From shlomi.agiv at cell-buddy.com Mon Dec 1 12:34:43 2014 From: shlomi.agiv at cell-buddy.com (Shlomi Agiv) Date: Mon, 1 Dec 2014 11:34:43 +0200 Subject: [Freeswitch-users] USB GSM Dongle with FreeSwitch for voice calls In-Reply-To: References: Message-ID: Hi, I will be leaving my current job soon, therefore will leave the freeswitch project. I put all my changes in https://pastebin.freeswitch.org/23657. It's done vs. version 1.4.4. The serial reconnect works pretty well. It also contains a libctb change to immediately detect errors using select, that also works well. I put my other email in the CC since I don't know for how long I will keep the current email, for any issues you can contact me there. Thanks and best regards, Shlomi Agiv On Tue, Sep 30, 2014 at 1:48 PM, Daniel Ivanov wrote: > Yes, it's in the protected zone message : pastebin / freeswitch > > On Tue, Sep 30, 2014 at 1:40 PM, rumodigital < > webmaster.rumodigital at gmail.com> wrote: > >> hello Shlomi >> >> is protected with password? >> >> 2014-09-30 12:03 GMT+01:00 Shlomi Agiv : >> >>> http://pastebin.freeswitch.org/23339 >>> >>> The code is based on gsmopen version 1.4.7, if I recall correctly. >>> >>> On Mon, Sep 29, 2014 at 9:33 PM, Du?an Dragi? >>> wrote: >>> >>>> If you've been working with git follow instructions in >>>> docs/SubmittingPatches (in fs repo) and >>>> https://confluence.freeswitch.org/display/FREESWITCH/Pull+Requests. >>>> Just follow the steps, push your branch to your stash repo and skip the >>>> last part (creating the pull request). Afterwards anyone can take a look at >>>> your changes (your stash repo is public). >>>> Or if it's easier just use "git diff" to create a patch and put it >>>> somewhere. >>>> >>>> >>>> >>>> >>>> >>>> On 29 September 2014 14:45, Shlomi Agiv >>>> wrote: >>>> >>>>> >>>>> Du?an, >>>>> The code I made works pretty well, though I can't say it works >>>>> perfectly. I just recently quashed a nasty bug there >>>>> I can post it , no problems. >>>>> Haven't posted any changes yet, what's the easiest way to do it? >>>>> >>>>> On Mon, Sep 29, 2014 at 3:16 PM, Giovanni Maruzzelli < >>>>> gmaruzz at gmail.com> wrote: >>>>> >>>>>> When you get more info, open a new, specific thread. >>>>>> >>>>>> I'm sure many persons will find it of interest. >>>>>> >>>>>> -giovanni >>>>>> >>>>>> Sent from Mobile >>>>>> >>>>>> Giovanni Maruzzelli >>>>>> +39 347 266 56 18 >>>>>> On Sep 29, 2014 2:14 PM, "Daniel Ivanov" wrote: >>>>>> >>>>>>> I would rather not hijack this thread with rpi specific issues. I >>>>>>> will investigate further as from tomorrow, truth is panics started >>>>>>> happening since i updated to the latest kern & picode. >>>>>>> >>>>>>> On Mon, Sep 29, 2014 at 2:02 PM, Giovanni Maruzzelli < >>>>>>> gmaruzz at gmail.com> wrote: >>>>>>> >>>>>>>> Kernel panics have probably to do with drivers, udev and kernel >>>>>>>> versions... >>>>>>>> >>>>>>>> What distro, kernel, drivers, udev, etc uses when successful and >>>>>>>> when panics? >>>>>>>> >>>>>>>> Please you all, list them here... >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> Sent from Mobile >>>>>>>> >>>>>>>> Giovanni Maruzzelli >>>>>>>> +39 347 266 56 18 >>>>>>>> On Sep 29, 2014 2:00 PM, "Daniel Ivanov" wrote: >>>>>>>> >>>>>>>>> I also use a hub with external power supply, but that hardly cures >>>>>>>>> the problem. On a normal x64 system, i seldom get usb errors, but the >>>>>>>>> kernel doesn't panic. On the Pi i get the same bus or mem errors and it >>>>>>>>> panics to need restart. >>>>>>>>> >>>>>>>>> On Mon, Sep 29, 2014 at 12:17 PM, Giovanni Maruzzelli < >>>>>>>>> gmaruzz at gmail.com> wrote: >>>>>>>>> >>>>>>>>>> nice!!! >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> Sincerely, >>>>>>>>>> >>>>>>>>>> Giovanni Maruzzelli >>>>>>>>>> Cell : +39-347-2665618 >>>>>>>>>> >>>>>>>>>> On Mon, Sep 29, 2014 at 12:14 PM, rumodigital < >>>>>>>>>> webmaster.rumodigital at gmail.com> wrote: >>>>>>>>>> >>>>>>>>>>> I decided not to add more than five to avoid heat and maintain >>>>>>>>>>> stability, so working with a margin of safety devices and distance to allow >>>>>>>>>>> for ventilation. >>>>>>>>>>> >>>>>>>>>>> Not heats and works very well. >>>>>>>>>>> >>>>>>>>>>> 2014-09-29 11:01 GMT+01:00 Giovanni Maruzzelli < >>>>>>>>>>> gmaruzz at gmail.com>: >>>>>>>>>>> >>>>>>>>>>>> seems a good one, with power enough to add another additional >>>>>>>>>>>> dongle, for a total of 5 concurrent. >>>>>>>>>>>> >>>>>>>>>>>> Do it get too much warm? >>>>>>>>>>>> >>>>>>>>>>>> -giovanni >>>>>>>>>>>> >>>>>>>>>>>> Sincerely, >>>>>>>>>>>> >>>>>>>>>>>> Giovanni Maruzzelli >>>>>>>>>>>> Cell : +39-347-2665618 >>>>>>>>>>>> >>>>>>>>>>>> On Mon, Sep 29, 2014 at 11:57 AM, rumodigital < >>>>>>>>>>>> webmaster.rumodigital at gmail.com> wrote: >>>>>>>>>>>> >>>>>>>>>>>>> Hi Giovanni, >>>>>>>>>>>>> >>>>>>>>>>>>> I'm in the testing phase in a Mini ITX board (Atom D2550 + >>>>>>>>>>>>> debian 7.6 32b) and DUB-H7 USB HUB?. I use 4 devices simultaneously (4x >>>>>>>>>>>>> 500ma = 2A) and yet everything perfect! >>>>>>>>>>>>> >>>>>>>>>>>>> 7-Port USB 2.0 Hub (500ma on all seven front ports, but 3A in >>>>>>>>>>>>> total) >>>>>>>>>>>>> >>>>>>>>>>>>> http://www.dlink.com/uk/en/home-solutions/connect/usb/dub-h7-7-port-usb-2-0-hub >>>>>>>>>>>>> >>>>>>>>>>>>> Do you recommend some other HUB? >>>>>>>>>>>>> >>>>>>>>>>>>> Doug >>>>>>>>>>>>> >>>>>>>>>>>>> 2014-09-29 10:23 GMT+01:00 Giovanni Maruzzelli < >>>>>>>>>>>>> gmaruzz at gmail.com>: >>>>>>>>>>>>> >>>>>>>>>>>>>> all this kind of problems come from lack of USB bus >>>>>>>>>>>>>> electrical power. >>>>>>>>>>>>>> >>>>>>>>>>>>>> Use a wall powered (eg: with a "brick" power supply) USB hub >>>>>>>>>>>>>> between FreeSWITCH and dongle. >>>>>>>>>>>>>> >>>>>>>>>>>>>> No software way to go around this. >>>>>>>>>>>>>> >>>>>>>>>>>>>> And be sure the hub deliver enough power! (check the specs >>>>>>>>>>>>>> for your dongles power needs, add them up and use an USB hub with more than >>>>>>>>>>>>>> that continous power delivery). >>>>>>>>>>>>>> >>>>>>>>>>>>>> Internet dongles are power hungry ! >>>>>>>>>>>>>> >>>>>>>>>>>>>> -giovanni >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> Sincerely, >>>>>>>>>>>>>> >>>>>>>>>>>>>> Giovanni Maruzzelli >>>>>>>>>>>>>> Cell : +39-347-2665618 >>>>>>>>>>>>>> >>>>>>>>>>>>>> On Mon, Sep 29, 2014 at 11:12 AM, rumodigital < >>>>>>>>>>>>>> webmaster.rumodigital at gmail.com> wrote: >>>>>>>>>>>>>> >>>>>>>>>>>>>>> Hi, >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> Really, can be a major headache, many problems can be solved >>>>>>>>>>>>>>> with the use of a self powered USB hub... yet I had no problems, but for >>>>>>>>>>>>>>> example, I tested with an old pc I have, and when he >>>>>>>>>>>>>>> received a call the module disconnected and blocked the USB port. >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> Which versions of Pi and what model of hardware that use? >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> Doug >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> _________________________________________________________________________ >>>>>>>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>>>>>>> consulting at freeswitch.org >>>>>>>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>>>>> http://confluence.freeswitch.org >>>>>>>>>>>>>>> http://www.cluecon.com >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>>>>>>>>>>> http://www.cudatel.com >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>>>>>>> UNSUBSCRIBE: >>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> _________________________________________________________________________ >>>>>>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>>>>>> consulting at freeswitch.org >>>>>>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>>>>>> >>>>>>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>>>> http://confluence.freeswitch.org >>>>>>>>>>>>>> http://www.cluecon.com >>>>>>>>>>>>>> >>>>>>>>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>>>>>>>>>> http://www.cudatel.com >>>>>>>>>>>>>> >>>>>>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>>>>>> UNSUBSCRIBE: >>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> _________________________________________________________________________ >>>>>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>>>>> consulting at freeswitch.org >>>>>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>>>>> >>>>>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>>> http://confluence.freeswitch.org >>>>>>>>>>>>> http://www.cluecon.com >>>>>>>>>>>>> >>>>>>>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>>>>>>>>> http://www.cudatel.com >>>>>>>>>>>>> >>>>>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>>>>> UNSUBSCRIBE: >>>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> _________________________________________________________________________ >>>>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>>>> consulting at freeswitch.org >>>>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>>>> >>>>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>> http://confluence.freeswitch.org >>>>>>>>>>>> http://www.cluecon.com >>>>>>>>>>>> >>>>>>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>>>>>>>> http://www.cudatel.com >>>>>>>>>>>> >>>>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>>>> UNSUBSCRIBE: >>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> _________________________________________________________________________ >>>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>>> consulting at freeswitch.org >>>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>>> >>>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>> http://confluence.freeswitch.org >>>>>>>>>>> http://www.cluecon.com >>>>>>>>>>> >>>>>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>>>>>>> http://www.cudatel.com >>>>>>>>>>> >>>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>>> UNSUBSCRIBE: >>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> _________________________________________________________________________ >>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>> consulting at freeswitch.org >>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>> >>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>> http://www.freeswitch.org >>>>>>>>>> http://confluence.freeswitch.org >>>>>>>>>> http://www.cluecon.com >>>>>>>>>> >>>>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>>>>>> http://www.cudatel.com >>>>>>>>>> >>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>> UNSUBSCRIBE: >>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>>> http://www.freeswitch.org >>>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> _________________________________________________________________________ >>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>> consulting at freeswitch.org >>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>> >>>>>>>>> Official FreeSWITCH Sites >>>>>>>>> http://www.freeswitch.org >>>>>>>>> http://confluence.freeswitch.org >>>>>>>>> http://www.cluecon.com >>>>>>>>> >>>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>>>>> http://www.cudatel.com >>>>>>>>> >>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>> UNSUBSCRIBE: >>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>> http://www.freeswitch.org >>>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> _________________________________________________________________________ >>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>> consulting at freeswitch.org >>>>>>>> http://www.freeswitchsolutions.com >>>>>>>> >>>>>>>> Official FreeSWITCH Sites >>>>>>>> http://www.freeswitch.org >>>>>>>> http://confluence.freeswitch.org >>>>>>>> http://www.cluecon.com >>>>>>>> >>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>>>> http://www.cudatel.com >>>>>>>> >>>>>>>> FreeSWITCH-users mailing list >>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>> UNSUBSCRIBE: >>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>> http://www.freeswitch.org >>>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://confluence.freeswitch.org >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>>> http://www.cudatel.com >>>>>>> >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://confluence.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>> http://www.cudatel.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>> http://www.cudatel.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> >>>> -- >>>> Du?an Dragi? >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>> http://www.cudatel.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>> http://www.cudatel.