around before I posted, but I thought since you guys >> do allow some flexibility with silence, tone generation and file playback >> all usable with the same app, I thought it would make more sense to take >> advantage of it and combine some things there. >> > >> > All the best, thanks for your response. >> > Emrah >> > On Aug 22, 2012, at 12:07 PM, Michael Collins >> wrote: >> > >> > > You could always take the sneaky approach and use sox to add some >> silence to the beginning of conf-alone.wav. >> > > -MC >> > > >> > > On Wed, Aug 22, 2012 at 8:56 AM, Emrah wrote: >> > > Thanks for your response. >> > > I don't think it would be something bad to implement. >> > > >> > > I also wanted to correct a little issueI experience both with FS and >> Asterisk. >> > > If you generate a tone, even if it's played in a file, and you >> execute a record action right after it is played, the last 30 ms (or so) of >> the tone is distorded and kind of faded out. >> > > In some sircompstances, you might even hear the hint of the tone when >> you execute the next playback. >> > > I wanted to replace my record tones with the tone followed by 30 MS >> silence. >> > > >> > > Best, >> > > Emrah >> > > >> > > On Aug 22, 2012, at 10:36 AM, Peter Olsson < >> peter.olsson at visionutveckling.se> wrote: >> > > >> > > > I don't believe this is possible. If you record 5 seconds of >> silence into a file you might be able to use file_string:// to play them >> both. >> > > > >> > > > /Peter >> > > > >> > > > >> > > > -----Ursprungligt meddelande----- >> > > > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] F?r Emrah >> > > > Skickat: den 22 augusti 2012 16:24 >> > > > Till: FreeSWITCH Users Help >> > > > ?mne: [Freeswitch-users] Concatenated multiple playback types >> > > > >> > > > Hello there, >> > > > >> > > > I am trying to add some silence before the alone sound file in my >> conference.cnf.xml and can't get it to work. >> > > > >> > > > Here is what I tried: >> > > > > value="silence_stream://5000!file_string://conference/conf-alone.wav"/> >> > > > > value="silence_stream://5000&file_string://conference/conf-alone.wav"/> >> > > > > value="silence_stream://5000|file_string://conference/conf-alone.wav"/> >> > > > > value="silence_stream://5000;file_string://conference/conf-alone.wav"/> >> > > > >> > > > Any idea on how to do this? >> > > > >> > > > Thanks! >> > > > Emrah >> > > > >> _________________________________________________________________________ >> > > > Professional FreeSWITCH Consulting Services: >> > > > consulting at freeswitch.org >> > > > http://www.freeswitchsolutions.com >> > > > >> > > > >> >> > > > >> > > > Official FreeSWITCH Sites >> > > > http://www.freeswitch.org >> > > > http://wiki.freeswitch.org >> > > > http://www.cluecon.com >> > > > >> > > > 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 >> > > > >> > > > !DSPAM:5034e9a632761303016055! >> > > > >> > > > >> > > > >> _________________________________________________________________________ >> > > > Professional FreeSWITCH Consulting Services: >> > > > consulting at freeswitch.org >> > > > http://www.freeswitchsolutions.com >> > > > >> > > > >> > > > >> > > > >> > > > Official FreeSWITCH Sites >> > > > http://www.freeswitch.org >> > > > http://wiki.freeswitch.org >> > > > http://www.cluecon.com >> > > > >> > > > 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 >> > > >> > > >> > > >> _________________________________________________________________________ >> > > Professional FreeSWITCH Consulting Services: >> > > consulting at freeswitch.org >> > > http://www.freeswitchsolutions.com >> > > >> > > >> > > >> > > >> > > Official FreeSWITCH Sites >> > > http://www.freeswitch.org >> > > http://wiki.freeswitch.org >> > > http://www.cluecon.com >> > > >> > > 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 >> > > >> > > >> > > >> > > -- >> > > Michael S Collins >> > > Twitter: @mercutioviz >> > > http://www.FreeSWITCH.org >> > > http://www.ClueCon.com >> > > http://www.OSTAG.org >> > > >> > > >> > > >> _________________________________________________________________________ >> > > Professional FreeSWITCH Consulting Services: >> > > consulting at freeswitch.org >> > > http://www.freeswitchsolutions.com >> > > >> > > >> > > >> > > >> > > Official FreeSWITCH Sites >> > > http://www.freeswitch.org >> > > http://wiki.freeswitch.org >> > > http://www.cluecon.com >> > > >> > > 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 >> > >> > >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > 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 >> > >> > >> > >> > -- >> > Michael S Collins >> > Twitter: @mercutioviz >> > http://www.FreeSWITCH.org >> > http://www.ClueCon.com >> > http://www.OSTAG.org >> > >> > >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > 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 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130226/a543b2e8/attachment-0001.html From bdfoster at endigotech.com Wed Feb 27 03:07:12 2013 From: bdfoster at endigotech.com (Brian Foster) Date: Tue, 26 Feb 2013 19:07:12 -0500 Subject: [Freeswitch-users] E169 question In-Reply-To: References: Message-ID: <4DF043DC-C88E-429A-B069-649A9F4B3862@endigotech.com> > Hi, > I was wondering if I could ask you some dongle questions. I've been > testing out Gsmopen with an E160. Do you have any experience with > that one and why do you prefer E169s? I personally have not. They were, however, recommended to me by a friend who has experience with Huawei-brand dongles. > I'm in NYC. I might be > interested in a group buy, but I'm still trying to figure out if it's > going to reliable for clients. I currently cannot verify anything about the Huawei devices. The group buy actually fell through, and I was planning on getting one or two to play with. If others are interested in doing a group buy I'd be more than happy to facilitate. In your setup, what is your general purpose for having them in production? I planned on using a few for emergency backup in case the Internet or something else went kaput. With the advent of cell phones these days, it's really not an issue now. Otherwise, to me it's a toy. Now if you we're in Europe, there are advantages to using gsmopen and a dongle with a few different mobile providers to get free or reduced rate calling (within the same network). I'm CC'ing the Freeswitch users mailing list, in hopes that someone can better answer your questions. If you haven't already, subscribe to the ML by going to http://lists.freeswitch.org/mailman/listinfo/freeswitch-users. Good luck in your future endeavors, and welcome to the wonderful world of Freeswitch. Regards, Brian Foster Sent from my iPhone > I'm waiting on an antenna at the > moment....I suspect that's my current problem. > > > Thanks in advance! > > Cheers, > > > Peter Hartmann > Hartmann Computer Consulting -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130226/2c36337e/attachment.html From cmrienzo at gmail.com Wed Feb 27 03:13:59 2013 From: cmrienzo at gmail.com (Christopher Rienzo) Date: Tue, 26 Feb 2013 19:13:59 -0500 Subject: [Freeswitch-users] Audio dropped using unimrcp tts. In-Reply-To: References: Message-ID: Sorry for the delay. I looked at the pcap and it confirmed my suspicion. The MRCP server doesn't reply with IN-PROGRESS until 1.35 sec of audio has already been transmitted. IN-PROGRESS tells mod_unimrcp that the request is has been accepted and is being processed. The module assumes the reply is immediately sent before the RTP is streamed. Chris On Mon, Feb 18, 2013 at 6:09 AM, Manuel Dur?n Aguete wrote: > Hello, > > I've update the dropbox folder with the pcap and mod_unimrcp_rx/tx files, > added the log file too. > > https://www.dropbox.com/sh/tcgnktzzby0h2ns/v7ViiUyObB > > I've tested with branch 1.2 and with master, the pcap is from master code. > > Thank you for your help. > > Regards. > > > > > 2013/2/15 Christopher Rienzo > >> Send PCAP with the RTSP signaling and RTP audio. I'm curious when the >> server starts streaming the audio. >> >> >> >> On Fri, Feb 15, 2013 at 4:20 AM, Manuel Dur?n Aguete wrote: >> >>> Hello, >>> >>> I've tried with different values for sleep, but the problem persists, >>> in http://pastebin.freeswitch.org/20594 i've posted the call log. >>> I've enabled MOD_UNIMRCP_DEBUG_AUDIO_QUEUE in mod_unircp.cm the audio >>> streams generated can be downloaded from >>> https://www.dropbox.com/sh/tcgnktzzby0h2ns/v7ViiUyObB. >>> >>> I've changed the speak data to >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> and only "3 4 5 6 7 8 9 0" is played in softphone. >>> >>> Thank you for your help. >>> >>> Regards. >>> >>> >>> >>> >>> >>> 2013/2/14 Michael Collins >>> >>>> Try adding a sleep in your dialplan prior to the call to "speak" >>>> application. Start with 1000 ms and see if it makes any difference, then >>>> tweak the sleep time from there. >>>> >>>> -MC >>>> >>>> On Thu, Feb 14, 2013 at 5:26 AM, Manuel Dur?n Aguete < >>>> manuel at aguete.org> wrote: >>>> >>>>> Hello, >>>>> >>>>> I'm a newbie testing freeswitch tts with loquendo mrcp1 server, >>>>> >>>>> mrcp_profile: >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> Dialplan code: >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> I can hear all text with first word dropped. I've captured the rtp >>>>> traffic and the audio between FS <-> Loquendo contains the word. After >>>>> apply the fixed recommended in >>>>> http://wiki.freeswitch.org/wiki/RTP_Issues#Dropped_Audio the issue >>>>> persists. >>>>> >>>>> Any idea, suggestion, etc ? >>>>> >>>>> Thank you. >>>>> >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> 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 >>>>> >>>>> >>>> >>>> >>>> -- >>>> Michael S Collins >>>> Twitter: @mercutioviz >>>> http://www.FreeSWITCH.org >>>> http://www.ClueCon.com >>>> http://www.OSTAG.org >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130226/93375609/attachment-0001.html From steveu at coppice.org Wed Feb 27 03:17:49 2013 From: steveu at coppice.org (Steve Underwood) Date: Wed, 27 Feb 2013 08:17:49 +0800 Subject: [Freeswitch-users] Fax document number of pages In-Reply-To: References: Message-ID: <512D50AD.2040206@coppice.org> Hi Tihomir, On 02/26/2013 11:42 PM, Tihomir Culjaga wrote: > hello, > i know its stupid, but please, is there a way to somehow print the > number of pages on a document in a / format > instead of p. ? > > i cannot find this in mod_spandsp so i assume its somewhere in spandsp > library... any idea where ? > > > regards, > Tihomir. > Spandsp just follows the practice in most FAX machines, and puts p. in the page headers. Spandsp doesn't actually know the total number of pages when it sends the first page, as it allows additional TIFF files to be added to the transmission while a call is in progress. / could be set up for the simple case of one call == one TIFF file, but that's not in the code right now. Steve From tculjaga at gmail.com Wed Feb 27 03:34:35 2013 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Wed, 27 Feb 2013 01:34:35 +0100 Subject: [Freeswitch-users] Fax document number of pages In-Reply-To: <512D50AD.2040206@coppice.org> References: <512D50AD.2040206@coppice.org> Message-ID: On Wed, Feb 27, 2013 at 1:17 AM, Steve Underwood wrote: > Hi Tihomir, > > On 02/26/2013 11:42 PM, Tihomir Culjaga wrote: > > hello, > > i know its stupid, but please, is there a way to somehow print the > > number of pages on a document in a / format > > instead of p. ? > > > > i cannot find this in mod_spandsp so i assume its somewhere in spandsp > > library... any idea where ? > > > > > > regards, > > Tihomir. > > > Spandsp just follows the practice in most FAX machines, and puts > p. in the page headers. Spandsp doesn't actually know the total > number of pages when it sends the first page, as it allows additional > TIFF files to be added to the transmission while a call is in progress. > / could be set up for the simple case of one call == one > TIFF file, but that's not in the code right now. > > Hi Steve, thanks for the prompt answer. yeah, i was afraid so... just for curiosity, where is the code that puts p. in the page header ? also, how do you send multiple tiff files during one call session ... something like txfax('file1','file2','file3'....) ? Regards, Tihomir. > Steve > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130227/3dcccff4/attachment.html From steveu at coppice.org Wed Feb 27 04:00:50 2013 From: steveu at coppice.org (Steve Underwood) Date: Wed, 27 Feb 2013 09:00:50 +0800 Subject: [Freeswitch-users] Fax document number of pages In-Reply-To: References: <512D50AD.2040206@coppice.org> Message-ID: <512D5AC2.1070109@coppice.org> On 02/27/2013 08:34 AM, Tihomir Culjaga wrote: > > > On Wed, Feb 27, 2013 at 1:17 AM, Steve Underwood > wrote: > > Hi Tihomir, > > On 02/26/2013 11:42 PM, Tihomir Culjaga wrote: > > hello, > > i know its stupid, but please, is there a way to somehow print the > > number of pages on a document in a / format > > instead of p. ? > > > > i cannot find this in mod_spandsp so i assume its somewhere in > spandsp > > library... any idea where ? > > > > > > regards, > > Tihomir. > > > Spandsp just follows the practice in most FAX machines, and puts > p. in the page headers. Spandsp doesn't actually know the > total > number of pages when it sends the first page, as it allows additional > TIFF files to be added to the transmission while a call is in > progress. > / could be set up for the simple case of one call == one > TIFF file, but that's not in the code right now. > > Hi Steve, thanks for the prompt answer. > > yeah, i was afraid so... just for curiosity, where is the code that > puts p. in the page header ? > > also, how do you send multiple tiff files during one call session ... > something like txfax('file1','file2','file3'....) ? > > Regards, > Tihomir. > A quick grep of the source shows it is in t4_tx.c. If you want to change it, be careful not to make your new string longer, or it won't fit. If the variable tiff.pages_in_file in the T.4 structure is > 0 it should contain the number of pages in the current TIFF file. However, that does not allow for the start_page and stop_page limitations, which are in the T.30 structure, and not directly available to the code in t4_tx.c. Steve From jefersonparanaense at gmail.com Wed Feb 27 04:16:21 2013 From: jefersonparanaense at gmail.com (Jeferson Rodrigo Almeida) Date: Tue, 26 Feb 2013 22:16:21 -0300 Subject: [Freeswitch-users] ESL phpmod for Windows? Message-ID: Does anybody have ever tried to compile the ESL phpmod in windows? Can we generate a dll file (esl.dll), and load it as a php extension? Is there any pre-compiled dll for this purpose? Any help will be greatly appreciated. Ps.: Sorry for the poor english... -- Jeferson Rodrigo Almeida Engenheiro de Computa??o jefersonparanaense at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130226/a2f58781/attachment.html From jpyle at fidelityvoice.com Wed Feb 27 05:01:16 2013 From: jpyle at fidelityvoice.com (Jeff Pyle) Date: Tue, 26 Feb 2013 21:01:16 -0500 Subject: [Freeswitch-users] t38 200OK timeout In-Reply-To: <20130226192614.c8767f514b9168395c611121@yandex.ru> References: <20130226192614.c8767f514b9168395c611121@yandex.ru> Message-ID: I also see this behavior with proxy-media enabled. With normal media processing it does not happen. Running 1.3.13b+git~20130208T191813Z~578f455741 (git 578f455 2013-02-08 19:18:13Z) on i386. - Jeff On Tue, Feb 26, 2013 at 10:26 AM, royj wrote: > > I wonder if anyone faced the issue that posted on jira - > http://jira.freeswitch.org/browse/FS-5120 > I understand employment of developers and hope Anthony or somebody else > will give the time soon, but may be here would be any thoughts. > > -- > regards > royj > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130226/9a875c16/attachment.html From anthony.minessale at gmail.com Wed Feb 27 08:07:40 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 26 Feb 2013 23:07:40 -0600 Subject: [Freeswitch-users] t38 200OK timeout In-Reply-To: References: <20130226192614.c8767f514b9168395c611121@yandex.ru> Message-ID: Its extra work to post both to the list and open a jira about the same thing. On Tue, Feb 26, 2013 at 8:01 PM, Jeff Pyle wrote: > I also see this behavior with proxy-media enabled. With normal media > processing it does not happen. > Running 1.3.13b+git~20130208T191813Z~578f455741 (git 578f455 2013-02-08 > 19:18:13Z) on i386. > > > - Jeff > > > On Tue, Feb 26, 2013 at 10:26 AM, royj wrote: > >> >> I wonder if anyone faced the issue that posted on jira - >> http://jira.freeswitch.org/browse/FS-5120 >> I understand employment of developers and hope Anthony or somebody else >> will give the time soon, but may be here would be any thoughts. >> >> -- >> regards >> royj >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130226/fe81544a/attachment-0001.html From anthony.minessale at gmail.com Wed Feb 27 08:32:21 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 26 Feb 2013 23:32:21 -0600 Subject: [Freeswitch-users] Silence suppression again In-Reply-To: <512CF844.5070509@digitalmail.com> References: <512CF844.5070509@digitalmail.com> Message-ID: It does not send any by default. Also, yes it only generates silence when the call is not getting any RTP. On Tue, Feb 26, 2013 at 12:00 PM, Alex Lake wrote: > Does Freeswitch enable any kind of silence suppression by default? If I > have bridge_generate_comfort_noise=true, does that only have an effect > when the audio-generating end decides to stop sending RTP? > > Any other tips for diagnosing silence (it's kind of like a slow noise > gate effect)? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130226/285d25fb/attachment.html From anthony.minessale at gmail.com Wed Feb 27 08:39:25 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 26 Feb 2013 23:39:25 -0600 Subject: [Freeswitch-users] new site, cant call OUT! In-Reply-To: References: <076e01ce0961$c14511d0$43cf3570$@bizfocused.com> <08f401ce0988$21b97e90$652c7bb0$@bizfocused.com> Message-ID: When it fails the phone is not getting the 407 triggering it to re-register with the auth info. The abandoned message is when it times out waiting for the 2nd INVITE. Try tracing it from where the phone is. On Tue, Feb 26, 2013 at 12:06 PM, Steven Schoch < schoch+freeswitch.org at xwin32.com> wrote: > On Tue, Feb 12, 2013 at 5:19 PM, Sean Devoy wrote: > >> Then I noticed the Cisco (which is in Gateway mode) has a WAN address of >> 10.1.10.10!!! I poked around and found the Comcast Business Router on the >> other side of the Cisco. Yippee DOUBLE NAT. I chose to just say ?NO.? >> > > We just got Comcast business class here. We have static addresses > assigned to our account, but the Comcast router/modem does both NAT and > regular routing at the same time. It's DHCP server will serve addresses > from 10.1.10.10 to 10.1.10.199. That's the Comcast fail safe - if you > "forget" to configure your interface to one of the static addresses they > give you, and leave it on DHCP, then it will still work. (At least until > you try to use VoIP!) > > -- > Steve > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130226/d18533c4/attachment.html From igor.cappello at edistar.com Wed Feb 27 12:01:14 2013 From: igor.cappello at edistar.com (Igor Cappello) Date: Wed, 27 Feb 2013 10:01:14 +0100 (CET) Subject: [Freeswitch-users] Problem setting up bind_digit_action In-Reply-To: Message-ID: <461554606.17147.1361955674129.JavaMail.root@mailserver.edistar.com> We managed to make it work, specifying the "peer" parameter in the data attribute of application bind_digit_action. Here's the working extension And here's the execution log http://pastebin.freeswitch.org/20635 We now face a related issue, though: we need to specify an extension to be executed on the bleg. Using execute_on_answer='execute_extension ...' seems to be the way to go, but we did not manage to make it work. Here are the extensions And here's the execution log: the extension MYSETUP doesn't seem to be executed properly on the bleg of the bridged call. http://pastebin.freeswitch.org/20636 Could you give us a hint? Thanks for your time, Igor Cappello > Please use http://pastebin.freeswitch.org/ > > On Tue, Feb 26, 2013 at 12:42 PM, Michael Collins > wrote: > > You need to set the digit action realm: > > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_digit_action_set_realm > > > > -MC > > From alex at digitalmail.com Wed Feb 27 13:31:27 2013 From: alex at digitalmail.com (Alex Lake) Date: Wed, 27 Feb 2013 10:31:27 +0000 Subject: [Freeswitch-users] Silence suppression again In-Reply-To: References: <512CF844.5070509@digitalmail.com> Message-ID: <512DE07F.3090503@digitalmail.com> Thanks. I've been looking at the channel variables of the call. And this is part of it: 10890919572121034455815480004323440018299081829908106391063900000 The thing that slightly concerns me here is 4323 That sounds as though the inbound audio path contains some comfort noise, suggesting that the handset the other end has some kind of silence suppression enabled. I'm afraid it's a dreaded Grandstream GXP2000 - which I know are not the handset of the cognoscenti! - but the customer tells me that he has silence suppression disabled. Do we think this is correct?! Rgds, Alex > It does not send any by default. Also, yes it only generates silence > when the call is not getting any RTP. > > > > On Tue, Feb 26, 2013 at 12:00 PM, Alex Lake > wrote: > > Does Freeswitch enable any kind of silence suppression by default? > If I > have bridge_generate_comfort_noise=true, does that only have an effect > when the audio-generating end decides to stop sending RTP? > > Any other tips for diagnosing silence (it's kind of like a slow noise > gate effect)? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > googletalk:conf+888 at conference.freeswitch.org > > pstn:+19193869900 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.2238 / Virus Database: 2641/5634 - Release Date: 02/26/13 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130227/95507f39/attachment-0001.html From 8f27e956 at gmail.com Wed Feb 27 13:33:06 2013 From: 8f27e956 at gmail.com (Scott) Date: Wed, 27 Feb 2013 05:33:06 -0500 Subject: [Freeswitch-users] Logical AND + NOT In-Reply-To: <1361865683.776896435@f250.mail.ru> References: <1361865683.776896435@f250.mail.ru> Message-ID: "regex#2", as written, returns TRUE providing that dest_num matches a '\d{3,16} pattern but is NOT 399 or 400. Therefore, 399 = false 400 = false 4001 = true 4005551212 = true 416 = true 4165551212 = true 1234 = true ANONYMOUS = false There is very, very little that CANNOT be done with native regex. If you need me to "explain" the regex construction, let me know. Cheers, /Scott On 26 February 2013 03:01, Dmitriy Shumaev wrote: > Hi > > I want to record all sessions, except calls to group numbers and to IVR. > So I need something like: > > > > > > > > > > > data="$${base_dir}/recordings/archive/${strftime(%Y-%m-%d-%H-%M-%S)}_${caller_id_name}_${caller_id_number}_2_${destination_number}.mp3"/> > > > > . But the syntax of condition does not allow the operator "not equal" or > "!=". What should I do? > > > With best regards, Shumaev DA, KBR Ltd. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130227/8d107e77/attachment.html From jdiaz at coinfru.com Wed Feb 27 14:44:39 2013 From: jdiaz at coinfru.com (Josue Diaz Cruz) Date: Wed, 27 Feb 2013 12:44:39 +0100 Subject: [Freeswitch-users] E169 question In-Reply-To: <4DF043DC-C88E-429A-B069-649A9F4B3862@endigotech.com> References: <4DF043DC-C88E-429A-B069-649A9F4B3862@endigotech.com> Message-ID: I am using it personally. I am in the first step setting up The system with Ubuntu 12.04 Server. I tried with CentOS. I work ussually with this distro, but i could not make it work because of my lack of understaning this gsmopen system. Yesterday i made it work with this: Ubuntu 12.04 Server AMD64 Huawey E169 Install all the Libs i found in the freeswitch wiki (all) I set up for the time been just only one dongle. ttyUSB1, ttyUSB2 Codec: G711 I made this dialplan: /* Because Spain Mobile */ /* Default config i have, changing just only tty ports */ My Experience: Calling with G711: no delay, audio quality same like with your own mobile (perfect), even with network operator low coverage were i am. Calling with G729: little delay (sensible). Probably i had the system not very well implemented. A very important thing we did: We try unlocking the dongle. We did it. With every system we test, no working. We soldered pinouts to a unlocking device. Reinstall complete firmware with this device. Then from this factory firmware we unlock it and activate the voice. I am experiencing with it. In my opinion for the time been works very nice. I need to stress the system with 8 dongles and live traffic to check all the issues that can happend. During this week i will do so many test. i will give you feed back. Cheers Josue Diaz Cruz Departamento Tecnico y Soporte jdiaz at coinfru.com www.coinfru.com Este e-mail contiene informaci?n confidencial, el contenido de la mismo se encuentra protegido por Ley. Cualquier persona distinta a su destinataria tiene prohibida su reproducci?n, uso, divulgaci?n o impresi?n total o parcial. Si ha recibido este mensaje por error, notifiquelo de inmediato al remitente borrando el mensaje original juntamente con sus ficheros anexos. Gracias. _____ De: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] En nombre de Brian Foster Enviado el: Wednesday, February 27, 2013 01:07 Para: Peter Hartmann CC: FreeSWITCH Users Help Asunto: Re: [Freeswitch-users] E169 question Hi, I was wondering if I could ask you some dongle questions. I've been testing out Gsmopen with an E160. Do you have any experience with that one and why do you prefer E169s? I personally have not. They were, however, recommended to me by a friend who has experience with Huawei-brand dongles. I'm in NYC. I might be interested in a group buy, but I'm still trying to figure out if it's going to reliable for clients. I currently cannot verify anything about the Huawei devices. The group buy actually fell through, and I was planning on getting one or two to play with. If others are interested in doing a group buy I'd be more than happy to facilitate. In your setup, what is your general purpose for having them in production? I planned on using a few for emergency backup in case the Internet or something else went kaput. With the advent of cell phones these days, it's really not an issue now. Otherwise, to me it's a toy. Now if you we're in Europe, there are advantages to using gsmopen and a dongle with a few different mobile providers to get free or reduced rate calling (within the same network). I'm CC'ing the Freeswitch users mailing list, in hopes that someone can better answer your questions. If you haven't already, subscribe to the ML by going to http://lists.freeswitch.org/mailman/listinfo/freeswitch-users. Good luck in your future endeavors, and welcome to the wonderful world of Freeswitch. Regards, Brian Foster Sent from my iPhone I'm waiting on an antenna at the moment....I suspect that's my current problem. Thanks in advance! Cheers, Peter Hartmann Hartmann Computer Consulting -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130227/b92af946/attachment-0001.html From mehroz.ashraf85 at gmail.com Wed Feb 27 17:47:05 2013 From: mehroz.ashraf85 at gmail.com (mehroz) Date: Wed, 27 Feb 2013 06:47:05 -0800 (PST) Subject: [Freeswitch-users] FS with SSL/TLS issues! In-Reply-To: <1361875157474-7588012.post@n2.nabble.com> References: <1361367187947-7587736.post@n2.nabble.com> <1361513236022-7587844.post@n2.nabble.com> <1361519970790-7587849.post@n2.nabble.com> <113D484C-EA7E-4611-B958-6514F5F2B5B9@freeswitch.org> <1361875157474-7588012.post@n2.nabble.com> Message-ID: <1361976425293-7588069.post@n2.nabble.com> TLS-NAT issue arises due to the fact that it tries to contact on other port (TCP session) as far as my understanding. Changing contact header can sort this issue and i have done it setting up for each directory (user). This helped and worked :) I suppose you should also have "ext-rtp-ip and "ext-sip-ip" = "nat-auto" in SIP profile. Cheeers! -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FS-with-SSL-TLS-issues-tp7587736p7588069.html Sent from the freeswitch-users mailing list archive at Nabble.com. From avi at avimarcus.net Wed Feb 27 03:14:43 2013 From: avi at avimarcus.net (Avi Marcus) Date: Wed, 27 Feb 2013 02:14:43 +0200 Subject: [Freeswitch-users] Monitoring Phone Endpoint Deregistration/Losing Connection Message-ID: I'd like to monitor if my user's sip phones are disconnected, to let me know there might be a problem in advance. I see there's a CUSTOM event called sofia::unregister. I presume that triggers whenever a phone doesn't re-register in enough time. However, with default registation of 60 minutes (or even 10) I'd like more granularity. I have the endpoints pinging FS, but I don't see an event or that information stored anywhere. How about setting nat-options-ping (or better, all-reg-options-ping since most NATed endpoint don't actually self-identify). Do I get an event if that fails? I see unregister-on-options-fail but that seemed too aggressive when I tried that in the past. What exactly does the less sever mark-dead-on-options-fail do? Does it remove it from the user/$NUMBER endpoint? Does it trigger an event for me? Other suggestions? -Avi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130227/47a09b6e/attachment.html From msc at freeswitch.org Wed Feb 27 19:50:26 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 27 Feb 2013 08:50:26 -0800 Subject: [Freeswitch-users] FreeSWITCH Conference Call Today Message-ID: Hello all. Today's conference call agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2013_02_27 We will have an update on the mod_ha_cluster discussion and then Ken Rice will be showing off his SwitchPi project. Talk to you soon! