- FreeSWITCH-users - FreeSWITCH Mailing Lists

LUA error won't start
by Charles Bujold 14 Nov '13

14 Nov '13

When I start Freeswitch I get the following error, How can I fix? Thanks cjb 2013-11-14 17:15:43.193926 [CRIT] switch_loadable_module.c:1446 Error Loading module /usr/local/freeswitch/mod/mod_lua.so **/usr/local/freeswitch/mod/mod_lua.so: undefined symbol: lua_callk**

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RTP problem after attended trasfer
by Miha 14 Nov '13

14 Nov '13

Hi, When call is being trasfered in MS lync I noticed very bad media quality. I have trace this with whireshak, so I could here which side is causing this. I noticed that MS lync is sending ok RTP but when auto is send from FS to UAC, RTP is broken. For now I have used proxy_media. I can send you whole sip trace with rtp. Please let me know how can I fix this. p.s.: sip signaling is ok. br, miha

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Re: [Freeswitch-users] Audio Quality of recordings
by Scott Ross 14 Nov '13

14 Nov '13

Ok, I'll collect the pcaps and try the milliwatt extension today then post results along with recordings and instructions into JIRA. Thanks ---------- Forwarded message ---------- > From: Brian West <brian(a)freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Tue, 12 Nov 2013 20:59:27 -0600 > Subject: Re: [Freeswitch-users] Audio Quality of recordings > Can you send me pcaps and recordings of this to my personal address please. > > -- > Brian West > brian(a)freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH_Wire , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > > > On Nov 12, 2013, at 8:37 PM, Scott Ross <slross16(a)gmail.com> wrote: > > > > > I wasn't advised to use ubuntu, just including it as another data point > (its my dev machine). > > > > I've tested against two physical servers. > > RedHat 6, Xeon 6 core 2.6ghz, 32g ram > > The dev machine running Ubuntu 13.04, decent Core i7, 12g ram > > I've tested on several XenServer instances running RedHat 6. > > All devises/hosts are on the same internal network. > > > > Seeing the same results regardless. > > > > The device on the other end of the calls is an external conference > bridge (physical host). There could be some device specific issues - but I > see the same results when testing against another freeswitch instance on > the same network set up with mod_conference. > > Using g711 and recording to .wav. > > The call sounds unsatisfactory in real time and in the recordings > > Call quality is near perfect when the two calls aren't bridged into a > freeswtich conference or our external conference bridge. But even setting > up a basic freeswitch conference to run the tests against resulted in the > lower quality. > > > > So "failure" case appears to be when two separately originated calls > being bridged into any single conferenced environment. > > > > Just to be clear - the quality is ok, but not great. For reference, I'm > expecting a PESQ score of 3.8 or greater - which is met with asterisk > (seeing 3.8 - 4.3), but Freeswitch in its current setup consistently > results in 0.5 - 1 point lower at around 2.9 - 3.7. > > > > I'm hoping there is some environment/configuration tweaking I can do to > fix this. I have tried running FS with the -hp flag. Do I need to make > configuration changes on my OS/network to account for that? > > > > Does this feel like an OS level issue? Do I need to try Debian 6? > > > > Thank you for the reply. Again, let me know what other info I can > provide. > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting(a)freeswitch.org > > http://www.freeswitchsolutions.com > > > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > > http://www.cudatel.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users(a)lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > ---------- Forwarded message ---------- > From: Anthony Minessale <anthony.minessale(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Tue, 12 Nov 2013 21:03:02 -0600 > Subject: Re: [Freeswitch-users] Audio Quality of recordings > In fact, > > Can you give us all the info to reproduce your problem so we can see if we > can create the same situation in our lab. > file a JIRA http://jira.freeswitch.org and attach the stuff that brian > has asked for and preferably the instructions to reproduce the calls (We > can do originate from cli so we don't need the event socket stuff) and the > PSEQ app you are using. > > If we can see what you mean on another sever maybe we can prove its > environmental or a persistent issue. > > > On Tue, Nov 12, 2013 at 8:37 PM, Scott Ross <slross16(a)gmail.com> wrote: > >> >> I wasn't advised to use ubuntu, just including it as another data point >> (its my dev machine). >> >> I've tested against two physical servers. >> RedHat 6, Xeon 6 core 2.6ghz, 32g ram >> The dev machine running Ubuntu 13.04, decent Core i7, 12g ram >> I've tested on several XenServer instances running RedHat 6. >> All devises/hosts are on the same internal network. >> >> Seeing the same results regardless. >> >> The device on the other end of the calls