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141201/c6e530a4/attachment-0001.html From vipkilla at gmail.com Mon Dec 1 16:04:26 2014 From: vipkilla at gmail.com (Vik Killa) Date: Mon, 1 Dec 2014 08:04:26 -0500 Subject: [Freeswitch-users] multi tenet user directory management best practice In-Reply-To: References: Message-ID: Use XML_CURL to push same configs dynamically to all FS servers. On Sun, Nov 30, 2014 at 6:25 AM, ik wrote: > Hello, > > This question raises a lot, but no simple answer, so I'm trying to ask it > differently. > > Let's say I have 5 FS servers and Opensips/Kamailio as a load balancer. > > > > I'm afraid that creating it using xml files is not dynamic enough, and > requires duplications. > mod_xml_rpc does not store that information on each update of the server > itself, and there is no API for providing database management that I could > find. > > So what is the best practice regarding of creating a user directory that > all of the PBX will be able to handle ? > > Thanks > Ido > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141201/c718dd73/attachment.html From brian at freeswitch.org Mon Dec 1 17:17:58 2014 From: brian at freeswitch.org (Brian West) Date: Mon, 1 Dec 2014 08:17:58 -0600 Subject: [Freeswitch-users] echo cancellation and mod_conference In-Reply-To: <21728.1417347781@ccs.covici.com> References: <21728.1417347781@ccs.covici.com> Message-ID: Acoustical Echo Cancel would be rather resource intensive, Why not instruct your users to acquire proper endpoints? :) Most Optimal... probably least favorable. I for one advocate for endpoints that handle their own nat traversal, echo cancel leaving the registrar/freeswitch no avoid that sort of heavy lifting. On Sun, Nov 30, 2014 at 5:43 AM, wrote: > Hi. I have some sip callers who call in on what are apparently bad > lines and get echo when calling into conferences -- not freeswitch ones > yet. So, I am wondering in mod_conference is there any attempt to do > echo cancellation, if so I will try to have them call into an fs one > instead. Its intermittent, so I wanted to find out first before I test. > > Thanks in advance for any suggestions. > > > -- > 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://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141201/bca01149/attachment.html From brian at freeswitch.org Mon Dec 1 17:19:30 2014 From: brian at freeswitch.org (Brian West) Date: Mon, 1 Dec 2014 08:19:30 -0600 Subject: [Freeswitch-users] FreeSWITCH TLS not able to receive calls In-Reply-To: <069E914C-296C-495C-8A49-54E1A74A866E@gmail.com> References: <069E914C-296C-495C-8A49-54E1A74A866E@gmail.com> Message-ID: What endpoints are involved? have you looked at 'sofia loglevel all 9' output and see if it gives you a clue? On Fri, Nov 28, 2014 at 6:24 PM, Ahmed Habiba wrote: > Dears, > > I?ve configured FreeSWITCH with the below version with TLS/SRTPas per the > recommendation in page ?https://wiki.freeswitch.org/wiki/SIP_TLS? and it > was strait forward, and I was able to connect and make make calls using > zoiper, but I was not able to receive any calls after enabling the TLS/SRTP. > > *"FreeSWITCH Version 1.4.13+git~20141103T195300Z~b942d0faa8~64bit (git > b942d0f 2014-11-03 19:53:00Z 64bit)**?* > > Your kind feedback will be appreciate. > > Thanks, > > Ahmed Habiba. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141201/3fe0b486/attachment.html From brian at freeswitch.org Mon Dec 1 17:25:53 2014 From: brian at freeswitch.org (Brian West) Date: Mon, 1 Dec 2014 08:25:53 -0600 Subject: [Freeswitch-users] mod_verto behind NAT In-Reply-To: References: Message-ID: Please contact consulting at freeswitch.org for a quote to build this into mod_verto, It would appear mod_verto has no concept of NAT like sofia does. On Wed, Nov 26, 2014 at 10:13 AM, Ciprian Dosoftei < ciprian.dosoftei at gmail.com> wrote: > Hi, > > I am dealing with a NAT situation related to mod_verto. The switch is > hosted in EC2 and the configuration works great with mod_sofia (both for > receiving calls from our SIP providers and via webRTC clients). > > However, the counterpart configuration for mod_verto fails to expose the > external IP address in the SDP. Here's the mod_verto configuration we're > using: > > http://pastebin.com/4PDMXvW6 > > I can confirm the external RTP IP is set properly, however here's the > returned SDP: > > http://pastebin.com/3vzyihRp > > The verto client is running behind NAT as well, but this doesn't seem to > be the root of the problem as Sofia's own implementation of webRTC works > fine. It is effectively sending its media to 10.178.x.x rather than the > switch's public IP address. > > Please advise, thank you. > > -- > Best Regards, > Ciprian Dosoftei > > The information transmitted is intended only for the addressee and may > contain privileged and/or confidential material. If you are not the > intended recipient, kindly contact the sender and delete the message. > > Any disclosure, distribution or copying of this message is strictly > prohibited without the expressed permission of the sender. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141201/c891ce17/attachment-0001.html From covici at ccs.covici.com Mon Dec 1 18:15:08 2014 From: covici at ccs.covici.com (covici at ccs.covici.com) Date: Mon, 01 Dec 2014 10:15:08 -0500 Subject: [Freeswitch-users] echo cancellation and mod_conference In-Reply-To: References: <21728.1417347781@ccs.covici.com> Message-ID: <19723.1417446908@ccs.covici.com> Its not the endpoint itself, but something in the pstn -- if they call in say a number of times, the echo will stop, it seems to be something in the Verizon equipment, this is why I was hoping for fs to do something. Thanks. Brian West wrote: > Acoustical Echo Cancel would be rather resource intensive, Why not instruct > your users to acquire proper endpoints? :) Most Optimal... probably least > favorable. I for one advocate for endpoints that handle their own nat > traversal, echo cancel leaving the registrar/freeswitch no avoid that sort > of heavy lifting. > > On Sun, Nov 30, 2014 at 5:43 AM, wrote: > > > Hi. I have some sip callers who call in on what are apparently bad > > lines and get echo when calling into conferences -- not freeswitch ones > > yet. So, I am wondering in mod_conference is there any attempt to do > > echo cancellation, if so I will try to have them call into an fs one > > instead. Its intermittent, so I wanted to find out first before I test. > > > > Thanks in advance for any suggestions. > > > > > > -- > > 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://confluence.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > > *Brian West* > brian at freeswitch.org > > > *Twitter: @FreeSWITCH , @briankwest* > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest > > ---------------------------------------------------- > Alternatives: > > ---------------------------------------------------- > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 brian at freeswitch.org Mon Dec 1 18:45:54 2014 From: brian at freeswitch.org (Brian West) Date: Mon, 1 Dec 2014 09:45:54 -0600 Subject: [Freeswitch-users] echo cancellation and mod_conference In-Reply-To: <19723.1417446908@ccs.covici.com> References: <21728.1417347781@ccs.covici.com> <19723.1417446908@ccs.covici.com> Message-ID: Sadly there probably doesn't exist a silver bullet to solve this problem... it will always be an uphill battle without a resolution... the real solution is to open tickets with upstream providers till its resolved, but I'm sure you could retire before they'd actually find/fix it unless complain daily for weeks on end. On Mon, Dec 1, 2014 at 9:15 AM, wrote: > Its not the endpoint itself, but something in the pstn -- if they call > in say a number of times, the echo will stop, it seems to be something > in the Verizon equipment, this is why I was hoping for fs to do > something. > > Thanks. > > Brian West wrote: > > > Acoustical Echo Cancel would be rather resource intensive, Why not > instruct > > your users to acquire proper endpoints? :) Most Optimal... probably > least > > favorable. I for one advocate for endpoints that handle their own nat > > traversal, echo cancel leaving the registrar/freeswitch no avoid that > sort > > of heavy lifting. > > > > On Sun, Nov 30, 2014 at 5:43 AM, wrote: > > > > > Hi. I have some sip callers who call in on what are apparently bad > > > lines and get echo when calling into conferences -- not freeswitch ones > > > yet. So, I am wondering in mod_conference is there any attempt to do > > > echo cancellation, if so I will try to have them call into an fs one > > > instead. Its intermittent, so I wanted to find out first before I > test. > > > > > > Thanks in advance for any suggestions. > > > > > > > > > -- > > > 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://confluence.freeswitch.org > > > http://www.cluecon.com > > > > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > > > > -- > > > > *Brian West* > > brian at freeswitch.org > > > > > > *Twitter: @FreeSWITCH , @briankwest* > > http://www.freeswitchbook.com > > http://www.freeswitchcookbook.com > > > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest > > > > ---------------------------------------------------- > > Alternatives: > > > > ---------------------------------------------------- > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://confluence.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141201/cf941e97/attachment.html From netcentrica at gmail.com Mon Dec 1 18:46:39 2014 From: netcentrica at gmail.com (Mateusz Bartczak) Date: Mon, 1 Dec 2014 16:46:39 +0100 Subject: [Freeswitch-users] Donate creation of mod_gearman module Message-ID: Hi I?m looking for experienced FreeSWITCH developer to build new module which will allow to send FS events using Gearman library. Something like existing mod_zmq module, but for other protocol. Additionally module should support of RPC calls from Gearman clients and invoke them as API (blocking) and BGAPI (non-blocking) commands inside FS. I have detailed specification of needed functionalities. Looking for someone ready to start promptly and make it fast. If someone from core FS dev team will respond, then I can declare to push back this module as open source available for all of the community Anybody interested please PM me: netcentrica at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141201/11622594/attachment.html From ssinyagin at gmail.com Mon Dec 1 18:55:03 2014 From: ssinyagin at gmail.com (Stanislav Sinyagin) Date: Mon, 1 Dec 2014 16:55:03 +0100 Subject: [Freeswitch-users] Donate creation of mod_gearman module In-Reply-To: References: Message-ID: why does it have to be a FreeSWITCH module? Why not making a separate daemon which talks to FreeSWITCH via ESL and communicates with Gearman? The benefit would be that you don't need anything new inside FreeSWITCH, so your development woudl be completely independent, and always compatible with future releases of FreeSWITCH. On Mon, Dec 1, 2014 at 4:46 PM, Mateusz Bartczak wrote: > Hi > > > I?m looking for experienced FreeSWITCH developer to build new module which > will allow to send FS events using Gearman library. Something like existing > mod_zmq module, but for other protocol. > > > Additionally module should support of RPC calls from Gearman clients and > invoke them as API (blocking) and BGAPI (non-blocking) commands inside FS. > > > I have detailed specification of needed functionalities. Looking for someone > ready to start promptly and make it fast. > > > If someone from core FS dev team will respond, then I can declare to push > back this module as open source available for all of the community > > > Anybody interested please PM me: netcentrica at gmail.com > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 Mon Dec 1 19:03:53 2014 From: brian at freeswitch.org (Brian West) Date: Mon, 1 Dec 2014 10:03:53 -0600 Subject: [Freeswitch-users] Donate creation of mod_gearman module In-Reply-To: References: Message-ID: well either way can work, depends on the functionality desired, can you outline or write up what exactly the use cases are? On Mon, Dec 1, 2014 at 9:55 AM, Stanislav Sinyagin wrote: > why does it have to be a FreeSWITCH module? Why not making a separate > daemon which talks to FreeSWITCH via ESL and communicates with > Gearman? > > The benefit would be that you don't need anything new inside > FreeSWITCH, so your development woudl be completely independent, and > always compatible with future releases of FreeSWITCH. > > > > > On Mon, Dec 1, 2014 at 4:46 PM, Mateusz Bartczak > wrote: > > Hi > > > > > > I?m looking for experienced FreeSWITCH developer to build new module > which > > will allow to send FS events using Gearman library. Something like > existing > > mod_zmq module, but for other protocol. > > > > > > Additionally module should support of RPC calls from Gearman clients and > > invoke them as API (blocking) and BGAPI (non-blocking) commands inside > FS. > > > > > > I have detailed specification of needed functionalities. Looking for > someone > > ready to start promptly and make it fast. > > > > > > If someone from core FS dev team will respond, then I can declare to push > > back this module as open source available for all of the community > > > > > > Anybody interested please PM me: netcentrica at gmail.com > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://confluence.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141201/b7ee61d5/attachment-0001.html From netcentrica at gmail.com Mon Dec 1 19:10:21 2014 From: netcentrica at gmail.com (Mateusz Bartczak) Date: Mon, 1 Dec 2014 17:10:21 +0100 Subject: [Freeswitch-users] Donate creation of mod_gearman module In-Reply-To: References: Message-ID: The main reason is performance. Some other reasons are specific to my company internal requirements and the architecture of project we are working on. I can't discuss on this public forum. As stated in the first message I'll be happy to pay for work needed to create this module 2014-12-01 16:55 GMT+01:00 Stanislav Sinyagin : > why does it have to be a FreeSWITCH module? Why not making a separate > daemon which talks to FreeSWITCH via ESL and communicates with > Gearman? > > The benefit would be that you don't need anything new inside > FreeSWITCH, so your development woudl be completely independent, and > always compatible with future releases of FreeSWITCH. > > > > > On Mon, Dec 1, 2014 at 4:46 PM, Mateusz Bartczak > wrote: > > Hi > > > > > > I?m looking for experienced FreeSWITCH developer to build new module > which > > will allow to send FS events using Gearman library. Something like > existing > > mod_zmq module, but for other protocol. > > > > > > Additionally module should support of RPC calls from Gearman clients and > > invoke them as API (blocking) and BGAPI (non-blocking) commands inside > FS. > > > > > > I have detailed specification of needed functionalities. Looking for > someone > > ready to start promptly and make it fast. > > > > > > If someone from core FS dev team will respond, then I can declare to push > > back this module as open source available for all of the community > > > > > > Anybody interested please PM me: netcentrica at gmail.com > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://confluence.