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130227/b528bdd1/attachment.html From alex at digitalmail.com Wed Feb 27 20:44:07 2013 From: alex at digitalmail.com (Alex Lake) Date: Wed, 27 Feb 2013 17:44:07 +0000 Subject: [Freeswitch-users] Silence suppression again In-Reply-To: <512DE07F.3090503@digitalmail.com> References: <512CF844.5070509@digitalmail.com> <512DE07F.3090503@digitalmail.com> Message-ID: <512E45E7.2070004@digitalmail.com> A quick update on this - I have verified that suppresion of silence suppression(!) on Grandstreams doesn't work when it is the B-Party of a call. I'm not sure if silence-suppression is part of the SIP handshake, but I know that other VOIP technologies seem to be able to prevent it. I'm calling in a handset specialist to advise.. One of the things that intrigues me is this extract from the log (particularly the reference to silenceSupp): 2013-02-27 17:28:05.151624 [DEBUG] switch_core_codec.c:244 sofia/internal/0253302 at 004-0253.sb12.dmclub.org Restore previous codec PCMA:8. 2013-02-27 17:28:05.151624 [DEBUG] mod_sofia.c:754 Local SDP sofia/internal/0253302 at 004-0253.sb12.dmclub.org: v=0 o=FreeSWITCH 1361975871 1361975873 IN IP4 176.58.88.201 s=FreeSWITCH c=IN IP4 176.58.88.201 t=0 0 m=audio 10212 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:30 a=sendrecv What does all this mean? From the "A" leg of the call (which is placed by a LinkSys SPA922) I have: v%3D0%0Ao%3DFreeSWITCH%201361974765%201361974767%20IN%20IP4%20176.58.88.201%0As%3DFreeSWITCH%0Ac%3DIN%20IP4%20176.58.88.201%0At%3D0%200%0Am%3Daudio%2011468%20RTP/AVP%208%20101%0Aa%3Drtpmap%3A8%20PCMA/8000%0Aa%3Drtpmap%3A101%20telephone-event/8000%0Aa%3Dfmtp%3A101%200-16%0Aa%3DsilenceSupp%3Aoff%20-%20-%20-%20-%0Aa%3Dptime%3A30%0Aa%3Dsendrecv%0A From the troublesome "B" leg of the call (placed by FreeSwitch to my GXP2000) I have: sip_local_sdp_str>v%3D0%0Ao%3DFreeSWITCH%201361974745%201361974746%20IN%20IP4%20176.58.88.201%0As%3DFreeSWITCH%0Ac%3DIN%20IP4%20176.58.88.201%0At%3D0%200%0Am%3Daudio%2011488%20RTP/AVP%208%20101%2013%0Aa%3Drtpmap%3A101%20telephone-event/8000%0Aa%3Dfmtp%3A101%200-16%0Aa%3Dptime%3A30%0Aa%3Dsendrecv%0Am%3Daudio%2011488%20RTP/AVP%208%203%20101%2013%0Aa%3Drtpmap%3A101%20telephone-event/8000%0Aa%3Dfmtp%3A101%200-16%0Aa%3Dptime%3A20%0Aa%3Dsendrecv%0A I was wondering if there's a way I can modify this to include a="silenceSupp:off - - - -" and whether that would make the slightest bit of difference to the handset's behaviour! Alex > Thanks. > I've been looking at the channel variables of the call. And this is > part of it: > > 1089091957212103445581548000432344001829908182990810639 _media_packet_count>1063900000 > > The thing that slightly concerns me here is > 4323 > > That sounds as though the inbound audio path contains some comfort > noise, suggesting that the handset the other end has some kind of > silence suppression enabled. > > I'm afraid it's a dreaded Grandstream GXP2000 - which I know are not > the handset of the cognoscenti! - but the customer tells me that he > has silence suppression disabled. Do we think this is correct?! > > Rgds, > Alex >> It does not send any by default. Also, yes it only generates silence >> when the call is not getting any RTP. >> >> >> >> On Tue, Feb 26, 2013 at 12:00 PM, Alex Lake > > wrote: >> >> Does Freeswitch enable any kind of silence suppression by >> default? If I >> have bridge_generate_comfort_noise=true, does that only have an >> effect >> when the audio-generating end decides to stop sending RTP? >> >> Any other tips for diagnosing silence (it's kind of like a slow noise >> gate effect)? >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> >> googletalk:conf+888 at conference.freeswitch.org >> >> pstn:+19193869900 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> >> No virus found in this message. >> Checked by AVG - www.avg.com >> Version: 2012.0.2238 / Virus Database: 2641/5634 - Release Date: 02/26/13 >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.2238 / Virus Database: 2641/5635 - Release Date: 02/26/13 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130227/7d02936d/attachment-0001.html From tomasz.szuster at gmail.com Wed Feb 27 21:08:59 2013 From: tomasz.szuster at gmail.com (Tomasz Szuster) Date: Wed, 27 Feb 2013 19:08:59 +0100 Subject: [Freeswitch-users] nibblebill, spidermonkey_odbc issue In-Reply-To: References: <361E98F99D3CC3439EED59BC1924ED6971B0B2@SERVER2003.SecuReachSystems.local> Message-ID: Thank you all for fast reply. In nibblebill.xml I've changed ** to ** and now it is working. Thank you for all advices. Regards. Tom. On Tue, Feb 26, 2013 at 11:14 PM, Ken Rice wrote: > Also, I didn?t read the whole email earlier.. Spidermonkey odbc and > nibblebill odbc are 2 different config settings... You need to check the > wiki for the proper odbc dsn syntax and make sure the DSN you configured in > your odbc.ini works from isql > > K > > > > On 2/26/13 4:00 PM, "Jason Moran" wrote: > > Can you make connections to your database using the ODBC connection > outside of FreeSWITCH (but from the same server that FS is installed on)? > I?ve often caught problems either in my firewall or a dumb typo in my ODBC > configurations. > > > *From:* Tomasz Szuster [mailto:tomasz.szuster at gmail.com] > > *Sent:* Tuesday, February 26, 2013 3:48 PM > *To:* FreeSWITCH Users Help > *Subject:* [Freeswitch-users] nibblebill, spidermonkey_odbc issue > > Hi, > > > > I'm struggling with making nibblebill working. > > What I've did till now is: > > > > Installed odbc: > > * libmyodbc > > * libodbc1 > > * odbcinst > > * odbcinst1debian2 > > * unixodbc > > * unixodbc-dev > > > > compile freeswitch using > > > > > > ./configure --enable-core-odbc-support > make; make install > > > My spidermonkey.conf file has: > > > > > > > > > > > > > > > > > > > > > > > > My odbc.ini: > > > > [nibblebill1] > > Driver = /usr/lib/x86_64-linux-gnu/odbc/libmyodbc.so > > SERVER = callcenter > > PORT = 3306 > > DATABASE = nibblebill1 > > OPTION = 67108864 > > USER = nibblebill1 > > PASSWORD = XXXXXXX > > > > > > > > > > ldd /usr/local/freeswitch/mod/mod_spidermonkey_odbc.so > > linux-vdso.so.1 => (0x00007fffbd7ff000) > > libfreeswitch.so.1 => /usr/local/freeswitch/lib/libfreeswitch.so.1 > (0x00007f2f193ee000) > > libjs.so.1 => /usr/local/freeswitch/lib/libjs.so.1 > (0x00007f2f19120000) > > libnspr4.so => /usr/local/freeswitch/lib/libnspr4.so > (0x00007f2f18eef000) > > libodbc.so.1 => /usr/lib/x86_64-linux-gnu/libodbc.so.1 > (0x00007f2f18c82000) > > libpthread.so.0 => /lib/x86_64-linux-gnu/libpthread.so.0 > (0x00007f2f18a65000) > > libc.so.6 => /lib/x86_64-linux-gnu/libc.so.6 (0x00007f2f186a5000) > > libdl.so.2 => /lib/x86_64-linux-gnu/libdl.so.2 (0x00007f2f184a1000) > > libcrypt.so.1 => /lib/x86_64-linux-gnu/libcrypt.so.1 > (0x00007f2f18268000) > > librt.so.1 => /lib/x86_64-linux-gnu/librt.so.1 (0x00007f2f1805f000) > > libssl.so.1.0.0 => /lib/x86_64-linux-gnu/libssl.so.1.0.0 > (0x00007f2f17e03000) > > libcrypto.so.1.0.0 => /lib/x86_64-linux-gnu/libcrypto.so.1.0.0 > (0x00007f2f17a3b000) > > libtinfo.so.5 => /lib/x86_64-linux-gnu/libtinfo.so.5 > (0x00007f2f17813000) > > libstdc++.so.6 => /usr/lib/x86_64-linux-gnu/libstdc++.so.6 > (0x00007f2f17513000) > > libm.so.6 => /lib/x86_64-linux-gnu/libm.so.6 (0x00007f2f17217000) > > libgcc_s.so.1 => /lib/x86_64-linux-gnu/libgcc_s.so.1 > (0x00007f2f17000000) > > libltdl.so.7 => /usr/lib/x86_64-linux-gnu/libltdl.so.7 > (0x00007f2f16df6000) > > /lib64/ld-linux-x86-64.so.2 (0x00007f2f19a48000) > > libz.so.1 => /lib/x86_64-linux-gnu/libz.so.1 (0x00007f2f16bde000) > > > > > FreeSWITCH (Version 1.2.6 git a424765 2013-01-04 15:45:59Z) > > > > When I try to run* load mod_spidermonkey_odbc* I get: > > > > [CRIT] switch_loadable_module.c:1330 Error Loading module > /usr/local/freeswitch/mod/mod_spidermonkey_odbc.so > **/usr/local/freeswitch/mod/mod_spidermonkey_odbc.so: undefined symbol: > mod_spidermonkey_odbc_module_interface** > > > > *load mod_nibblebill: > * > > > 2013-02-26 21:46:40.116678 [ERR] switch_odbc.c:365 STATE: IM002 CODE 0 > ERROR: [unixODBC][Driver Manager]Data source name not found, and no default > driver specified > > > > 2013-02-26 21:46:40.116678 [CRIT] mod_nibblebill.c:220 Cannot connect to > ODBC driver/database odbc://callcenter (user: nibblebill1 / pass XXXXX)! > > 2013-02-26 21:46:40.116678 [CONSOLE] switch_loadable_module.c:1348 > Successfully Loaded [mod_nibblebill] > > > > > > Also from time to time in logs I've see: > > > > [ERR] switch_odbc.c:365 STATE: IM002 CODE 0 ERROR: [unixODBC][Driver > Manager]Data source name not found, and no default driver specified > > > > Will you be able to help with this issue ? > > Thank you. > > > -- > Ken > *http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > *irc.freenode.net #freeswitch > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Pozdrawiam Tomasz -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130227/328172d4/attachment-0001.html From vittorio.guglielmo at gmail.com Thu Feb 28 00:32:57 2013 From: vittorio.guglielmo at gmail.com (Vittorio Guglielmo) Date: Wed, 27 Feb 2013 22:32:57 +0100 Subject: [Freeswitch-users] FS with SSL/TLS issues! In-Reply-To: <1361976425293-7588069.post@n2.nabble.com> References: <1361367187947-7587736.post@n2.nabble.com> <1361513236022-7587844.post@n2.nabble.com> <1361519970790-7587849.post@n2.nabble.com> <113D484C-EA7E-4611-B958-6514F5F2B5B9@freeswitch.org> <1361875157474-7588012.post@n2.nabble.com> <1361976425293-7588069.post@n2.nabble.com> Message-ID: <512E7B89.6030903@gmail.com> Hi, I've done all you said but I'm in trouble indeed: TLS/SRTP enabled on two extension; If I try to dial between the two extensions I have: /mod_dptools.c:3052 Originate Failed. Cause: INCOMPATIBLE_DESTINATION/ If I call outbound, it's ok, and the call is encrypted. Can you help me? Vic On 27/02/13 15:47, mehroz wrote: > TLS-NAT issue arises due to the fact that it tries to contact on other port > (TCP session) as far as my understanding. > > Changing contact header can sort this issue and i have done it setting up > > > for each directory (user). > > This helped and worked :) > > I suppose you should also have "ext-rtp-ip and "ext-sip-ip" = "nat-auto" > in SIP profile. > Cheeers! > > > > -- > View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FS-with-SSL-TLS-issues-tp7587736p7588069.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130227/2e94376d/attachment.html From anthony.minessale at gmail.com Thu Feb 28 00:54:02 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 27 Feb 2013 15:54:02 -0600 Subject: [Freeswitch-users] FS with SSL/TLS issues! In-Reply-To: <512E7B89.6030903@gmail.com> References: <1361367187947-7587736.post@n2.nabble.com> <1361513236022-7587844.post@n2.nabble.com> <1361519970790-7587849.post@n2.nabble.com> <113D484C-EA7E-4611-B958-6514F5F2B5B9@freeswitch.org> <1361875157474-7588012.post@n2.nabble.com> <1361976425293-7588069.post@n2.nabble.com> <512E7B89.6030903@gmail.com> Message-ID: This will work better in 1.4 but the code in 1.2 is frozen so we can't improve it there. In the meantime you can just do this: make an acl that is default allow called tls add it to apply-nat-acl param in sofia profile where you have tls Now it will record the rport for every client in the contact path and it should resolve the existing connection. On Wed, Feb 27, 2013 at 3:32 PM, Vittorio Guglielmo < vittorio.guglielmo at gmail.com> wrote: > Hi, > > I've done all you said but I'm in trouble indeed: > > TLS/SRTP enabled on two extension; > > If I try to dial between the two extensions I have: > > *mod_dptools.c:3052 Originate Failed. Cause: INCOMPATIBLE_DESTINATION* > > If I call outbound, it's ok, and the call is encrypted. > > Can you help me? > > Vic > > > > > On 27/02/13 15:47, mehroz wrote: > > TLS-NAT issue arises due to the fact that it tries to contact on other port > (TCP session) as far as my understanding. > > Changing contact header can sort this issue and i have done it setting up > > > for each directory (user). > > This helped and worked :) > > I suppose you should also have "ext-rtp-ip and "ext-sip-ip" = "nat-auto" > in SIP profile. > Cheeers! > > > > -- > View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FS-with-SSL-TLS-issues-tp7587736p7588069.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130227/f5eb5137/attachment.html From vittorio.guglielmo at gmail.com Thu Feb 28 01:13:42 2013 From: vittorio.guglielmo at gmail.com (Vittorio Guglielmo) Date: Wed, 27 Feb 2013 23:13:42 +0100 Subject: [Freeswitch-users] FS with SSL/TLS issues! In-Reply-To: References: <1361367187947-7587736.post@n2.nabble.com> <1361513236022-7587844.post@n2.nabble.com> <1361519970790-7587849.post@n2.nabble.com> <113D484C-EA7E-4611-B958-6514F5F2B5B9@freeswitch.org> <1361875157474-7588012.post@n2.nabble.com> <1361976425293-7588069.post@n2.nabble.com> <512E7B89.6030903@gmail.com> Message-ID: <512E8516.1060008@gmail.com> I did it, now I have: /mod_dptools.c:3052 Originate Failed. Cause: INCOMPATIBLE_DESTINATION/ The only way I succeded was to add /bypass_media=true/ in the dialplan for each extension, but I have a huge latency in the voice (like 2/3 seconds). Vic On 27/02/13 22:54, Anthony Minessale wrote: > This will work better in 1.4 but the code in 1.2 is frozen so we can't > improve it there. > In the meantime you can just do this: > > make an acl that is default allow called tls > add it to apply-nat-acl param in sofia profile where you have tls > > Now it will record the rport for every client in the contact path and > it should resolve the existing connection. > > > > > On Wed, Feb 27, 2013 at 3:32 PM, Vittorio Guglielmo > > > wrote: > > Hi, > > I've done all you said but I'm in trouble indeed: > > TLS/SRTP enabled on two extension; > > If I try to dial between the two extensions I have: > > /mod_dptools.c:3052 Originate Failed. Cause: > INCOMPATIBLE_DESTINATION/ > > If I call outbound, it's ok, and the call is encrypted. > > Can you help me? > > Vic > > > > > On 27/02/13 15:47, mehroz wrote: >> TLS-NAT issue arises due to the fact that it tries to contact on other port >> (TCP session) as far as my understanding. >> >> Changing contact header can sort this issue and i have done it setting up >> >> >> for each directory (user). >> >> This helped and worked :) >> >> I suppose you should also have "ext-rtp-ip and "ext-sip-ip" = "nat-auto" >> in SIP profile. >> Cheeers! >> >> >> >> -- >> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FS-with-SSL-TLS-issues-tp7587736p7588069.html >> Sent from the freeswitch-users mailing list archive at Nabble.com. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > googletalk:conf+888 at conference.freeswitch.org > > pstn:+19193869900 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130227/26181511/attachment-0001.html From bdfoster at endigotech.com Thu Feb 28 01:16:08 2013 From: bdfoster at endigotech.com (Brian Foster) Date: Wed, 27 Feb 2013 17:16:08 -0500 Subject: [Freeswitch-users] E169 question In-Reply-To: References: <4DF043DC-C88E-429A-B069-649A9F4B3862@endigotech.com> Message-ID: > Hey Brian, > Thanks for getting back to me. > >> In your setup, what is your general purpose for having them in production? > > A POTS replacement for small businesses (PBX-users) that doesn't > piggy-back on another service like VOIP does. That's the dream > anyway. :P I don't think I would use that for the only means of dialing out/in, but you know your clients better than I do. It's a great backup for VoIP services though. Read below for a response from the ML. Take care! - BDF >> I'm CC'ing the Freeswitch users mailing list, in hopes that someone can >> better answer your questions. If you haven't already, subscribe to the ML by >> going to http://lists.freeswitch.org/mailman/listinfo/freeswitch-users. > > > Thanks! > >> Good luck in your future endeavors, and welcome to the wonderful world of >> Freeswitch. > > > Thank you, I'm loving it! > > > > Peter Hartmann > Hartmann Computer Consulting > > > On Tue, Feb 26, 2013 at 7:07 PM, Brian Foster wrote: >> >> >> Hi, >> I was wondering if I could ask you some dongle questions. I've been >> testing out Gsmopen with an E160. Do you have any experience with >> that one and why do you prefer E169s? >> >> >> I personally have not. They were, however, recommended to me by a friend who >> has experience with Huawei-brand dongles. >> >> I'm in NYC. I might be >> interested in a group buy, but I'm still trying to figure out if it's >> going to reliable for clients. >> >> >> I currently cannot verify anything about the Huawei devices. The group buy >> actually fell through, and I was planning on getting one or two to play >> with. If others are interested in doing a group buy I'd be more than happy >> to facilitate. >> >> In your setup, what is your general purpose for having them in production? I >> planned on using a few for emergency backup in case the Internet or >> something else went kaput. With the advent of cell phones these days, it's >> really not an issue now. Otherwise, to me it's a toy. Now if you we're in >> Europe, there are advantages to using gsmopen and a dongle with a few >> different mobile providers to get free or reduced rate calling (within the >> same network). >> >> I'm CC'ing the Freeswitch users mailing list, in hopes that someone can >> better answer your questions. If you haven't already, subscribe to the ML by >> going to http://lists.freeswitch.org/mailman/listinfo/freeswitch-users. >> >> Good luck in your future endeavors, and welcome to the wonderful world of >> Freeswitch. >> >> Regards, >> >> Brian Foster >> >> Sent from my iPhone >> >> I'm waiting on an antenna at the >> moment....I suspect that's my current problem. >> >> >> Thanks in advance! >> >> Cheers, >> >> >> Peter Hartmann >> Hartmann Computer Consulting From anthony.minessale at gmail.com Thu Feb 28 01:20:02 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 27 Feb 2013 16:20:02 -0600 Subject: [Freeswitch-users] Silence suppression again In-Reply-To: <512E45E7.2070004@digitalmail.com> References: <512CF844.5070509@digitalmail.com> <512DE07F.3090503@digitalmail.com> <512E45E7.2070004@digitalmail.com> Message-ID: suppress_cng=true variable or supress-cng profile param set to true. On Wed, Feb 27, 2013 at 11:44 AM, Alex Lake wrote: > A quick update on this - I have verified that suppresion of silence > suppression(!) on Grandstreams doesn't work when it is the B-Party of a > call. I'm not sure if silence-suppression is part of the SIP handshake, but > I know that other VOIP technologies seem to be able to prevent it. > I'm calling in a handset specialist to advise.. > > One of the things that intrigues me is this extract from the log > (particularly the reference to silenceSupp): > > 2013-02-27 17:28:05.151624 [DEBUG] switch_core_codec.c:244 > sofia/internal/0253302 at 004-0253.sb12.dmclub.org Restore previous codec > PCMA:8. > 2013-02-27 17:28:05.151624 [DEBUG] mod_sofia.c:754 Local SDP > sofia/internal/0253302 at 004-0253.sb12.dmclub.org: > v=0 > o=FreeSWITCH 1361975871 1361975873 IN IP4 176.58.88.201 > s=FreeSWITCH > c=IN IP4 176.58.88.201 > t=0 0 > m=audio 10212 RTP/AVP 8 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:30 > a=sendrecv > > What does all this mean? > > From the "A" leg of the call (which is placed by a LinkSys SPA922) I have: > > > v%3D0%0Ao%3DFreeSWITCH%201361974765%201361974767%20IN%20IP4%20176.58.88.201%0As%3DFreeSWITCH%0Ac%3DIN%20IP4%20176.58.88.201%0At%3D0%200%0Am%3Daudio%2011468%20RTP/AVP%208%20101%0Aa%3Drtpmap%3A8%20PCMA/8000%0Aa%3Drtpmap%3A101%20telephone-event/8000%0Aa%3Dfmtp%3A101%200-16%0Aa%3DsilenceSupp%3Aoff%20-%20-%20-%20-%0Aa%3Dptime%3A30%0Aa%3Dsendrecv%0A > > From the troublesome "B" leg of the call (placed by FreeSwitch to my > GXP2000) I have: > > > sip_local_sdp_str>v%3D0%0Ao%3DFreeSWITCH%201361974745%201361974746%20IN%20IP4%20176.58.88.201%0As%3DFreeSWITCH%0Ac%3DIN%20IP4%20176.58.88.201%0At%3D0%200%0Am%3Daudio%2011488%20RTP/AVP%208%20101%2013%0Aa%3Drtpmap%3A101%20telephone-event/8000%0Aa%3Dfmtp%3A101%200-16%0Aa%3Dptime%3A30%0Aa%3Dsendrecv%0Am%3Daudio%2011488%20RTP/AVP%208%203%20101%2013%0Aa%3Drtpmap%3A101%20telephone-event/8000%0Aa%3Dfmtp%3A101%200-16%0Aa%3Dptime%3A20%0Aa%3Dsendrecv%0A > > I was wondering if there's a way I can modify this to include > a="silenceSupp:off - - - -" and whether that would make the slightest bit > of difference to the handset's behaviour! > > > > Alex > > Thanks. > I've been looking at the channel variables of the call. And this is part > of it: > > 1089091957212103445581548000432344001829908182990810639 _media_packet_count>1063900000 > > The thing that slightly concerns me here is > 4323 > > That sounds as though the inbound audio path contains some comfort noise, > suggesting that the handset the other end has some kind of silence > suppression enabled. > > I'm afraid it's a dreaded Grandstream GXP2000 - which I know are not the > handset of the cognoscenti! - but the customer tells me that he has silence > suppression disabled. Do we think this is correct?! > > Rgds, > Alex > > It does not send any by default. Also, yes it only generates silence when > the call is not getting any RTP. > > > > On Tue, Feb 26, 2013 at 12:00 PM, Alex Lake wrote: > >> Does Freeswitch enable any kind of silence suppression by default? If I >> have bridge_generate_comfort_noise=true, does that only have an effect >> when the audio-generating end decides to stop sending RTP? >> >> Any other tips for diagnosing silence (it's kind of like a slow noise >> gate effect)? >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.2238 / Virus Database: 2641/5634 - Release Date: 02/26/13 > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.2238 / Virus Database: 2641/5635 - Release Date: 02/26/13 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130227/4143013c/attachment-0001.html From anthony.minessale at gmail.com Thu Feb 28 01:22:29 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 27 Feb 2013 16:22:29 -0600 Subject: [Freeswitch-users] FS with SSL/TLS issues! In-Reply-To: <512E8516.1060008@gmail.com> References: <1361367187947-7587736.post@n2.nabble.com> <1361513236022-7587844.post@n2.nabble.com> <1361519970790-7587849.post@n2.nabble.com> <113D484C-EA7E-4611-B958-6514F5F2B5B9@freeswitch.org> <1361875157474-7588012.post@n2.nabble.com> <1361976425293-7588069.post@n2.nabble.com> <512E7B89.6030903@gmail.com> <512E8516.1060008@gmail.com> Message-ID: Now you are just stuck with ordinary problems with how you have things configured. One line is not enough to tell you exactly why but a good guess is you have codecs that don't match. This is unrelated to TLS. On Wed, Feb 27, 2013 at 4:13 PM, Vittorio Guglielmo < vittorio.guglielmo at gmail.com> wrote: > I did it, > now I have: > > > *mod_dptools.c:3052 Originate Failed. Cause: INCOMPATIBLE_DESTINATION* > > The only way I succeded was to add *bypass_media=true* in the dialplan > for each extension, > but I have a huge latency in the voice (like 2/3 seconds). > > > Vic > > > > > On 27/02/13 22:54, Anthony Minessale wrote: > > This will work better in 1.4 but the code in 1.2 is frozen so we can't > improve it there. > In the meantime you can just do this: > > make an acl that is default allow called tls > add it to apply-nat-acl param in sofia profile where you have tls > > Now it will record the rport for every client in the contact path and it > should resolve the existing connection. > > > > > On Wed, Feb 27, 2013 at 3:32 PM, Vittorio Guglielmo < > vittorio.guglielmo at gmail.com> wrote: > >> Hi, >> >> I've done all you said but I'm in trouble indeed: >> >> TLS/SRTP enabled on two extension; >> >> If I try to dial between the two extensions I have: >> >> *mod_dptools.c:3052 Originate Failed. Cause: INCOMPATIBLE_DESTINATION* >> >> If I call outbound, it's ok, and the call is encrypted. >> >> Can you help me? >> >> Vic >> >> >> >> >> On 27/02/13 15:47, mehroz wrote: >> >> TLS-NAT issue arises due to the fact that it tries to contact on other port >> (TCP session) as far as my understanding. >> >> Changing contact header can sort this issue and i have done it setting up >> >> >> for each directory (user). >> >> This helped and worked :) >> >> I suppose you should also have "ext-rtp-ip and "ext-sip-ip" = "nat-auto" >> in SIP profile. >> Cheeers! >> >> >> >> -- >> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FS-with-SSL-TLS-issues-tp7587736p7588069.html >> Sent from the freeswitch-users mailing list archive at Nabble.com. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >> >> >> >> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com >> >> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130227/ec03ef6a/attachment.html From jmoran at secureachsystems.com Thu Feb 28 01:36:12 2013 From: jmoran at secureachsystems.com (Jason Moran) Date: Wed, 27 Feb 2013 17:36:12 -0500 Subject: [Freeswitch-users] Nuance Speech Server / VocalizerMRCPv2mod_unimrcp simultaneous licenses not releasing quickly enough. References: <361E98F99D3CC3439EED59BC1924ED6971B08A@SERVER2003.SecuReachSystems.local><95382631-D364-43A5-B93F-E553F68248F9@gmail.com><361E98F99D3CC3439EED59BC1924ED6971B0AE@SERVER2003.SecuReachSystems.local> Message-ID: <361E98F99D3CC3439EED59BC1924ED6971B0BC@SERVER2003.SecuReachSystems.local> MRCPv1 works - it wasn't my MRCP config at all. I was issuing "reloadxml" and that did not appear to do the trick, but fully stopping FS and restarting it worked fine (after about 10 hours of trying hundreds of permutations of configs..) I think we'll be using MRCPv1 over MRCPv2. However, going back to the initial problem of MRCPv2 eating connections quickly: Changing the following settings in NSSserver.cfg with Nuance Speech Server made MRCPv2 stop eating the connections: server.mrcp2.sip.contact.useHostIPAddress 1 server.mrcp2.rsspeechsynth.sendRtcpByeOnPromptDone 1 Jason From: Christopher Rienzo [mailto:cmrienzo at gmail.com] Sent: Tuesday, February 26, 2013 6:48 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Nuance Speech Server / VocalizerMRCPv2mod_unimrcp simultaneous licenses not releasing quickly enough. I used MRCPv1 with remote servers. 3 MRCP servers handling requests from 12 FS servers. Start with the example nuance MRCPv1 configuration in conf/mrcp_profiles and replace the rtp-ip, server-ip, and server-port. Use IP address, not hostnames, not "auto". If that doesn't work, reply with your configuration. Chris On Tue, Feb 26, 2013 at 2:33 PM, Jason Moran wrote: After trying MRCPv1 - I don't think anybody out there is using Nuance for TTS remotely / externally! The audio is always blank / silence(both Nuance and FreeSWITCH seem to think it is working ok). I can get it to work with Nuance on localhost/127.0.0.1, but I believe there is a problem with FreeSWITCH/UniMRCP determining the correct rtp-ip when in MRCPv1 mode when using a server other than localhost. As shown in earlier posts in this thread, I can get it to work with MRCPv2 on the same external server. Regardless of all of these settings (client-ext-ip , client-ip , rtp-ip , rtp-ext-ip being set to an IP address other than 127.0.0.1, RTP is still trying to use 127.0.0.1, although RTSP is using the external IP address provided. mpf_rtp_stream.c from uniMRCP returns several references to 127.0.0.1 although it should not: mpf_rtp_stream.c:495 () Close RTP Receiver 127.0.0.1:4018 <- 10.0.0.67:7914 [r:0 l:0 j:0 d:0 i:0] and mpf_rtp_stream.c:1294 () Send Compound RTCP Packet [BYE] [76 bytes] 127.0.0.1:4019 -> 10.0.0.67:7915 mpf_rtp_stream.c:1306 () Failed to Send Compound RTCP Packet [BYE] [0 bytes] 127.0.0.1:4019 -> 10.0.0.67:7915 I believe this explains why I cannot hear the audio that is generated, as it should be using a 10.0.0.x IP address to receive and sent over RTP. To be safe I also changed the external_rtp_ip in vars.xml, but that hasn't fixed the issue either. Does anybody know where mpf_rtp_stream.c is getting this RTP IP Address from? Why does it insist on 127.0.0.1 despite all settings saying otherwise? Jason From: Christopher Rienzo [mailto:cmrienzo at gmail.com] Sent: Monday, February 25, 2013 8:34 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Nuance Speech Server / Vocalizer MRCPv2mod_unimrcp simultaneous licenses not releasing quickly enough. You can also try MRCPv1 (RTSP) and see if there is better performance. Using SIP to set up an MRCP session is silly. Chris On Mon, Feb 25, 2013 at 6:42 PM, Michael Collins wrote: Another case of DRM hurting only those who've paid for it... *sigh* I recommend calling your VAR or Nuance and gently but firmly insisting on a resolution. -MC On Mon, Feb 25, 2013 at 2:21 PM, wrote: We also had an issue with nuance eating licenses. We had enough room to tolerate this but did have to occasionally reboot servers when too many were consumed. Chris On Feb 25, 2013, at 3:21 PM, Rob Hutton wrote: We have a similar situation on a Cisco UCCX platform that we ended up having to write a script to bounce the services every couple of hours to clear. We haven't seen a fix from Nuance yet. On Mon, Feb 25, 2013 at 2:32 PM, Jason Moran wrote: In connecting to Nuance Speech Server's MRCPv2 for text-to-speech (Nuance Vocalizer) using mod_unimrcp, a single phone call which has several different TTS phrases should only use up 1 single simultaneous license. However, I am seeing a call use up 2,3,4 (etc) like it is not releasing the connection/license. I looked this up and some forums have noted that Nuance "does things wrong" and uniMRCP needs to send Nuance the "RTCP BYE" to tell Nuance that it is done with the synthesizer speech. But I tried that, and it still eats up the licenses. Within a couple of calls I'm out of TTS ports and the calls fail (but they clear again after several more seconds). Ideas? Experience? -Jason ________________________________________________________________________ _ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 ________________________________________________________________________ _ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 ________________________________________________________________________ _ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org ________________________________________________________________________ _ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 ________________________________________________________________________ _ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130227/dd42905b/attachment-0001.html From mikebemail at gmail.com Thu Feb 28 00:32:53 2013 From: mikebemail at gmail.com (mikeborschow) Date: Wed, 27 Feb 2013 13:32:53 -0800 (PST) Subject: [Freeswitch-users] Freeswitch/FusionPBX error opening *.xml Message-ID: <1362000773371-7588074.post@n2.nabble.com> I am getting this error using Freeswitch / Fusion. Any ideas. I used Fusion default install script. [ERR] switch_xml.c:1331 Couldnt open /usr/local/freeswitch/conf/dialplan/public/*.xml (No such file or directory) Running on Ubuntu 12.04 Any help will be appreciated. Thanks. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-FusionPBX-error-opening-xml-tp7588074.html Sent from the freeswitch-users mailing list archive at Nabble.com. From jeff at jefflenk.com Thu Feb 28 01:48:18 2013 From: jeff at jefflenk.com (Jeff Lenk) Date: Wed, 27 Feb 2013 14:48:18 -0800 (PST) Subject: [Freeswitch-users] ESL phpmod for Windows? In-Reply-To: References: Message-ID: <1362005298077-7588082.post@n2.nabble.com> This does not exist as far as I know unless someone contributes it. Lua and managed c# have existing windows support. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/ESL-phpmod-for-Windows-tp7588060p7588082.html Sent from the freeswitch-users mailing list archive at Nabble.com. From steveayre at gmail.com Thu Feb 28 01:49:17 2013 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 27 Feb 2013 22:49:17 +0000 Subject: [Freeswitch-users] Freeswitch/FusionPBX error opening *.xml In-Reply-To: <1362000773371-7588074.post@n2.nabble.com> References: <1362000773371-7588074.post@n2.nabble.com> Message-ID: The XML files in numerous places include files in directories. Looks like public dialplan (/usr/local/freeswitch/conf/dialplan/public.xml) is including extensions in /usr/local/freeswitch/conf/dialplan/public but no files exist in that directory. -Steve On 27 February 2013 21:32, mikeborschow wrote: > I am getting this error using Freeswitch / Fusion. Any ideas. I used > Fusion > default install script. > > [ERR] switch_xml.c:1331 Couldnt open > /usr/local/freeswitch/conf/dialplan/public/*.xml (No such file or > directory) > > Running on Ubuntu 12.04 > > Any help will be appreciated. Thanks. > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-FusionPBX-error-opening-xml-tp7588074.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130227/b0985572/attachment.html From jnvines at gmail.com Thu Feb 28 01:54:22 2013 From: jnvines at gmail.com (Nick Vines) Date: Wed, 27 Feb 2013 14:54:22 -0800 Subject: [Freeswitch-users] Freeswitch/FusionPBX error opening *.xml In-Reply-To: <1362000773371-7588074.post@n2.nabble.com> References: <1362000773371-7588074.post@n2.nabble.com> Message-ID: Are there any xml files in that directory? On Wed, Feb 27, 2013 at 1:32 PM, mikeborschow wrote: > I am getting this error using Freeswitch / Fusion. Any ideas. I used > Fusion > default install script. > > [ERR] switch_xml.c:1331 Couldnt open > /usr/local/freeswitch/conf/dialplan/public/*.xml (No such file or > directory) > > Running on Ubuntu 12.04 > > Any help will be appreciated. Thanks. > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-FusionPBX-error-opening-xml-tp7588074.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130227/e727dff6/attachment.html From vittorio.guglielmo at gmail.com Thu Feb 28 01:56:08 2013 From: vittorio.guglielmo at gmail.com (Vittorio Guglielmo) Date: Wed, 27 Feb 2013 23:56:08 +0100 Subject: [Freeswitch-users] FS with SSL/TLS issues! In-Reply-To: References: <1361367187947-7587736.post@n2.nabble.com> <1361513236022-7587844.post@n2.nabble.com> <1361519970790-7587849.post@n2.nabble.com> <113D484C-EA7E-4611-B958-6514F5F2B5B9@freeswitch.org> <1361875157474-7588012.post@n2.nabble.com> <1361976425293-7588069.post@n2.nabble.com> <512E7B89.6030903@gmail.com> <512E8516.1060008@gmail.com> Message-ID: <512E8F08.40204@gmail.com> I thought that codec could be the cause, but all the extensions have only G.711 (alaw and ulaw). These is the log when I try to call between the extensions: 2013-02-27 23:53:13.085555 [NOTICE] switch_channel.c:968 New Channel sofia/internal/101 at voice.wicube.net [771747c6-8130-11e2-a4a8-efe939e44db4] 2013-02-27 23:53:13.085555 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/101 at voice.wicube.net [BREAK] 2013-02-27 23:53:13.085555 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/101 at voice.wicube.net [BREAK] 2013-02-27 23:53:13.085555 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/101 at voice.wicube.net) Running State Change CS_NEW 2013-02-27 23:53:13.085555 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/101 at voice.wicube.net) State NEW 2013-02-27 23:53:13.105591 [DEBUG] sofia.c:7729 IP 82.85.60.13 Rejected by acl "domains". Falling back to Digest auth. 2013-02-27 23:53:13.105591 [WARNING] sofia_reg.c:1502 SIP auth challenge (INVITE) on sofia profile 'internal' for [103 at voice.wicube.net] from ip 82.85.60.13 2013-02-27 23:53:13.105591 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/101 at voice.wicube.net [BREAK] 2013-02-27 23:53:13.105591 [DEBUG] sofia.c:1750 detaching session 771747c6-8130-11e2-a4a8-efe939e44db4 2013-02-27 23:53:13.145610 [DEBUG] sofia.c:1842 Re-attaching to session 771747c6-8130-11e2-a4a8-efe939e44db4 2013-02-27 23:53:13.145610 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/101 at voice.wicube.net [BREAK] 2013-02-27 23:53:13.145610 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/101 at voice.wicube.net [BREAK] 2013-02-27 23:53:13.145610 [DEBUG] sofia.c:7729 IP 82.85.60.13 Rejected by acl "domains". Falling back to Digest auth. 2013-02-27 23:53:13.165556 [DEBUG] sofia.c:5599 Channel sofia/internal/101 at voice.wicube.net entering state [received][100] 2013-02-27 23:53:13.165556 [DEBUG] sofia.c:5610 Remote SDP: v=0 o=- 3570994402 3570994402 IN IP4 10.10.5.101 s=cpc_med c=IN IP4 10.10.5.101 t=0 0 m=audio 53526 RTP/SAVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:G5TO9FRh3y0+WkCMn336K2K4X3iaYBFRdNsPF9f5 a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:Fn6vvWfGAQH497IRM9HaO+MEoMw+tjpeyAdA4kId a=ice-ufrag:75f854d0 a=ice-pwd:02886032 a=candidate:Ha0a0565 1 UDP 2130706431 10.10.5.101 53526 typ host a=candidate:H5587298 1 UDP 2130706431 5.88.114.152 53526 typ host a=candidate:Ha0a0565 2 UDP 2130706430 10.10.5.101 64080 typ host a=candidate:H5587298 2 UDP 2130706430 5.88.114.152 64080 typ host 2013-02-27 23:53:13.165556 [DEBUG] sofia_glue.c:4978 Set Remote Key [1 AES_CM_128_HMAC_SHA1_80 inline:G5TO9FRh3y0+WkCMn336K2K4X3iaYBFRdNsPF9f5] 2013-02-27 23:53:13.165556 [DEBUG] sofia_glue.c:3155 Set Local Key [1 AES_CM_128_HMAC_SHA1_80 inline:kbdihjLnWO2Mnu0tsc+VpZKyIwsw7eKVp7ODyoxx] 2013-02-27 23:53:13.165556 [DEBUG] sofia_glue.c:5137 Audio Codec Compare [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000] 2013-02-27 23:53:13.165556 [DEBUG] sofia_glue.c:5137 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2013-02-27 23:53:13.165556 [DEBUG] sofia_glue.c:3093 Set Codec sofia/internal/101 at voice.wicube.net PCMU/8000 20 ms 160 samples 64000 bits 2013-02-27 23:53:13.165556 [DEBUG] switch_core_codec.c:111 sofia/internal/101 at voice.wicube.net Original read codec set to PCMU:0 2013-02-27 23:53:13.165556 [DEBUG] sofia_glue.c:5266 Set 2833 dtmf send/recv payload to 101 2013-02-27 23:53:13.165556 [DEBUG] sofia.c:5827 (sofia/internal/101 at voice.wicube.net) State Change CS_NEW -> CS_INIT 2013-02-27 23:53:13.165556 [DEBUG] switch_core_session.c:1283 Send signal sofia/internal/101 at voice.wicube.net [BREAK] 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/101 at voice.wicube.net) Running State Change CS_INIT 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:454 (sofia/internal/101 at voice.wicube.net) State INIT 2013-02-27 23:53:13.165556 [DEBUG] mod_sofia.c:86 sofia/internal/101 at voice.wicube.net SOFIA INIT 2013-02-27 23:53:13.165556 [DEBUG] mod_sofia.c:126 (sofia/internal/101 at voice.wicube.net) State Change CS_INIT -> CS_ROUTING 2013-02-27 23:53:13.165556 [DEBUG] switch_core_session.c:1283 Send signal sofia/internal/101 at voice.wicube.net [BREAK] 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:454 (sofia/internal/101 at voice.wicube.net) State INIT going to sleep 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/101 at voice.wicube.net) Running State Change CS_ROUTING 2013-02-27 23:53:13.165556 [DEBUG] switch_channel.c:2003 (sofia/internal/101 at voice.wicube.net) Callstate Change DOWN -> RINGING 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:470 (sofia/internal/101 at voice.wicube.net) State ROUTING 2013-02-27 23:53:13.165556 [DEBUG] mod_sofia.c:149 sofia/internal/101 at voice.wicube.net SOFIA ROUTING 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:117 sofia/internal/101 at voice.wicube.net Standard ROUTING 2013-02-27 23:53:13.165556 [INFO] mod_dialplan_xml.c:557 Processing 101 <101>->103 in context default Dialplan: sofia/internal/101 at voice.wicube.net parsing [default->101] continue=false Dialplan: sofia/internal/101 at voice.wicube.net Regex (FAIL) [101] destination_number(103) =~ /^101$/ break=on-false Dialplan: sofia/internal/101 at voice.wicube.net parsing [default->102] continue=false Dialplan: sofia/internal/101 at voice.wicube.net Regex (FAIL) [102] destination_number(103) =~ /^102$/ break=on-false Dialplan: sofia/internal/101 at voice.wicube.net parsing [default->103] continue=false Dialplan: sofia/internal/101 at voice.wicube.net Regex (PASS) [103] destination_number(103) =~ /^103$/ break=on-false Dialplan: sofia/internal/101 at voice.wicube.net Action bridge(USER/103 at 82.85.60.237) 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:167 (sofia/internal/101 at voice.wicube.net) State Change CS_ROUTING -> CS_EXECUTE 2013-02-27 23:53:13.165556 [DEBUG] switch_core_session.c:1283 Send signal sofia/internal/101 at voice.wicube.net [BREAK] 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:470 (sofia/internal/101 at voice.wicube.net) State ROUTING going to sleep 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/101 at voice.wicube.net) Running State Change CS_EXECUTE 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:477 (sofia/internal/101 at voice.wicube.net) State EXECUTE 2013-02-27 23:53:13.165556 [DEBUG] mod_sofia.c:242 sofia/internal/101 at voice.wicube.net SOFIA EXECUTE 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:209 sofia/internal/101 at voice.wicube.net Standard EXECUTE EXECUTE sofia/internal/101 at voice.wicube.net bridge(USER/103 at 82.85.60.237) 2013-02-27 23:53:13.165556 [DEBUG] switch_channel.c:1089 sofia/internal/101 at voice.wicube.net EXPORTING[export_vars] [domain_name]=[82.85.60.237] to event 2013-02-27 23:53:13.165556 [DEBUG] switch_ivr_originate.c:2022 Parsing global variables 2013-02-27 23:53:13.165556 [DEBUG] switch_channel.c:1089 sofia/internal/101 at voice.wicube.net EXPORTING[export_vars] [domain_name]=[82.85.60.237] to event 2013-02-27 23:53:13.165556 [DEBUG] switch_ivr_originate.c:2022 Parsing global variables 2013-02-27 23:53:13.165556 [DEBUG] switch_event.c:1608 Parsing variable [sip_invite_domain]=[82.85.60.237] 2013-02-27 23:53:13.165556 [DEBUG] switch_event.c:1608 Parsing variable [presence_id]=[103 at 82.85.60.237] 2013-02-27 23:53:13.165556 [NOTICE] switch_channel.c:968 New Channel sofia/internal/sip:103 at 82.85.60.13:37595 [77239eea-8130-11e2-a4b2-efe939e44db4] 2013-02-27 23:53:13.165556 [DEBUG] mod_sofia.c:4970 (sofia/internal/sip:103 at 82.85.60.13:37595) State Change CS_NEW -> CS_INIT 2013-02-27 23:53:13.165556 [DEBUG] switch_core_session.c:1283 Send signal sofia/internal/sip:103 at 82.85.60.13:37595 [BREAK] 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/sip:103 at 82.85.60.13:37595) Running State Change CS_INIT 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:454 (sofia/internal/sip:103 at 82.85.60.13:37595) State INIT 2013-02-27 23:53:13.165556 [DEBUG] mod_sofia.c:86 sofia/internal/sip:103 at 82.85.60.13:37595 SOFIA INIT 2013-02-27 23:53:13.165556 [DEBUG] sofia_glue.c:2618 sip:103 at 82.85.60.13:37595;transport=TLS;ob Setting proxy route to sofia/internal/sip:103 at 82.85.60.13:37595 2013-02-27 23:53:13.165556 [DEBUG] sofia_glue.c:2647 Local SDP: v=0 o=FreeSWITCH 1361974937 1361974938 IN IP4 82.85.60.237 s=FreeSWITCH c=IN IP4 82.85.60.237 t=0 0 m=audio 30656 RTP/AVP 0 9 8 3 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv 2013-02-27 23:53:13.165556 [DEBUG] mod_sofia.c:126 (sofia/internal/sip:103 at 82.85.60.13:37595) State Change CS_INIT -> CS_ROUTING 2013-02-27 23:53:13.165556 [DEBUG] switch_core_session.c:1283 Send signal sofia/internal/sip:103 at 82.85.60.13:37595 [BREAK] 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:454 (sofia/internal/sip:103 at 82.85.60.13:37595) State INIT going to sleep 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/sip:103 at 82.85.60.13:37595) Running State Change CS_ROUTING 2013-02-27 23:53:13.165556 [DEBUG] switch_channel.c:2003 (sofia/internal/sip:103 at 82.85.60.13:37595) Callstate Change DOWN -> RINGING 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:470 (sofia/internal/sip:103 at 82.85.60.13:37595) State ROUTING 2013-02-27 23:53:13.165556 [DEBUG] mod_sofia.c:149 sofia/internal/sip:103 at 82.85.60.13:37595 SOFIA ROUTING 2013-02-27 23:53:13.165556 [DEBUG] switch_ivr_originate.c:67 (sofia/internal/sip:103 at 82.85.60.13:37595) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2013-02-27 23:53:13.165556 [DEBUG] switch_core_session.c:1283 Send signal sofia/internal/sip:103 at 82.85.60.13:37595 [BREAK] 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:470 (sofia/internal/sip:103 at 82.85.60.13:37595) State ROUTING going to sleep 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/sip:103 at 82.85.60.13:37595) Running State Change CS_CONSUME_MEDIA 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:489 (sofia/internal/sip:103 at 82.85.60.13:37595) State CONSUME_MEDIA 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:489 (sofia/internal/sip:103 at 82.85.60.13:37595) State CONSUME_MEDIA going to sleep 2013-02-27 23:53:13.165556 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/sip:103 at 82.85.60.13:37595 [BREAK] 2013-02-27 23:53:13.165556 [DEBUG] sofia.c:5599 Channel sofia/internal/sip:103 at 82.85.60.13:37595 entering state [calling][0] 2013-02-27 23:53:14.705571 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/sip:103 at 82.85.60.13:37595 [BREAK] 2013-02-27 23:53:14.705571 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/sip:103 at 82.85.60.13:37595 [BREAK] 2013-02-27 23:53:14.705571 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/sip:103 at 82.85.60.13:37595 [BREAK] 2013-02-27 23:53:14.705571 [DEBUG] sofia.c:5599 Channel sofia/internal/sip:103 at 82.85.60.13:37595 entering state [terminated][488] 2013-02-27 23:53:14.705571 [DEBUG] switch_channel.c:2994 (sofia/internal/sip:103 at 82.85.60.13:37595) Callstate Change RINGING -> HANGUP 2013-02-27 23:53:14.705571 [NOTICE] sofia.c:6383 Hangup sofia/internal/sip:103 at 82.85.60.13:37595 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] 2013-02-27 23:53:14.705571 [DEBUG] switch_channel.c:3017 Send signal sofia/internal/sip:103 at 82.85.60.13:37595 [KILL] 2013-02-27 23:53:14.705571 [DEBUG] switch_core_session.c:1283 Send signal sofia/internal/sip:103 at 82.85.60.13:37595 [BREAK] 2013-02-27 23:53:14.705571 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/sip:103 at 82.85.60.13:37595) Running State Change CS_HANGUP 2013-02-27 23:53:14.705571 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/sip:103 at 82.85.60.13:37595) State HANGUP 2013-02-27 23:53:14.705571 [DEBUG] mod_sofia.c:503 Channel sofia/internal/sip:103 at 82.85.60.13:37595 hanging up, cause: INCOMPATIBLE_DESTINATION 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:48 sofia/internal/sip:103 at 82.85.60.13:37595 Standard HANGUP, cause: INCOMPATIBLE_DESTINATION 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/sip:103 at 82.85.60.13:37595) State HANGUP going to sleep 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:446 (sofia/internal/sip:103 at 82.85.60.13:37595) State Change CS_HANGUP -> CS_REPORTING 2013-02-27 23:53:14.725551 [DEBUG] switch_core_session.c:1283 Send signal sofia/internal/sip:103 at 82.85.60.13:37595 [BREAK] 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/sip:103 at 82.85.60.13:37595) Running State Change CS_REPORTING 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/sip:103 at 82.85.60.13:37595) State REPORTING 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:92 sofia/internal/sip:103 at 82.85.60.13:37595 Standard REPORTING, cause: INCOMPATIBLE_DESTINATION 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/sip:103 at 82.85.60.13:37595) State REPORTING going to sleep 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/sip:103 at 82.85.60.13:37595) State Change CS_REPORTING -> CS_DESTROY 2013-02-27 23:53:14.725551 [DEBUG] switch_core_session.c:1283 Send signal sofia/internal/sip:103 at 82.85.60.13:37595 [BREAK] 2013-02-27 23:53:14.725551 [DEBUG] switch_core_session.c:1488 Session 10 (sofia/internal/sip:103 at 82.85.60.13:37595) Locked, Waiting on external entities 2013-02-27 23:53:14.725551 [DEBUG] switch_ivr_originate.c:3533 Originate Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION] 2013-02-27 23:53:14.725551 [NOTICE] switch_ivr_originate.c:2608 Cannot create outgoing channel of type [USER] cause: [INCOMPATIBLE_DESTINATION] 2013-02-27 23:53:14.725551 [DEBUG] switch_ivr_originate.c:3533 Originate Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION] 2013-02-27 23:53:14.725551 [INFO] mod_dptools.c:3052 Originate Failed. Cause: INCOMPATIBLE_DESTINATION 2013-02-27 23:53:14.725551 [DEBUG] switch_channel.c:2994 (sofia/internal/101 at voice.wicube.net) Callstate Change RINGING -> HANGUP 2013-02-27 23:53:14.725551 [NOTICE] mod_dptools.c:3172 Hangup sofia/internal/101 at voice.wicube.net [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] 2013-02-27 23:53:14.725551 [DEBUG] switch_channel.c:3017 Send signal sofia/internal/101 at voice.wicube.net [KILL] 2013-02-27 23:53:14.725551 [DEBUG] switch_core_session.c:1283 Send signal sofia/internal/101 at voice.wicube.net [BREAK] 2013-02-27 23:53:14.725551 [DEBUG] switch_core_session.c:2678 sofia/internal/101 at voice.wicube.net skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:477 (sofia/internal/101 at voice.wicube.net) State EXECUTE going to sleep 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/101 at voice.wicube.net) Running State Change CS_HANGUP 2013-02-27 23:53:14.725551 [NOTICE] switch_core_session.c:1506 Session 10 (sofia/internal/sip:103 at 82.85.60.13:37595) Ended 2013-02-27 23:53:14.725551 [NOTICE] switch_core_session.c:1510 Close Channel sofia/internal/sip:103 at 82.85.60.13:37595 [CS_DESTROY] 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:556 (sofia/internal/sip:103 at 82.85.60.13:37595) Callstate Change HANGUP -> DOWN 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:559 (sofia/internal/sip:103 at 82.85.60.13:37595) Running State Change CS_DESTROY 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/sip:103 at 82.85.60.13:37595) State DESTROY 2013-02-27 23:53:14.725551 [DEBUG] mod_sofia.c:396 sofia/internal/sip:103 at 82.85.60.13:37595 SOFIA DESTROY 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:99 sofia/internal/sip:103 at 82.85.60.13:37595 Standard DESTROY 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/sip:103 at 82.85.60.13:37595) State DESTROY going to sleep 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/101 at voice.wicube.net) State HANGUP 2013-02-27 23:53:14.725551 [DEBUG] mod_sofia.c:497 sofia/internal/101 at voice.wicube.net Overriding SIP cause 488 with 488 from the other leg 2013-02-27 23:53:14.725551 [DEBUG] mod_sofia.c:503 Channel sofia/internal/101 at voice.wicube.net hanging up, cause: INCOMPATIBLE_DESTINATION 2013-02-27 23:53:14.725551 [DEBUG] mod_sofia.c:633 Responding to INVITE with: 488 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:48 sofia/internal/101 at voice.wicube.net Standard HANGUP, cause: INCOMPATIBLE_DESTINATION 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/101 at voice.wicube.net) State HANGUP going to sleep 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:446 (sofia/internal/101 at voice.wicube.net) State Change CS_HANGUP -> CS_REPORTING 2013-02-27 23:53:14.725551 [DEBUG] switch_core_session.c:1283 Send signal sofia/internal/101 at voice.wicube.net [BREAK] 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/101 at voice.wicube.net) Running State Change CS_REPORTING 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/101 at voice.wicube.net) State REPORTING 2013-02-27 23:53:14.765550 [DEBUG] switch_core_state_machine.c:92 sofia/internal/101 at voice.wicube.net Standard REPORTING, cause: INCOMPATIBLE_DESTINATION 2013-02-27 23:53:14.765550 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/101 at voice.wicube.net) State REPORTING going to sleep 2013-02-27 23:53:14.765550 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/101 at voice.wicube.net) State Change CS_REPORTING -> CS_DESTROY 2013-02-27 23:53:14.765550 [DEBUG] switch_core_session.c:1283 Send signal sofia/internal/101 at voice.wicube.net [BREAK] 2013-02-27 23:53:14.765550 [DEBUG] switch_core_session.c:1488 Session 9 (sofia/internal/101 at voice.wicube.net) Locked, Waiting on external entities 2013-02-27 23:53:14.765550 [NOTICE] switch_core_session.c:1506 Session 9 (sofia/internal/101 at voice.wicube.net) Ended 2013-02-27 23:53:14.765550 [NOTICE] switch_core_session.c:1510 Close Channel sofia/internal/101 at voice.wicube.net [CS_DESTROY] 2013-02-27 23:53:14.765550 [DEBUG] switch_core_state_machine.c:556 (sofia/internal/101 at voice.wicube.net) Callstate Change HANGUP -> DOWN 2013-02-27 23:53:14.765550 [DEBUG] switch_core_state_machine.c:559 (sofia/internal/101 at voice.wicube.net) Running State Change CS_DESTROY 2013-02-27 23:53:14.765550 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/101 at voice.wicube.net) State DESTROY 2013-02-27 23:53:14.765550 [DEBUG] mod_sofia.c:396 sofia/internal/101 at voice.wicube.net SOFIA DESTROY 2013-02-27 23:53:14.765550 [DEBUG] switch_core_state_machine.c:99 sofia/internal/101 at voice.wicube.net Standard DESTROY 2013-02-27 23:53:14.765550 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/101 at voice.wicube.net) State DESTROY going to sleep Vic On 27/02/13 23:22, Anthony Minessale wrote: > Now you are just stuck with ordinary problems with how you have things > configured. > One line is not enough to tell you exactly why but a good guess is you > have codecs that don't match. This is unrelated to TLS. > > > > On Wed, Feb 27, 2013 at 4:13 PM, Vittorio Guglielmo > > > wrote: > > I did it, > now I have: > > > /mod_dptools.c:3052 Originate Failed. Cause: > INCOMPATIBLE_DESTINATION/ > > The only way I succeded was to add /bypass_media=true/ in the > dialplan for each extension, > but I have a huge latency in the voice (like 2/3 seconds). > > > Vic > > > > > On 27/02/13 22:54, Anthony Minessale wrote: >> This will work better in 1.4 but the code in 1.2 is frozen so we >> can't improve it there. >> In the meantime you can just do this: >> >> make an acl that is default allow called tls >> add it to apply-nat-acl param in sofia profile where you have tls >> >> Now it will record the rport for every client in the contact path >> and it should resolve the existing connection. >> >> >> >> >> On Wed, Feb 27, 2013 at 3:32 PM, Vittorio Guglielmo >> > > wrote: >> >> Hi, >> >> I've done all you said but I'm in trouble indeed: >> >> TLS/SRTP enabled on two extension; >> >> If I try to dial between the two extensions I have: >> >> /mod_dptools.c:3052 Originate Failed. Cause: >> INCOMPATIBLE_DESTINATION/ >> >> If I call outbound, it's ok, and the call is encrypted. >> >> Can you help me? >> >> Vic >> >> >> >> >> On 27/02/13 15:47, mehroz wrote: >>> TLS-NAT issue arises due to the fact that it tries to contact on other port >>> (TCP session) as far as my understanding. >>> >>> Changing contact header can sort this issue and i have done it setting up >>> >>> >>> for each directory (user). >>> >>> This helped and worked :) >>> >>> I suppose you should also have "ext-rtp-ip and "ext-sip-ip" = "nat-auto" >>> in SIP profile. >>> Cheeers! >>> >>> >>> >>> -- >>> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FS-with-SSL-TLS-issues-tp7587736p7588069.html >>> Sent from the freeswitch-users mailing list archive at Nabble.com. >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> >> googletalk:conf+888 at conference.freeswitch.org >> >> pstn:+19193869900 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > googletalk:conf+888 at conference.freeswitch.org > > pstn:+19193869900 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130227/468f5d33/attachment-0001.html From anthony.minessale at gmail.com Thu Feb 28 02:23:00 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 27 Feb 2013 17:23:00 -0600 Subject: [Freeswitch-users] FS with SSL/TLS issues! In-Reply-To: <512E8F08.40204@gmail.com> References: <1361367187947-7587736.post@n2.nabble.com> <1361513236022-7587844.post@n2.nabble.com> <1361519970790-7587849.post@n2.nabble.com> <113D484C-EA7E-4611-B958-6514F5F2B5B9@freeswitch.org> <1361875157474-7588012.post@n2.nabble.com> <1361976425293-7588069.post@n2.nabble.com> <512E7B89.6030903@gmail.com> <512E8516.1060008@gmail.com> <512E8F08.40204@gmail.com> Message-ID: You should turn on the sip trace too sofia global siptrace on Then you can see the 488 you are getting and try to figure out why. On Wed, Feb 27, 2013 at 4:56 PM, Vittorio Guglielmo < vittorio.guglielmo at gmail.com> wrote: > I thought that codec could be the cause, > but all the extensions have only G.711 (alaw and ulaw). > > These is the log when I try to call between the extensions: > > 2013-02-27 23:53:13.085555 [NOTICE] switch_channel.c:968 New Channel > sofia/internal/101 at voice.wicube.net [771747c6-8130-11e2-a4a8-efe939e44db4] > 2013-02-27 23:53:13.085555 [DEBUG] switch_core_session.c:975 Send signal > sofia/internal/101 at voice.wicube.net [BREAK] > 2013-02-27 23:53:13.085555 [DEBUG] switch_core_session.c:975 Send signal > sofia/internal/101 at voice.wicube.net [BREAK] > 2013-02-27 23:53:13.085555 [DEBUG] switch_core_state_machine.c:415 ( > sofia/internal/101 at voice.wicube.net) Running State Change CS_NEW > 2013-02-27 23:53:13.085555 [DEBUG] switch_core_state_machine.c:433 ( > sofia/internal/101 at voice.wicube.net) State NEW > 2013-02-27 23:53:13.105591 [DEBUG] sofia.c:7729 IP 82.85.60.13 Rejected > by acl "domains". Falling back to Digest auth. > 2013-02-27 23:53:13.105591 [WARNING] sofia_reg.c:1502 SIP auth challenge > (INVITE) on sofia profile 'internal' for [103 at voice.wicube.net] from ip > 82.85.60.13 > 2013-02-27 23:53:13.105591 [DEBUG] switch_core_session.c:975 Send signal > sofia/internal/101 at voice.wicube.net [BREAK] > 2013-02-27 23:53:13.105591 [DEBUG] sofia.c:1750 detaching session > 771747c6-8130-11e2-a4a8-efe939e44db4 > 2013-02-27 23:53:13.145610 [DEBUG] sofia.c:1842 Re-attaching to session > 771747c6-8130-11e2-a4a8-efe939e44db4 > 2013-02-27 23:53:13.145610 [DEBUG] switch_core_session.c:975 Send signal > sofia/internal/101 at voice.wicube.net [BREAK] > 2013-02-27 23:53:13.145610 [DEBUG] switch_core_session.c:975 Send signal > sofia/internal/101 at voice.wicube.net [BREAK] > 2013-02-27 23:53:13.145610 [DEBUG] sofia.c:7729 IP 82.85.60.13 Rejected by > acl "domains". Falling back to Digest auth. > 2013-02-27 23:53:13.165556 [DEBUG] sofia.c:5599 Channel > sofia/internal/101 at voice.wicube.net entering state [received][100] > 2013-02-27 23:53:13.165556 [DEBUG] sofia.c:5610 Remote SDP: > v=0 > o=- 3570994402 3570994402 IN IP4 10.10.5.101 > s=cpc_med > c=IN IP4 10.10.5.101 > t=0 0 > m=audio 53526 RTP/SAVP 0 8 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=crypto:1 AES_CM_128_HMAC_SHA1_80 > inline:G5TO9FRh3y0+WkCMn336K2K4X3iaYBFRdNsPF9f5 > a=crypto:2 AES_CM_128_HMAC_SHA1_32 > inline:Fn6vvWfGAQH497IRM9HaO+MEoMw+tjpeyAdA4kId > a=ice-ufrag:75f854d0 > a=ice-pwd:02886032 > a=candidate:Ha0a0565 1 UDP 2130706431 10.10.5.101 53526 typ host > a=candidate:H5587298 1 UDP 2130706431 5.88.114.152 53526 typ host > a=candidate:Ha0a0565 2 UDP 2130706430 10.10.5.101 64080 typ host > a=candidate:H5587298 2 UDP 2130706430 5.88.114.152 64080 typ host > > 2013-02-27 23:53:13.165556 [DEBUG] sofia_glue.c:4978 Set Remote Key [1 > AES_CM_128_HMAC_SHA1_80 inline:G5TO9FRh3y0+WkCMn336K2K4X3iaYBFRdNsPF9f5] > 2013-02-27 23:53:13.165556 [DEBUG] sofia_glue.c:3155 Set Local Key [1 > AES_CM_128_HMAC_SHA1_80 inline:kbdihjLnWO2Mnu0tsc+VpZKyIwsw7eKVp7ODyoxx] > 2013-02-27 23:53:13.165556 [DEBUG] sofia_glue.c:5137 Audio Codec Compare > [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000] > 2013-02-27 23:53:13.165556 [DEBUG] sofia_glue.c:5137 Audio Codec Compare > [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] > 2013-02-27 23:53:13.165556 [DEBUG] sofia_glue.c:3093 Set Codec > sofia/internal/101 at voice.wicube.net PCMU/8000 20 ms 160 samples 64000 bits > 2013-02-27 23:53:13.165556 [DEBUG] switch_core_codec.c:111 > sofia/internal/101 at voice.wicube.net Original read codec set to PCMU:0 > 2013-02-27 23:53:13.165556 [DEBUG] sofia_glue.c:5266 Set 2833 dtmf > send/recv payload to 101 > 2013-02-27 23:53:13.165556 [DEBUG] sofia.c:5827 ( > sofia/internal/101 at voice.wicube.net) State Change CS_NEW -> CS_INIT > 2013-02-27 23:53:13.165556 [DEBUG] switch_core_session.c:1283 Send signal > sofia/internal/101 at voice.wicube.net [BREAK] > 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:415 ( > sofia/internal/101 at voice.wicube.net) Running State Change CS_INIT > 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:454 ( > sofia/internal/101 at voice.wicube.net) State INIT > 2013-02-27 23:53:13.165556 [DEBUG] mod_sofia.c:86 > sofia/internal/101 at voice.wicube.net SOFIA INIT > 2013-02-27 23:53:13.165556 [DEBUG] mod_sofia.c:126 ( > sofia/internal/101 at voice.wicube.net) State Change CS_INIT -> CS_ROUTING > 2013-02-27 23:53:13.165556 [DEBUG] switch_core_session.c:1283 Send signal > sofia/internal/101 at voice.wicube.net [BREAK] > 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:454 ( > sofia/internal/101 at voice.wicube.net) State INIT going to sleep > 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:415 ( > sofia/internal/101 at voice.wicube.net) Running State Change CS_ROUTING > 2013-02-27 23:53:13.165556 [DEBUG] switch_channel.c:2003 ( > sofia/internal/101 at voice.wicube.net) Callstate Change DOWN -> RINGING > 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:470 ( > sofia/internal/101 at voice.wicube.net) State ROUTING > 2013-02-27 23:53:13.165556 [DEBUG] mod_sofia.c:149 > sofia/internal/101 at voice.wicube.net SOFIA ROUTING > 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:117 > sofia/internal/101 at voice.wicube.net Standard ROUTING > 2013-02-27 23:53:13.165556 [INFO] mod_dialplan_xml.c:557 Processing 101 > <101>->103 in context default > Dialplan: sofia/internal/101 at voice.wicube.net parsing [default->101] > continue=false > Dialplan: sofia/internal/101 at voice.wicube.net Regex (FAIL) [101] > destination_number(103) =~ /^101$/ break=on-false > Dialplan: sofia/internal/101 at voice.wicube.net parsing [default->102] > continue=false > Dialplan: sofia/internal/101 at voice.wicube.net Regex (FAIL) [102] > destination_number(103) =~ /^102$/ break=on-false > Dialplan: sofia/internal/101 at voice.wicube.net parsing [default->103] > continue=false > Dialplan: sofia/internal/101 at voice.wicube.net Regex (PASS) [103] > destination_number(103) =~ /^103$/ break=on-false > Dialplan: sofia/internal/101 at voice.wicube.net Action bridge( > USER/103 at 82.85.60.237) > 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:167 ( > sofia/internal/101 at voice.wicube.net) State Change CS_ROUTING -> CS_EXECUTE > 2013-02-27 23:53:13.165556 [DEBUG] switch_core_session.c:1283 Send signal > sofia/internal/101 at voice.wicube.net [BREAK] > 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:470 ( > sofia/internal/101 at voice.wicube.net) State ROUTING going to sleep > 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:415 ( > sofia/internal/101 at voice.wicube.net) Running State Change CS_EXECUTE > 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:477 ( > sofia/internal/101 at voice.wicube.net) State EXECUTE > 2013-02-27 23:53:13.165556 [DEBUG] mod_sofia.c:242 > sofia/internal/101 at voice.wicube.net SOFIA EXECUTE > 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:209 > sofia/internal/101 at voice.wicube.net Standard EXECUTE > EXECUTE sofia/internal/101 at voice.wicube.net bridge(USER/103 at 82.85.60.237) > 2013-02-27 23:53:13.165556 [DEBUG] switch_channel.c:1089 > sofia/internal/101 at voice.wicube.net EXPORTING[export_vars] > [domain_name]=[82.85.60.237] to event > 2013-02-27 23:53:13.165556 [DEBUG] switch_ivr_originate.c:2022 Parsing > global variables > 2013-02-27 23:53:13.165556 [DEBUG] switch_channel.c:1089 > sofia/internal/101 at voice.wicube.net EXPORTING[export_vars] > [domain_name]=[82.85.60.237] to event > 2013-02-27 23:53:13.165556 [DEBUG] switch_ivr_originate.c:2022 Parsing > global variables > 2013-02-27 23:53:13.165556 [DEBUG] switch_event.c:1608 Parsing variable > [sip_invite_domain]=[82.85.60.237] > 2013-02-27 23:53:13.165556 [DEBUG] switch_event.c:1608 Parsing variable > [presence_id]=[103 at 82.85.60.237] > 2013-02-27 23:53:13.165556 [NOTICE] switch_channel.c:968 New Channel > sofia/internal/sip:103 at 82.85.60.13:37595[77239eea-8130-11e2-a4b2-efe939e44db4] > 2013-02-27 23:53:13.165556 [DEBUG] mod_sofia.c:4970 (sofia/internal/ > sip:103 at 82.85.60.13:37595) State Change CS_NEW -> CS_INIT > 2013-02-27 23:53:13.165556 [DEBUG] switch_core_session.c:1283 Send signal > sofia/internal/sip:103 at 82.85.60.13:37595 [BREAK] > 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:415 > (sofia/internal/sip:103 at 82.85.60.13:37595) Running State Change CS_INIT > 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:454 > (sofia/internal/sip:103 at 82.85.60.13:37595) State INIT > 2013-02-27 23:53:13.165556 [DEBUG] mod_sofia.c:86 sofia/internal/ > sip:103 at 82.85.60.13:37595 SOFIA INIT > 2013-02-27 23:53:13.165556 [DEBUG] sofia_glue.c:2618 > sip:103 at 82.85.60.13:37595;transport=TLS;ob Setting proxy route to > sofia/internal/sip:103 at 82.85.60.13:37595 > 2013-02-27 23:53:13.165556 [DEBUG] sofia_glue.c:2647 Local SDP: > v=0 > o=FreeSWITCH 1361974937 1361974938 IN IP4 82.85.60.237 > s=FreeSWITCH > c=IN IP4 82.85.60.237 > t=0 0 > m=audio 30656 RTP/AVP 0 9 8 3 101 13 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > a=sendrecv > > 2013-02-27 23:53:13.165556 [DEBUG] mod_sofia.c:126 (sofia/internal/ > sip:103 at 82.85.60.13:37595) State Change CS_INIT -> CS_ROUTING > 2013-02-27 23:53:13.165556 [DEBUG] switch_core_session.c:1283 Send signal > sofia/internal/sip:103 at 82.85.60.13:37595 [BREAK] > 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:454 > (sofia/internal/sip:103 at 82.85.60.13:37595) State INIT going to sleep > 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:415 > (sofia/internal/sip:103 at 82.85.60.13:37595) Running State Change CS_ROUTING > 2013-02-27 23:53:13.165556 [DEBUG] switch_channel.c:2003 (sofia/internal/ > sip:103 at 82.85.60.13:37595) Callstate Change DOWN -> RINGING > 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:470 > (sofia/internal/sip:103 at 82.85.60.13:37595) State ROUTING > 2013-02-27 23:53:13.165556 [DEBUG] mod_sofia.c:149 sofia/internal/ > sip:103 at 82.85.60.13:37595 SOFIA ROUTING > 2013-02-27 23:53:13.165556 [DEBUG] switch_ivr_originate.c:67 > (sofia/internal/sip:103 at 82.85.60.13:37595) State Change CS_ROUTING -> > CS_CONSUME_MEDIA > 2013-02-27 23:53:13.165556 [DEBUG] switch_core_session.c:1283 Send signal > sofia/internal/sip:103 at 82.85.60.13:37595 [BREAK] > 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:470 > (sofia/internal/sip:103 at 82.85.60.13:37595) State ROUTING going to sleep > 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:415 > (sofia/internal/sip:103 at 82.85.60.13:37595) Running State Change > CS_CONSUME_MEDIA > 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:489 > (sofia/internal/sip:103 at 82.85.60.13:37595) State CONSUME_MEDIA > 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:489 > (sofia/internal/sip:103 at 82.85.60.13:37595) State CONSUME_MEDIA going to > sleep > 2013-02-27 23:53:13.165556 [DEBUG] switch_core_session.c:975 Send signal > sofia/internal/sip:103 at 82.85.60.13:37595 [BREAK] > 2013-02-27 23:53:13.165556 [DEBUG] sofia.c:5599 Channel sofia/internal/ > sip:103 at 82.85.60.13:37595 entering state [calling][0] > 2013-02-27 23:53:14.705571 [DEBUG] switch_core_session.c:975 Send signal > sofia/internal/sip:103 at 82.85.60.13:37595 [BREAK] > 2013-02-27 23:53:14.705571 [DEBUG] switch_core_session.c:975 Send signal > sofia/internal/sip:103 at 82.85.60.13:37595 [BREAK] > 2013-02-27 23:53:14.705571 [DEBUG] switch_core_session.c:975 Send signal > sofia/internal/sip:103 at 82.85.60.13:37595 [BREAK] > 2013-02-27 23:53:14.705571 [DEBUG] sofia.c:5599 Channel sofia/internal/ > sip:103 at 82.85.60.13:37595 entering state [terminated][488] > 2013-02-27 23:53:14.705571 [DEBUG] switch_channel.c:2994 (sofia/internal/ > sip:103 at 82.85.60.13:37595) Callstate Change RINGING -> HANGUP > 2013-02-27 23:53:14.705571 [NOTICE] sofia.c:6383 Hangup sofia/internal/ > sip:103 at 82.85.60.13:37595 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] > 2013-02-27 23:53:14.705571 [DEBUG] switch_channel.c:3017 Send signal > sofia/internal/sip:103 at 82.85.60.13:37595 [KILL] > 2013-02-27 23:53:14.705571 [DEBUG] switch_core_session.c:1283 Send signal > sofia/internal/sip:103 at 82.85.60.13:37595 [BREAK] > 2013-02-27 23:53:14.705571 [DEBUG] switch_core_state_machine.c:415 > (sofia/internal/sip:103 at 82.85.60.13:37595) Running State Change CS_HANGUP > 2013-02-27 23:53:14.705571 [DEBUG] switch_core_state_machine.c:667 > (sofia/internal/sip:103 at 82.85.60.13:37595) State HANGUP > 2013-02-27 23:53:14.705571 [DEBUG] mod_sofia.c:503 Channel sofia/internal/ > sip:103 at 82.85.60.13:37595 hanging up, cause: INCOMPATIBLE_DESTINATION > 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:48 > sofia/internal/sip:103 at 82.85.60.13:37595 Standard HANGUP, cause: > INCOMPATIBLE_DESTINATION > 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:667 > (sofia/internal/sip:103 at 82.85.60.13:37595) State HANGUP going to sleep > 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:446 > (sofia/internal/sip:103 at 82.85.60.13:37595) State Change CS_HANGUP -> > CS_REPORTING > 2013-02-27 23:53:14.725551 [DEBUG] switch_core_session.c:1283 Send signal > sofia/internal/sip:103 at 82.85.60.13:37595 [BREAK] > 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:415 > (sofia/internal/sip:103 at 82.85.60.13:37595) Running State Change > CS_REPORTING > 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:749 > (sofia/internal/sip:103 at 82.85.60.13:37595) State REPORTING > 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:92 > sofia/internal/sip:103 at 82.85.60.13:37595 Standard REPORTING, cause: > INCOMPATIBLE_DESTINATION > 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:749 > (sofia/internal/sip:103 at 82.85.60.13:37595) State REPORTING going to sleep > 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:440 > (sofia/internal/sip:103 at 82.85.60.13:37595) State Change CS_REPORTING -> > CS_DESTROY > 2013-02-27 23:53:14.725551 [DEBUG] switch_core_session.c:1283 Send signal > sofia/internal/sip:103 at 82.85.60.13:37595 [BREAK] > 2013-02-27 23:53:14.725551 [DEBUG] switch_core_session.c:1488 Session 10 > (sofia/internal/sip:103 at 82.85.60.13:37595) Locked, Waiting on external > entities > 2013-02-27 23:53:14.725551 [DEBUG] switch_ivr_originate.c:3533 Originate > Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION] > 2013-02-27 23:53:14.725551 [NOTICE] switch_ivr_originate.c:2608 Cannot > create outgoing channel of type [USER] cause: [INCOMPATIBLE_DESTINATION] > 2013-02-27 23:53:14.725551 [DEBUG] switch_ivr_originate.c:3533 Originate > Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION] > 2013-02-27 23:53:14.725551 [INFO] mod_dptools.c:3052 Originate Failed. > Cause: INCOMPATIBLE_DESTINATION > 2013-02-27 23:53:14.725551 [DEBUG] switch_channel.c:2994 ( > sofia/internal/101 at voice.wicube.net) Callstate Change RINGING -> HANGUP > 2013-02-27 23:53:14.725551 [NOTICE] mod_dptools.c:3172 Hangup > sofia/internal/101 at voice.wicube.net [CS_EXECUTE] > [INCOMPATIBLE_DESTINATION] > 2013-02-27 23:53:14.725551 [DEBUG] switch_channel.c:3017 Send signal > sofia/internal/101 at voice.wicube.net [KILL] > 2013-02-27 23:53:14.725551 [DEBUG] switch_core_session.c:1283 Send signal > sofia/internal/101 at voice.wicube.net [BREAK] > 2013-02-27 23:53:14.725551 [DEBUG] switch_core_session.c:2678 > sofia/internal/101 at voice.wicube.net skip receive message > [APPLICATION_EXEC_COMPLETE] (channel is hungup already) > 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:477 ( > sofia/internal/101 at voice.wicube.net) State EXECUTE going to sleep > 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:415 ( > sofia/internal/101 at voice.wicube.net) Running State Change CS_HANGUP > 2013-02-27 23:53:14.725551 [NOTICE] switch_core_session.c:1506 Session 10 > (sofia/internal/sip:103 at 82.85.60.13:37595) Ended > 2013-02-27 23:53:14.725551 [NOTICE] switch_core_session.c:1510 Close > Channel sofia/internal/sip:103 at 82.85.60.13:37595 [CS_DESTROY] > 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:556 > (sofia/internal/sip:103 at 82.85.60.13:37595) Callstate Change HANGUP -> DOWN > 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:559 > (sofia/internal/sip:103 at 82.85.60.13:37595) Running State Change CS_DESTROY > 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:569 > (sofia/internal/sip:103 at 82.85.60.13:37595) State DESTROY > 2013-02-27 23:53:14.725551 [DEBUG] mod_sofia.c:396 sofia/internal/ > sip:103 at 82.85.60.13:37595 SOFIA DESTROY > 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:99 > sofia/internal/sip:103 at 82.85.60.13:37595 Standard DESTROY > 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:569 > (sofia/internal/sip:103 at 82.85.60.13:37595) State DESTROY going to sleep > 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:667 ( > sofia/internal/101 at voice.wicube.net) State HANGUP > 2013-02-27 23:53:14.725551 [DEBUG] mod_sofia.c:497 > sofia/internal/101 at voice.wicube.net Overriding SIP cause 488 with 488 > from the other leg > 2013-02-27 23:53:14.725551 [DEBUG] mod_sofia.c:503 Channel > sofia/internal/101 at voice.wicube.net hanging up, cause: > INCOMPATIBLE_DESTINATION > 2013-02-27 23:53:14.725551 [DEBUG] mod_sofia.c:633 Responding to INVITE > with: 488 > 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:48 > sofia/internal/101 at voice.wicube.net Standard HANGUP, cause: > INCOMPATIBLE_DESTINATION > 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:667 ( > sofia/internal/101 at voice.wicube.net) State HANGUP going to sleep > 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:446 ( > sofia/internal/101 at voice.wicube.net) State Change CS_HANGUP -> > CS_REPORTING > 2013-02-27 23:53:14.725551 [DEBUG] switch_core_session.c:1283 Send signal > sofia/internal/101 at voice.wicube.net [BREAK] > 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:415 ( > sofia/internal/101 at voice.wicube.net) Running State Change CS_REPORTING > 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:749 ( > sofia/internal/101 at voice.wicube.net) State REPORTING > 2013-02-27 23:53:14.765550 [DEBUG] switch_core_state_machine.c:92 > sofia/internal/101 at voice.wicube.net Standard REPORTING, cause: > INCOMPATIBLE_DESTINATION > 2013-02-27 23:53:14.765550 [DEBUG] switch_core_state_machine.c:749 ( > sofia/internal/101 at voice.wicube.net) State REPORTING going to sleep > 2013-02-27 23:53:14.765550 [DEBUG] switch_core_state_machine.c:440 ( > sofia/internal/101 at voice.wicube.net) State Change CS_REPORTING -> > CS_DESTROY > 2013-02-27 23:53:14.765550 [DEBUG] switch_core_session.c:1283 Send signal > sofia/internal/101 at voice.wicube.net [BREAK] > 2013-02-27 23:53:14.765550 [DEBUG] switch_core_session.c:1488 Session 9 ( > sofia/internal/101 at voice.wicube.net) Locked, Waiting on external entities > 2013-02-27 23:53:14.765550 [NOTICE] switch_core_session.c:1506 Session 9 ( > sofia/internal/101 at voice.wicube.net) Ended > 2013-02-27 23:53:14.765550 [NOTICE] switch_core_session.c:1510 Close > Channel sofia/internal/101 at voice.wicube.net [CS_DESTROY] > 2013-02-27 23:53:14.765550 [DEBUG] switch_core_state_machine.c:556 ( > sofia/internal/101 at voice.wicube.net) Callstate Change HANGUP -> DOWN > 2013-02-27 23:53:14.765550 [DEBUG] switch_core_state_machine.c:559 ( > sofia/internal/101 at voice.wicube.net) Running State Change CS_DESTROY > 2013-02-27 23:53:14.765550 [DEBUG] switch_core_state_machine.c:569 ( > sofia/internal/101 at voice.wicube.net) State DESTROY > 2013-02-27 23:53:14.765550 [DEBUG] mod_sofia.c:396 > sofia/internal/101 at voice.wicube.net SOFIA DESTROY > 2013-02-27 23:53:14.765550 [DEBUG] switch_core_state_machine.c:99 > sofia/internal/101 at voice.wicube.net Standard DESTROY > 2013-02-27 23:53:14.765550 [DEBUG] switch_core_state_machine.c:569 ( > sofia/internal/101 at voice.wicube.net) State DESTROY going to sleep > > > Vic > > > On 27/02/13 23:22, Anthony Minessale wrote: > > Now you are just stuck with ordinary problems with how you have things > configured. > One line is not enough to tell you exactly why but a good guess is you > have codecs that don't match. This is unrelated to TLS. > > > > On Wed, Feb 27, 2013 at 4:13 PM, Vittorio Guglielmo < > vittorio.guglielmo at gmail.com> wrote: > >> I did it, >> now I have: >> >> >> *mod_dptools.c:3052 Originate Failed. Cause: INCOMPATIBLE_DESTINATION* >> >> The only way I succeded was to add *bypass_media=true* in the dialplan >> for each extension, >> but I have a huge latency in the voice (like 2/3 seconds). >> >> >> Vic >> >> >> >> >> On 27/02/13 22:54, Anthony Minessale wrote: >> >> This will work better in 1.4 but the code in 1.2 is frozen so we can't >> improve it there. >> In the meantime you can just do this: >> >> make an acl that is default allow called tls >> add it to apply-nat-acl param in sofia profile where you have tls >> >> Now it will record the rport for every client in the contact path and >> it should resolve the existing connection. >> >> >> >> >> On Wed, Feb 27, 2013 at 3:32 PM, Vittorio Guglielmo < >> vittorio.guglielmo at gmail.com> wrote: >> >>> Hi, >>> >>> I've done all you said but I'm in trouble indeed: >>> >>> TLS/SRTP enabled on two extension; >>> >>> If I try to dial between the two extensions I have: >>> >>> *mod_dptools.c:3052 Originate Failed. Cause: INCOMPATIBLE_DESTINATION* >>> >>> If I call outbound, it's ok, and the call is encrypted. >>> >>> Can you help me? >>> >>> Vic >>> >>> >>> >>> >>> On 27/02/13 15:47, mehroz wrote: >>> >>> TLS-NAT issue arises due to the fact that it tries to contact on other port >>> (TCP session) as far as my understanding. >>> >>> Changing contact header can sort this issue and i have done it setting up >>> >>> >>> for each directory (user). >>> >>> This helped and worked :) >>> >>> I suppose you should also have "ext-rtp-ip and "ext-sip-ip" = "nat-auto" >>> in SIP profile. >>> Cheeers! >>> >>> >>> >>> -- >>> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FS-with-SSL-TLS-issues-tp7587736p7588069.html >>> Sent from the freeswitch-users mailing list archive at Nabble.com. >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >>> >>> >>> >>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com >>> >>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >> >> >> >> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com >> >> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130227/624c1ece/attachment-0001.html From msc at freeswitch.org Thu Feb 28 03:26:32 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 27 Feb 2013 16:26:32 -0800 Subject: [Freeswitch-users] Problem setting up bind_digit_action In-Reply-To: <461554606.17147.1361955674129.JavaMail.root@mailserver.edistar.com> References: <461554606.17147.1361955674129.JavaMail.root@mailserver.edistar.com> Message-ID: It looks to me like your "MYSETUP" extension is getting executed on the answer: 1. 