is an external conference bridge >> (physical host). There could be some device specific issues - but I see the >> same results when testing against another freeswitch instance on the same >> network set up with mod_conference. >> Using g711 and recording to .wav. >> The call sounds unsatisfactory in real time and in the recordings >> Call quality is near perfect when the two calls aren't bridged into a >> freeswtich conference or our external conference bridge. But even setting >> up a basic freeswitch conference to run the tests against resulted in the >> lower quality. >> >> So "failure" case appears to be when two separately originated calls >> being bridged into any single conferenced environment. >> >> Just to be clear - the quality is ok, but not great. For reference, I'm >> expecting a PESQ score of 3.8 or greater - which is met with asterisk >> (seeing 3.8 - 4.3), but Freeswitch in its current setup consistently >> results in 0.5 - 1 point lower at around 2.9 - 3.7. >> >> I'm hoping there is some environment/configuration tweaking I can do to >> fix this. I have tried running FS with the -hp flag. Do I need to make >> configuration changes on my OS/network to account for that? >> >> Does this feel like an OS level issue? Do I need to try Debian 6? >> >> Thank you for the reply. Again, let me know what other info I can >> provide. >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale@hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888@conference.freeswitch.org > googletalk:conf+888@conference.freeswitch.org > pstn:+19193869900 > > > ---------- Forwarded message ---------- > From: Anthony Minessale <anthony.minessale(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Tue, 12 Nov 2013 21:06:18 -0600 > Subject: Re: [Freeswitch-users] Audio Quality of recordings > p.s. > > You didn't respond to the part about milliwatt.... doing that gives you a > visual feedback on the recording as well. > > > > On Tue, Nov 12, 2013 at 9:03 PM, Anthony Minessale < > anthony.minessale(a)gmail.com> wrote: > >> In fact, >> >> Can you give us all the info to reproduce your problem so we can see if >> we can create the same situation in our lab. >> file a JIRA http://jira.freeswitch.org and attach the stuff that brian >> has asked for and preferably the instructions to reproduce the calls (We >> can do originate from cli so we don't need the event socket stuff) and the >> PSEQ app you are using. >> >> If we can see what you mean on another sever maybe we can prove its >> environmental or a persistent issue. >> >> >> On Tue, Nov 12, 2013 at 8:37 PM, Scott Ross <slross16(a)gmail.com> wrote: >> >>> >>> I wasn't advised to use ubuntu, just including it as another data point >>> (its my dev machine). >>> >>> I've tested against two physical servers. >>> RedHat 6, Xeon 6 core 2.6ghz, 32g ram >>> The dev machine running Ubuntu 13.04, decent Core i7, 12g ram >>> I've tested on several XenServer instances running RedHat 6. >>> All devises/hosts are on the same internal network. >>> >>> Seeing the same results regardless. >>> >>> The device on the other end of the calls is an external conference >>> bridge (physical host). There could be some device specific issues - but I >>> see the same results when testing against another freeswitch instance on >>> the same network set up with mod_conference. >>> Using g711 and recording to .wav. >>> The call sounds unsatisfactory in real time and in the recordings >>> Call quality is near perfect when the two calls aren't bridged into a >>> freeswtich conference or our external conference bridge. But even setting >>> up a basic freeswitch conference to run the tests against resulted in the >>> lower quality. >>> >>> So "failure" case appears to be when two separately originated calls >>> being bridged into any single conferenced environment. >>> >>> Just to be clear - the quality is ok, but not great. For reference, I'm >>> expecting a PESQ score of 3.8 or greater - which is met with asterisk >>> (seeing 3.8 - 4.3), but Freeswitch in its current setup consistently >>> results in 0.5 - 1 point lower at around 2.9 - 3.7. >>> >>> I'm hoping there is some environment/configuration tweaking I can do to >>> fix this. I have tried running FS with the -hp flag. Do I need to make >>> configuration changes on my OS/network to account for that? >>> >>> Does this feel like an OS level issue? Do I need to try Debian 6? >>> >>> Thank you for the reply. Again, let me know what other info I can >>> provide. >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting(a)freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>> http://www.cudatel.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users(a)lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale@hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888@conference.freeswitch.org >> googletalk:conf+888@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@hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888@conference.freeswitch.org > googletalk:conf+888@conference.freeswitch.org > pstn:+19193869900 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >