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141201/47b05a56/attachment.html From mike at jerris.com Mon Dec 1 19:54:11 2014 From: mike at jerris.com (Michael Jerris) Date: Mon, 1 Dec 2014 11:54:11 -0500 Subject: [Freeswitch-users] Donate creation of mod_gearman module In-Reply-To: References: Message-ID: <155F1183-78B4-4A3F-A8FA-F880E8E2CBE7@jerris.com> we can help you at consulting at freeswitch.org. > On Dec 1, 2014, at 11:10 AM, Mateusz Bartczak wrote: > > The main reason is performance. Some other reasons are specific to my company internal requirements and the architecture of project we are working on. I can't discuss on this public forum. > As stated in the first message I'll be happy to pay for work needed to create this module > > 2014-12-01 16:55 GMT+01:00 Stanislav Sinyagin >: > why does it have to be a FreeSWITCH module? Why not making a separate > daemon which talks to FreeSWITCH via ESL and communicates with > Gearman? > > The benefit would be that you don't need anything new inside > FreeSWITCH, so your development woudl be completely independent, and > always compatible with future releases of FreeSWITCH. > > > > > On Mon, Dec 1, 2014 at 4:46 PM, Mateusz Bartczak > wrote: > > Hi > > > > > > I?m looking for experienced FreeSWITCH developer to build new module which > > will allow to send FS events using Gearman library. Something like existing > > mod_zmq module, but for other protocol. > > > > > > Additionally module should support of RPC calls from Gearman clients and > > invoke them as API (blocking) and BGAPI (non-blocking) commands inside FS. > > > > > > I have detailed specification of needed functionalities. Looking for someone > > ready to start promptly and make it fast. > > > > > > If someone from core FS dev team will respond, then I can declare to push > > back this module as open source available for all of the community > > > > > > Anybody interested please PM me: netcentrica at gmail.com > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141201/800b0b78/attachment.html From krice at freeswitch.org Mon Dec 1 21:53:40 2014 From: krice at freeswitch.org (Ken Rice) Date: Mon, 01 Dec 2014 18:53:40 +0000 Subject: [Freeswitch-users] =?utf-8?q?Verto=E2=84=A2_Not_just_for_call_sig?= =?utf-8?q?naling?= Message-ID: <547cb9344e5a0_d385faf3342313e@ip-10-156-165-183.mail> New Post on freeswitch.org from anthm check it out at http://ift.tt/1vL90Zr Verto? Not just for call signaling Verto is not only focused on call signaling; There are a few other key components that are important to understand. One key component of Verto is event channels. Event channels are bi-directional subscription-based logical data channels used to relay information in an organized fashion. This is a low level building block for many features. For instance, call or chat presence. Event channels can be subscribed to from JS and granular permissions may be set from FS on-the-fly or ahead of time. Modules inside FreeSWITCH fire json events and every connection who has a subscription will get a copy of the data allowing action to be taken. At the same time each browser connection is able to send a request back to the server on a particular event channel which can trigger action varying from sending more events, a full sync of saved data or even pushing the events into another event bus to send to another external application. Our demo site http://ift.tt/12g7Iff and call 3500 you can see an example of the data synchronization functionality as it will generate events that sync and update in real time a data structure in FreeSWITCH and one in the page of anyone subscribed. Because 3500 is a conference, the conference module will grant you access to the data about who else is called into the conference. It sends an event scoped to the exact uuid of the connection of the calling verto client. This event says that there is an available data synchronization for the particular conference. Using the data from this event, you can initialize an object that uses event channels to subscribe to the conference data and request a full sync and then keeps the object in sync with update events as long as you are on the call. This allows a table to be drawn in real time reflecting the changes going on in the conference. When you hangup, it sends a similar event telling you that you should unregister from this service and removes your permission. If you wanted, you could allow permissions to one or all conference data from your FS config and use verto to connect and provide info about who?s in the conference even when you are not connected directly. Another important feature of the module is the ability to communicate with FreeSWITCH and invent your own api modules. The lower level json-rpc in FS contains 2 important commands, fsapi and jsapi. fsapi allows you to execute any FreeSWITCH FSAPI command (basically all the commands you can type at the cli) This is set on a per command permission basis and is of course best used with security in mind. The other command, jsapi, is used to interface with a new type of module interface in FreeSWITCH called the JSON API Interface (JSONAPI) This interface is similar to FSAPI only its designed with a JSON in JSON out structure. One could write a C module defining a series of JSONAPI designed to allow your html5 application to communicate directly with your module in the same fashion that mod_verto does. This allows mod_verto to host the server logic and just pass commands up to your module and communicate in basic json exchanges. There some examples in FreeSWITCH now that you can test from your cli There is a FSAPI command called json that just assumes the input is a chunk of json and passes it right into JSONAPI then gets the results and converts it back to text so you can see it on the screen. Paste this command into your terminal: json {?command?: ?status?, ?format?: ?pretty?, ?data?: {}} Had it been over jsonrpc it would have been sent with method of jsapi and the json below would have been the params element: { ?jsonrpc?: ?2.0?, ?method?: ?jsapi?, ?params?:{?command?: ?status?, ?format?: ?pretty?, ?data?: {}} ?id?: 2 } -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141201/4489f324/attachment.html From k.presler at megafit.su Mon Dec 1 22:51:41 2014 From: k.presler at megafit.su (=?koi8-r?B?68nSyczMIPDSxdPMxdI=?=) Date: Mon, 1 Dec 2014 19:51:41 +0000 Subject: [Freeswitch-users] Freeswitch Python scripting Message-ID: <20905bdd9aa94c639fbcf2e901f0290c@AM3PR06MB433.eurprd06.prod.outlook.com> We want to use dialplan python script. Script working but call setup is lasting too long ( <10sec to >40sec when 60 concurent calls on switch) Script just route call from user to gateway. I've not a lot of experience in FS and programming, so I need your help. Ps I'm not good in English, excuse me. Code part sessionData = 'sofia/gateway/gwname/dest' new_session = Session(sessionData,session) if new_session.ready(): bridge(session,new_session) if new_session.ready(): new_session.hangup() hcause = new_session.hangupCause() My native language is -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141201/bbec86eb/attachment-0001.html From ahabiba at gmail.com Mon Dec 1 23:56:24 2014 From: ahabiba at gmail.com (Ahmed habiba) Date: Mon, 1 Dec 2014 23:56:24 +0300 Subject: [Freeswitch-users] FreeSWITCH TLS not able to receive calls Message-ID: I?m using "zoiper 1.19.7? on android, the issue appears only one I enable SRTP, for TLS only it is working very fine. The issue as shown below it keeps registering till finally zipper restart on mobile. One more point I?m configuring Zoiper without using RPORT for either SRTP or TLS. *?2014-12-01 20:57:24.003230 [NOTICE] switch_ivr_originate.c:3467 Hangup sofia/internal/sip:1000 at 222.248.102.244 <1000 at 222.248.102.244>:38614 [CS_CONSUME_MEDIA] [NO_ANSWER]"* recv 1290 bytes from tls/[222.215.195.234]:2896 at 20:56:54.344450: ------------------------------------------------------------------------ INVITE sip:1000 at abc-xyz.com SIP/2.0 v: SIP/2.0/TLS 192.168.8.109:41659 ;rport;branch=z9hG4bKPjvnvRZjpAXahzIV7S8SRgS5vwKmYI9WFS;alias Max-Forwards: 70 f: ;tag=weeMqzInPcNLPd6s5IX63c8be3DMvGkU t: m: i: sjprJQszC2LHvY6jRr-lm9rau9V1PlYX CSeq: 32737 INVITE Route: k: replaces, 100rel, timer, norefersub x: 1800 Min-SE: 90 User-Agent: CSipSimple_GT-S5830-10/r2450 Proxy-Authorization: Digest username="1006", realm="abc-xyz.com", nonce="329456ee-7994-11e4-bf79-1182ebca2429", uri="sip:1000 at abc-xyz.com", response="ab3a0f2fe9207ced62eebd8c9b8c32b4", algorithm=MD5, cnonce="iLa-C.wpnBSeTQjWiXZK3H3PGYhi6EGt", qop=auth, nc=00000001 c: application/sdp l: 469 v=0 o=- 3626452487 3626452487 IN IP4 192.168.8.109 s=pjmedia c=IN IP4 192.168.8.109 t=0 0 m=audio 4000 RTP/SAVP 98 0 8 101 c=IN IP4 192.168.8.109 a=rtcp:4001 IN IP4 192.168.8.109 a=sendrecv a=rtpmap:98 SILK/16000 a=fmtp:98 useinbandfec=0 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:Vci7cNdSFTkkLnqz+qRkqCctPvRT6jIOrNc5KMbz a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:nW3ECq5YHHnrY2q5u7XZE4fvPfMOAWKQ+ehOp7zV ------------------------------------------------------------------------ send 405 bytes to tls/[222.215.195.234]:2896 at 20:56:54.344687: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/TLS 192.168.8.109:41659 ;rport=2896;branch=z9hG4bKPjvnvRZjpAXahzIV7S8SRgS5vwKmYI9WFS;alias;received=222.215.195.234 f: ;tag=weeMqzInPcNLPd6s5IX63c8be3DMvGkU t: i: sjprJQszC2LHvY6jRr-lm9rau9V1PlYX CSeq: 32737 INVITE User-Agent: FreeSWITCH-mod_sofia/1.4.13+git~20141103T195300Z~b942d0faa8~64bit Content-Length: 0 send 2133 bytes to tls/[222.248.102.244]:38614 at 20:56:54.360608: ------------------------------------------------------------------------ INVITE sip:1000 at 222.248.102.244:38614;rinstance=98c51296f2d531cc;transport=TLS SIP/2.0 Via: SIP/2.0/TLS 176.31.245.17;branch=z9hG4bKjXZvg3cX76Krj Max-Forwards: 69 From: "Extension 1006" ;tag=15DcKN9tUZeaK To: < sip:1000 at 222.248.102.244:38614;rinstance=98c51296f2d531cc;transport=TLS> Call-ID: 0a3e07c5-f437-1232-9bba-d2ab2784dc6a CSeq: 68395843 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.4.13+git~20141103T195300Z~b942d0faa8~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 1076 X-FS-Support: update_display,send_info Remote-Party-ID: "Extension 1006" ;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1417445672 1417445673 IN IP4 176.31.245.17 s=FreeSWITCH c=IN IP4 176.31.245.17 t=0 0 m=audio 18142 RTP/SAVP 0 8 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=crypto:1 AEAD_AES_256_GCM_8 inline:6p/uXB7tN5SG9vHP/DMurij6BaZIOKcP4Tle0kjo2lwL1Oa4z9WGyZtk4zs a=crypto:2 AEAD_AES_128_GCM_8 inline:hGeJA6XQV1V35oTu2141o7mD64dFDRbeCbcf6g a=crypto:3 AES_CM_256_HMAC_SHA1_80 inline:XdZwwcl6kMRrWY7Vkrbozf/o1awka9lef0UjQRwvnOnnrbAgzXIm8s3yu7o8DQ a=crypto:4 AES_CM_192_HMAC_SHA1_80 inline:KtM92m0ac2musYPoXLDJb4rUbEMYAwMegPH9RMtzwzPb+0O93e0 a=crypto:5 AES_CM_128_HMAC_SHA1_80 inline:PtOpdIk2LTSCSDVzPaSN6AV2xosBrJUTd4mvKIGh a=crypto:6 AES_CM_256_HMAC_SHA1_32 inline:ZHzWxajRlQp8IuzX5CuFM1SRkL0huC62ukX583+Vg0LBiK9t2NrpP8FNwdkckQ a=crypto:7 AES_CM_192_HMAC_SHA1_32 inline:S9BWrDxGFYpU3P9o3HD0Z97W5jwo8vD+E1sJi821r4/B+cdukTw a=crypto:8 AES_CM_128_HMAC_SHA1_32 inline:l7C0wU29AjF2ZMnCTvks+1ytd/UiXHqN8UIsM6vf a=crypto:9 AES_CM_128_NULL_AUTH inline:tv97QsO7+S9NLTXrD02omyWoyM7I2KaWaimUgq99 a=ptime:20 ------------------------------------------------------------------------ REGISTER sip:abc-xyz.com;transport=TLS SIP/2.0 Via: SIP/2.0/TLS 192.168.10.2:41120 ;branch=z9hG4bK-d8754z-903949acb1d10083-1---d8754z-;rport Max-Forwards: 70 Contact: < sip:1000 at 192.168.10.2:41120;rinstance=57d4e16c725a51e6;transport=TLS> To: From: ;tag=4a2eae22 Call-ID: N2RiMjE2ZTQxNzYzMGY3ZmM3ZGJhODM4NTMzZmQ1YTA. CSeq: 1 REGISTER Expires: 60 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Zoiper r27147 Allow-Events: presence, kpml Content-Length: 0 ------------------------------------------------------------------------ send 716 bytes to tls/[222.248.102.244]:38620 at 20:57:01.806433: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/TLS 192.168.10.2:41120 ;branch=z9hG4bK-d8754z-903949acb1d10083-1---d8754z-;rport=38620;received=222.248.102.244 From: ;tag=4a2eae22 To: ;tag=2e74mgtyr84ve Call-ID: N2RiMjE2ZTQxNzYzMGY3ZmM3ZGJhODM4NTMzZmQ1YTA. CSeq: 1 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.4.13+git~20141103T195300Z~b942d0faa8~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces WWW-Authenticate: Digest realm="abc-xyz.com", nonce="374fdd98-7994-11e4-bf9c-1182ebca2429", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ recv 944 bytes from tls/[222.248.102.244]:38620 at 20:57:01.995662: ------------------------------------------------------------------------ REGISTER sip:abc-xyz.com;transport=TLS SIP/2.0 Via: SIP/2.0/TLS 192.168.10.2:41120 ;branch=z9hG4bK-d8754z-8ca0e782336414eb-1---d8754z-;rport Max-Forwards: 70 Contact: < sip:1000 at 192.168.10.2:41120;rinstance=57d4e16c725a51e6;transport=TLS> To: From: ;tag=4a2eae22 Call-ID: N2RiMjE2ZTQxNzYzMGY3ZmM3ZGJhODM4NTMzZmQ1YTA. CSeq: 2 REGISTER Expires: 60 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Zoiper r27147 Authorization: Digest username="1000",realm="abc-xyz.com ",nonce="374fdd98-7994-11e4-bf9c-1182ebca2429",uri=" sip:abc-xyz.com;transport=TLS ",response="8d7e108ff98035d37424c32166fe0253",cnonce="107cf281c3b57888a4bfd1a9c3776098",nc=00000001,qop=auth,algorithm=MD5 Allow-Events: presence, kpml Content-Length: 0 ------------------------------------------------------------------------ send 712 bytes to tls/[222.248.102.244]:38620 at 20:57:01.997931: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/TLS 192.168.10.2:41120 ;branch=z9hG4bK-d8754z-8ca0e782336414eb-1---d8754z-;rport=38620;received=222.248.102.244 From: ;tag=4a2eae22 To: ;tag=3Q0XpBB2NHUFa Call-ID: N2RiMjE2ZTQxNzYzMGY3ZmM3ZGJhODM4NTMzZmQ1YTA. CSeq: 2 REGISTER Contact: < sip:1000 at 192.168.10.2:41120;rinstance=57d4e16c725a51e6;transport=TLS >;expires=60 Date: Mon, 01 Dec 2014 19:57:01 GMT User-Agent: FreeSWITCH-mod_sofia/1.4.13+git~20141103T195300Z~b942d0faa8~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Content-Length: 0 ------------------------------------------------------------------------ recv 944 bytes from tls/[222.248.102.244]:38620 at 20:57:02.205719: ------------------------------------------------------------------------ REGISTER sip:abc-xyz.com;transport=TLS SIP/2.0 Via: SIP/2.0/TLS 222.248.102.244:38620 ;branch=z9hG4bK-d8754z-59d493abef215091-1---d8754z-;rport Max-Forwards: 70 Contact: < sip:1000 at 192.168.10.2:41120;rinstance=57d4e16c725a51e6;transport=TLS >;expires=0 To: From: ;tag=4a2eae22 Call-ID: N2RiMjE2ZTQxNzYzMGY3ZmM3ZGJhODM4NTMzZmQ1YTA. CSeq: 3 REGISTER Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Zoiper r27147 Authorization: Digest username="1000",realm="abc-xyz.com ",nonce="374fdd98-7994-11e4-bf9c-1182ebca2429",uri=" sip:abc-xyz.com;transport=TLS ",response="5168bf9afffe9adacef7d898bd6bec9b",cnonce="80108fa519303990cfd2d46639d767ee",nc=00000002,qop=auth,algorithm=MD5 Allow-Events: presence, kpml Content-Length: 0 ------------------------------------------------------------------------ send 598 bytes to tls/[222.248.102.244]:38620 at 20:57:02.207249: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/TLS 222.248.102.244:38620 ;branch=z9hG4bK-d8754z-59d493abef215091-1---d8754z-;rport=38620 From: ;tag=4a2eae22 To: ;tag=40Spr6U5jtH2N Call-ID: N2RiMjE2ZTQxNzYzMGY3ZmM3ZGJhODM4NTMzZmQ1YTA. CSeq: 3 REGISTER Date: Mon, 01 Dec 2014 19:57:02 GMT User-Agent: FreeSWITCH-mod_sofia/1.4.13+git~20141103T195300Z~b942d0faa8~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Content-Length: 0 ------------------------------------------------------------------------ recv 681 bytes from tls/[222.248.102.244]:38620 at 20:57:02.400564: ------------------------------------------------------------------------ REGISTER sip:abc-xyz.com;transport=TLS SIP/2.0 Via: SIP/2.0/TLS 222.248.102.244:38620 ;branch=z9hG4bK-d8754z-d738a1a836608006-1---d8754z-;rport Max-Forwards: 70 Contact: < sip:1000 at 222.248.102.244:38620;rinstance=43166fee0aaac1bc;transport=TLS> To: From: ;tag=979e5f46 Call-ID: MTJhNjIzMDM1MDEwMDVjZDRkOGMyYjBlMWU4ZDEzY2Y. CSeq: 1 REGISTER Expires: 60 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Zoiper r27147 Allow-Events: presence, kpml Content-Length: 0 ------------------------------------------------------------------------ send 694 bytes to tls/[222.248.102.244]:38620 at 20:57:02.401167: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/TLS 222.248.102.244:38620 ;branch=z9hG4bK-d8754z-d738a1a836608006-1---d8754z-;rport=38620 From: ;tag=979e5f46 To: ;tag=59jFt1c9F37mH Call-ID: MTJhNjIzMDM1MDEwMDVjZDRkOGMyYjBlMWU4ZDEzY2Y. CSeq: 1 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.4.13+git~20141103T195300Z~b942d0faa8~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces WWW-Authenticate: Digest realm="abc-xyz.com", nonce="37aaa44e-7994-11e4-bf9d-1182ebca2429", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ recv 950 bytes from tls/[222.248.102.244]:38620 at 20:57:02.655864: ------------------------------------------------------------------------ REGISTER sip:abc-xyz.com;transport=TLS SIP/2.0 Via: SIP/2.0/TLS 222.248.102.244:38620 ;branch=z9hG4bK-d8754z-dfaa7130675d6cfa-1---d8754z-;rport Max-Forwards: 70 Contact: < sip:1000 at 222.248.102.244:38620;rinstance=43166fee0aaac1bc;transport=TLS> To: From: ;tag=979e5f46 Call-ID: MTJhNjIzMDM1MDEwMDVjZDRkOGMyYjBlMWU4ZDEzY2Y. CSeq: 2 REGISTER Expires: 60 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Zoiper r27147 Authorization: Digest username="1000",realm="abc-xyz.com ",nonce="37aaa44e-7994-11e4-bf9d-1182ebca2429",uri=" sip:abc-xyz.com;transport=TLS ",response="f1667094626fc10ca76a070438497e6f",cnonce="dc790bffe285311f0b903bd515a3e741",nc=00000001,qop=auth,algorithm=MD5 Allow-Events: presence, kpml Content-Length: 0 ------------------------------------------------------------------------ send 693 bytes to tls/[222.248.102.244]:38620 at 20:57:02.657688: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/TLS 222.248.102.244:38620 ;branch=z9hG4bK-d8754z-dfaa7130675d6cfa-1---d8754z-;rport=38620 From: ;tag=979e5f46 To: ;tag=6jc8UvXcDcy7c Call-ID: MTJhNjIzMDM1MDEwMDVjZDRkOGMyYjBlMWU4ZDEzY2Y. CSeq: 2 REGISTER Contact: < sip:1000 at 222.248.102.244:38620;rinstance=43166fee0aaac1bc;transport=TLS >;expires=60 Date: Mon, 01 Dec 2014 19:57:02 GMT User-Agent: FreeSWITCH-mod_sofia/1.4.13+git~20141103T195300Z~b942d0faa8~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Content-Length: 0 *From: *Brian West *To: *FreeSWITCH Users Help *Date: *December 1, 2014 at 5:19:30 PM GMT+3 *Reply-To: *FreeSWITCH Users Help *Subject: **Re: [Freeswitch-users] FreeSWITCH TLS not able to receive calls* What endpoints are involved? have you looked at 'sofia loglevel all 9' output and see if it gives you a clue? On Fri, Nov 28, 2014 at 6:24 PM, Ahmed Habiba wrote: > Dears, > > I?ve configured FreeSWITCH with the below version with TLS/SRTPas per the > recommendation in page ?https://wiki.freeswitch.org/wiki/SIP_TLS? and it > was strait forward, and I was able to connect and make make calls using > zoiper, but I was not able to receive any calls after enabling the TLS/SRTP. > > *"FreeSWITCH Version 1.4.13+git~20141103T195300Z~b942d0faa8~64bit (git > b942d0f 2014-11-03 19:53:00Z 64bit)**?* > > Your kind feedback will be appreciate. > > Thanks, > > Ahmed Habiba. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -- Thanks and Best Regards, Ahmed Habiba Mob: +20 10 37 82 970 *Success: believe (Vision) plus commitment (Action)* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141201/5b7d4eef/attachment-0001.html From AWalters at cawst.org Tue Dec 2 02:50:32 2014 From: AWalters at cawst.org (Allison Walters) Date: Mon, 1 Dec 2014 23:50:32 +0000 Subject: [Freeswitch-users] Questions before starting new install Message-ID: <20b779910393484881146823fb74258f@BN1PR0101MB0817.prod.exchangelabs.com> We're wanting to replace our current freeswitch server (it's fairly old), and I have a few things I'm curious about before I start trying to build the new one. I inherited the current one and am relatively unfamiliar to all of this, so I appreciate your patience. 1. I was told by the consultant that set up the current box that freeswitch doesn't play well with fusionpbx or really any of the GUI's - there have been instances (and indeed one happened a couple of months ago while I was here) of the config files getting randomly overwritten with old settings, which supposedly is the fault of trying to run fusion. Also I was told it doesn't provide nice clean XML, etc... I was strongly encouraged to go with no GUI. If it were just me, I'd be okay with that, but we have a couple of other people who might be helping to maintain the server and they're far more comfortable with GUIs. Has anyone else had any of these issues, and/or have any recommendations on the subject? 2. I'm trying to install on Ubuntu. When I go to the wiki, I get this page: https://freeswitch.org/confluence/display/FREESWITCH/Linux , which just gives an apt-get line for Ubuntu. However, that line just seems to have pre-reqs and not freeswitch itself. Anyone know where I could find current Ubuntu instructions? I googled but the most recent pages I could find were from 2012, and most were from 2011. Is that the most up to date? 3. Are there any guidelines for migration from an old server running an old version to a new server with a current version? Thank you! -- Allison Walters IT Coordinator CAWST - Centre for Affordable Water and Sanitation Technology 424 Aviation Road NE, Calgary, Alberta, T2E 8H6, Canada tel: +1-403-243-3285 ext.264 web: www.cawst.org Join us in reaching 20 million people by 2020 with safe drinking water and sanitation. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141201/f58e8ffe/attachment.html From max at nysolutions.com Tue Dec 2 03:31:58 2014 From: max at nysolutions.com (Moishe Grunstein) Date: Tue, 2 Dec 2014 00:31:58 +0000 Subject: [Freeswitch-users] Questions before starting new install In-Reply-To: <20b779910393484881146823fb74258f@BN1PR0101MB0817.prod.exchangelabs.com> References: <20b779910393484881146823fb74258f@BN1PR0101MB0817.prod.exchangelabs.com> Message-ID: 1) You cannot migrate a working freeswitch to FusionPBX. You would need to recreate all the configs in the FusionPBX web interface. 2) Ubuntu install is the same as debian https://freeswitch.org/confluence/display/FREESWITCH/Debian if you are going with fusionpbx you would be better off with one of their scripts. 3) If you are staying with native freeswitch, you can copy over almost all configs to the new server. What version are you migrating from? Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support at nysolutions.com Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [cid:image001.jpg at 01C72F94.9EE45D60] Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Allison Walters Sent: Monday, December 1, 2014 6:51 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Questions before starting new install We're wanting to replace our current freeswitch server (it's fairly old), and I have a few things I'm curious about before I start trying to build the new one. I inherited the current one and am relatively unfamiliar to all of this, so I appreciate your patience. 1. I was told by the consultant that set up the current box that freeswitch doesn't play well with fusionpbx or really any of the GUI's - there have been instances (and indeed one happened a couple of months ago while I was here) of the config files getting randomly overwritten with old settings, which supposedly is the fault of trying to run fusion. Also I was told it doesn't provide nice clean XML, etc... I was strongly encouraged to go with no GUI. If it were just me, I'd be okay with that, but we have a couple of other people who might be helping to maintain the server and they're far more comfortable with GUIs. Has anyone else had any of these issues, and/or have any recommendations on the subject? 2. I'm trying to install on Ubuntu. When I go to the wiki, I get this page: https://freeswitch.org/confluence/display/FREESWITCH/Linux , which just gives an apt-get line for Ubuntu. However, that line just seems to have pre-reqs and not freeswitch itself. Anyone know where I could find current Ubuntu instructions? I googled but the most recent pages I could find were from 2012, and most were from 2011. Is that the most up to date? 3. Are there any guidelines for migration from an old server running an old version to a new server with a current version? Thank you! -- Allison Walters IT Coordinator CAWST - Centre for Affordable Water and Sanitation Technology 424 Aviation Road NE, Calgary, Alberta, T2E 8H6, Canada tel: +1-403-243-3285 ext.264 web: www.cawst.org Join us in reaching 20 million people by 2020 with safe drinking water and sanitation. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141202/bd3bbeaf/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 2424 bytes Desc: image001.jpg Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141202/bd3bbeaf/attachment-0001.jpg From aademattia at comcast.net Tue Dec 2 04:37:18 2014 From: aademattia at comcast.net (Andrew) Date: Mon, 1 Dec 2014 20:37:18 -0500 Subject: [Freeswitch-users] Send_DTMF Message-ID: <01f601d00dd0$86da3370$948e9a50$@comcast.net> Hi, I am making an outbound call then I am doing a bridge. Once I bridge the call I need to send dtmf to second call (bridge call) When doing a trace it looks like the DTMF is going to my first call and not the bridge call. I am using UUID send dtmf using both UUID but both don't seem to work. I do see the DTMF in wire shark but again looks like its going the wrong direction. Andrew -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141201/3831cf32/attachment.html From AWalters at cawst.org Tue Dec 2 04:51:06 2014 From: AWalters at cawst.org (Allison Walters) Date: Tue, 2 Dec 2014 01:51:06 +0000 Subject: [Freeswitch-users] Questions before starting new install In-Reply-To: References: <20b779910393484881146823fb74258f@BN1PR0101MB0817.prod.exchangelabs.com> Message-ID: <78cb8a4d983743bc94c0f63a732a1c10@BN1PR0101MB0817.prod.exchangelabs.com> Whoops ? accidentally had mail delivery disabled. So I?ll respond to Moishe?s points with a manual reply/quote: > 1) You cannot migrate a working freeswitch to FusionPBX. You would need to recreate all the configs in the FusionPBX web interface. I didn?t change anything ? they were both already installed. Mind you, I don?t know how the person who installed it did it; maybe that?s related to the issues they were seeing? Regardless, the Fusion interface seems to read all of the correct info, including when things are changed in the XML files (which is how I was encouraged to do changes?). Now, I?m confused ? Fusion keeps its own copy of the config files? It?s not just a wrapper that modifies the existing freeswitch ones? (Just going off what I was told?) > 2) Ubuntu install is the same as debian https://freeswitch.org/confluence/display/FREESWITCH/Debian if you are going with fusionpbx you would be better off with one of their scripts. I looked at their script, but it didn?t seem to have been updated since 2012, which makes me a bit nervous. I?ll take another peek, though. > 3) If you are staying with native freeswitch, you can copy over almost all configs to the new server. What version are you migrating from? The current server is running 1.3.0. Thanks! -- Allison Walters IT Coordinator CAWST - Centre for Affordable Water and Sanitation Technology 424 Aviation Road NE, Calgary, Alberta, T2E 8H6, Canada tel: +1-403-243-3285 ext.264 web: www.cawst.org Join us in reaching 20 million people by 2020 with safe drinking water and sanitation. From: Allison Walters > Date: Mon, Dec 1, 2014 at 3:50 PM Subject: [Freeswitch-users] Questions before starting new install To: "freeswitch-users at lists.freeswitch.org" > We?re wanting to replace our current freeswitch server (it?s fairly old), and I have a few things I?m curious about before I start trying to build the new one. I inherited the current one and am relatively unfamiliar to all of this, so I appreciate your patience. 