2013-02-27 09:24:08.511899 [NOTICE] sofia.c:6326 Channel [ sofia/external/723331] has been answered 2. EXECUTE sofia/external/723331 execute_extension(MYSETUP XML public) 3. 2013-02-27 09:24:08.511899 [INFO] mod_dialplan_xml.c:557 Processing Igor Cappello ->MYSETUP in context public 1. Dialplan: sofia/external/723331 Regex (PASS) [setup_bind_digit_action]destination_number (MYSETUP) =~ /^MYSETUP$/ break=on-false 2. 2013-02-27 09:24:08.511899 [NOTICE] switch_core_session.c:2788Execute set (outside_call=true) 3. EXECUTE sofia/external/723331 set(outside_call=true) 4. 2013-02-27 09:24:08.511899 [DEBUG] mod_dptools.c:1344 sofia/external/ 723331 SET [outside_call]=[true] 5. 2013-02-27 09:24:08.511899 [NOTICE] switch_core_session.c:2788Execute set (RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) 6. EXECUTE sofia/external/723331 set(RFC2822_DATE=Wed, 27 Feb 2013 09:24: 08 +0100) It looks like your MYSETUP extension has just two "set" apps and that's it. -MC On Wed, Feb 27, 2013 at 1:01 AM, Igor Cappello wrote: > We managed to make it work, specifying the "peer" parameter in the data > attribute of application bind_digit_action. > > Here's the working extension > > > expression="^test_bind_dtmf$"> > > > data="test,00,exec:log,INFO TEST,peer"/> > > > data="{execute_on_answer='start_dtmf'}sofia/gateway/voipcard1/723331"/> > > > > > > And here's the execution log > http://pastebin.freeswitch.org/20635 > > We now face a related issue, though: we need to specify an extension to be > executed on the bleg. Using execute_on_answer='execute_extension ...' seems > to be the way to go, but we did not manage to make it work. > > > Here are the extensions > > > > > > > > > > expression="^test_bind_dtmf$"> > > data="test,00,exec:log,INFO TEST,peer"/> > > data="{execute_on_answer='execute_extension MYSETUP XML > public'}sofia/gateway/voipcard1/723331"/> > > > > > And here's the execution log: the extension MYSETUP doesn't seem to be > executed properly on the bleg of the bridged call. > http://pastebin.freeswitch.org/20636 > > > Could you give us a hint? > > Thanks for your time, > Igor Cappello > > > > Please use http://pastebin.freeswitch.org/ > > > > On Tue, Feb 26, 2013 at 12:42 PM, Michael Collins > > wrote: > > > You need to set the digit action realm: > > > > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_digit_action_set_realm > > > > > > -MC > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130227/3a090f66/attachment.html From schoch+freeswitch.org at xwin32.com Thu Feb 28 03:40:03 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Wed, 27 Feb 2013 16:40:03 -0800 Subject: [Freeswitch-users] IVR XML: Can a timeout action be specified? Message-ID: Looking through the IVR XML documentation tells me that I can make an IVR play a sound and exit after a timeout, but there doesn't seem to be a way to execute an action after a timeout. Is this possible, or will I have to write my IVR in a script? -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130227/f224ae3c/attachment.html From mikebemail at gmail.com Thu Feb 28 03:40:03 2013 From: mikebemail at gmail.com (mikeborschow) Date: Wed, 27 Feb 2013 16:40:03 -0800 (PST) Subject: [Freeswitch-users] Freeswitch/FusionPBX error opening *.xml In-Reply-To: References: <1362000773371-7588074.post@n2.nabble.com> Message-ID: <1362012003407-7588088.post@n2.nabble.com> in that directory is this file. 00_inbound_did.xml.noload contents of file: -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-FusionPBX-error-opening-xml-tp7588074p7588088.html Sent from the freeswitch-users mailing list archive at Nabble.com. From msc at freeswitch.org Thu Feb 28 03:48:17 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 27 Feb 2013 16:48:17 -0800 Subject: [Freeswitch-users] IVR XML: Can a timeout action be specified? In-Reply-To: References: Message-ID: There is no "action on timeout" in the IVR, but when the ivr app ends it will move on to the next action in your dialplan. -MC On Wed, Feb 27, 2013 at 4:40 PM, Steven Schoch < schoch+freeswitch.org at xwin32.com> wrote: > Looking through the IVR XML documentation tells me that I can make an IVR > play a sound and exit after a timeout, but there doesn't seem to be a way > to execute an action after a timeout. Is this possible, or will I have to > write my IVR in a script? > > -- > Steve > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130227/78b8e238/attachment-0001.html From msc at freeswitch.org Thu Feb 28 03:50:25 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 27 Feb 2013 16:50:25 -0800 Subject: [Freeswitch-users] Freeswitch/FusionPBX error opening *.xml In-Reply-To: <1362012003407-7588088.post@n2.nabble.com> References: <1362000773371-7588074.post@n2.nabble.com> <1362012003407-7588088.post@n2.nabble.com> Message-ID: This is a harmless error and you can ignore it. If it is causing you undue concern then just add a real simple xml file like this: Make sure the file is named with a .xml suffix and you should be good to go. -MC On Wed, Feb 27, 2013 at 4:40 PM, mikeborschow wrote: > in that directory is this file. > > 00_inbound_did.xml.noload > > contents of file: > > > > > > > > > > > > > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-FusionPBX-error-opening-xml-tp7588074p7588088.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130227/4bfac7fa/attachment.html From krice at freeswitch.org Thu Feb 28 03:51:16 2013 From: krice at freeswitch.org (Ken Rice) Date: Wed, 27 Feb 2013 18:51:16 -0600 Subject: [Freeswitch-users] Freeswitch/FusionPBX error opening *.xml In-Reply-To: <1362012003407-7588088.post@n2.nabble.com> Message-ID: That file will not get loaded... Check the way the configs work and look at the error message... It explicitly says /some/path/*.xml Its an error in the fact you told the preprocessor to load some files that's don't exist... Other wise its an error to safely ignore most of the time On 2/27/13 6:40 PM, "mikeborschow" wrote: > in that directory is this file. > > 00_inbound_did.xml.noload > > contents of file: > > > > > > > > > > > > > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-FusionPBX-error-openi > ng-xml-tp7588074p7588088.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch From schoch+freeswitch.org at xwin32.com Thu Feb 28 03:55:03 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Wed, 27 Feb 2013 16:55:03 -0800 Subject: [Freeswitch-users] FreeSWITCH and HylaFax Message-ID: The problem was that HylaFax, which runs as the user 'uucp' on CentOS, couldn't open /dev/FS0, because the permissions were wrong. Since I didn't see any way to make FreeSwitch change the permissions, I added one line to the /etc/init.d/freeswitch file: *** 46,51 **** --- 46,52 ---- RETVAL=$? echo [ $RETVAL -eq 0 ] && touch $LOCK_FILE; + (sleep 120; chgrp uucp /dev/FS*; chmod g+r /dev/FS*) & echo return $RETVAL } -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130227/cc905977/attachment.html From jnvines at gmail.com Thu Feb 28 03:56:21 2013 From: jnvines at gmail.com (Nick Vines) Date: Wed, 27 Feb 2013 16:56:21 -0800 Subject: [Freeswitch-users] Freeswitch/FusionPBX error opening *.xml In-Reply-To: <1362012003407-7588088.post@n2.nabble.com> References: <1362000773371-7588074.post@n2.nabble.com> <1362012003407-7588088.post@n2.nabble.com> Message-ID: That file won't be found by *.xml. If you got rid of the .noload it would. I ignored that error for a long time before I tidied up my installations. On Wed, Feb 27, 2013 at 4:40 PM, mikeborschow wrote: > in that directory is this file. > > 00_inbound_did.xml.noload > > contents of file: > > > > > > > > > > > > > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-FusionPBX-error-opening-xml-tp7588074p7588088.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130227/397ac251/attachment.html From schoch+freeswitch.org at xwin32.com Thu Feb 28 04:01:23 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Wed, 27 Feb 2013 17:01:23 -0800 Subject: [Freeswitch-users] IVR XML: Can a timeout action be specified? In-Reply-To: References: Message-ID: On Wed, Feb 27, 2013 at 4:48 PM, Michael Collins wrote: > There is no "action on timeout" in the IVR, but when the ivr app ends it > will move on to the next action in your dialplan. D'oh! (That should have been obvious.) Thank you. -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130227/8707da4c/attachment.html From mikebemail at gmail.com Thu Feb 28 04:23:06 2013 From: mikebemail at gmail.com (mikeborschow) Date: Wed, 27 Feb 2013 17:23:06 -0800 (PST) Subject: [Freeswitch-users] Freeswitch/FusionPBX error opening *.xml In-Reply-To: <1362000773371-7588074.post@n2.nabble.com> References: <1362000773371-7588074.post@n2.nabble.com> Message-ID: <1362014586922-7588096.post@n2.nabble.com> OK ... will ignore the error. Thanks. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-FusionPBX-error-opening-xml-tp7588074p7588096.html Sent from the freeswitch-users mailing list archive at Nabble.com. From kkgp20 at gmail.com Thu Feb 28 10:34:02 2013 From: kkgp20 at gmail.com (K K) Date: Thu, 28 Feb 2013 08:34:02 +0100 Subject: [Freeswitch-users] Repository: where is the Video-Media-Bug brunch? Message-ID: Hello, I would like to use the product parts which are located in the video-media-bug branch but I am not able to find anywhere this branch and address from which I can download it. Also in the Wiki there is written that the repository have only two branches: Master and v1.2.stable. So I am confused. Thank you for help. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130228/f2b5064c/attachment-0001.html From denis.gasparin at edistar.com Thu Feb 28 11:20:17 2013 From: denis.gasparin at edistar.com (Denis Gasparin) Date: Thu, 28 Feb 2013 09:20:17 +0100 (CET) Subject: [Freeswitch-users] Problem setting up bind_digit_action In-Reply-To: Message-ID: <509746653.18043.1362039617472.JavaMail.root@mailserver.edistar.com> Freeswitch writes in the log that it's processing the extension MYSETUP. The extension MYSETUP is: I'd expect that in the log, after the following lines, 1. EXECUTE sofia/external/ 723331 execute_extension ( MYSETUP XML public ) 2. 2013 -02 -27 09 : 24 : 08.511899 [ INFO ] mod_dialplan_xml.c: 557 Processing Igor Cappello ->MYSETUP in context public Freeswitch should write an INFO log message with text "MYSETUP exec". The log message actually is not written. It seems that Freeswitch "processes" the extension but actually doesn't execute it... Thank you Denis ----- Messaggio originale ----- Da: "Michael Collins" A: "FreeSWITCH Users Help" Inviato: Gioved?, 28 febbraio 2013 1:26:32 Oggetto: Re: [Freeswitch-users] Problem setting up bind_digit_action It looks to me like your "MYSETUP" extension is getting executed on the answer: 1. 2013 -02 -27 09 : 24 : 08.511899 [ NOTICE ] sofia.c: 6326 Channel [ sofia/external/ 723331 ] has been answered 2. EXECUTE sofia/external/ 723331 execute_extension ( MYSETUP XML public ) 3. 2013 -02 -27 09 : 24 : 08.511899 [ INFO ] mod_dialplan_xml.c: 557 Processing Igor Cappello ->MYSETUP in context public 1. Dialplan: sofia/external/ 723331 Regex ( PASS ) [ setup_bind_digit_action ] destination_number ( MYSETUP ) =~ /^MYSETUP$/ break=on-false 2. 2013 -02 -27 09 : 24 : 08.511899 [ NOTICE ] switch_core_session.c: 2788 Execute set ( outside_call=true ) 3. EXECUTE sofia/external/ 723331 set ( outside_call=true ) 4. 2013 -02 -27 09 : 24 : 08.511899 [ DEBUG ] mod_dptools.c: 1344 sofia/external/ 723331 SET [ outside_call ] = [ true ] 5. 2013 -02 -27 09 : 24 : 08.511899 [ NOTICE ] switch_core_session.c: 2788 Execute set ( RFC2822_DATE=$ { strftime ( %a, %d %b %Y %T %z ) } ) 6. EXECUTE sofia/external/ 723331 set ( RFC2822_DATE=Wed, 27 Feb 2013 09 : 24 : 08 +0100 ) It looks like your MYSETUP extension has just two "set" apps and that's it. -MC On Wed, Feb 27, 2013 at 1:01 AM, Igor Cappello < igor.cappello at edistar.com > wrote: We managed to make it work, specifying the "peer" parameter in the data attribute of application bind_digit_action. Here's the working extension And here's the execution log http://pastebin.freeswitch.org/20635 We now face a related issue, though: we need to specify an extension to be executed on the bleg. Using execute_on_answer='execute_extension ...' seems to be the way to go, but we did not manage to make it work. Here are the extensions And here's the execution log: the extension MYSETUP doesn't seem to be executed properly on the bleg of the bridged call. http://pastebin.freeswitch.org/20636 Could you give us a hint? Thanks for your time, Igor Cappello > Please use http://pastebin.freeswitch.org/ > > On Tue, Feb 26, 2013 at 12:42 PM, Michael Collins < msc at freeswitch.org > > wrote: > > You need to set the digit action realm: > > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_digit_action_set_realm > > > > -MC > > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130228/48b1b425/attachment.html From avi at avimarcus.net Thu Feb 28 11:34:11 2013 From: avi at avimarcus.net (Avi Marcus) Date: Thu, 28 Feb 2013 10:34:11 +0200 Subject: [Freeswitch-users] Freeswitch/FusionPBX error opening *.xml In-Reply-To: References: <1362000773371-7588074.post@n2.nabble.com> <1362012003407-7588088.post@n2.nabble.com> Message-ID: MC: You can actually do: "touch empty.xml" or create a blank file some other way and it's fine. FreeSWITCH just looks to include a *file* - it doesn't care what's in it. -Avi On Thu, Feb 28, 2013 at 2:50 AM, Michael Collins wrote: > This is a harmless error and you can ignore it. If it is causing you undue > concern then just add a real simple xml file like this: > > > > Make sure the file is named with a .xml suffix and you should be good to > go. > > -MC > > > On Wed, Feb 27, 2013 at 4:40 PM, mikeborschow wrote: > >> in that directory is this file. >> >> 00_inbound_did.xml.noload >> >> contents of file: >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> -- >> View this message in context: >> http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-FusionPBX-error-opening-xml-tp7588074p7588088.html >> Sent from the freeswitch-users mailing list archive at Nabble.com. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130228/50b560df/attachment-0001.html From lists at kavun.ch Thu Feb 28 12:00:26 2013 From: lists at kavun.ch (Emrah) Date: Thu, 28 Feb 2013 04:00:26 -0500 Subject: [Freeswitch-users] Concatenated multiple playback types In-Reply-To: References: <1FFF97C269757C458224B7C895F35F1514B3F0@cantor.std.visionutv.se> <55F547E7-F986-4268-9BBD-C4210D997D12@kavun.ch> Message-ID: <95DB739D-42B1-4B6B-AACC-918DE2B69CD9@kavun.ch> Christopher, Thanks a bunch, that worked! It looks funny, but it worked. :) Thanks! On Feb 26, 2013, at 6:50 PM, Christopher Rienzo wrote: > file_string:// does the concatenation.. so you need: > > file_string://silence_stream://6000!conference/conf-alone.wav > > > On Tue, Feb 26, 2013 at 2:53 PM, Michael Collins wrote: > What about this? > > > > -MC > > > On Mon, Feb 25, 2013 at 9:36 PM, Emrah wrote: > Hi there, > > Just wanted to inquire about this again. > FS is ever so flexible in so many ways that I cannot understand why this wouldn't work. > I would like to apply the following to my alone sound in conference.conf.xml. > silence_stream://6000!file_string:///path-to-alone-sound.wav > > I am imagining other scenarios where I would want to use some silence before generating a ringback. > > What is your take on this? > Cheers > On Aug 22, 2012, at 12:40 PM, Michael Collins wrote: > > > One thing I like about the sox method is that it requires very little in the way of processing/resources. > > -MC > > > > On Wed, Aug 22, 2012 at 9:27 AM, Emrah wrote: > > Hey Michael, > > > > That was my work around before I posted, but I thought since you guys do allow some flexibility with silence, tone generation and file playback all usable with the same app, I thought it would make more sense to take advantage of it and combine some things there. > > > > All the best, thanks for your response. > > Emrah > > On Aug 22, 2012, at 12:07 PM, Michael Collins wrote: > > > > > You could always take the sneaky approach and use sox to add some silence to the beginning of conf-alone.wav. > > > -MC > > > > > > On Wed, Aug 22, 2012 at 8:56 AM, Emrah wrote: > > > Thanks for your response. > > > I don't think it would be something bad to implement. > > > > > > I also wanted to correct a little issueI experience both with FS and Asterisk. > > > If you generate a tone, even if it's played in a file, and you execute a record action right after it is played, the last 30 ms (or so) of the tone is distorded and kind of faded out. > > > In some sircompstances, you might even hear the hint of the tone when you execute the next playback. > > > I wanted to replace my record tones with the tone followed by 30 MS silence. > > > > > > Best, > > > Emrah > > > > > > On Aug 22, 2012, at 10:36 AM, Peter Olsson wrote: > > > > > > > I don't believe this is possible. If you record 5 seconds of silence into a file you might be able to use file_string:// to play them both. > > > > > > > > /Peter > > > > > > > > > > > > -----Ursprungligt meddelande----- > > > > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Emrah > > > > Skickat: den 22 augusti 2012 16:24 > > > > Till: FreeSWITCH Users Help > > > > ?mne: [Freeswitch-users] Concatenated multiple playback types > > > > > > > > Hello there, > > > > > > > > I am trying to add some silence before the alone sound file in my conference.cnf.xml and can't get it to work. > > > > > > > > Here is what I tried: > > > > > > > > > > > > > > > > > > > > > > > > Any idea on how to do this? > > > > > > > > Thanks! > > > > Emrah > > > > _________________________________________________________________________ > > > > Professional FreeSWITCH Consulting Services: > > > > consulting at freeswitch.org > > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > > http://www.freeswitch.org > > > > http://wiki.freeswitch.org > > > > http://www.cluecon.com > > > > > > > > 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 > > > > > > > > !DSPAM:5034e9a632761303016055! > > > > > > > > > > > > _________________________________________________________________________ > > > > Professional FreeSWITCH Consulting Services: > > > > consulting at freeswitch.org > > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > > http://www.freeswitch.org > > > > http://wiki.freeswitch.org > > > > http://www.cluecon.com > > > > > > > > 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 > > > > > > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > > > > > 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 > > > > > > > > > > > > -- > > > Michael S Collins > > > Twitter: @mercutioviz > > > http://www.FreeSWITCH.org > > > http://www.ClueCon.com > > > http://www.OSTAG.org > > > > > > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > > > > > 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 > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > 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 > > > > > > > > -- > > Michael S Collins > > Twitter: @mercutioviz > > http://www.FreeSWITCH.org > > http://www.ClueCon.com > > http://www.OSTAG.org > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 alex at digitalmail.com Thu Feb 28 11:57:49 2013 From: alex at digitalmail.com (Alex Lake) Date: Thu, 28 Feb 2013 08:57:49 +0000 Subject: [Freeswitch-users] Silence suppression again In-Reply-To: References: <512CF844.5070509@digitalmail.com> <512DE07F.3090503@digitalmail.com> <512E45E7.2070004@digitalmail.com> Message-ID: <512F1C0D.6090308@digitalmail.com> Ah! I think that has fixed it. Many thanks. I'll see if I can update a bit of wiki somewhere... > suppress_cng=true variable or supress-cng profile param set to true. > > > On Wed, Feb 27, 2013 at 11:44 AM, Alex Lake > wrote: > > A quick update on this - I have verified that suppresion of > silence suppression(!) on Grandstreams doesn't work when it is the > B-Party of a call. I'm not sure if silence-suppression is part of > the SIP handshake, but I know that other VOIP technologies seem to > be able to prevent it. > I'm calling in a handset specialist to advise.. > > One of the things that intrigues me is this extract from the log > (particularly the reference to silenceSupp): > > 2013-02-27 17:28:05.151624 [DEBUG] switch_core_codec.c:244 > sofia/internal/0253302 at 004-0253.sb12.dmclub.org > Restore > previous codec PCMA:8. > 2013-02-27 17:28:05.151624 [DEBUG] mod_sofia.c:754 Local SDP > sofia/internal/0253302 at 004-0253.sb12.dmclub.org > : > v=0 > o=FreeSWITCH 1361975871 1361975873 IN IP4 176.58.88.201 > s=FreeSWITCH > c=IN IP4 176.58.88.201 > t=0 0 > m=audio 10212 RTP/AVP 8 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:30 > a=sendrecv > > What does all this mean? > > From the "A" leg of the call (which is placed by a LinkSys SPA922) > I have: > > v%3D0%0Ao%3DFreeSWITCH%201361974765%201361974767%20IN%20IP4%20176.58.88.201%0As%3DFreeSWITCH%0Ac%3DIN%20IP4%20176.58.88.201%0At%3D0%200%0Am%3Daudio%2011468%20RTP/AVP%208%20101%0Aa%3Drtpmap%3A8%20PCMA/8000%0Aa%3Drtpmap%3A101%20telephone-event/8000%0Aa%3Dfmtp%3A101%200-16%0Aa%3DsilenceSupp%3Aoff%20-%20-%20-%20-%0Aa%3Dptime%3A30%0Aa%3Dsendrecv%0A > > From the troublesome "B" leg of the call (placed by FreeSwitch to > my GXP2000) I have: > > sip_local_sdp_str>v%3D0%0Ao%3DFreeSWITCH%201361974745%201361974746%20IN%20IP4%20176.58.88.201%0As%3DFreeSWITCH%0Ac%3DIN%20IP4%20176.58.88.201%0At%3D0%200%0Am%3Daudio%2011488%20RTP/AVP%208%20101%2013%0Aa%3Drtpmap%3A101%20telephone-event/8000%0Aa%3Dfmtp%3A101%200-16%0Aa%3Dptime%3A30%0Aa%3Dsendrecv%0Am%3Daudio%2011488%20RTP/AVP%208%203%20101%2013%0Aa%3Drtpmap%3A101%20telephone-event/8000%0Aa%3Dfmtp%3A101%200-16%0Aa%3Dptime%3A20%0Aa%3Dsendrecv%0A > > I was wondering if there's a way I can modify this to include > a="silenceSupp:off - - - -" and whether that would make the > slightest bit of difference to the handset's behaviour! > > > > Alex >> Thanks. >> I've been looking at the channel variables of the call. And this >> is part of it: >> >> 1089091957212103445581548000432344001829908182990810639> _media_packet_count>1063900000 >> >> The thing that slightly concerns me here is >> 4323 >> >> That sounds as though the inbound audio path contains some >> comfort noise, suggesting that the handset the other end has some >> kind of silence suppression enabled. >> >> I'm afraid it's a dreaded Grandstream GXP2000 - which I know are >> not the handset of the cognoscenti! - but the customer tells me >> that he has silence suppression disabled. Do we think this is >> correct?! >> >> Rgds, >> Alex >>> It does not send any by default. Also, yes it only generates >>> silence when the call is not getting any RTP. >>> >>> >>> >>> On Tue, Feb 26, 2013 at 12:00 PM, Alex Lake >>> > wrote: >>> >>> Does Freeswitch enable any kind of silence suppression by >>> default? If I >>> have bridge_generate_comfort_noise=true, does that only have >>> an effect >>> when the audio-generating end decides to stop sending RTP? >>> >>> Any other tips for diagnosing silence (it's kind of like a >>> slow noise >>> gate effect)? >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >>> >>> >>> -- >>> Anthony Minessale II >>> >>> FreeSWITCH http://www.freeswitch.org/ >>> ClueCon http://www.cluecon.com/ >>> Twitter: http://twitter.com/FreeSWITCH_wire >>> >>> AIM: anthm >>> MSN:anthony_minessale at hotmail.com >>> >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>> >>> IRC: irc.freenode.net #freeswitch >>> >>> FreeSWITCH Developer Conference >>> sip:888 at conference.freeswitch.org >>> >>> googletalk:conf+888 at conference.freeswitch.org >>> >>> pstn:+19193869900 >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >>> No virus found in this message. >>> Checked by AVG - www.avg.com >>> Version: 2012.0.2238 / Virus Database: 2641/5634 - Release Date: >>> 02/26/13 >>> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> >> No virus found in this message. >> Checked by AVG - www.avg.com >> Version: 2012.0.2238 / Virus Database: 2641/5635 - Release Date: >> 02/26/13 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > googletalk:conf+888 at conference.freeswitch.org > > pstn:+19193869900 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.2238 / Virus Database: 2641/5636 - Release Date: 02/27/13 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130228/3d2854b7/attachment-0001.html From dujinfang at gmail.com Thu Feb 28 12:03:35 2013 From: dujinfang at gmail.com (Seven Du) Date: Thu, 28 Feb 2013 17:03:35 +0800 Subject: [Freeswitch-users] Repository: where is the Video-Media-Bug brunch? In-Reply-To: References: Message-ID: git clone git at git.freeswitch.org/freeswitch.git git checkout video-media-bug http://wiki.freeswitch.org/wiki/Installation_Guide -- Seven Du http://www.freeswitch.org.cn http://about.me/dujinfang http://www.dujinfang.com Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Thursday, February 28, 2013 at 3:34 PM, K K wrote: > Hello, > > I would like to use the product parts which are located in the video-media-bug branch but I am not able to find anywhere this branch and address from which I can download it. > Also in the Wiki there is written that the repository have only two branches: Master and v1.2.stable. So I am confused. Thank you for help. > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto: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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130228/beff74c7/attachment.html From emamirazavi at gmail.com Thu Feb 28 12:19:42 2013 From: emamirazavi at gmail.com (Sayyed Mohammad Emami Razavi) Date: Thu, 28 Feb 2013 12:49:42 +0330 Subject: [Freeswitch-users] Supervisor for calls Message-ID: How to set supervisor for calls. For example one of my agents is calling with one of my customers. I want to listen their call so that no one understands i am listening to him/her. Or i want to call just with my agent so that customer doesn't understand that i am calling with him or her. Or i want to call just with the customer so that the agent doesn't understand that i am calling with it's customer! Can anybody help me? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130228/bb1c030e/attachment.html From krice at freeswitch.org Thu Feb 28 12:36:58 2013 From: krice at freeswitch.org (Ken Rice) Date: Thu, 28 Feb 2013 03:36:58 -0600 Subject: [Freeswitch-users] Supervisor for calls In-Reply-To: Message-ID: You are looking for the eavesdrop functionality... See the examples in the wike or the default example dialplan On 2/28/13 3:19 AM, "Sayyed Mohammad Emami Razavi" wrote: > How to set supervisor for calls. > For example one of my agents is calling with one of my customers. > I want to listen their call so that no one understands i am listening to > him/her. Or i want to call just with my agent so that customer doesn't > understand that i am calling with him or her. Or i want to call just with the > customer so that the agent doesn't understand that i am calling with it's > customer! > Can anybody help me? > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130228/0e90e6ec/attachment.html From vittorio.guglielmo at gmail.com Thu Feb 28 15:36:36 2013 From: vittorio.guglielmo at gmail.com (Vittorio Guglielmo) Date: Thu, 28 Feb 2013 13:36:36 +0100 Subject: [Freeswitch-users] FS with SSL/TLS issues! In-Reply-To: References: <1361367187947-7587736.post@n2.nabble.com> <1361513236022-7587844.post@n2.nabble.com> <1361519970790-7587849.post@n2.nabble.com> <113D484C-EA7E-4611-B958-6514F5F2B5B9@freeswitch.org> <1361875157474-7588012.post@n2.nabble.com> <1361976425293-7588069.post@n2.nabble.com> <512E7B89.6030903@gmail.com> <512E8516.1060008@gmail.com> <512E8F08.40204@gmail.com> Message-ID: <512F4F54.8020903@gmail.com> Anthony, sofia global siptrace on and in attachment are the two siptrace: 1. *101* ( Bria iOS 2.3.4 on Iphone 3GS) *===> 102* (Bria 3 release 3.5.0b on Mac OSX 10.7.5) 2. *102* *===> 101* Both of them have forced only ulaw (first) and alaw (second), and without TSL/SRTP they call each other without problems. The behavior is different in the two above cases: When I call from 101 to 102 I get a *Originate Resulted in Error Cause: 79 [SERVICE_NOT_IMPLEMENTED]* due to, I believe a: *SIP/2.0 415 Unsupported Media Type* When I instead call from 102 to 101 I get a *Originate Failed. Cause: INCOMPATIBLE_DESTINATION* with a : *SIP/2.0 488 Not Acceptable Here* Vic On 28/02/13 00:23, Anthony Minessale wrote: > sofia global siptrace on > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130228/88d44a57/attachment-0001.html -------------- next part -------------- freeswitch at internal> recv 1333 bytes from tls/[85.18.36.68]:51187 at 12:27:39.121764: ------------------------------------------------------------------------ INVITE sip:102 at voice.wicube.net SIP/2.0 Via: SIP/2.0/TLS 85.18.36.68:51187;rport;branch=z9hG4bKPjbSUoRnakXwBPZqfm76TRKwoxH-KVdl9a;alias Max-Forwards: 70 From: "101" ;tag=RQ-SQdhcZuhI5SYNSNKTY9F-9-.unsww To: Contact: ;+sip.ice Call-ID: 5xUNyMbEzeDbgO48GR2uPiITtdGkVHm9 CSeq: 31360 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub User-Agent: Bria iOS 2.3.4 Content-Type: application/sdp Content-Length: 717 v=0 o=- 3571043268 3571043268 IN IP4 172.22.55.178 s=cpc_med c=IN IP4 172.22.55.178 t=0 0 m=audio 53530 RTP/SAVP 0 8 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:fqwD6Slr4snIn2sJPbAtfR+sLr/VRQbTOmtdi7JH a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:FvJ8slt1jidvluNFDVItIgmux+WiqgIFoAp+ItS5 a=ice-ufrag:7bec20f6 a=ice-pwd:4b683609 a=candidate:Hac1637b2 1 UDP 2130706431 172.22.55.178 53530 typ host a=candidate:H55ad1d9 1 UDP 2130706431 5.90.209.217 53530 typ host a=candidate:Hac1637b2 2 UDP 2130706430 172.22.55.178 58436 typ host a=candidate:H55ad1d9 2 UDP 2130706430 5.90.209.217 58436 typ host ------------------------------------------------------------------------ send 388 bytes to tls/[85.18.36.68]:51187 at 12:27:39.122154: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/TLS 85.18.36.68:51187;rport=51187;branch=z9hG4bKPjbSUoRnakXwBPZqfm76TRKwoxH-KVdl9a;alias From: "101" ;tag=RQ-SQdhcZuhI5SYNSNKTY9F-9-.unsww To: Call-ID: 5xUNyMbEzeDbgO48GR2uPiITtdGkVHm9 CSeq: 31360 INVITE User-Agent: FreeSWITCH-mod_sofia/1.2.6+git~20130104T154559Z~a4247651ca Content-Length: 0 ------------------------------------------------------------------------ 2013-02-28 13:07:04.438143 [NOTICE] switch_channel.c:968 New Channel sofia/internal/101 at voice.wicube.net [3d835816-81a2-11e2-ac9e-ad22efd2ace4] 2013-02-28 13:07:04.438143 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/101 at voice.wicube.net [BREAK] 2013-02-28 13:07:04.438143 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/101 at voice.wicube.net [BREAK] 2013-02-28 13:07:04.458126 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/101 at voice.wicube.net) Running State Change CS_NEW 2013-02-28 13:07:04.458126 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/101 at voice.wicube.net) State NEW 2013-02-28 13:07:04.478139 [DEBUG] sofia.c:7729 IP 85.18.36.68 Rejected by acl "domains". Falling back to Digest auth. 2013-02-28 13:07:04.478139 [WARNING] sofia_reg.c:1502 SIP auth challenge (INVITE) on sofia profile 'internal' for [102 at voice.wicube.net] from ip 85.18.36.68 send 891 bytes to tls/[85.18.36.68]:51187 at 12:27:39.150958: ------------------------------------------------------------------------ SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/TLS 85.18.36.68:51187;rport=51187;branch=z9hG4bKPjbSUoRnakXwBPZqfm76TRKwoxH-KVdl9a;alias From: "101" ;tag=RQ-SQdhcZuhI5SYNSNKTY9F-9-.unsww To: ;tag=Dm4763BDQjZ6S Call-ID: 5xUNyMbEzeDbgO48GR2uPiITtdGkVHm9 CSeq: 31360 INVITE User-Agent: FreeSWITCH-mod_sofia/1.2.6+git~20130104T154559Z~a4247651ca Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Proxy-Authenticate: Digest realm="voice.wicube.net", nonce="3d86ab38-81a2-11e2-ac9f-ad22efd2ace4", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ 2013-02-28 13:07:04.478139 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/101 at voice.wicube.net [BREAK] 2013-02-28 13:07:04.478139 [DEBUG] sofia.c:1750 detaching session 3d835816-81a2-11e2-ac9e-ad22efd2ace4 recv 362 bytes from tls/[85.18.36.68]:51187 at 12:27:39.207888: ------------------------------------------------------------------------ ACK sip:102 at voice.wicube.net SIP/2.0 Via: SIP/2.0/TLS 85.18.36.68:51187;rport;branch=z9hG4bKPjbSUoRnakXwBPZqfm76TRKwoxH-KVdl9a;alias Max-Forwards: 70 From: "101" ;tag=RQ-SQdhcZuhI5SYNSNKTY9F-9-.unsww To: ;tag=Dm4763BDQjZ6S Call-ID: 5xUNyMbEzeDbgO48GR2uPiITtdGkVHm9 CSeq: 31360 ACK Content-Length: 0 ------------------------------------------------------------------------ recv 1607 bytes from tls/[85.18.36.68]:51187 at 12:27:39.281632: ------------------------------------------------------------------------ INVITE sip:102 at voice.wicube.net SIP/2.0 Via: SIP/2.0/TLS 85.18.36.68:51187;rport;branch=z9hG4bKPjVkpmhYfPXXJ.MmmYQ62ufvtZD0K5i8Lf;alias Max-Forwards: 70 From: "101" ;tag=RQ-SQdhcZuhI5SYNSNKTY9F-9-.unsww To: Contact: ;+sip.ice Call-ID: 5xUNyMbEzeDbgO48GR2uPiITtdGkVHm9 CSeq: 31361 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub User-Agent: Bria iOS 2.3.4 Proxy-Authorization: Digest username="101", realm="voice.wicube.net", nonce="3d86ab38-81a2-11e2-ac9f-ad22efd2ace4", uri="sip:102 at voice.wicube.net", response="6b0c7609274244fe82cf620ca92f32c4", algorithm=MD5, cnonce="GEwL.oSmgLolHXaIddxqHalqr5vm.JPF", qop=auth, nc=00000001 Content-Type: application/sdp Content-Length: 717 v=0 o=- 3571043268 3571043268 IN IP4 172.22.55.178 s=cpc_med c=IN IP4 172.22.55.178 t=0 0 m=audio 53530 RTP/SAVP 0 8 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:fqwD6Slr4snIn2sJPbAtfR+sLr/VRQbTOmtdi7JH a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:FvJ8slt1jidvluNFDVItIgmux+WiqgIFoAp+ItS5 a=ice-ufrag:7bec20f6 a=ice-pwd:4b683609 a=candidate:Hac1637b2 1 UDP 2130706431 172.22.55.178 53530 typ host a=candidate:H55ad1d9 1 UDP 2130706431 5.90.209.217 53530 typ host a=candidate:Hac1637b2 2 UDP 2130706430 172.22.55.178 58436 typ host a=candidate:H55ad1d9 2 UDP 2130706430 5.90.209.217 58436 typ host ------------------------------------------------------------------------ send 388 bytes to tls/[85.18.36.68]:51187 at 12:27:39.282486: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/TLS 85.18.36.68:51187;rport=51187;branch=z9hG4bKPjVkpmhYfPXXJ.MmmYQ62ufvtZD0K5i8Lf;alias From: "101" ;tag=RQ-SQdhcZuhI5SYNSNKTY9F-9-.unsww To: Call-ID: 5xUNyMbEzeDbgO48GR2uPiITtdGkVHm9 CSeq: 31361 INVITE User-Agent: FreeSWITCH-mod_sofia/1.2.6+git~20130104T154559Z~a4247651ca Content-Length: 0 ------------------------------------------------------------------------ 2013-02-28 13:07:04.598195 [DEBUG] sofia.c:1842 Re-attaching to session 3d835816-81a2-11e2-ac9e-ad22efd2ace4 2013-02-28 13:07:04.598195 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/101 at voice.wicube.net [BREAK] 2013-02-28 13:07:04.598195 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/101 at voice.wicube.net [BREAK] 2013-02-28 13:07:04.618122 [DEBUG] sofia.c:7729 IP 85.18.36.68 Rejected by acl "domains". Falling back to Digest auth. 2013-02-28 13:07:04.618122 [DEBUG] sofia.c:5599 Channel sofia/internal/101 at voice.wicube.net entering state [received][100] 2013-02-28 13:07:04.618122 [DEBUG] sofia.c:5610 Remote SDP: v=0 o=- 3571043268 3571043268 IN IP4 172.22.55.178 s=cpc_med c=IN IP4 172.22.55.178 t=0 0 m=audio 53530 RTP/SAVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:fqwD6Slr4snIn2sJPbAtfR+sLr/VRQbTOmtdi7JH a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:FvJ8slt1jidvluNFDVItIgmux+WiqgIFoAp+ItS5 a=ice-ufrag:7bec20f6 a=ice-pwd:4b683609 a=candidate:Hac1637b2 1 UDP 2130706431 172.22.55.178 53530 typ host a=candidate:H55ad1d9 1 UDP 2130706431 5.90.209.217 53530 typ host a=candidate:Hac1637b2 2 UDP 2130706430 172.22.55.178 58436 typ host a=candidate:H55ad1d9 2 UDP 2130706430 5.90.209.217 58436 typ host 2013-02-28 13:07:04.618122 [DEBUG] sofia_glue.c:4978 Set Remote Key [1 AES_CM_128_HMAC_SHA1_80 inline:fqwD6Slr4snIn2sJPbAtfR+sLr/VRQbTOmtdi7JH] 2013-02-28 13:07:04.618122 [DEBUG] sofia_glue.c:3155 Set Local Key [1 AES_CM_128_HMAC_SHA1_80 inline:L+JO67wbK0cKnCTZ7xDdwJy7QKIJfK4x3VB07M5s] 2013-02-28 13:07:04.618122 [DEBUG] sofia_glue.c:5137 Audio Codec Compare [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000] 2013-02-28 13:07:04.618122 [DEBUG] sofia_glue.c:5137 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2013-02-28 13:07:04.618122 [DEBUG] sofia_glue.c:3093 Set Codec sofia/internal/101 at voice.wicube.net PCMU/8000 20 ms 160 samples 64000 bits 2013-02-28 13:07:04.618122 [DEBUG] switch_core_codec.c:111 sofia/internal/101 at voice.wicube.net Original read codec set to PCMU:0 2013-02-28 13:07:04.618122 [DEBUG] sofia_glue.c:5266 Set 2833 dtmf send/recv payload to 101 2013-02-28 13:07:04.618122 [DEBUG] sofia.c:5827 (sofia/internal/101 at voice.wicube.net) State Change CS_NEW -> CS_INIT 2013-02-28 13:07:04.618122 [DEBUG] switch_core_session.c:1283 Send signal sofia/internal/101 at voice.wicube.net [BREAK] 2013-02-28 13:07:04.618122 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/101 at voice.wicube.net) Running State Change CS_INIT 2013-02-28 13:07:04.618122 [DEBUG] switch_core_state_machine.c:454 (sofia/internal/101 at voice.wicube.net) State INIT 2013-02-28 13:07:04.618122 [DEBUG] mod_sofia.c:86 sofia/internal/101 at voice.wicube.net SOFIA INIT 2013-02-28 13:07:04.618122 [DEBUG] mod_sofia.c:126 (sofia/internal/101 at voice.wicube.net) State Change CS_INIT -> CS_ROUTING 2013-02-28 13:07:04.618122 [DEBUG] switch_core_session.c:1283 Send signal sofia/internal/101 at voice.wicube.net [BREAK] 2013-02-28 13:07:04.618122 [DEBUG] switch_core_state_machine.c:454 (sofia/internal/101 at voice.wicube.net) State INIT going to sleep 2013-02-28 13:07:04.618122 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/101 at voice.wicube.net) Running State Change CS_ROUTING 2013-02-28 13:07:04.618122 [DEBUG] switch_channel.c:2003 (sofia/internal/101 at voice.wicube.net) Callstate Change DOWN -> RINGING 2013-02-28 13:07:04.618122 [DEBUG] switch_core_state_machine.c:470 (sofia/internal/101 at voice.wicube.net) State ROUTING 2013-02-28 13:07:04.618122 [DEBUG] mod_sofia.c:149 sofia/internal/101 at voice.wicube.net SOFIA ROUTING 2013-02-28 13:07:04.618122 [DEBUG] switch_core_state_machine.c:117 sofia/internal/101 at voice.wicube.net Standard ROUTING 2013-02-28 13:07:04.618122 [INFO] mod_dialplan_xml.c:557 Processing 101 <101>->102 in context default Dialplan: sofia/internal/101 at voice.wicube.net parsing [default->101] continue=false Dialplan: sofia/internal/101 at voice.wicube.net Regex (FAIL) [101] destination_number(102) =~ /^101$/ break=on-false Dialplan: sofia/internal/101 at voice.wicube.net parsing [default->102] continue=false Dialplan: sofia/internal/101 at voice.wicube.net Regex (PASS) [102] destination_number(102) =~ /^102$/ break=on-false Dialplan: sofia/internal/101 at voice.wicube.net Action bridge(USER/102 at 82.85.60.237) 2013-02-28 13:07:04.618122 [DEBUG] switch_core_state_machine.c:167 (sofia/internal/101 at voice.wicube.net) State Change CS_ROUTING -> CS_EXECUTE 2013-02-28 13:07:04.618122 [DEBUG] switch_core_session.c:1283 Send signal sofia/internal/101 at voice.wicube.net [BREAK] 2013-02-28 13:07:04.618122 [DEBUG] switch_core_state_machine.c:470 (sofia/internal/101 at voice.wicube.net) State ROUTING going to sleep 2013-02-28 13:07:04.618122 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/101 at voice.wicube.net) Running State Change CS_EXECUTE 2013-02-28 13:07:04.618122 [DEBUG] switch_core_state_machine.c:477 (sofia/internal/101 at voice.wicube.net) State EXECUTE 2013-02-28 13:07:04.618122 [DEBUG] mod_sofia.c:242 sofia/internal/101 at voice.wicube.net SOFIA EXECUTE 2013-02-28 13:07:04.618122 [DEBUG] switch_core_state_machine.c:209 sofia/internal/101 at voice.wicube.net Standard EXECUTE EXECUTE sofia/internal/101 at voice.wicube.net bridge(USER/102 at 82.85.60.237) 2013-02-28 13:07:04.639117 [DEBUG] switch_channel.c:1089 sofia/internal/101 at voice.wicube.net EXPORTING[export_vars] [domain_name]=[82.85.60.237] to event 2013-02-28 13:07:04.639117 [DEBUG] switch_ivr_originate.c:2022 Parsing global variables 2013-02-28 13:07:04.639117 [DEBUG] switch_channel.c:1089 sofia/internal/101 at voice.wicube.net EXPORTING[export_vars] [domain_name]=[82.85.60.237] to event 2013-02-28 13:07:04.639117 [DEBUG] switch_ivr_originate.c:2022 Parsing global variables 2013-02-28 13:07:04.639117 [DEBUG] switch_event.c:1608 Parsing variable [sip_invite_domain]=[82.85.60.237] 2013-02-28 13:07:04.639117 [DEBUG] switch_event.c:1608 Parsing variable [presence_id]=[102 at 82.85.60.237] 2013-02-28 13:07:04.639117 [NOTICE] switch_channel.c:968 New Channel sofia/internal/sip:102 at 85.18.36.68:55272 [3d9f6736-81a2-11e2-aca8-ad22efd2ace4] 2013-02-28 13:07:04.639117 [DEBUG] mod_sofia.c:4970 (sofia/internal/sip:102 at 85.18.36.68:55272) State Change CS_NEW -> CS_INIT 2013-02-28 13:07:04.639117 [DEBUG] switch_core_session.c:1283 Send signal sofia/internal/sip:102 at 85.18.36.68:55272 [BREAK] 2013-02-28 13:07:04.639117 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/sip:102 at 85.18.36.68:55272) Running State Change CS_INIT 2013-02-28 13:07:04.639117 [DEBUG] switch_core_state_machine.c:454 (sofia/internal/sip:102 at 85.18.36.68:55272) State INIT 2013-02-28 13:07:04.639117 [DEBUG] mod_sofia.c:86 sofia/internal/sip:102 at 85.18.36.68:55272 SOFIA INIT 2013-02-28 13:07:04.639117 [DEBUG] sofia_glue.c:2618 sip:102 at 85.18.36.68:55272;transport=tls;rinstance=2804dee11a135346 Setting proxy route to sofia/internal/sip:102 at 85.18.36.68:55272 2013-02-28 13:07:04.639117 [DEBUG] sofia_glue.c:2647 Local SDP: v=0 o=FreeSWITCH 1362024826 1362024827 IN IP4 82.85.60.237 s=FreeSWITCH c=IN IP4 82.85.60.237 t=0 0 m=audio 28398 RTP/AVP 0 9 8 3 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv 2013-02-28 13:07:04.639117 [DEBUG] mod_sofia.c:126 (sofia/internal/sip:102 at 85.18.36.68:55272) State Change CS_INIT -> CS_ROUTING 2013-02-28 13:07:04.639117 [DEBUG] switch_core_session.c:1283 Send signal sofia/internal/sip:102 at 85.18.36.68:55272 [BREAK] 2013-02-28 13:07:04.639117 [DEBUG] switch_core_state_machine.c:454 (sofia/internal/sip:102 at 85.18.36.68:55272) State INIT going to sleep 2013-02-28 13:07:04.639117 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/sip:102 at 85.18.36.68:55272) Running State Change CS_ROUTING 2013-02-28 13:07:04.639117 [DEBUG] switch_channel.c:2003 (sofia/internal/sip:102 at 85.18.36.68:55272) Callstate Change DOWN -> RINGING 2013-02-28 13:07:04.639117 [DEBUG] switch_core_state_machine.c:470 (sofia/internal/sip:102 at 85.18.36.68:55272) State ROUTING 2013-02-28 13:07:04.639117 [DEBUG] mod_sofia.c:149 sofia/internal/sip:102 at 85.18.36.68:55272 SOFIA ROUTING 2013-02-28 13:07:04.639117 [DEBUG] switch_ivr_originate.c:67 (sofia/internal/sip:102 at 85.18.36.68:55272) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2013-02-28 13:07:04.639117 [DEBUG] switch_core_session.c:1283 Send signal sofia/internal/sip:102 at 85.18.36.68:55272 [BREAK] 2013-02-28 13:07:04.639117 [DEBUG] switch_core_state_machine.c:470 (sofia/internal/sip:102 at 85.18.36.68:55272) State ROUTING going to sleep 2013-02-28 13:07:04.639117 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/sip:102 at 85.18.36.68:55272) Running State Change CS_CONSUME_MEDIA 2013-02-28 13:07:04.639117 [DEBUG] switch_core_state_machine.c:489 (sofia/internal/sip:102 at 85.18.36.68:55272) State CONSUME_MEDIA 2013-02-28 13:07:04.639117 [DEBUG] switch_core_state_machine.c:489 (sofia/internal/sip:102 at 85.18.36.68:55272) State CONSUME_MEDIA going to sleep send 1228 bytes to tls/[85.18.36.68]:55272 at 12:27:39.310041: ------------------------------------------------------------------------ INVITE sip:102 at 85.18.36.68:55272;transport=tls;rinstance=2804dee11a135346 SIP/2.0 Via: SIP/2.0/TLS 82.85.60.237;branch=z9hG4bK18Q1U05eeZBrB Max-Forwards: 69 From: "Vic Iphone" ;tag=F6pSatDmH4BcH To: Call-ID: 14fdf9be-fc45-1230-f395-005056b93f51 CSeq: 40691165 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.2.6+git~20130104T154559Z~a4247651ca Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 207 X-FS-Support: update_display,send_info Remote-Party-ID: "Vic Iphone" ;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1362024826 1362024827 IN IP4 82.85.60.237 s=FreeSWITCH c=IN IP4 82.85.60.237 t=0 0 m=audio 28398 RTP/AVP 0 9 8 3 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 ------------------------------------------------------------------------ 2013-02-28 13:07:04.639117 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/sip:102 at 85.18.36.68:55272 [BREAK] 2013-02-28 13:07:04.639117 [DEBUG] sofia.c:5599 Channel sofia/internal/sip:102 at 85.18.36.68:55272 entering state [calling][0] recv 405 bytes from tls/[85.18.36.68]:55272 at 12:27:39.335865: ------------------------------------------------------------------------ SIP/2.0 415 Unsupported Media Type Via: SIP/2.0/TLS 82.85.60.237;branch=z9hG4bK18Q1U05eeZBrB To: ;tag=d85c4e58 From: "Vic Iphone";tag=F6pSatDmH4BcH Call-ID: 14fdf9be-fc45-1230-f395-005056b93f51 CSeq: 40691165 INVITE Accept: application/sdp User-Agent: Bria 3 release 3.5.0b stamp 69410 Content-Length: 0 ------------------------------------------------------------------------ send 393 bytes to tls/[85.18.36.68]:55272 at 12:27:39.336009: ------------------------------------------------------------------------ ACK sip:102 at 85.18.36.68:55272;transport=tls;rinstance=2804dee11a135346 SIP/2.0 Via: SIP/2.0/TLS 82.85.60.237;branch=z9hG4bK18Q1U05eeZBrB Max-Forwards: 69 From: "Vic Iphone" ;tag=F6pSatDmH4BcH To: ;tag=d85c4e58 Call-ID: 14fdf9be-fc45-1230-f395-005056b93f51 CSeq: 40691165 ACK Content-Length: 0 ------------------------------------------------------------------------ 2013-02-28 13:07:04.658143 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/sip:102 at 85.18.36.68:55272 [BREAK] 2013-02-28 13:07:04.658143 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/sip:102 at 85.18.36.68:55272 [BREAK] 2013-02-28 13:07:04.658143 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/sip:102 at 85.18.36.68:55272 [BREAK] 2013-02-28 13:07:04.658143 [DEBUG] sofia.c:5599 Channel sofia/internal/sip:102 at 85.18.36.68:55272 entering state [terminated][415] 2013-02-28 13:07:04.658143 [DEBUG] switch_channel.c:2994 (sofia/internal/sip:102 at 85.18.36.68:55272) Callstate Change RINGING -> HANGUP 2013-02-28 13:07:04.658143 [NOTICE] sofia.c:6383 Hangup sofia/internal/sip:102 at 85.18.36.68:55272 [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED] 2013-02-28 13:07:04.658143 [DEBUG] switch_channel.c:3017 Send signal sofia/internal/sip:102 at 85.18.36.68:55272 [KILL] 2013-02-28 13:07:04.658143 [DEBUG] switch_core_session.c:1283 Send signal sofia/internal/sip:102 at 85.18.36.68:55272 [BREAK] 2013-02-28 13:07:04.658143 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/sip:102 at 85.18.36.68:55272) Running State Change CS_HANGUP 2013-02-28 13:07:04.658143 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/sip:102 at 85.18.36.68:55272) State HANGUP 2013-02-28 13:07:04.658143 [DEBUG] mod_sofia.c:503 Channel sofia/internal/sip:102 at 85.18.36.68:55272 hanging up, cause: SERVICE_NOT_IMPLEMENTED 2013-02-28 13:07:04.658143 [DEBUG] switch_core_state_machine.c:48 sofia/internal/sip:102 at 85.18.36.68:55272 Standard HANGUP, cause: SERVICE_NOT_IMPLEMENTED 2013-02-28 13:07:04.658143 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/sip:102 at 85.18.36.68:55272) State HANGUP going to sleep 2013-02-28 13:07:04.658143 [DEBUG] switch_core_state_machine.c:446 (sofia/internal/sip:102 at 85.18.36.68:55272) State Change CS_HANGUP -> CS_REPORTING 2013-02-28 13:07:04.658143 [DEBUG] switch_core_session.c:1283 Send signal sofia/internal/sip:102 at 85.18.36.68:55272 [BREAK] 2013-02-28 13:07:04.658143 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/sip:102 at 85.18.36.68:55272) Running State Change CS_REPORTING 2013-02-28 13:07:04.658143 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/sip:102 at 85.18.36.68:55272) State REPORTING 2013-02-28 13:07:04.658143 [DEBUG] switch_core_state_machine.c:92 sofia/internal/sip:102 at 85.18.36.68:55272 Standard REPORTING, cause: SERVICE_NOT_IMPLEMENTED 2013-02-28 13:07:04.658143 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/sip:102 at 85.18.36.68:55272) State REPORTING going to sleep 2013-02-28 13:07:04.678496 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/sip:102 at 85.18.36.68:55272) State Change CS_REPORTING -> CS_DESTROY 2013-02-28 13:07:04.678496 [DEBUG] switch_core_session.c:1283 Send signal sofia/internal/sip:102 at 85.18.36.68:55272 [BREAK] 2013-02-28 13:07:04.678496 [DEBUG] switch_core_session.c:1488 Session 22 (sofia/internal/sip:102 at 85.18.36.68:55272) Locked, Waiting on external entities 2013-02-28 13:07:04.678496 [DEBUG] switch_ivr_originate.c:3533 Originate Resulted in Error Cause: 79 [SERVICE_NOT_IMPLEMENTED] 2013-02-28 13:07:04.678496 [NOTICE] switch_ivr_originate.c:2608 Cannot create outgoing channel of type [USER] cause: [SERVICE_NOT_IMPLEMENTED] 2013-02-28 13:07:04.678496 [DEBUG] switch_ivr_originate.c:3533 Originate Resulted in Error Cause: 79 [SERVICE_NOT_IMPLEMENTED] 2013-02-28 13:07:04.678496 [NOTICE] switch_core_session.c:1506 Session 22 (sofia/internal/sip:102 at 85.18.36.68:55272) Ended 2013-02-28 13:07:04.678496 [NOTICE] switch_core_session.c:1510 Close Channel sofia/internal/sip:102 at 85.18.36.68:55272 [CS_DESTROY] 2013-02-28 13:07:04.678496 [INFO] mod_dptools.c:3052 Originate Failed. Cause: SERVICE_NOT_IMPLEMENTED 2013-02-28 13:07:04.678496 [DEBUG] switch_channel.c:2994 (sofia/internal/101 at voice.wicube.net) Callstate Change RINGING -> HANGUP 2013-02-28 13:07:04.678496 [NOTICE] mod_dptools.c:3172 Hangup sofia/internal/101 at voice.wicube.net [CS_EXECUTE] [SERVICE_NOT_IMPLEMENTED] 2013-02-28 13:07:04.678496 [DEBUG] switch_channel.c:3017 Send signal sofia/internal/101 at voice.wicube.net [KILL] 2013-02-28 13:07:04.678496 [DEBUG] switch_core_session.c:1283 Send signal sofia/internal/101 at voice.wicube.net [BREAK] 2013-02-28 13:07:04.678496 [DEBUG] switch_core_session.c:2678 sofia/internal/101 at voice.wicube.net skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2013-02-28 13:07:04.678496 [DEBUG] switch_core_state_machine.c:477 (sofia/internal/101 at voice.wicube.net) State EXECUTE going to sleep 2013-02-28 13:07:04.678496 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/101 at voice.wicube.net) Running State Change CS_HANGUP 2013-02-28 13:07:04.678496 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/101 at voice.wicube.net) State HANGUP 2013-02-28 13:07:04.678496 [DEBUG] mod_sofia.c:497 sofia/internal/101 at voice.wicube.net Overriding SIP cause 501 with 415 from the other leg 2013-02-28 13:07:04.678496 [DEBUG] mod_sofia.c:503 Channel sofia/internal/101 at voice.wicube.net hanging up, cause: SERVICE_NOT_IMPLEMENTED 2013-02-28 13:07:04.678496 [DEBUG] switch_core_state_machine.c:556 (sofia/internal/sip:102 at 85.18.36.68:55272) Callstate Change HANGUP -> DOWN 2013-02-28 13:07:04.678496 [DEBUG] switch_core_state_machine.c:559 (sofia/internal/sip:102 at 85.18.36.68:55272) Running State Change CS_DESTROY 2013-02-28 13:07:04.678496 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/sip:102 at 85.18.36.68:55272) State DESTROY 2013-02-28 13:07:04.678496 [DEBUG] mod_sofia.c:396 sofia/internal/sip:102 at 85.18.36.68:55272 SOFIA DESTROY 2013-02-28 13:07:04.678496 [DEBUG] switch_core_state_machine.c:99 sofia/internal/sip:102 at 85.18.36.68:55272 Standard DESTROY 2013-02-28 13:07:04.678496 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/sip:102 at 85.18.36.68:55272) State DESTROY going to sleep 2013-02-28 13:07:04.678496 [DEBUG] mod_sofia.c:633 Responding to INVITE with: 415 2013-02-28 13:07:04.678496 [DEBUG] switch_core_state_machine.c:48 sofia/internal/101 at voice.wicube.net Standard HANGUP, cause: SERVICE_NOT_IMPLEMENTED 2013-02-28 13:07:04.678496 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/101 at voice.wicube.net) State HANGUP going to sleep 2013-02-28 13:07:04.678496 [DEBUG] switch_core_state_machine.c:446 (sofia/internal/101 at voice.wicube.net) State Change CS_HANGUP -> CS_REPORTING 2013-02-28 13:07:04.678496 [DEBUG] switch_core_session.c:1283 Send signal sofia/internal/101 at voice.wicube.net [BREAK] 2013-02-28 13:07:04.678496 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/101 at voice.wicube.net) Running State Change CS_REPORTING 2013-02-28 13:07:04.678496 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/101 at voice.wicube.net) State REPORTING send 918 bytes to tls/[85.18.36.68]:51187 at 12:27:39.358707: ------------------------------------------------------------------------ SIP/2.0 415 Unsupported Media Type Via: SIP/2.0/TLS 85.18.36.68:51187;rport=51187;branch=z9hG4bKPjVkpmhYfPXXJ.MmmYQ62ufvtZD0K5i8Lf;alias Max-Forwards: 70 From: "101" ;tag=RQ-SQdhcZuhI5SYNSNKTY9F-9-.unsww To: ;tag=eXX08yvgmUNSN Call-ID: 5xUNyMbEzeDbgO48GR2uPiITtdGkVHm9 CSeq: 31361 INVITE User-Agent: FreeSWITCH-mod_sofia/1.2.6+git~20130104T154559Z~a4247651ca Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Reason: Q.850;cause=79;text="SERVICE_NOT_IMPLEMENTED" Content-Length: 0 Remote-Party-ID: "102" ;party=calling;privacy=off;screen=no ------------------------------------------------------------------------ 2013-02-28 13:07:04.718652 [DEBUG] switch_core_state_machine.c:92 sofia/internal/101 at voice.wicube.net Standard REPORTING, cause: SERVICE_NOT_IMPLEMENTED 2013-02-28 13:07:04.718652 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/101 at voice.wicube.net) State REPORTING going to sleep 2013-02-28 13:07:04.718652 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/101 at voice.wicube.net) State Change CS_REPORTING -> CS_DESTROY 2013-02-28 13:07:04.718652 [DEBUG] switch_core_session.c:1283 Send signal sofia/internal/101 at voice.wicube.net [BREAK] 2013-02-28 13:07:04.718652 [DEBUG] switch_core_session.c:1488 Session 21 (sofia/internal/101 at voice.wicube.net) Locked, Waiting on external entities 2013-02-28 13:07:04.718652 [NOTICE] switch_core_session.c:1506 Session 21 (sofia/internal/101 at voice.wicube.net) Ended 2013-02-28 13:07:04.718652 [NOTICE] switch_core_session.c:1510 Close Channel sofia/internal/101 at voice.wicube.net [CS_DESTROY] 2013-02-28 13:07:04.718652 [DEBUG] switch_core_state_machine.c:556 (sofia/internal/101 at voice.wicube.net) Callstate Change HANGUP -> DOWN 2013-02-28 13:07:04.718652 [DEBUG] switch_core_state_machine.c:559 (sofia/internal/101 at voice.wicube.net) Running State Change CS_DESTROY 2013-02-28 13:07:04.718652 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/101 at voice.wicube.net) State DESTROY 2013-02-28 13:07:04.718652 [DEBUG] mod_sofia.c:396 sofia/internal/101 at voice.wicube.net SOFIA DESTROY 2013-02-28 13:07:04.718652 [DEBUG] switch_core_state_machine.c:99 sofia/internal/101 at voice.wicube.net Standard DESTROY 2013-02-28 13:07:04.718652 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/101 at voice.wicube.net) State DESTROY going to sleep recv 362 bytes from tls/[85.18.36.68]:51187 at 12:27:39.415726: ------------------------------------------------------------------------ ACK sip:102 at voice.wicube.net SIP/2.0 Via: SIP/2.0/TLS 85.18.36.68:51187;rport;branch=z9hG4bKPjVkpmhYfPXXJ.MmmYQ62ufvtZD0K5i8Lf;alias Max-Forwards: 70 From: "101" ;tag=RQ-SQdhcZuhI5SYNSNKTY9F-9-.unsww To: ;tag=eXX08yvgmUNSN Call-ID: 5xUNyMbEzeDbgO48GR2uPiITtdGkVHm9 CSeq: 31361 ACK Content-Length: 0 ------------------------------------------------------------------------ -------------- next part -------------- freeswitch at internal> recv 967 bytes from tls/[85.18.36.68]:55272 at 12:33:08.114408: ------------------------------------------------------------------------ INVITE sip:101 at voice.wicube.net;transport=tls SIP/2.0 Via: SIP/2.0/TLS 172.22.55.229:23309;branch=z9hG4bK-d8754z-925e617a4fdb6345-1---d8754z-;rport Max-Forwards: 70 Contact: To: From: ;tag=6b71e108 Call-ID: NGEyMTUxZDI0YjUxNTI4MDljYjQ2NGQ2Y2ExMzRjODE CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: replaces User-Agent: Bria 3 release 3.5.0b stamp 69410 Content-Length: 393 v=0 o=- 1362054788086402 1 IN IP4 172.22.55.229 s=Bria 3 release 3.5.0b stamp 69410 c=IN IP4 172.22.55.229 b=AS:2064 t=0 0 m=audio 57780 RTP/SAVP 8 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:R2NgmDwTbaaxPRUFErm1Jcm1Ci074YlExkSiMN3Q a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:R2NgmDwTbaaxPRUFErm1Jcm1Ci074YlExkSiMN3Q a=sendrecv ------------------------------------------------------------------------ send 384 bytes to tls/[85.18.36.68]:55272 at 12:33:08.114775: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/TLS 172.22.55.229:23309;branch=z9hG4bK-d8754z-925e617a4fdb6345-1---d8754z-;rport=55272;received=85.18.36.68 From: ;tag=6b71e108 To: Call-ID: NGEyMTUxZDI0YjUxNTI4MDljYjQ2NGQ2Y2ExMzRjODE CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.2.6+git~20130104T154559Z~a4247651ca Content-Length: 0 ------------------------------------------------------------------------ 2013-02-28 13:12:33.438185 [NOTICE] switch_channel.c:968 New Channel sofia/internal/102 at voice.wicube.net [019b7184-81a3-11e2-acb9-ad22efd2ace4] 2013-02-28 13:12:33.438185 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/102 at voice.wicube.net [BREAK] 2013-02-28 13:12:33.438185 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/102 at voice.wicube.net [BREAK] 2013-02-28 13:12:33.438185 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/102 at voice.wicube.net) Running State Change CS_NEW 2013-02-28 13:12:33.438185 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/102 at voice.wicube.net) State NEW 2013-02-28 13:12:33.458149 [DEBUG] sofia.c:7729 IP 85.18.36.68 Rejected by acl "domains". Falling back to Digest auth. 2013-02-28 13:12:33.458149 [WARNING] sofia_reg.c:1502 SIP auth challenge (INVITE) on sofia profile 'internal' for [101 at voice.wicube.net] from ip 85.18.36.68 send 887 bytes to tls/[85.18.36.68]:55272 at 12:33:08.141400: ------------------------------------------------------------------------ SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/TLS 172.22.55.229:23309;branch=z9hG4bK-d8754z-925e617a4fdb6345-1---d8754z-;rport=55272;received=85.18.36.68 From: ;tag=6b71e108 To: ;tag=QFaSHmF1X86Qg Call-ID: NGEyMTUxZDI0YjUxNTI4MDljYjQ2NGQ2Y2ExMzRjODE CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.2.6+git~20130104T154559Z~a4247651ca Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Proxy-Authenticate: Digest realm="voice.wicube.net", nonce="019ea14c-81a3-11e2-acba-ad22efd2ace4", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ 2013-02-28 13:12:33.458149 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/102 at voice.wicube.net [BREAK] 2013-02-28 13:12:33.458149 [DEBUG] sofia.c:1750 detaching session 019b7184-81a3-11e2-acb9-ad22efd2ace4 recv 350 bytes from tls/[85.18.36.68]:55272 at 12:33:08.165867: ------------------------------------------------------------------------ ACK sip:101 at voice.wicube.net;transport=tls SIP/2.0 Via: SIP/2.0/TLS 172.22.55.229:23309;branch=z9hG4bK-d8754z-925e617a4fdb6345-1---d8754z-;rport Max-Forwards: 70 To: ;tag=QFaSHmF1X86Qg From: ;tag=6b71e108 Call-ID: NGEyMTUxZDI0YjUxNTI4MDljYjQ2NGQ2Y2ExMzRjODE CSeq: 1 ACK Content-Length: 0 ------------------------------------------------------------------------ recv 1247 bytes from tls/[85.18.36.68]:55272 at 12:33:08.228403: ------------------------------------------------------------------------ INVITE sip:101 at voice.wicube.net;transport=tls SIP/2.0 Via: SIP/2.0/TLS 172.22.55.229:23309;branch=z9hG4bK-d8754z-1ed3ac4e4ed2fb0d-1---d8754z-;rport Max-Forwards: 70 Contact: To: From: ;tag=6b71e108 Call-ID: NGEyMTUxZDI0YjUxNTI4MDljYjQ2NGQ2Y2ExMzRjODE CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Proxy-Authorization: Digest username="102",realm="voice.wicube.net",nonce="019ea14c-81a3-11e2-acba-ad22efd2ace4",uri="sip:101 at voice.wicube.net;transport=tls",response="c4c265a7ed630f5baf61861f213ffa85",cnonce="c2d87f66e274b12b96063e5f6d1afa92",nc=00000001,qop=auth,algorithm=MD5 Supported: replaces User-Agent: Bria 3 release 3.5.0b stamp 69410 Content-Length: 393 v=0 o=- 1362054788086402 1 IN IP4 172.22.55.229 s=Bria 3 release 3.5.0b stamp 69410 c=IN IP4 172.22.55.229 b=AS:2064 t=0 0 m=audio 57780 RTP/SAVP 8 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:R2NgmDwTbaaxPRUFErm1Jcm1Ci074YlExkSiMN3Q a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:R2NgmDwTbaaxPRUFErm1Jcm1Ci074YlExkSiMN3Q a=sendrecv ------------------------------------------------------------------------ send 384 bytes to tls/[85.18.36.68]:55272 at 12:33:08.228742: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/TLS 172.22.55.229:23309;branch=z9hG4bK-d8754z-1ed3ac4e4ed2fb0d-1---d8754z-;rport=55272;received=85.18.36.68 From: ;tag=6b71e108 To: Call-ID: NGEyMTUxZDI0YjUxNTI4MDljYjQ2NGQ2Y2ExMzRjODE CSeq: 2 INVITE User-Agent: FreeSWITCH-mod_sofia/1.2.6+git~20130104T154559Z~a4247651ca Content-Length: 0 ------------------------------------------------------------------------ 2013-02-28 13:12:33.558196 [DEBUG] sofia.c:1842 Re-attaching to session 019b7184-81a3-11e2-acb9-ad22efd2ace4 2013-02-28 13:12:33.558196 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/102 at voice.wicube.net [BREAK] 2013-02-28 13:12:33.558196 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/102 at voice.wicube.net [BREAK] 2013-02-28 13:12:33.558196 [DEBUG] sofia.c:7729 IP 85.18.36.68 Rejected by acl "domains". Falling back to Digest auth. 2013-02-28 13:12:33.578143 [DEBUG] sofia.c:5599 Channel sofia/internal/102 at voice.wicube.net entering state [received][100] 2013-02-28 13:12:33.578143 [DEBUG] sofia.c:5610 Remote SDP: v=0 o=- 1362054788086402 1 IN IP4 172.22.55.229 s=Bria 3 release 3.5.0b stamp 69410 c=IN IP4 172.22.55.229 b=AS:2064 t=0 0 m=audio 57780 RTP/SAVP 8 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:R2NgmDwTbaaxPRUFErm1Jcm1Ci074YlExkSiMN3Q a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:R2NgmDwTbaaxPRUFErm1Jcm1Ci074YlExkSiMN3Q 2013-02-28 13:12:33.578143 [DEBUG] sofia_glue.c:4978 Set Remote Key [1 AES_CM_128_HMAC_SHA1_80 inline:R2NgmDwTbaaxPRUFErm1Jcm1Ci074YlExkSiMN3Q] 2013-02-28 13:12:33.578143 [DEBUG] sofia_glue.c:3155 Set Local Key [1 AES_CM_128_HMAC_SHA1_80 inline:ctGA+qVyHINVQ/LypSDfJ+OdCPJPp57nGMPtm/nw] 2013-02-28 13:12:33.578143 [DEBUG] sofia_glue.c:5137 Audio Codec Compare [PCMA:8:8000:20:64000]/[G722:9:8000:20:64000] 2013-02-28 13:12:33.578143 [DEBUG] sofia_glue.c:5137 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMU:0:8000:20:64000] 2013-02-28 13:12:33.578143 [DEBUG] sofia_glue.c:5137 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMA:8:8000:20:64000] 2013-02-28 13:12:33.578143 [DEBUG] sofia_glue.c:3093 Set Codec sofia/internal/102 at voice.wicube.net PCMA/8000 20 ms 160 samples 64000 bits 2013-02-28 13:12:33.578143 [DEBUG] switch_core_codec.c:111 sofia/internal/102 at voice.wicube.net Original read codec set to PCMA:8 2013-02-28 13:12:33.578143 [DEBUG] sofia_glue.c:5266 Set 2833 dtmf send/recv payload to 101 2013-02-28 13:12:33.578143 [DEBUG] sofia.c:5827 (sofia/internal/102 at voice.wicube.net) State Change CS_NEW -> CS_INIT 2013-02-28 13:12:33.578143 [DEBUG] switch_core_session.c:1283 Send signal sofia/internal/102 at voice.wicube.net [BREAK] 2013-02-28 13:12:33.578143 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/102 at voice.wicube.net) Running State Change CS_INIT 2013-02-28 13:12:33.578143 [DEBUG] switch_core_state_machine.c:454 (sofia/internal/102 at voice.wicube.net) State INIT 2013-02-28 13:12:33.578143 [DEBUG] mod_sofia.c:86 sofia/internal/102 at voice.wicube.net SOFIA INIT 2013-02-28 13:12:33.578143 [DEBUG] mod_sofia.c:126 (sofia/internal/102 at voice.wicube.net) State Change CS_INIT -> CS_ROUTING 2013-02-28 13:12:33.578143 [DEBUG] switch_core_session.c:1283 Send signal sofia/internal/102 at voice.wicube.net [BREAK] 2013-02-28 13:12:33.578143 [DEBUG] switch_core_state_machine.c:454 (sofia/internal/102 at voice.wicube.net) State INIT going to sleep 2013-02-28 13:12:33.578143 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/102 at voice.wicube.net) Running State Change CS_ROUTING 2013-02-28 13:12:33.578143 [DEBUG] switch_channel.c:2003 (sofia/internal/102 at voice.wicube.net) Callstate Change DOWN -> RINGING 2013-02-28 13:12:33.578143 [DEBUG] switch_core_state_machine.c:470 (sofia/internal/102 at voice.wicube.net) State ROUTING 2013-02-28 13:12:33.578143 [DEBUG] mod_sofia.c:149 sofia/internal/102 at voice.wicube.net SOFIA ROUTING 2013-02-28 13:12:33.578143 [DEBUG] switch_core_state_machine.c:117 sofia/internal/102 at voice.wicube.net Standard ROUTING 2013-02-28 13:12:33.578143 [INFO] mod_dialplan_xml.c:557 Processing 102 <102>->101 in context default Dialplan: sofia/internal/102 at voice.wicube.net parsing [default->101] continue=false Dialplan: sofia/internal/102 at voice.wicube.net Regex (PASS) [101] destination_number(101) =~ /^101$/ break=on-false Dialplan: sofia/internal/102 at voice.wicube.net Action bridge(USER/101 at 82.85.60.237) 2013-02-28 13:12:33.578143 [DEBUG] switch_core_state_machine.c:167 (sofia/internal/102 at voice.wicube.net) State Change CS_ROUTING -> CS_EXECUTE 2013-02-28 13:12:33.578143 [DEBUG] switch_core_session.c:1283 Send signal sofia/internal/102 at voice.wicube.net [BREAK] 2013-02-28 13:12:33.578143 [DEBUG] switch_core_state_machine.c:470 (sofia/internal/102 at voice.wicube.net) State ROUTING going to sleep 2013-02-28 13:12:33.578143 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/102 at voice.wicube.net) Running State Change CS_EXECUTE 2013-02-28 13:12:33.578143 [DEBUG] switch_core_state_machine.c:477 (sofia/internal/102 at voice.wicube.net) State EXECUTE 2013-02-28 13:12:33.578143 [DEBUG] mod_sofia.c:242 sofia/internal/102 at voice.wicube.net SOFIA EXECUTE 2013-02-28 13:12:33.578143 [DEBUG] switch_core_state_machine.c:209 sofia/internal/102 at voice.wicube.net Standard EXECUTE EXECUTE sofia/internal/102 at voice.wicube.net bridge(USER/101 at 82.85.60.237) 2013-02-28 13:12:33.578143 [DEBUG] switch_channel.c:1089 sofia/internal/102 at voice.wicube.net EXPORTING[export_vars] [domain_name]=[82.85.60.237] to event 2013-02-28 13:12:33.578143 [DEBUG] switch_ivr_originate.c:2022 Parsing global variables 2013-02-28 13:12:33.578143 [DEBUG] switch_channel.c:1089 sofia/internal/102 at voice.wicube.net EXPORTING[export_vars] [domain_name]=[82.85.60.237] to event 2013-02-28 13:12:33.578143 [DEBUG] switch_ivr_originate.c:2022 Parsing global variables 2013-02-28 13:12:33.578143 [DEBUG] switch_event.c:1608 Parsing variable [sip_invite_domain]=[82.85.60.237] 2013-02-28 13:12:33.578143 [DEBUG] switch_event.c:1608 Parsing variable [presence_id]=[101 at 82.85.60.237] 2013-02-28 13:12:33.578143 [NOTICE] switch_channel.c:968 New Channel sofia/internal/sip:101 at 85.18.36.68:51187 [01b12f2e-81a3-11e2-acc3-ad22efd2ace4] 2013-02-28 13:12:33.578143 [DEBUG] mod_sofia.c:4970 (sofia/internal/sip:101 at 85.18.36.68:51187) State Change CS_NEW -> CS_INIT 2013-02-28 13:12:33.578143 [DEBUG] switch_core_session.c:1283 Send signal sofia/internal/sip:101 at 85.18.36.68:51187 [BREAK] 2013-02-28 13:12:33.578143 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/sip:101 at 85.18.36.68:51187) Running State Change CS_INIT 2013-02-28 13:12:33.578143 [DEBUG] switch_core_state_machine.c:454 (sofia/internal/sip:101 at 85.18.36.68:51187) State INIT 2013-02-28 13:12:33.578143 [DEBUG] mod_sofia.c:86 sofia/internal/sip:101 at 85.18.36.68:51187 SOFIA INIT 2013-02-28 13:12:33.578143 [DEBUG] sofia_glue.c:2618 sip:101 at 85.18.36.68:51187;transport=TLS;ob Setting proxy route to sofia/internal/sip:101 at 85.18.36.68:51187 2013-02-28 13:12:33.578143 [DEBUG] sofia_glue.c:2647 Local SDP: v=0 o=FreeSWITCH 1362036675 1362036676 IN IP4 82.85.60.237 s=FreeSWITCH c=IN IP4 82.85.60.237 t=0 0 m=audio 16878 RTP/AVP 8 9 0 3 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv 2013-02-28 13:12:33.578143 [DEBUG] mod_sofia.c:126 (sofia/internal/sip:101 at 85.18.36.68:51187) State Change CS_INIT -> CS_ROUTING 2013-02-28 13:12:33.578143 [DEBUG] switch_core_session.c:1283 Send signal sofia/internal/sip:101 at 85.18.36.68:51187 [BREAK] 2013-02-28 13:12:33.578143 [DEBUG] switch_core_state_machine.c:454 (sofia/internal/sip:101 at 85.18.36.68:51187) State INIT going to sleep 2013-02-28 13:12:33.578143 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/sip:101 at 85.18.36.68:51187) Running State Change CS_ROUTING 2013-02-28 13:12:33.578143 [DEBUG] switch_channel.c:2003 (sofia/internal/sip:101 at 85.18.36.68:51187) Callstate Change DOWN -> RINGING 2013-02-28 13:12:33.578143 [DEBUG] switch_core_state_machine.c:470 (sofia/internal/sip:101 at 85.18.36.68:51187) State ROUTING 2013-02-28 13:12:33.578143 [DEBUG] mod_sofia.c:149 sofia/internal/sip:101 at 85.18.36.68:51187 SOFIA ROUTING 2013-02-28 13:12:33.578143 [DEBUG] switch_ivr_originate.c:67 (sofia/internal/sip:101 at 85.18.36.68:51187) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2013-02-28 13:12:33.578143 [DEBUG] switch_core_session.c:1283 Send signal sofia/internal/sip:101 at 85.18.36.68:51187 [BREAK] 2013-02-28 13:12:33.578143 [DEBUG] switch_core_state_machine.c:470 (sofia/internal/sip:101 at 85.18.36.68:51187) State ROUTING going to sleep 2013-02-28 13:12:33.578143 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/sip:101 at 85.18.36.68:51187) Running State Change CS_CONSUME_MEDIA 2013-02-28 13:12:33.578143 [DEBUG] switch_core_state_machine.c:489 (sofia/internal/sip:101 at 85.18.36.68:51187) State CONSUME_MEDIA 2013-02-28 13:12:33.578143 [DEBUG] switch_core_state_machine.c:489 (sofia/internal/sip:101 at 85.18.36.68:51187) State CONSUME_MEDIA going to sleep send 1182 bytes to tls/[85.18.36.68]:51187 at 12:33:08.259885: ------------------------------------------------------------------------ INVITE sip:101 at 85.18.36.68:51187;transport=TLS;ob SIP/2.0 Via: SIP/2.0/TLS 82.85.60.237;branch=z9hG4bK3taKZp7N8grXj Max-Forwards: 69 From: "Vic Mac Pro" ;tag=S1vaNaH8QtKXQ To: Call-ID: d90fbc70-fc45-1230-f395-005056b93f51 CSeq: 40691330 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.2.6+git~20130104T154559Z~a4247651ca Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 207 X-FS-Support: update_display,send_info Remote-Party-ID: "Vic Mac Pro" ;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1362036675 1362036676 IN IP4 82.85.60.237 s=FreeSWITCH c=IN IP4 82.85.60.237 t=0 0 m=audio 16878 RTP/AVP 8 9 0 3 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 ------------------------------------------------------------------------ 2013-02-28 13:12:33.578143 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/sip:101 at 85.18.36.68:51187 [BREAK] 2013-02-28 13:12:33.578143 [DEBUG] sofia.c:5599 Channel sofia/internal/sip:101 at 85.18.36.68:51187 entering state [calling][0] recv 341 bytes from tls/[85.18.36.68]:51187 at 12:33:08.332434: ------------------------------------------------------------------------ SIP/2.0 488 Not Acceptable Here Via: SIP/2.0/TLS 82.85.60.237;received=82.85.60.237;branch=z9hG4bK3taKZp7N8grXj;alias Call-ID: d90fbc70-fc45-1230-f395-005056b93f51 From: "Vic Mac Pro" ;tag=S1vaNaH8QtKXQ To: ;tag=GSUFFx7sd2US6fRO3fWnz40WRNdua4Eo CSeq: 40691330 INVITE Content-Length: 0 ------------------------------------------------------------------------ send 350 bytes to tls/[85.18.36.68]:51187 at 12:33:08.332571: ------------------------------------------------------------------------ ACK sip:101 at 85.18.36.68:51187;transport=TLS;ob SIP/2.0 Via: SIP/2.0/TLS 82.85.60.237;branch=z9hG4bK3taKZp7N8grXj Max-Forwards: 69 From: "Vic Mac Pro" ;tag=S1vaNaH8QtKXQ To: ;tag=GSUFFx7sd2US6fRO3fWnz40WRNdua4Eo Call-ID: d90fbc70-fc45-1230-f395-005056b93f51 CSeq: 40691330 ACK Content-Length: 0 ------------------------------------------------------------------------ 2013-02-28 13:12:33.658123 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/sip:101 at 85.18.36.68:51187 [BREAK] 2013-02-28 13:12:33.658123 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/sip:101 at 85.18.36.68:51187 [BREAK] 2013-02-28 13:12:33.658123 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/sip:101 at 85.18.36.68:51187 [BREAK] 2013-02-28 13:12:33.658123 [DEBUG] sofia.c:5599 Channel sofia/internal/sip:101 at 85.18.36.68:51187 entering state [terminated][488] 2013-02-28 13:12:33.658123 [DEBUG] switch_channel.c:2994 (sofia/internal/sip:101 at 85.18.36.68:51187) Callstate Change RINGING -> HANGUP 2013-02-28 13:12:33.658123 [NOTICE] sofia.c:6383 Hangup sofia/internal/sip:101 at 85.18.36.68:51187 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] 2013-02-28 13:12:33.658123 [DEBUG] switch_channel.c:3017 Send signal sofia/internal/sip:101 at 85.18.36.68:51187 [KILL] 2013-02-28 13:12:33.658123 [DEBUG] switch_core_session.c:1283 Send signal sofia/internal/sip:101 at 85.18.36.68:51187 [BREAK] 2013-02-28 13:12:33.658123 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/sip:101 at 85.18.36.68:51187) Running State Change CS_HANGUP 2013-02-28 13:12:33.658123 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/sip:101 at 85.18.36.68:51187) State HANGUP 2013-02-28 13:12:33.658123 [DEBUG] mod_sofia.c:503 Channel sofia/internal/sip:101 at 85.18.36.68:51187 hanging up, cause: INCOMPATIBLE_DESTINATION 2013-02-28 13:12:33.658123 [DEBUG] switch_core_state_machine.c:48 sofia/internal/sip:101 at 85.18.36.68:51187 Standard HANGUP, cause: INCOMPATIBLE_DESTINATION 2013-02-28 13:12:33.658123 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/sip:101 at 85.18.36.68:51187) State HANGUP going to sleep 2013-02-28 13:12:33.658123 [DEBUG] switch_core_state_machine.c:446 (sofia/internal/sip:101 at 85.18.36.68:51187) State Change CS_HANGUP -> CS_REPORTING 2013-02-28 13:12:33.658123 [DEBUG] switch_core_session.c:1283 Send signal sofia/internal/sip:101 at 85.18.36.68:51187 [BREAK] 2013-02-28 13:12:33.658123 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/sip:101 at 85.18.36.68:51187) Running State Change CS_REPORTING 2013-02-28 13:12:33.658123 [DEBUG] switch_ivr_originate.c:3533 Originate Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION] 2013-02-28 13:12:33.658123 [NOTICE] switch_ivr_originate.c:2608 Cannot create outgoing channel of type [USER] cause: [INCOMPATIBLE_DESTINATION] 2013-02-28 13:12:33.658123 [DEBUG] switch_ivr_originate.c:3533 Originate Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION] 2013-02-28 13:12:33.658123 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/sip:101 at 85.18.36.68:51187) State REPORTING 2013-02-28 13:12:33.658123 [DEBUG] switch_core_state_machine.c:92 sofia/internal/sip:101 at 85.18.36.68:51187 Standard REPORTING, cause: INCOMPATIBLE_DESTINATION 2013-02-28 13:12:33.658123 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/sip:101 at 85.18.36.68:51187) State REPORTING going to sleep 2013-02-28 13:12:33.658123 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/sip:101 at 85.18.36.68:51187) State Change CS_REPORTING -> CS_DESTROY 2013-02-28 13:12:33.658123 [DEBUG] switch_core_session.c:1283 Send signal sofia/internal/sip:101 at 85.18.36.68:51187 [BREAK] 2013-02-28 13:12:33.658123 [DEBUG] switch_core_session.c:1488 Session 24 (sofia/internal/sip:101 at 85.18.36.68:51187) Locked, Waiting on external entities 2013-02-28 13:12:33.658123 [NOTICE] switch_core_session.c:1506 Session 24 (sofia/internal/sip:101 at 85.18.36.68:51187) Ended 2013-02-28 13:12:33.658123 [NOTICE] switch_core_session.c:1510 Close Channel sofia/internal/sip:101 at 85.18.36.68:51187 [CS_DESTROY] 2013-02-28 13:12:33.658123 [DEBUG] switch_core_state_machine.c:556 (sofia/internal/sip:101 at 85.18.36.68:51187) Callstate Change HANGUP -> DOWN 2013-02-28 13:12:33.658123 [INFO] mod_dptools.c:3052 Originate Failed. Cause: INCOMPATIBLE_DESTINATION 2013-02-28 13:12:33.658123 [DEBUG] switch_channel.c:2994 (sofia/internal/102 at voice.wicube.net) Callstate Change RINGING -> HANGUP 2013-02-28 13:12:33.658123 [NOTICE] mod_dptools.c:3172 Hangup sofia/internal/102 at voice.wicube.net [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] 2013-02-28 13:12:33.658123 [DEBUG] switch_channel.c:3017 Send signal sofia/internal/102 at voice.wicube.net [KILL] 2013-02-28 13:12:33.658123 [DEBUG] switch_core_session.c:1283 Send signal sofia/internal/102 at voice.wicube.net [BREAK] 2013-02-28 13:12:33.658123 [DEBUG] switch_core_state_machine.c:559 (sofia/internal/sip:101 at 85.18.36.68:51187) Running State Change CS_DESTROY 2013-02-28 13:12:33.658123 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/sip:101 at 85.18.36.68:51187) State DESTROY 2013-02-28 13:12:33.658123 [DEBUG] mod_sofia.c:396 sofia/internal/sip:101 at 85.18.36.68:51187 SOFIA DESTROY 2013-02-28 13:12:33.658123 [DEBUG] switch_core_state_machine.c:99 sofia/internal/sip:101 at 85.18.36.68:51187 Standard DESTROY 2013-02-28 13:12:33.658123 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/sip:101 at 85.18.36.68:51187) State DESTROY going to sleep 2013-02-28 13:12:33.658123 [DEBUG] switch_core_session.c:2678 sofia/internal/102 at voice.wicube.net skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2013-02-28 13:12:33.658123 [DEBUG] switch_core_state_machine.c:477 (sofia/internal/102 at voice.wicube.net) State EXECUTE going to sleep 2013-02-28 13:12:33.658123 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/102 at voice.wicube.net) Running State Change CS_HANGUP 2013-02-28 13:12:33.658123 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/102 at voice.wicube.net) State HANGUP 2013-02-28 13:12:33.658123 [DEBUG] mod_sofia.c:497 sofia/internal/102 at voice.wicube.net Overriding SIP cause 488 with 488 from the other leg 2013-02-28 13:12:33.658123 [DEBUG] mod_sofia.c:503 Channel sofia/internal/102 at voice.wicube.net hanging up, cause: INCOMPATIBLE_DESTINATION 2013-02-28 13:12:33.678270 [DEBUG] mod_sofia.c:633 Responding to INVITE with: 488 2013-02-28 13:12:33.678270 [DEBUG] switch_core_state_machine.c:48 sofia/internal/102 at voice.wicube.net Standard HANGUP, cause: INCOMPATIBLE_DESTINATION 2013-02-28 13:12:33.678270 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/102 at voice.wicube.net) State HANGUP going to sleep 2013-02-28 13:12:33.678270 [DEBUG] switch_core_state_machine.c:446 (sofia/internal/102 at voice.wicube.net) State Change CS_HANGUP -> CS_REPORTING 2013-02-28 13:12:33.678270 [DEBUG] switch_core_session.c:1283 Send signal sofia/internal/102 at voice.wicube.net [BREAK] 2013-02-28 13:12:33.678270 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/102 at voice.wicube.net) Running State Change CS_REPORTING 2013-02-28 13:12:33.678270 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/102 at voice.wicube.net) State REPORTING send 912 bytes to tls/[85.18.36.68]:55272 at 12:33:08.356137: ------------------------------------------------------------------------ SIP/2.0 488 Not Acceptable Here Via: SIP/2.0/TLS 172.22.55.229:23309;branch=z9hG4bK-d8754z-1ed3ac4e4ed2fb0d-1---d8754z-;rport=55272;received=85.18.36.68 Max-Forwards: 70 From: ;tag=6b71e108 To: ;tag=rr3HKF04tHXac Call-ID: NGEyMTUxZDI0YjUxNTI4MDljYjQ2NGQ2Y2ExMzRjODE CSeq: 2 INVITE User-Agent: FreeSWITCH-mod_sofia/1.2.6+git~20130104T154559Z~a4247651ca Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" Content-Length: 0 Remote-Party-ID: "101" ;party=calling;privacy=off;screen=no ------------------------------------------------------------------------ 2013-02-28 13:12:33.721161 [DEBUG] switch_core_state_machine.c:92 sofia/internal/102 at voice.wicube.net Standard REPORTING, cause: INCOMPATIBLE_DESTINATION 2013-02-28 13:12:33.721161 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/102 at voice.wicube.net) State REPORTING going to sleep 2013-02-28 13:12:33.721161 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/102 at voice.wicube.net) State Change CS_REPORTING -> CS_DESTROY 2013-02-28 13:12:33.721161 [DEBUG] switch_core_session.c:1283 Send signal sofia/internal/102 at voice.wicube.net [BREAK] 2013-02-28 13:12:33.721161 [DEBUG] switch_core_session.c:1488 Session 23 (sofia/internal/102 at voice.wicube.net) Locked, Waiting on external entities 2013-02-28 13:12:33.721161 [NOTICE] switch_core_session.c:1506 Session 23 (sofia/internal/102 at voice.wicube.net) Ended 2013-02-28 13:12:33.721161 [NOTICE] switch_core_session.c:1510 Close Channel sofia/internal/102 at voice.wicube.net [CS_DESTROY] 2013-02-28 13:12:33.721161 [DEBUG] switch_core_state_machine.c:556 (sofia/internal/102 at voice.wicube.net) Callstate Change HANGUP -> DOWN 2013-02-28 13:12:33.721161 [DEBUG] switch_core_state_machine.c:559 (sofia/internal/102 at voice.wicube.net) Running State Change CS_DESTROY 2013-02-28 13:12:33.721161 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/102 at voice.wicube.net) State DESTROY 2013-02-28 13:12:33.721161 [DEBUG] mod_sofia.c:396 sofia/internal/102 at voice.wicube.net SOFIA DESTROY 2013-02-28 13:12:33.721161 [DEBUG] switch_core_state_machine.c:99 sofia/internal/102 at voice.wicube.net Standard DESTROY 2013-02-28 13:12:33.721161 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/102 at voice.wicube.net) State DESTROY going to sleep recv 350 bytes from tls/[85.18.36.68]:55272 at 12:33:08.426513: ------------------------------------------------------------------------ ACK sip:101 at voice.wicube.net;transport=tls SIP/2.0 Via: SIP/2.0/TLS 172.22.55.229:23309;branch=z9hG4bK-d8754z-1ed3ac4e4ed2fb0d-1---d8754z-;rport Max-Forwards: 70 To: ;tag=rr3HKF04tHXac From: ;tag=6b71e108 Call-ID: NGEyMTUxZDI0YjUxNTI4MDljYjQ2NGQ2Y2ExMzRjODE CSeq: 2 ACK Content-Length: 0 ------------------------------------------------------------------------ From manuel at aguete.org Thu Feb 28 16:08:49 2013 From: manuel at aguete.org (=?ISO-8859-1?Q?Manuel_Dur=E1n_Aguete?