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(no subject)
by bill vaxevanopoulos 14 Nov '13

14 Nov '13

hello i am using esl function for event socket in freeswitch .unfortunately i can not make dl function to work with php 5.3.3the option for php 5.2 is not an option.. and using Extension Loading Directives instead. is not very clear for my. please kindly tell me how to check if the neccessary librady is installed for the esl file to work .and i am receiving the following error.although i am not sure about the esl.so files how can i check i f they are exists? PHP Fatal error: Call to undefined function dl() in .... also is there any possibility to use another version of esl.php file instead that is not dependant on dl ?? Thank you !Best regards.

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shared line music on hold problem
by Simone 14 Nov '13

14 Nov '13

Hi all, recently I have installed the latest version of Freeswitch (1.2.14) and I have found the following Issue. If I call an extension connected to only a phone and transfer the call I can hear the music on hold. Instead when I call a Shared line (extension shared between two phones) and I try to transfer the call I can't hear the music on hold. I think is something related to the configuration of the shared line because with a normal line works good. Does someone have an Idea about what is going wrong? I attach both logs of when I try to call a shared line and when i try to call a normal line. Best Regards Simone

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Issue with audio file playback in early media
by Samir Doshi 14 Nov '13

14 Nov '13

Hi Guys, I need to playback audio file in early media and for that I have created below dialplan. But when I am calling number, its not playing audio and call seems getting stuck at playback like. And after some time fs is giving media_timeout error. <extension name="12127773456"> <condition field="destination_number" expression="12127773456"> <action application="set" data="effective_destination_number=12127773456"/> <action application="playback" data="/usr/local/freeswitch/sounds/en/us/callie/audio.wav"/> <action application="say" data="en CURRENCY PRONOUNCED 1 "/> <action application="bridge" data="sofia/gateway/Yourgateway/12127773456"/> </condition> </extension> Can anybody please help me to get this working? Thanks in advance. ----- Thanks, Samir -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Issue-with-audio-file-playbac… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Update callee id in pickup
by Andrew Cassidy 14 Nov '13

14 Nov '13

Hi guys, I've just posted this on Jira (http://jira.freeswitch.org/browse/FS-5967) but I might get more discussion here: I want to update the calling phone's display with the details of the user that picked up the call when using directed pickup via the pickup endpoint and application. What I've done in mod_dptools.c is: - check a new variable pickup_update_aleg - read effective caller id from channel - update effective callee id on pickup channel I've also added 2 log statement that show the variables do have the correct values, and that the test for the variable worked, but the original caller's display is not updated. Thanks, -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T <info(a)cassidywebservices.co.uk> *03300 100 960 *F <info(a)cassidywebservices.co.uk> *03300 100 961 *E <info(a)cassidywebservices.co.uk> *andrew(a)cassidywebservices.co.uk *W <info(a)cassidywebservices.co.uk> *www.cassidywebservices.co.uk

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Re: [Freeswitch-users] How to add PCMA/H263 to outgoing INVITE SDP
by Eugene Prokopiev 14 Nov '13

14 Nov '13

> How to add PCMA/H263 to FreeSWITCH INVITE? verbose_sdp=true helps -- WBR, Eugene Prokopiev

1 0

FS limit
by Juan Antonio Ibañez Santorum 13 Nov '13

13 Nov '13

Hello guys, I am doing some test in my FS box and I found it does not accept more than 300 calls. It is a 12 core E5-1650 Xeon server and I can see in 'top' that FS cpu goes to 140% (over 1200%). Once 300 calls is reached, FS starts responding 480 Temporary Unavailable: http://i.tinyuploads.com/to7Amt.png Do you know where the bottleneck is? I can see low disk activity, low memory consuption... Regards

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lua 5.2
by François 13 Nov '13

13 Nov '13

Thanks Anthony for THE lua update we've been waiting for! Will it be included in 1.4-stable? François.

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