1. I was told by the consultant that set up the current box that freeswitch doesn?t play well with fusionpbx or really any of the GUI?s ? there have been instances (and indeed one happened a couple of months ago while I was here) of the config files getting randomly overwritten with old settings, which supposedly is the fault of trying to run fusion. Also I was told it doesn?t provide nice clean XML, etc? I was strongly encouraged to go with no GUI. If it were just me, I?d be okay with that, but we have a couple of other people who might be helping to maintain the server and they?re far more comfortable with GUIs. Has anyone else had any of these issues, and/or have any recommendations on the subject? 2. I?m trying to install on Ubuntu. When I go to the wiki, I get this page: https://freeswitch.org/confluence/display/FREESWITCH/Linux , which just gives an apt-get line for Ubuntu. However, that line just seems to have pre-reqs and not freeswitch itself. Anyone know where I could find current Ubuntu instructions? I googled but the most recent pages I could find were from 2012, and most were from 2011. Is that the most up to date? 3. Are there any guidelines for migration from an old server running an old version to a new server with a current version? Thank you! -- Allison Walters IT Coordinator CAWST - Centre for Affordable Water and Sanitation Technology 424 Aviation Road NE, Calgary, Alberta, T2E 8H6, Canada tel: +1-403-243-3285 ext.264 web: www.cawst.org Join us in reaching 20 million people by 2020 with safe drinking water and sanitation. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141202/e0264b13/attachment-0001.html From max at nysolutions.com Tue Dec 2 07:37:15 2014 From: max at nysolutions.com (Moishe Grunstein) Date: Tue, 2 Dec 2014 04:37:15 +0000 Subject: [Freeswitch-users] Questions before starting new install In-Reply-To: <78cb8a4d983743bc94c0f63a732a1c10@BN1PR0101MB0817.prod.exchangelabs.com> References: <20b779910393484881146823fb74258f@BN1PR0101MB0817.prod.exchangelabs.com> <78cb8a4d983743bc94c0f63a732a1c10@BN1PR0101MB0817.prod.exchangelabs.com> Message-ID: 1) FusionPBX renames Freeswitch default configs after installation, it uses and then creates its own XML files (unless you use their XML handler which writes to db instead). If you then modify XML files directly, the next time you make a change in the GUI they will overwrite your changes. They have an IRC channel #FusionPBX and they use googlecode for svn and issue tracker.. 2) Not sure what scripts you are referring to be dated, FusionPBX scripts have been last modified this month http://fusionpbx.googlecode.com/svn/trunk/scripts/install/ubuntu/install_fusionpbx.sh Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support at nysolutions.com Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [cid:image001.jpg at 01C72F94.9EE45D60] Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Allison Walters Sent: Monday, December 1, 2014 8:51 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Questions before starting new install Whoops ? accidentally had mail delivery disabled. So I?ll respond to Moishe?s points with a manual reply/quote: > 1) You cannot migrate a working freeswitch to FusionPBX. You would need to recreate all the configs in the FusionPBX web interface. I didn?t change anything ? they were both already installed. Mind you, I don?t know how the person who installed it did it; maybe that?s related to the issues they were seeing? Regardless, the Fusion interface seems to read all of the correct info, including when things are changed in the XML files (which is how I was encouraged to do changes?). Now, I?m confused ? Fusion keeps its own copy of the config files? It?s not just a wrapper that modifies the existing freeswitch ones? (Just going off what I was told?) > 2) Ubuntu install is the same as debian https://freeswitch.org/confluence/display/FREESWITCH/Debian if you are going with fusionpbx you would be better off with one of their scripts. I looked at their script, but it didn?t seem to have been updated since 2012, which makes me a bit nervous. I?ll take another peek, though. > 3) If you are staying with native freeswitch, you can copy over almost all configs to the new server. What version are you migrating from? The current server is running 1.3.0. Thanks! -- Allison Walters IT Coordinator CAWST - Centre for Affordable Water and Sanitation Technology 424 Aviation Road NE, Calgary, Alberta, T2E 8H6, Canada tel: +1-403-243-3285 ext.264 web: www.cawst.org Join us in reaching 20 million people by 2020 with safe drinking water and sanitation. From: Allison Walters > Date: Mon, Dec 1, 2014 at 3:50 PM Subject: [Freeswitch-users] Questions before starting new install To: "freeswitch-users at lists.freeswitch.org" > We?re wanting to replace our current freeswitch server (it?s fairly old), and I have a few things I?m curious about before I start trying to build the new one. I inherited the current one and am relatively unfamiliar to all of this, so I appreciate your patience. 1. I was told by the consultant that set up the current box that freeswitch doesn?t play well with fusionpbx or really any of the GUI?s ? there have been instances (and indeed one happened a couple of months ago while I was here) of the config files getting randomly overwritten with old settings, which supposedly is the fault of trying to run fusion. Also I was told it doesn?t provide nice clean XML, etc? I was strongly encouraged to go with no GUI. If it were just me, I?d be okay with that, but we have a couple of other people who might be helping to maintain the server and they?re far more comfortable with GUIs. Has anyone else had any of these issues, and/or have any recommendations on the subject? 2. I?m trying to install on Ubuntu. When I go to the wiki, I get this page: https://freeswitch.org/confluence/display/FREESWITCH/Linux , which just gives an apt-get line for Ubuntu. However, that line just seems to have pre-reqs and not freeswitch itself. Anyone know where I could find current Ubuntu instructions? I googled but the most recent pages I could find were from 2012, and most were from 2011. Is that the most up to date? 3. Are there any guidelines for migration from an old server running an old version to a new server with a current version? Thank you! -- Allison Walters IT Coordinator CAWST - Centre for Affordable Water and Sanitation Technology 424 Aviation Road NE, Calgary, Alberta, T2E 8H6, Canada tel: +1-403-243-3285 ext.264 web: www.cawst.org Join us in reaching 20 million people by 2020 with safe drinking water and sanitation. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141202/244dbd56/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 2424 bytes Desc: image001.jpg Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141202/244dbd56/attachment-0001.jpg From jaybinks at gmail.com Tue Dec 2 07:54:31 2014 From: jaybinks at gmail.com (jay binks) Date: Tue, 2 Dec 2014 14:54:31 +1000 Subject: [Freeswitch-users] Crazy Cause Code with "INVALID PROFILE" Message-ID: So im doing some integration testing, I am sending a call to sofia profile A , and its routed out profile B. if I take profile B down, I can see freeswitch logs "mod_sofia.c:4417 Invalid Profile", which is fine.. the profile is stopped. but what I did not expect is that Freeswitch responds ( on profile A ) with : SIP/2.0 502 Bad Gateway Reason: SIP;cause=611;text="INVALID_PROFILE" which isnt even listed in https://freeswitch.org/confluence/display/FREESWITCH/Hangup+Cause+Code+Table as a valid hangup cause !! What I want to know is, how can I make freeswitch send a Q850 cause 41 in this case ?? ( without patching mod_sofia :P ) -- Sincerely Jay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141202/b8269d31/attachment.html From steveayre at gmail.com Tue Dec 2 12:10:51 2014 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 2 Dec 2014 09:10:51 +0000 Subject: [Freeswitch-users] Crazy Cause Code with "INVALID PROFILE" In-Reply-To: References: Message-ID: You can have it return to the dialplan on failure and (conditionally?) hangup with a 41 from there. On 2 December 2014 at 04:54, jay binks wrote: > So im doing some integration testing, I am sending a call to sofia profile > A , and its routed out profile B. > > if I take profile B down, I can see freeswitch logs "mod_sofia.c:4417 > Invalid Profile", which is fine.. the profile is stopped. > > but what I did not expect is that Freeswitch responds ( on profile A ) > with : > > SIP/2.0 502 Bad Gateway > Reason: SIP;cause=611;text="INVALID_PROFILE" > > > which isnt even listed in > https://freeswitch.org/confluence/display/FREESWITCH/Hangup+Cause+Code+Table > as a valid hangup cause !! > > What I want to know is, how can I make freeswitch send a Q850 cause 41 in > this case ?? > ( without patching mod_sofia :P ) > > -- > Sincerely > > Jay > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141202/5c1bf0b6/attachment.html From telishisheer at gmail.com Tue Dec 2 07:25:53 2014 From: telishisheer at gmail.com (Shisheer Teli) Date: Tue, 2 Dec 2014 09:55:53 +0530 Subject: [Freeswitch-users] Ldap alise and freeswitch integration problem Message-ID: Hi, I am able to bind with any alise on ldap server except userPassword (MD5) alise. when i bind password with userPassword , authentication fails. any MD5 authentication configuration in freeswitch? Configuration file: -- Regards, Shisheer T Phone: +91-022 2278 2763 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141202/55be9619/attachment.html From brian at freeswitch.org Tue Dec 2 15:12:03 2014 From: brian at freeswitch.org (Brian West) Date: Tue, 2 Dec 2014 06:12:03 -0600 Subject: [Freeswitch-users] Crazy Cause Code with "INVALID PROFILE" In-Reply-To: References: Message-ID: Those were added as part of FS-3321, Try sending your calls out a valid profile? :P On Mon, Dec 1, 2014 at 10:54 PM, jay binks wrote: > So im doing some integration testing, I am sending a call to sofia profile > A , and its routed out profile B. > > if I take profile B down, I can see freeswitch logs "mod_sofia.c:4417 > Invalid Profile", which is fine.. the profile is stopped. > > but what I did not expect is that Freeswitch responds ( on profile A ) > with : > > SIP/2.0 502 Bad Gateway > Reason: SIP;cause=611;text="INVALID_PROFILE" > > > which isnt even listed in > https://freeswitch.org/confluence/display/FREESWITCH/Hangup+Cause+Code+Table > as a valid hangup cause !! > > What I want to know is, how can I make freeswitch send a Q850 cause 41 in > this case ?? > ( without patching mod_sofia :P ) > > -- > Sincerely > > Jay > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141202/dcd9d6de/attachment.html From jaybinks at gmail.com Tue Dec 2 15:37:36 2014 From: jaybinks at gmail.com (jay binks) Date: Tue, 2 Dec 2014 22:37:36 +1000 Subject: [Freeswitch-users] Crazy Cause Code with "INVALID PROFILE" In-Reply-To: References: Message-ID: yea your probably right that im never really going to have that scenario... however I wanted to test what would happen ( on profile A side ) if link was down on the interface for profile B. hmmm have to test that better :P On 2 December 2014 at 22:12, Brian West wrote: > Those were added as part of FS-3321, Try sending your calls out a valid > profile? :P > > On Mon, Dec 1, 2014 at 10:54 PM, jay binks wrote: > >> So im doing some integration testing, I am sending a call to sofia >> profile A , and its routed out profile B. >> >> if I take profile B down, I can see freeswitch logs "mod_sofia.c:4417 >> Invalid Profile", which is fine.. the profile is stopped. >> >> but what I did not expect is that Freeswitch responds ( on profile A ) >> with : >> >> SIP/2.0 502 Bad Gateway >> Reason: SIP;cause=611;text="INVALID_PROFILE" >> >> >> which isnt even listed in >> https://freeswitch.org/confluence/display/FREESWITCH/Hangup+Cause+Code+Table >> as a valid hangup cause !! >> >> What I want to know is, how can I make freeswitch send a Q850 cause 41 >> in this case ?? >> ( without patching mod_sofia :P ) >> >> -- >> Sincerely >> >> Jay >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > > *Brian West* > brian at freeswitch.org > > > *Twitter: @FreeSWITCH , @briankwest* > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20141202/6905c928/attachment-0001.html From achinthau at gmail.com Tue Dec 2 16:11:24 2014 From: achinthau at gmail.com (Achintha) Date: Tue, 2 Dec 2014 18:41:24 +0530 Subject: [Freeswitch-users] Fwd: mod_verto configuration Message-ID: Hi I used freeswitch on centos 6 . I downloaded freeswitch from ?git clone https://freeswitch.org/stash/scm/fs/freeswitch.git And configure properly without errors Then I try to configure mod_verto , https://freeswitch.org/confluence/display/FREESWITCH/mod_verto First I compile the module and then I did configurations from mentioned link below, But I couldn?t create ssl certificate from that instructions Then I used following command to create the certificate and key openssl req -x509 -nodes -days 365 -newkey rsa:2048 -keyout /usr/local/freeswitch/certs/wss.key -out /usr/local/freeswitch/certs/wss.crt cat wss.crt wss.key > /usr/local/freeswitch/certs/wss.pem cat wss.crt > /usr/local/freeswitch/tls/dtls-srtp.crt then I changed ssl.conf file (/etc/httpd/conf.d/ssl.conf) as follows ServerName myserverip:443 SSLEngine on SSLCertificateFile /usr/local/freeswitch/certs/wss.crt SSLCertificateKeyFile /usr/local/freeswitch/certs/wss.key Then restarted httpd server Get the firefox and put https:// myserverip/verto Ask certificate and after add the certificate Loaded login page but can?t log in freeswitch console print following message 2014-12-02 18:11:40.150201 [INFO] mod_verto.c:3688 192.168.1.195:4870 Client Connect. 