=) Date: Thu, 28 Feb 2013 14:08:49 +0100 Subject: [Freeswitch-users] Audio dropped using unimrcp tts. In-Reply-To: References: Message-ID: Thank you Christopher, I'm going to review loquendo server for misconfgured settings or network issues. Regards. 2013/2/27 Christopher Rienzo > Sorry for the delay. I looked at the pcap and it confirmed my suspicion. > The MRCP server doesn't reply with IN-PROGRESS until 1.35 sec of audio has > already been transmitted. > > IN-PROGRESS tells mod_unimrcp that the request is has been accepted and is > being processed. The module assumes the reply is immediately sent before > the RTP is streamed. > > Chris > > > > On Mon, Feb 18, 2013 at 6:09 AM, Manuel Dur?n Aguete wrote: > >> Hello, >> >> I've update the dropbox folder with the pcap and mod_unimrcp_rx/tx files, >> added the log file too. >> >> https://www.dropbox.com/sh/tcgnktzzby0h2ns/v7ViiUyObB >> >> I've tested with branch 1.2 and with master, the pcap is from master code. >> >> Thank you for your help. >> >> Regards. >> >> >> >> >> 2013/2/15 Christopher Rienzo >> >>> Send PCAP with the RTSP signaling and RTP audio. I'm curious when the >>> server starts streaming the audio. >>> >>> >>> >>> On Fri, Feb 15, 2013 at 4:20 AM, Manuel Dur?n Aguete wrote: >>> >>>> Hello, >>>> >>>> I've tried with different values for sleep, but the problem persists, >>>> in http://pastebin.freeswitch.org/20594 i've posted the call log. >>>> I've enabled MOD_UNIMRCP_DEBUG_AUDIO_QUEUE in mod_unircp.cm the audio >>>> streams generated can be downloaded from >>>> https://www.dropbox.com/sh/tcgnktzzby0h2ns/v7ViiUyObB. >>>> >>>> I've changed the speak data to >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> and only "3 4 5 6 7 8 9 0" is played in softphone. >>>> >>>> Thank you for your help. >>>> >>>> Regards. >>>> >>>> >>>> >>>> >>>> >>>> 2013/2/14 Michael Collins >>>> >>>>> Try adding a sleep in your dialplan prior to the call to "speak" >>>>> application. Start with 1000 ms and see if it makes any difference, then >>>>> tweak the sleep time from there. >>>>> >>>>> -MC >>>>> >>>>> On Thu, Feb 14, 2013 at 5:26 AM, Manuel Dur?n Aguete < >>>>> manuel at aguete.org> wrote: >>>>> >>>>>> Hello, >>>>>> >>>>>> I'm a newbie testing freeswitch tts with loquendo mrcp1 server, >>>>>> >>>>>> mrcp_profile: >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Dialplan code: >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> I can hear all text with first word dropped. I've captured the rtp >>>>>> traffic and the audio between FS <-> Loquendo contains the word. After >>>>>> apply the fixed recommended in >>>>>> http://wiki.freeswitch.org/wiki/RTP_Issues#Dropped_Audio the issue >>>>>> persists. >>>>>> >>>>>> Any idea, suggestion, etc ? >>>>>> >>>>>> Thank you. >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> 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 >>>>>> >>>>>> >>>>> >>>>> >>>>> -- >>>>> Michael S Collins >>>>> Twitter: @mercutioviz >>>>> http://www.FreeSWITCH.org >>>>> http://www.ClueCon.com >>>>> http://www.OSTAG.org >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> 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 >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130228/f490bcd1/attachment.html From x.liu at hw.ac.uk Thu Feb 28 17:04:00 2013 From: x.liu at hw.ac.uk (x.liu) Date: Thu, 28 Feb 2013 14:04:00 +0000 Subject: [Freeswitch-users] issues in simultaneous speech recognition and TTS via mod_unimrcp In-Reply-To: <511BB4D8.70505@hw.ac.uk> References: <511BB4D8.70505@hw.ac.uk> Message-ID: <512F63D0.30601@hw.ac.uk> I can use play_and_detect_speech app and set Recognition-Mode=hotword to do the hotword barge-in. This is working. I am still trying detect_speech and speak for simultaneous speech recognition. I can receive the events like DETECTED_SPEECH or BEGIN_SPEAKING if I speak after the TTS finishes The problem is that I can not receive those events in my ESL client if I speak during the TTS is playing, but I can see there are some recognition results from FS console. Tried various things, didn't get it work. Does anybody have any clue on this? Thanks! Xing On 02/13/2013 03:44 PM, x.liu wrote: > Hello, > > I am trying to do simultaneous speech recognition and TTS via > mod_unimrcp using detect_speech and speak. > > What I want to achieve is to enable a user to say something like "shut > up" to stop TTS rather than using standard barge-in to stop TTS. > > When I spoke something during playing a sentence, I noticed there were > some recognition results from FS log, > but my ESL app didn't get any DETECTED_SPEECH or BEGIN_SPEAKING events. > > Has any body had similar experiences before? > > From Arsen Chaloyan's comments at 11 July 2011: > > "If I'm not mistaken, neither Asterisk nor FreeSWITCH are currently > capable to play a TTS prompt while making a recognition. However, I > don't see any conceptual problem here. This is somewhat not > implemented yet for those open source projects. Other UniMRCP based > implementations do this job very well." > > I am wondering if the current mod_unimrcp does not support concurrent > recognition and tts > or if I am not using it properly? > > > Thanks! > > Xing > > > ------------------------------------------------------------------------ > > Sunday Times Scottish University of the Year 2011-2013 > Top in the UK for student experience > Fourth university in the UK and top in Scotland (National Student > Survey 2012) > > We invite research leaders and ambitious early career researchers to > join us in leading and driving research in key inter-disciplinary > themes. Please see www.hw.ac.uk/researchleaders for further > information and how to apply. > > Heriot-Watt University is a Scottish charity registered under charity > number SC000278. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 ----- Sunday Times Scottish University of the Year 2011-2013 Top in the UK for student experience Fourth university in the UK and top in Scotland (National Student Survey 2012) We invite research leaders and ambitious early career researchers to join us in leading and driving research in key inter-disciplinary themes. Please see www.hw.ac.uk/researchleaders for further information and how to apply. Heriot-Watt University is a Scottish charity registered under charity number SC000278. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130228/19761f3a/attachment-0001.html From anthony.minessale at gmail.com Thu Feb 28 19:42:36 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 28 Feb 2013 10:42:36 -0600 Subject: [Freeswitch-users] FS with SSL/TLS issues! In-Reply-To: <512F4F54.8020903@gmail.com> References: <1361367187947-7587736.post@n2.nabble.com> <1361513236022-7587844.post@n2.nabble.com> <1361519970790-7587849.post@n2.nabble.com> <113D484C-EA7E-4611-B958-6514F5F2B5B9@freeswitch.org> <1361875157474-7588012.post@n2.nabble.com> <1361976425293-7588069.post@n2.nabble.com> <512E7B89.6030903@gmail.com> <512E8516.1060008@gmail.com> <512E8F08.40204@gmail.com> <512F4F54.8020903@gmail.com> Message-ID: They probably do not accept calls over TLS that are not SAVP On Thu, Feb 28, 2013 at 6:36 AM, Vittorio Guglielmo < vittorio.guglielmo at gmail.com> wrote: > Anthony, > > sofia global siptrace on > > and in attachment are the two siptrace: > > > 1. *101* ( Bria iOS 2.3.4 on Iphone 3GS) *===> 102* (Bria 3 release > 3.5.0b on Mac OSX 10.7.5) > 2. *102* *===> 101* > > Both of them have forced only ulaw (first) and alaw (second), and without > TSL/SRTP they call each other without problems. > > > The behavior is different in the two above cases: > > When I call from 101 to 102 I get a *Originate Resulted in Error Cause: > 79 [SERVICE_NOT_IMPLEMENTED]* > due to, I believe a: *SIP/2.0 415 Unsupported Media Type* > > When I instead call from 102 to 101 I get a *Originate Failed. Cause: > INCOMPATIBLE_DESTINATION* > with a : * SIP/2.0 488 Not Acceptable Here* > > > Vic > > On 28/02/13 00:23, Anthony Minessale wrote: > > sofia global siptrace on > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130228/e7e4e56f/attachment.html From the_saids at hotmail.com Thu Feb 28 20:00:58 2013 From: the_saids at hotmail.com (Hussein Said) Date: Thu, 28 Feb 2013 12:00:58 -0500 Subject: [Freeswitch-users] dialplan action when sending a REFER to FreeSWITCH In-Reply-To: References: <1361367187947-7587736.post@n2.nabble.com>, , <1361513236022-7587844.post@n2.nabble.com>, <1361519970790-7587849.post@n2.nabble.com>, <113D484C-EA7E-4611-B958-6514F5F2B5B9@freeswitch.org>, <1361875157474-7588012.post@n2.nabble.com>, <1361976425293-7588069.post@n2.nabble.com>, <512E7B89.6030903@gmail.com>, , <512E8516.1060008@gmail.com>, , <512E8F08.40204@gmail.com>, , <512F4F54.8020903@gmail.com>, Message-ID: Sorry for the repost. I had to change my email to be able to respond to the list. I did post in pastebin a snippet of the dial plan. Pretty much, every thing is set to the default installation. To clarify my issue, I need to be able to execute a dial plan when FreeSWITCH sends an INVITE in response to a REFER. So far, I am not able to locate a setting that would allow me to do that. The switch_ivr_originate.c file transitions the channel from CS_ROUTING to CS_CONSUME_MEDIA which I believe is bypassing the CS_EXECUTE to trigger executing the dialplan. thanks ------------------------------------- Michael Collins msc at freeswitch.org Tue Feb 26 22:17:12 MSK 2013 Previous message: [Freeswitch-users] dialplan action when sending a REFER to FreeSWITCH Next message: [Freeswitch-users] nibblebill, spidermonkey_odbc issue Messages sorted by: [ date ] [ thread ] [ subject ] [ author ] Would you mind putting a snippet of your dialplan into our pastebin? ( pastebin.freeswitch.org) I'm sure the folks here can help you figure it out. With FreeSWITCH there's almost always a way to do it and it's just a matter of figuring out what works for your scenario. -MC On Tue, Feb 26, 2013 at 10:44 AM, Hussein Said < Hussein.Said at cassidiancommunications.com> wrote: > Hi,**** > > I have an application that routes a call to an ad-hoc conference. That > application then hunts an extension to answer the call by sending a REFER > to FreeSWITCH which will send an INVITE from the conference to join. **** > > I need a way to make the ?hunted? extension the moderator so that it can > control the conference. **** > > I did set a moderator controls profile and can see it works if I change > the dial plan(public.xml) to set the ?member-flag=moderator?.**** > > Also, ?mod-wait|rfc4579? are set in the conference.conf.xml.**** > > My challenge is that when I hunt for an extension and find one, I send a > REFER to FREESwitch so that it sends an INVITE to the extension that needs > to answer the call, and I could not find a variable that worked in the > dialplan that I can use as a condition to set the ?member-flag=moderator? > for the channel that is about to join the conference. Is what I am > attempting feasible?**** > > ** ** > > Thanks**** > > ** ** > > ** ** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130226/e243b2df/attachment.html -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130228/146722ba/attachment.html From vittorio.guglielmo at gmail.com Thu Feb 28 20:03:10 2013 From: vittorio.guglielmo at gmail.com (Vittorio Guglielmo) Date: Thu, 28 Feb 2013 18:03:10 +0100 Subject: [Freeswitch-users] FS with SSL/TLS issues! In-Reply-To: References: <1361367187947-7587736.post@n2.nabble.com> <1361513236022-7587844.post@n2.nabble.com> <1361519970790-7587849.post@n2.nabble.com> <113D484C-EA7E-4611-B958-6514F5F2B5B9@freeswitch.org> <1361875157474-7588012.post@n2.nabble.com> <1361976425293-7588069.post@n2.nabble.com> <512E7B89.6030903@gmail.com> <512E8516.1060008@gmail.com> <512E8F08.40204@gmail.com> <512F4F54.8020903@gmail.com> Message-ID: <512F8DCE.8020804@gmail.com> Can you please explain to me what does it mean ? And, in case, how can I demonstrate it, cause I need to buy a lot of clients to build a secured voip network. Thank you very much for your time and your effort on a so huge piece of software ( I bought all of your books ). Vic On 28/02/13 17:42, Anthony Minessale wrote: > They probably do not accept calls over TLS that are not SAVP > > > On Thu, Feb 28, 2013 at 6:36 AM, Vittorio Guglielmo > > > wrote: > > Anthony, > > sofia global siptrace on > > and in attachment are the two siptrace: > > 1. *101* ( Bria iOS 2.3.4 on Iphone 3GS) *===> 102* (Bria 3 > release 3.5.0b on Mac OSX 10.7.5) > 2. *102* *===> 101* > > Both of them have forced only ulaw (first) and alaw (second), and > without TSL/SRTP they call each other without problems. > > > The behavior is different in the two above cases: > > When I call from 101 to 102 I get a *Originate Resulted in Error > Cause: 79 [SERVICE_NOT_IMPLEMENTED]* > due to, I believe a: *SIP/2.0 415 Unsupported Media Type* > > When I instead call from 102 to 101 I get a *Originate Failed. > Cause: INCOMPATIBLE_DESTINATION* > with a : *SIP/2.0 488 Not Acceptable Here* > > > Vic > > On 28/02/13 00:23, Anthony Minessale wrote: >> sofia global siptrace on >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > googletalk:conf+888 at conference.freeswitch.org > > pstn:+19193869900 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130228/234ec07d/attachment-0001.html From anthony.minessale at gmail.com Thu Feb 28 20:29:29 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 28 Feb 2013 11:29:29 -0600 Subject: [Freeswitch-users] FS with SSL/TLS issues! In-Reply-To: <512F8DCE.8020804@gmail.com> References: <1361367187947-7587736.post@n2.nabble.com> <1361513236022-7587844.post@n2.nabble.com> <1361519970790-7587849.post@n2.nabble.com> <113D484C-EA7E-4611-B958-6514F5F2B5B9@freeswitch.org> <1361875157474-7588012.post@n2.nabble.com> <1361976425293-7588069.post@n2.nabble.com> <512E7B89.6030903@gmail.com> <512E8516.1060008@gmail.com> <512E8F08.40204@gmail.com> <512F4F54.8020903@gmail.com> <512F8DCE.8020804@gmail.com> Message-ID: I mean, the phone may expect SRTP to be offered over the TLS socket. You were offering regular RTP profile "AVP" with no crypto in it and they rejected it as unsupported. That makes me guess they require you to enable srtp "SAVP" + crypto tags or they will drop the call. try adding {sip_secure_media=true} in the outbound call to the phone. On Thu, Feb 28, 2013 at 11:03 AM, Vittorio Guglielmo < vittorio.guglielmo at gmail.com> wrote: > Can you please explain to me what does it mean ? > > And, in case, how can I demonstrate it, cause I need to buy a lot of > clients to build a secured voip network. > > Thank you very much for your time and your effort on a so huge piece of > software ( I bought all of your books ). > > > Vic > > > > > On 28/02/13 17:42, Anthony Minessale wrote: > > They probably do not accept calls over TLS that are not SAVP > > > On Thu, Feb 28, 2013 at 6:36 AM, Vittorio Guglielmo < > vittorio.guglielmo at gmail.com> wrote: > >> Anthony, >> >> sofia global siptrace on >> >> and in attachment are the two siptrace: >> >> >> 1. *101* ( Bria iOS 2.3.4 on Iphone 3GS) *===> 102* (Bria 3 release >> 3.5.0b on Mac OSX 10.7.5) >> 2. *102* *===> 101* >> >> Both of them have forced only ulaw (first) and alaw (second), and without >> TSL/SRTP they call each other without problems. >> >> >> The behavior is different in the two above cases: >> >> When I call from 101 to 102 I get a *Originate Resulted in Error Cause: >> 79 [SERVICE_NOT_IMPLEMENTED]* >> due to, I believe a: *SIP/2.0 415 Unsupported Media Type* >> >> When I instead call from 102 to 101 I get a *Originate Failed. Cause: >> INCOMPATIBLE_DESTINATION* >> with a : * SIP/2.0 488 Not Acceptable Here* >> >> >> Vic >> >> On 28/02/13 00:23, Anthony Minessale wrote: >> >> sofia global siptrace on >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130228/54ec0a4d/attachment.html From a.venugopan at mundio.com Thu Feb 28 20:55:30 2013 From: a.venugopan at mundio.com (Archana Venugopan) Date: Thu, 28 Feb 2013 17:55:30 +0000 Subject: [Freeswitch-users] [ERR] switch_ivr_play_say.c:142 Can't find macro Message-ID: <592A9CF93E12394E8472A6CC66E66BF23AE0C7@Mail-Kilo.squay.com> Hi , I have created new sounds folder inside en folder and placed new_message_count and I have called this 'new_message_count' in my js script. The wav files inside this macro is being read correctly in development environment whereas in my production which is almost all same as development environment except for the src folder(/usr/srcFreeswitch) does not recognise this 'new_message_count' and throwing this error(2013-02-28 17:37:47.099253 [ERR] switch_ivr_play_say.c:142 Can't find macro new_message_count.) I even gave reload mod_say_en but this 1 is not being picked up. /usr/local/freeswitch/conf/lang/en/vect/sounds.xml Can anyone please let me know what went wrong in my other server and why this function is not being picked up. Regards, Archana -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130228/35cc2afa/attachment.html From vittorio.guglielmo at gmail.com Thu Feb 28 21:03:19 2013 From: vittorio.guglielmo at gmail.com (Vittorio Guglielmo) Date: Thu, 28 Feb 2013 19:03:19 +0100 Subject: [Freeswitch-users] FS with SSL/TLS issues! In-Reply-To: References: <1361367187947-7587736.post@n2.nabble.com> <1361513236022-7587844.post@n2.nabble.com> <1361519970790-7587849.post@n2.nabble.com> <113D484C-EA7E-4611-B958-6514F5F2B5B9@freeswitch.org> <1361875157474-7588012.post@n2.nabble.com> <1361976425293-7588069.post@n2.nabble.com> <512E7B89.6030903@gmail.com> <512E8516.1060008@gmail.com> <512E8F08.40204@gmail.com> <512F4F54.8020903@gmail.com> <512F8DCE.8020804@gmail.com> Message-ID: <512F9BE7.6050707@gmail.com> Thank you so much Anthony, that solved the problem !!! Vic On 28/02/13 18:29, Anthony Minessale wrote: > I mean, the phone may expect SRTP to be offered over the TLS socket. > You were offering regular RTP profile "AVP" with no crypto in it and > they rejected it as unsupported. > That makes me guess they require you to enable srtp "SAVP" + crypto > tags or they will drop the call. > > try adding {sip_secure_media=true} in the outbound call to the phone. > > > > On Thu, Feb 28, 2013 at 11:03 AM, Vittorio Guglielmo > > > wrote: > > Can you please explain to me what does it mean ? > > And, in case, how can I demonstrate it, cause I need to buy a lot > of clients to build a secured voip network. > > Thank you very much for your time and your effort on a so huge > piece of software ( I bought all of your books ). > > > Vic > > > > > On 28/02/13 17:42, Anthony Minessale wrote: >> They probably do not accept calls over TLS that are not SAVP >> >> >> On Thu, Feb 28, 2013 at 6:36 AM, Vittorio Guglielmo >> > > wrote: >> >> Anthony, >> >> sofia global siptrace on >> >> and in attachment are the two siptrace: >> >> 1. *101* ( Bria iOS 2.3.4 on Iphone 3GS) *===> 102* (Bria >> 3 release 3.5.0b on Mac OSX 10.7.5) >> 2. *102* *===> 101* >> >> Both of them have forced only ulaw (first) and alaw (second), >> and without TSL/SRTP they call each other without problems. >> >> >> The behavior is different in the two above cases: >> >> When I call from 101 to 102 I get a *Originate Resulted in >> Error Cause: 79 [SERVICE_NOT_IMPLEMENTED]* >> due to, I believe a: *SIP/2.0 415 Unsupported Media Type* >> >> When I instead call from 102 to 101 I get a *Originate >> Failed. Cause: INCOMPATIBLE_DESTINATION* >> with a : *SIP/2.0 488 Not Acceptable Here* >> >> >> Vic >> >> On 28/02/13 00:23, Anthony Minessale wrote: >>> sofia global siptrace on >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> >> googletalk:conf+888 at conference.freeswitch.org >> >> pstn:+19193869900 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > googletalk:conf+888 at conference.freeswitch.org > > pstn:+19193869900 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130228/46046474/attachment-0001.html From schoch+freeswitch.org at xwin32.com Thu Feb 28 21:05:18 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Thu, 28 Feb 2013 10:05:18 -0800 Subject: [Freeswitch-users] Problem setting up bind_digit_action In-Reply-To: <509746653.18043.1362039617472.JavaMail.root@mailserver.edistar.com> References: <509746653.18043.1362039617472.JavaMail.root@mailserver.edistar.com> Message-ID: On Thu, Feb 28, 2013 at 12:20 AM, Denis Gasparin wrote: > > The extension MYSETUP is: > > > > > I'm rather new at this, so I could be wrong, but I believe an action must be contained inside a condition. So extension from above should be: -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130228/56e8d117/attachment.html From brian at freeswitch.org Thu Feb 28 21:09:23 2013 From: brian at freeswitch.org (Brian West) Date: Thu, 28 Feb 2013 12:09:23 -0600 Subject: [Freeswitch-users] Changing codec during calls In-Reply-To: References: <6EF8B8E9-4CC5-480C-B1C1-09CBC533470C@kavun.ch> <35AF9E79-771D-440A-BEBB-A856D831161C@kavun.ch> Message-ID: tsk tsk, gotta slow down and read. :) -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire T: +1.918.420.9266 | F: +1.918.420.9267 | M: +1.918.424.WEST iNUM: +883 5100 1420 9266 ISN: 410*543 On Feb 20, 2013, at 3:01 PM, Dmitry Lysenko wrote: > > PB-20616 > > > 2013/2/20 Michael Collins > Haha, you don't need an account. Re-read the challenge dialog on pastebin.freeswitch.org. > > -MC From schoch+freeswitch.org at xwin32.com Thu Feb 28 21:43:42 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Thu, 28 Feb 2013 10:43:42 -0800 Subject: [Freeswitch-users] Freeswitch Installation In-Reply-To: <512B9962.9030609@puzzled.xs4all.nl> References: <512B9962.9030609@puzzled.xs4all.nl> Message-ID: On Mon, Feb 25, 2013 at 9:03 AM, Patrick Lists < freeswitch-list at puzzled.xs4all.nl> wrote: > On 02/25/2013 03:45 PM, Lloyd Aloysius wrote: > > > SELINUX=disabled > > Disabling is a bad idea. > I just checked my selinux/config file. I don't remember setting it this way, but it's set to: SELINUX=permissive SELINUXTYPE=targeted The comments say that this makes it only protect targeted daemons, and that it just prints warnings instead of actually protecting, so I guess it's almost like having it disabled. I looked through the audit logs and didn't find anything, bad, so I'm going to try to turn it on and see what happens. -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130228/8516bcfb/attachment.html From msc at freeswitch.org Thu Feb 28 21:52:31 2013 From: msc at freeswitch.org (Michael Collins) Date: Thu, 28 Feb 2013 10:52:31 -0800 Subject: [Freeswitch-users] Friday Free For All Topic: GSoC 2013 Message-ID: Hey all! We thought that maybe a good idea for a discussion topic for tomorrow's Friday Free For All (F3A) would be potential projects that FreeSWITCH/OSTAG could sponsor as a mentoring organization. Please bring your fresh ideas and we'll talk about them! Thanks! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130228/163148ec/attachment.html From venkateshwaran54 at gmail.com Thu Feb 28 21:53:28 2013 From: venkateshwaran54 at gmail.com (Venkateshwaran Thirugnanam) Date: Fri, 1 Mar 2013 00:23:28 +0530 Subject: [Freeswitch-users] codec Message-ID: Hi All, What the difference between Transcodable codecs and Pass-Through Codec? Pls explain Thanks, Kumaran T -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130301/618c89f0/attachment.html