2014-12-02 18:11:40.150201 [INFO] mod_verto.c:1827 192.168.1.195:4870 Starting client thread. 2014-12-02 18:11:41.169676 [DEBUG] mod_verto.c:1739 192.168.1.195:4870 WS SETUP FAILED [] 2014-12-02 18:11:41.169676 [INFO] mod_verto.c:1853 192.168.1.195:4870 Ending client thread. 2014-12-02 18:11:41.169676 [INFO] mod_verto.c:1860 192.168.1.195:4870 Thread ended Kindly help me to configure this thankyou ach -------------- next part -------------- An HTML attachment was scrubbed... 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Name: wss.key Type: application/octet-stream Size: 1703 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141202/5545978e/attachment-0001.obj From avi at avimarcus.net Tue Dec 2 16:22:22 2014 From: avi at avimarcus.net (Avi Marcus) Date: Tue, 2 Dec 2014 13:22:22 +0000 Subject: [Freeswitch-users] Newbie -- Help Needed Transferring Inbound Caller ID to external SIP Gateway URI In-Reply-To: References: Message-ID: <0000014a0b2a7f3a-e057e705-e811-4903-b2b9-bbd020abb4a7-000000@email.amazonses.com> Hi - did you figure this out yet? One comment: -- I don't know if $1 is available anymore. You might want to just set that as part of the actual number to route, e.g. add a 1212 prefix and match it again or set it as a channel variable. You can see in the logs if the $1 is resolving correctly. -Avi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141202/d0212cc2/attachment.html From dujinfang at gmail.com Tue Dec 2 16:52:10 2014 From: dujinfang at gmail.com (Seven Du) Date: Tue, 2 Dec 2014 21:52:10 +0800 Subject: [Freeswitch-users] Send_DTMF In-Reply-To: <01f601d00dd0$86da3370$948e9a50$@comcast.net> References: <01f601d00dd0$86da3370$948e9a50$@comcast.net> Message-ID: <855745AA993749539E5AF564EA1F61A4@gmail.com> try uuid_recv_dtmf -- Seven Du http://about.me/dujinfang http://www.dujinfang.com http://www.freeswitch.org.cn Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Tuesday, December 2, 2014 at 9:37 AM, Andrew wrote: > Hi, > I am making an outbound call then I am doing a bridge. > Once I bridge the call I need to send dtmf to second call (bridge call) > When doing a trace it looks like the DTMF is going to my first call and not the bridge call. > I am using UUID send dtmf using both UUID but both don?t seem to work. > > I do see the DTMF in wire shark but again looks like its going the wrong direction. > > Andrew > > > > _________________________________________________________________________ > 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://confluence.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/20141202/cbcf12ba/attachment.html From aademattia at comcast.net Tue Dec 2 18:04:31 2014 From: aademattia at comcast.net (=?utf-8?B?YWFkZW1hdHRpYUBjb21jYXN0Lm5ldA==?=) Date: Tue, 02 Dec 2014 10:04:31 -0500 Subject: [Freeswitch-users] =?utf-8?q?Send=5FDTMF?= Message-ID: I still see the dtmf going to the wrong destination. I am sending dtmf on both uuid and can't seem to get the ivr I am calling to see my dtmf. But if I hit dtmf on my phone it works. Sent from my HTC ----- Reply message ----- From: "Seven Du" To: "FreeSWITCH Users Help" Subject: [Freeswitch-users] Send_DTMF Date: Tue, Dec 2, 2014 8:52 AM try uuid_recv_dtmf -- Seven Du http://about.me/dujinfang http://www.dujinfang.com http://www.freeswitch.org.cn Sent with Sparrow On Tuesday, December 2, 2014 at 9:37 AM, Andrew wrote: Hi,I am making an outbound call then I am doing a bridge.Once I bridge the call I need to send dtmf to second call (bridge call)When doing a trace it looks like the DTMF is going to my first call and not the bridge call.I am using UUID send dtmf using both UUID but both don?t seem to work. I do see the DTMF in wire shark but again looks like its going the wrong direction. Andrew _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch..org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141202/ce64b799/attachment.html From steveayre at gmail.com Tue Dec 2 18:08:13 2014 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 2 Dec 2014 15:08:13 +0000 Subject: [Freeswitch-users] Crazy Cause Code with "INVALID PROFILE" In-Reply-To: References: Message-ID: If the link is down the profile will still exist but packets won't route. Unless you mean when it's down on startup? On 2 December 2014 at 12:37, jay binks wrote: > yea your probably right that im never really going to have that scenario... > however I wanted to test what would happen ( on profile A side ) if link > was down on the interface for profile B. > > hmmm have to test that better :P > > On 2 December 2014 at 22:12, Brian West wrote: > >> Those were added as part of FS-3321, Try sending your calls out a valid >> profile? :P >> >> On Mon, Dec 1, 2014 at 10:54 PM, jay binks wrote: >> >>> So im doing some integration testing, I am sending a call to sofia >>> profile A , and its routed out profile B. >>> >>> if I take profile B down, I can see freeswitch logs "mod_sofia.c:4417 >>> Invalid Profile", which is fine.. the profile is stopped. >>> >>> but what I did not expect is that Freeswitch responds ( on profile A ) >>> with : >>> >>> SIP/2.0 502 Bad Gateway >>> Reason: SIP;cause=611;text="INVALID_PROFILE" >>> >>> >>> which isnt even listed in >>> https://freeswitch.org/confluence/display/FREESWITCH/Hangup+Cause+Code+Table >>> as a valid hangup cause !! >>> >>> What I want to know is, how can I make freeswitch send a Q850 cause 41 >>> in this case ?? >>> ( without patching mod_sofia :P ) >>> >>> -- >>> Sincerely >>> >>> Jay >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> >> *Brian West* >> brian at freeswitch.org >> >> >> *Twitter: @FreeSWITCH , @briankwest* >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141202/f2de8a5e/attachment.html From msc at freeswitch.org Tue Dec 2 18:57:54 2014 From: msc at freeswitch.org (Michael Collins) Date: Tue, 2 Dec 2014 07:57:54 -0800 Subject: [Freeswitch-users] UniFi phones and FreeSWITCH Message-ID: Hello FreeSWITCHers! Does anybody have any experience with these phones? http://www.ubnt.com/unifi-voip/uvp/ I'm curious to know how they work in a FS environment. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141202/dfa6f51e/attachment.html From max at nysolutions.com Tue Dec 2 19:12:56 2014 From: max at nysolutions.com (Moishe Grunstein) Date: Tue, 2 Dec 2014 16:12:56 +0000 Subject: [Freeswitch-users] UniFi phones and FreeSWITCH In-Reply-To: References: Message-ID: They are only shipping the basic model at the moment. Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support at nysolutions.com Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [cid:image001.jpg at 01C72F94.9EE45D60] Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Tuesday, December 2, 2014 10:58 AM To: FreeSWITCH Users Help Subject: [Freeswitch-users] UniFi phones and FreeSWITCH Hello FreeSWITCHers! Does anybody have any experience with these phones? http://www.ubnt.com/unifi-voip/uvp/ I'm curious to know how they work in a FS environment. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141202/4087dfca/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 2424 bytes Desc: image001.jpg Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141202/4087dfca/attachment-0001.jpg From AWalters at cawst.org Tue Dec 2 19:41:46 2014 From: AWalters at cawst.org (Allison Walters) Date: Tue, 2 Dec 2014 16:41:46 +0000 Subject: [Freeswitch-users] Questions before starting new install In-Reply-To: References: <20b779910393484881146823fb74258f@BN1PR0101MB0817.prod.exchangelabs.com> <78cb8a4d983743bc94c0f63a732a1c10@BN1PR0101MB0817.prod.exchangelabs.com> Message-ID: <70694fb57953496b88aae1666accac7e@BN1PR0101MB0817.prod.exchangelabs.com> Hmmm?that?s a different version of the script than the link I found, I think. Thanks! So it?s starting to sound like the problem may be that Fusion was installed after the fact on the current server, and so things don?t integrate quite properly (or at least as expected). Maybe I?ll give a test install of both a go, then. Thanks! -- Allison Walters IT Coordinator CAWST - Centre for Affordable Water and Sanitation Technology 424 Aviation Road NE, Calgary, Alberta, T2E 8H6, Canada tel: +1-403-243-3285 ext.264 web: www.cawst.org Join us in reaching 20 million people by 2020 with safe drinking water and sanitation. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Moishe Grunstein Sent: Monday, December 01, 2014 9:37 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Questions before starting new install 1) FusionPBX renames Freeswitch default configs after installation, it uses and then creates its own XML files (unless you use their XML handler which writes to db instead). If you then modify XML files directly, the next time you make a change in the GUI they will overwrite your changes. They have an IRC channel #FusionPBX and they use googlecode for svn and issue tracker.. 2) Not sure what scripts you are referring to be dated, FusionPBX scripts have been last modified this month http://fusionpbx.googlecode.com/svn/trunk/scripts/install/ubuntu/install_fusionpbx.sh Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support at nysolutions.com Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [cid:image001.jpg at 01C72F94.9EE45D60] Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Allison Walters Sent: Monday, December 1, 2014 8:51 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Questions before starting new install Whoops ? accidentally had mail delivery disabled. So I?ll respond to Moishe?s points with a manual reply/quote: > 1) You cannot migrate a working freeswitch to FusionPBX. You would need to recreate all the configs in the FusionPBX web interface. I didn?t change anything ? they were both already installed. Mind you, I don?t know how the person who installed it did it; maybe that?s related to the issues they were seeing? Regardless, the Fusion interface seems to read all of the correct info, including when things are changed in the XML files (which is how I was encouraged to do changes?). Now, I?m confused ? Fusion keeps its own copy of the config files? It?s not just a wrapper that modifies the existing freeswitch ones? (Just going off what I was told?) > 2) Ubuntu install is the same as debian https://freeswitch.org/confluence/display/FREESWITCH/Debian if you are going with fusionpbx you would be better off with one of their scripts. I looked at their script, but it didn?t seem to have been updated since 2012, which makes me a bit nervous. I?ll take another peek, though. > 3) If you are staying with native freeswitch, you can copy over almost all configs to the new server. What version are you migrating from? The current server is running 1.3.0. Thanks! -- Allison Walters IT Coordinator CAWST - Centre for Affordable Water and Sanitation Technology 424 Aviation Road NE, Calgary, Alberta, T2E 8H6, Canada tel: +1-403-243-3285 ext.264 web: www.cawst.org Join us in reaching 20 million people by 2020 with safe drinking water and sanitation. From: Allison Walters > Date: Mon, Dec 1, 2014 at 3:50 PM Subject: [Freeswitch-users] Questions before starting new install To: "freeswitch-users at lists.freeswitch.org" > We?re wanting to replace our current freeswitch server (it?s fairly old), and I have a few things I?m curious about before I start trying to build the new one. I inherited the current one and am relatively unfamiliar to all of this, so I appreciate your patience. 1. I was told by the consultant that set up the current box that freeswitch doesn?t play well with fusionpbx or really any of the GUI?s ? there have been instances (and indeed one happened a couple of months ago while I was here) of the config files getting randomly overwritten with old settings, which supposedly is the fault of trying to run fusion. Also I was told it doesn?t provide nice clean XML, etc? I was strongly encouraged to go with no GUI. If it were just me, I?d be okay with that, but we have a couple of other people who might be helping to maintain the server and they?re far more comfortable with GUIs. Has anyone else had any of these issues, and/or have any recommendations on the subject? 2. I?m trying to install on Ubuntu. When I go to the wiki, I get this page: https://freeswitch.org/confluence/display/FREESWITCH/Linux , which just gives an apt-get line for Ubuntu. However, that line just seems to have pre-reqs and not freeswitch itself. Anyone know where I could find current Ubuntu instructions? I googled but the most recent pages I could find were from 2012, and most were from 2011. Is that the most up to date? 3. Are there any guidelines for migration from an old server running an old version to a new server with a current version? Thank you! -- Allison Walters IT Coordinator CAWST - Centre for Affordable Water and Sanitation Technology 424 Aviation Road NE, Calgary, Alberta, T2E 8H6, Canada tel: +1-403-243-3285 ext.264 web: www.cawst.org Join us in reaching 20 million people by 2020 with safe drinking water and sanitation. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141202/8ce9cf1d/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 2424 bytes Desc: image001.jpg Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141202/8ce9cf1d/attachment-0001.jpg From jlamanna at gmail.com Tue Dec 2 19:49:10 2014 From: jlamanna at gmail.com (James Lamanna) Date: Tue, 2 Dec 2014 11:49:10 -0500 Subject: [Freeswitch-users] Missing CALLID occasionally on SpanDSP modems Message-ID: Hi, I'm using FreeSwitch SpanDSP Modems in conjunction with Hylafax, and I'm noticing that occasionally Hylafax gets no CALLID information: Dec 02 08:13:22.45: [ 7439]: SESSION BEGIN 000000080 2292330412 Dec 02 08:13:22.45: [ 7439]: HylaFAX (tm) Version 5.5.5 Dec 02 08:13:22.45: [ 7439]: CallID: '' '' '' '' This doesn't happen consistently, but I've already seen it on around 3 faxes out of 40. Freeswitch prints out an appropriate callerId to my logs. This is what my Hylafax config file looks like: CountryCode: 1 AreaCode: 000 FAXNumber: +1.999.555.1212 LongDistancePrefix: 1 InternationalPrefix: 011 DialStringRules: etc/dialrules ServerTracing: 1 SessionTracing: 0xFFF RecvFileMode: 0660 LogFileMode: 0660 DeviceMode: 0660 RingsBeforeAnswer: 2 SpeakerVolume: off GettyArgs: "-h %l dx_%s" LocalIdentifier: "NothingSetup" TagLineFont: etc/LiberationSans-25.pcf TagLineFormat: "From %%l|%c|Page %%P of %%T" MaxRecvPages: 500 ModemType: Class1 # use this to supply a hint ModemResetCmds: AT+VCID=1 # enables CallID display PagerTTYParity: none Class1AdaptRecvCmd: AT+FAR=1 Class1TMConnectDelay: 400 # counteract quick CONNECT response CallIDPattern: "NMBR=" CallIDPattern: "NAME=" CallIDPattern: "ANID=" CallIDPattern: "NDID=" DynamicConfig: "/usr/local/bin/faxreceiveconfig" MaxBatchJobs: "1" BadPageHandlingMethod: DCN Any ideas? Thanks. (please CC me directly, I am viewing in digest mode) -- James -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141202/601cd214/attachment.html From raphael.lechner at gmail.com Tue Dec 2 19:49:57 2014 From: raphael.lechner at gmail.com (Raphael Lechner) Date: Tue, 2 Dec 2014 17:49:57 +0100 Subject: [Freeswitch-users] table sip_presence is always empty Message-ID: Hi, I?m trying to enable presence but for now the sip_presence table is always empty and I don?t know If that is ok or not. This is my internal.xml https://pastebin.freeswitch.org/23666 I tried with the sqlite db and postgresql but nothing has changed. The table exists but is always empty: select * from sip_presence ; sip_user | sip_host | status | rpid | expires | user_agent | profile_name | hostname | network_ip | network_port | open_closed ----------+----------+--------+------+---------+------------+--------------+----------+------------+--------------+------------- (0 rows) The table sip_subscriptions contains entries like: sip | 99 | 192.168.130.130 | 12 | 192.168.130.130 | 192.168.130.130,192.168.130.130,test-pbx | presence | "Raphael" | N2RiMDQ2OTkwMWVhZjYzMjIwMDM0MGZlZjU4NDM1ZDY | "Raphael" ;tag=a4cb4203 | SIP/2.0/UDP 192.168.130.131:27050;branch=z9hG4bK-d8754z-53ed9718e7866617-1---d8754z-;rport=27050 | 1417537731 | Bria 4 4.1.1 74256-688c7d71-M | multipart/related, application/rlmi+xml, application/pidf+xml | internal | test-pbx | 27050 | 192.168.130.131 | 16 | | ;tag=wkBl3G43vfhs and with sofia_presence_data the status is always unknown: freeswitch at internal> sofia_presence_data list */99 status|rpid|user_agent|network_ip|network_port unknown|unknown|Bria 4 4.1.1 74256-688c7d71-M|192.168.130.131|27050 +OK I have enabled debugging for presence: https://pastebin.freeswitch.org/23667 FreeSWITCH Version: FreeSWITCH Version 1.4.13+git~20141103T195300Z~b942d0faa8~64bit (git b942d0f 2014-11-03 19:53:00Z 64bit) Any hint what I have configured wrong? Thank you, Raphael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141202/a381827f/attachment.html From krice at freeswitch.org Tue Dec 2 20:09:26 2014 From: krice at freeswitch.org (Ken Rice) Date: Tue, 02 Dec 2014 17:09:26 +0000 Subject: [Freeswitch-users] Freeswitch Week in Review (Master Branch) November 23rd-29th Message-ID: <547df2467a147_3aa175b32c65017@ip-10-16-129-130.mail> New Post on freeswitch.org from kathleen check it out at http://ift.tt/11P0e1O Freeswitch Week in Review (Master Branch) November 23rd-29th Hello, again. This week in the FreeSWITCH master branch we had 7 commits. It was a quiet week with the Thanksgiving holiday falling on Thursday, and the majority of the work done was geared toward build support for unimrcp. Improvements in cross platform build supports: 32c27b3 Added a Debian dependency to the CentOS6 makefile f4876d5 FS-7031 [unimrcp] update sofia-sip.m4 so that it can build when relative path is used in configure.gnu ?with-sofia-sip [Jira: http://ift.tt/1wkDnZk] 061f3cb FS-7031 #resolve #comment [unimrcp] update library again to pull in upstream fix for ?with-sofia-sip=../sofia-sip [Jira: http://ift.tt/1wkDnZk] 382e683 Use FTDM_UINT64_FMT macro to log uint64_t values, in order to not break x86 builds. The following bugs were squashed: 5bbef7f FS-7015 Fix for inbound call on hold issue in mod_sofia [Jira: http://ift.tt/1wkDp3C] Miscellaneous commits: 0d636af FS-7031 [unimrcp] revert configure.gnu change- doesn?t work when using non-source build dir. [Jira: http://ift.tt/1wkDnZk] The complete list of commits can be found here:2014_11_23-2014_11_30new -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141202/ee3fdb7d/attachment.html From AWalters at cawst.org Tue Dec 2 21:40:47 2014 From: AWalters at cawst.org (Allison Walters) Date: Tue, 2 Dec 2014 18:40:47 +0000 Subject: [Freeswitch-users] Questions before starting new install In-Reply-To: References: <20b779910393484881146823fb74258f@BN1PR0101MB0817.prod.exchangelabs.com> <78cb8a4d983743bc94c0f63a732a1c10@BN1PR0101MB0817.prod.exchangelabs.com> Message-ID: I tried that version of the script and it says it was written for 10.04 and 12.04, and it says it doesn?t always work outside of that, but it?ll ?give it a go?. And then after a bit, it failed out with a git error: installing stable v1.4 of FreeSWITCH /usr/bin/time: cannot run /usr/bin/git: No such file or directory Command exited with non-zero status 127 0.00user 0.00system 0:00.00elapsed 0%CPU (0avgtext+0avgdata 1220maxresident)k 0inputs+0outputs (0major+27minor)pagefaults 0swaps ./install_fusionpbx.sh: line 1114: cd: /usr/src/freeswitch: No such file or directory ./install_fusionpbx.sh: line 1115: /usr/bin/git: No such file or directory GIT ERROR I suspect this is because freeswitch moved to stash in July, apparently. Anyway, I?ll see if I can find anyone on the Fusion forum or IRC or somewhere that can help with this part. Thanks! -- Allison Walters IT Coordinator CAWST - Centre for Affordable Water and Sanitation Technology 424 Aviation Road NE, Calgary, Alberta, T2E 8H6, Canada tel: +1-403-243-3285 ext.264 web: www.cawst.org Join us in reaching 20 million people by 2020 with safe drinking water and sanitation. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Moishe Grunstein Sent: Monday, December 01, 2014 9:37 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Questions before starting new install 1) FusionPBX renames Freeswitch default configs after installation, it uses and then creates its own XML files (unless you use their XML handler which writes to db instead). If you then modify XML files directly, the next time you make a change in the GUI they will overwrite your changes. They have an IRC channel #FusionPBX and they use googlecode for svn and issue tracker.. 2) Not sure what scripts you are referring to be dated, FusionPBX scripts have been last modified this month http://fusionpbx.googlecode.com/svn/trunk/scripts/install/ubuntu/install_fusionpbx.sh Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support at nysolutions.com Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [cid:image001.jpg at 01C72F94.9EE45D60] Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Allison Walters Sent: Monday, December 1, 2014 8:51 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Questions before starting new install Whoops ? accidentally had mail delivery disabled. So I?ll respond to Moishe?s points with a manual reply/quote: > 1) You cannot migrate a working freeswitch to FusionPBX. You would need to recreate all the configs in the FusionPBX web interface. I didn?t change anything ? they were both already installed. Mind you, I don?t know how the person who installed it did it; maybe that?s related to the issues they were seeing? Regardless, the Fusion interface seems to read all of the correct info, including when things are changed in the XML files (which is how I was encouraged to do changes?). Now, I?m confused ? Fusion keeps its own copy of the config files? It?s not just a wrapper that modifies the existing freeswitch ones? (Just going off what I was told?) > 2) Ubuntu install is the same as debian https://freeswitch.org/confluence/display/FREESWITCH/Debian if you are going with fusionpbx you would be better off with one of their scripts. I looked at their script, but it didn?t seem to have been updated since 2012, which makes me a bit nervous. I?ll take another peek, though. > 3) If you are staying with native freeswitch, you can copy over almost all configs to the new server. What version are you migrating from? The current server is running 1.3.0. Thanks! -- Allison Walters IT Coordinator CAWST - Centre for Affordable Water and Sanitation Technology 424 Aviation Road NE, Calgary, Alberta, T2E 8H6, Canada tel: +1-403-243-3285 ext.264 web: www.cawst.org Join us in reaching 20 million people by 2020 with safe drinking water and sanitation. From: Allison Walters > Date: Mon, Dec 1, 2014 at 3:50 PM Subject: [Freeswitch-users] Questions before starting new install To: "freeswitch-users at lists.freeswitch.org" > We?re wanting to replace our current freeswitch server (it?s fairly old), and I have a few things I?m curious about before I start trying to build the new one. I inherited the current one and am relatively unfamiliar to all of this, so I appreciate your patience. 1. I was told by the consultant that set up the current box that freeswitch doesn?t play well with fusionpbx or really any of the GUI?s ? there have been instances (and indeed one happened a couple of months ago while I was here) of the config files getting randomly overwritten with old settings, which supposedly is the fault of trying to run fusion. Also I was told it doesn?t provide nice clean XML, etc? I was strongly encouraged to go with no GUI. If it were just me, I?d be okay with that, but we have a couple of other people who might be helping to maintain the server and they?re far more comfortable with GUIs. Has anyone else had any of these issues, and/or have any recommendations on the subject? 2. I?m trying to install on Ubuntu. When I go to the wiki, I get this page: https://freeswitch.org/confluence/display/FREESWITCH/Linux , which just gives an apt-get line for Ubuntu. However, that line just seems to have pre-reqs and not freeswitch itself. Anyone know where I could find current Ubuntu instructions? I googled but the most recent pages I could find were from 2012, and most were from 2011. Is that the most up to date? 3. Are there any guidelines for migration from an old server running an old version to a new server with a current version? Thank you! -- Allison Walters IT Coordinator CAWST - Centre for Affordable Water and Sanitation Technology 424 Aviation Road NE, Calgary, Alberta, T2E 8H6, Canada tel: +1-403-243-3285 ext.264 web: www.cawst.org Join us in reaching 20 million people by 2020 with safe drinking water and sanitation. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141202/9beb6e04/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 2424 bytes Desc: image001.jpg Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141202/9beb6e04/attachment-0001.jpg From nneul at mst.edu Tue Dec 2 21:48:05 2014 From: nneul at mst.edu (Nathan Neulinger) Date: Tue, 02 Dec 2014 12:48:05 -0600 Subject: [Freeswitch-users] Questions before starting new install In-Reply-To: References: <20b779910393484881146823fb74258f@BN1PR0101MB0817.prod.exchangelabs.com> <78cb8a4d983743bc94c0f63a732a1c10@BN1PR0101MB0817.prod.exchangelabs.com> Message-ID: <547E0965.6050909@mst.edu> Do you have git installed on your system? Sure looks like it's complaining that you don't have it installed. -- Nathan On 12/02/2014 12:40 PM, Allison Walters wrote: > I tried that version of the script and it says it was written for 10.04 and 12.04, and it says it doesn?t always work > outside of that, but it?ll ?give it a go?. And then after a bit, it failed out with a git error: > > installing stable v1.4 of FreeSWITCH > > /usr/bin/time: cannot run /usr/bin/git: No such file or directory > > Command exited with non-zero status 127 > > 0.00user 0.00system 0:00.00elapsed 0%CPU (0avgtext+0avgdata 1220maxresident)k > > 0inputs+0outputs (0major+27minor)pagefaults 0swaps > > ./install_fusionpbx.sh: line 1114: cd: /usr/src/freeswitch: No such file or directory > > ./install_fusionpbx.sh: line 1115: /usr/bin/git: No such file or directory > > GIT ERROR > > I suspect this is because freeswitch moved to stash in July, apparently. Anyway, I?ll see if I can find anyone on the > Fusion forum or IRC or somewhere that can help with this part. Thanks! > > -- > > *Allison Walters* > > /IT Coordinator/ > > CAWST - Centre for Affordable Water and Sanitation Technology > > 424 Aviation Road NE, Calgary, Alberta, T2E 8H6, Canada > > tel: +1-403-243-3285 ext.264 > > web: www.cawst.org > > *//* > > */Join us in reaching 20 million people by 2020 with safe drinking water and sanitation. /* > > *From:*freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf > Of *Moishe Grunstein > *Sent:* Monday, December 01, 2014 9:37 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Questions before starting new install > > 1)FusionPBX renames Freeswitch default configs after installation, it uses and then creates its own XML files (unless > you use their XML handler which writes to db instead). If you then modify XML files directly, the next time you make a > change in the GUI they will overwrite your changes. They have an IRC channel #FusionPBX and they use googlecode for svn > and issue tracker.. > > 2)Not sure what scripts you are referring to be dated, FusionPBX scripts have been last modified this month > http://fusionpbx.googlecode.com/svn/trunk/scripts/install/ubuntu/install_fusionpbx.sh > > Thanks, > > Moishe Grunstein > > Tornado Computer Systems, Inc. > > 212.400.7650 888.IPPBX.US > *Service Request Email: support at nysolutions.com * > > Polycom Certified VAR > Microsoft Small Business Specialist, Cisco SMB Select Certified > > cid:image001.jpg at 01C72F94.9EE45D60 > > Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * > Disaster Recovery * Network Security * Site Surveys * CMS > > *From:*freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Allison Walters > *Sent:* Monday, December 1, 2014 8:51 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Questions before starting new install > > Whoops ? accidentally had mail delivery disabled. So I?ll respond to Moishe?s points with a manual reply/quote: > >> 1) You cannot migrate a working freeswitch to FusionPBX. You would need to recreate all the configs in the FusionPBX web interface. > > I didn?t change anything ? they were both already installed. Mind you, I don?t know how the person who installed it did > it; maybe that?s related to the issues they were seeing? Regardless, the Fusion interface seems to read all of the > correct info, including when things are changed in the XML files (which is how I was encouraged to do changes?). Now, > I?m confused ? Fusion keeps its own copy of the config files? It?s not just a wrapper that modifies the existing > freeswitch ones? (Just going off what I was told?) > >> 2) Ubuntu install is the same as debian https://freeswitch.org/confluence/display/FREESWITCH/Debian if you are going with fusionpbx you would be better off > with one of their scripts. > > I looked at their script, but it didn?t seem to have been updated since 2012, which makes me a bit nervous. I?ll take > another peek, though. > >> 3) If you are staying with native freeswitch, you can copy over almost all configs to the new server. What version are you migrating from? > > The current server is running 1.3.0. > > Thanks! > > -- > > *Allison Walters* > > /IT Coordinator/ > > CAWST - Centre for Affordable Water and Sanitation Technology > > 424 Aviation Road NE, Calgary, Alberta, T2E 8H6, Canada > > tel: +1-403-243-3285 ext.264 > > web: www.cawst.org > > *//* > > */Join us in reaching 20 million people by 2020 with safe drinking water and sanitation. /* > > From: *Allison Walters* > > Date: Mon, Dec 1, 2014 at 3:50 PM > Subject: [Freeswitch-users] Questions before starting new install > To: "freeswitch-users at lists.freeswitch.org " > > > > We?re wanting to replace our current freeswitch server (it?s fairly old), and I have a few things I?m curious about > before I start trying to build the new one. I inherited the current one and am relatively unfamiliar to all of this, so > I appreciate your patience. > > 1.I was told by the consultant that set up the current box that freeswitch doesn?t play well with fusionpbx or really > any of the GUI?s ? there have been instances (and indeed one happened a couple of months ago while I was here) of the > config files getting randomly overwritten with old settings, which supposedly is the fault of trying to run fusion. > Also I was told it doesn?t provide nice clean XML, etc? I was strongly encouraged to go with no GUI. If it were just > me, I?d be okay with that, but we have a couple of other people who might be helping to maintain the server and they?re > far more comfortable with GUIs. Has anyone else had any of these issues, and/or have any recommendations on the subject? > > 2.I?m trying to install on Ubuntu. When I go to the wiki, I get this page: > https://freeswitch.org/confluence/display/FREESWITCH/Linux , which just gives an apt-get line for Ubuntu. However, that > line just seems to have pre-reqs and not freeswitch itself. Anyone know where I could find current Ubuntu > instructions? I googled but the most recent pages I could find were from 2012, and most were from 2011. Is that the > most up to date? > > 3.Are there any guidelines for migration from an old server running an old version to a new server with a current version? > > Thank you! > > -- > > *Allison Walters* > > /IT Coordinator/ > > CAWST - Centre for Affordable Water and Sanitation Technology > > 424 Aviation Road NE, Calgary, Alberta, T2E 8H6, Canada > > tel: +1-403-243-3285 ext.264 > > web: www.cawst.org > > *//* > > */Join us in reaching 20 million people by 2020 with safe drinking water and sanitation. /* > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From AWalters at cawst.org Tue Dec 2 21:55:34 2014 From: AWalters at cawst.org (Allison Walters) Date: Tue, 2 Dec 2014 18:55:34 +0000 Subject: [Freeswitch-users] Questions before starting new install In-Reply-To: <547E0965.6050909@mst.edu> References: <20b779910393484881146823fb74258f@BN1PR0101MB0817.prod.exchangelabs.com> <78cb8a4d983743bc94c0f63a732a1c10@BN1PR0101MB0817.prod.exchangelabs.com> <547E0965.6050909@mst.edu> Message-ID: *facepalm* Sorry - I was installing on the wrong test VM. Ugh....long morning. I was kind of prepared for it to not work after the warning message so jumped to wrong conclusion. :-( My bad! Time to grab some food and come back to this, I think... -- Allison Walters IT Coordinator CAWST - Centre for Affordable Water and Sanitation Technology 424 Aviation Road NE, Calgary, Alberta, T2E 8H6, Canada tel: +1-403-243-3285 ext.264 web:?www.cawst.org ? Join us in reaching 20 million people by 2020 with safe drinking water and sanitation.? -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nathan Neulinger Sent: Tuesday, December 02, 2014 11:48 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Questions before starting new install Do you have git installed on your system? Sure looks like it's complaining that you don't have it installed. -- Nathan On 12/02/2014 12:40 PM, Allison Walters wrote: > I tried that version of the script and it says it was written for > 10.04 and 12.04, and it says it doesn't always work outside of that, but it'll "give it a go". And then after a bit, it failed out with a git error: > > installing stable v1.4 of FreeSWITCH > > /usr/bin/time: cannot run /usr/bin/git: No such file or directory > > Command exited with non-zero status 127 > > 0.00user 0.00system 0:00.00elapsed 0%CPU (0avgtext+0avgdata > 1220maxresident)k > > 0inputs+0outputs (0major+27minor)pagefaults 0swaps > > ./install_fusionpbx.sh: line 1114: cd: /usr/src/freeswitch: No such > file or directory > > ./install_fusionpbx.sh: line 1115: /usr/bin/git: No such file or > directory > > GIT ERROR > > I suspect this is because freeswitch moved to stash in July, > apparently. Anyway, I'll see if I can find anyone on the Fusion forum or IRC or somewhere that can help with this part. Thanks! > > -- > > *Allison Walters* > > /IT Coordinator/ > > CAWST - Centre for Affordable Water and Sanitation Technology > > 424 Aviation Road NE, Calgary, Alberta, T2E 8H6, Canada > > tel: +1-403-243-3285 ext.264 > > web: www.cawst.org > > *//* > > */Join us in reaching 20 million people by 2020 with safe drinking > water and sanitation. /* > > *From:*freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of > *Moishe Grunstein > *Sent:* Monday, December 01, 2014 9:37 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Questions before starting new > install > > 1)FusionPBX renames Freeswitch default configs after installation, it > uses and then creates its own XML files (unless you use their XML > handler which writes to db instead). If you then modify XML files > directly, the next time you make a change in the GUI they will overwrite your changes. They have an IRC channel #FusionPBX and they use googlecode for svn and issue tracker.. > > 2)Not sure what scripts you are referring to be dated, FusionPBX > scripts have been last modified this month > http://fusionpbx.googlecode.com/svn/trunk/scripts/install/ubuntu/insta > ll_fusionpbx.sh > > Thanks, > > Moishe Grunstein > > Tornado Computer Systems, Inc. > > 212.400.7650 888.IPPBX.US > *Service Request Email: support at nysolutions.com > * > > Polycom Certified VAR > Microsoft Small Business Specialist, Cisco SMB Select Certified > > cid:image001.jpg at 01C72F94.9EE45D60 > > Computer Networking * Managed Services * IP Video Surveillance * > Network Assessments * Web Solutions * Voice over IP * Disaster > Recovery * Network Security * Site Surveys * CMS > > *From:*freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of > *Allison Walters > *Sent:* Monday, December 1, 2014 8:51 PM > *To:* freeswitch-users at lists.freeswitch.org > > *Subject:* Re: [Freeswitch-users] Questions before starting new > install > > Whoops - accidentally had mail delivery disabled. So I'll respond to Moishe's points with a manual reply/quote: > >> 1) You cannot migrate a working freeswitch to FusionPBX. You would need to recreate all the configs in the FusionPBX web interface. > > I didn't change anything - they were both already installed. Mind > you, I don't know how the person who installed it did it; maybe that's > related to the issues they were seeing? Regardless, the Fusion > interface seems to read all of the correct info, including when things > are changed in the XML files (which is how I was encouraged to do > changes.). Now, I'm confused - Fusion keeps its own copy of the > config files? It's not just a wrapper that modifies the existing > freeswitch ones? (Just going off what I was told.) > >> 2) Ubuntu install is the same as debian https://freeswitch.org/confluence/display/FREESWITCH/Debian if you are going with fusionpbx you would be better off > with one of their scripts. > > I looked at their script, but it didn't seem to have been updated > since 2012, which makes me a bit nervous. I'll take another peek, though. > >> 3) If you are staying with native freeswitch, you can copy over almost all configs to the new server. What version are you migrating from? > > The current server is running 1.3.0. > > Thanks! > > -- > > *Allison Walters* > > /IT Coordinator/ > > CAWST - Centre for Affordable Water and Sanitation Technology > > 424 Aviation Road NE, Calgary, Alberta, T2E 8H6, Canada > > tel: +1-403-243-3285 ext.264 > > web: www.cawst.org > > *//* > > */Join us in reaching 20 million people by 2020 with safe drinking > water and sanitation. /* > > From: *Allison Walters* > > Date: Mon, Dec 1, 2014 at 3:50 PM > Subject: [Freeswitch-users] Questions before starting new install > To: "freeswitch-users at lists.freeswitch.org " > > > > We're wanting to replace our current freeswitch server (it's fairly > old), and I have a few things I'm curious about before I start trying > to build the new one. I inherited the current one and am relatively unfamiliar to all of this, so I appreciate your patience. > > 1.I was told by the consultant that set up the current box that > freeswitch doesn't play well with fusionpbx or really any of the GUI's > - there have been instances (and indeed one happened a couple of months ago while I was here) of the config files getting randomly overwritten with old settings, which supposedly is the fault of trying to run fusion. > Also I was told it doesn't provide nice clean XML, etc. I was > strongly encouraged to go with no GUI. If it were just me, I'd be > okay with that, but we have a couple of other people who might be helping to maintain the server and they're far more comfortable with GUIs. Has anyone else had any of these issues, and/or have any recommendations on the subject? > > 2.I'm trying to install on Ubuntu. When I go to the wiki, I get this page: > https://freeswitch.org/confluence/display/FREESWITCH/Linux , which > just gives an apt-get line for Ubuntu. However, that line just seems > to have pre-reqs and not freeswitch itself. Anyone know where I could > find current Ubuntu instructions? I googled but the most recent pages I could find were from 2012, and most were from 2011. Is that the most up to date? > > 3.Are there any guidelines for migration from an old server running an old version to a new server with a current version? > > Thank you! > > -- > > *Allison Walters* > > /IT Coordinator/ > > CAWST - Centre for Affordable Water and Sanitation Technology > > 424 Aviation Road NE, Calgary, Alberta, T2E 8H6, Canada > > tel: +1-403-243-3285 ext.264 > > web: www.cawst.org > > *//* > > */Join us in reaching 20 million people by 2020 with safe drinking > water and sanitation. /* > > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users 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://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users 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 > -- ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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 aademattia at comcast.net Tue Dec 2 23:25:20 2014 From: aademattia at comcast.net (=?utf-8?B?YWFkZW1hdHRpYUBjb21jYXN0Lm5ldA==?=) Date: Tue, 02 Dec 2014 15:25:20 -0500 Subject: [Freeswitch-users] =?utf-8?q?Send=5FDTMF?= Message-ID: After looking at another sdk and wireshark it looks like the only difference is Marker:false not working but on a working dtmf it's true. Not sure if this means anything. Sent from my HTC ----- Reply message ----- From: "aademattia at comcast.net" To: "FreeSWITCH Users Help" Subject: [Freeswitch-users]Send_DTMF Date: Tue, Dec 2, 2014 10:04 AM I still see the dtmf going to the wrong destination. I am sending dtmf on both uuid and can't seem to get the ivr I am calling to see my dtmf. But if I hit dtmf on my phone it works. Sent from my HTC ----- Reply message ----- From: "Seven Du" To: "FreeSWITCH Users Help" Subject: [Freeswitch-users] Send_DTMF Date: Tue, Dec 2, 2014 8:52 AM try uuid_recv_dtmf -- Seven Du http://about.me/dujinfang http://www.dujinfang.com http://www.freeswitch.org.cn Sent with Sparrow On Tuesday, December 2, 2014 at 9:37 AM, Andrew wrote: Hi,I am making an outbound call then I am doing a bridge.Once I bridge the call I need to send dtmf to second call (bridge call)When doing a trace it looks like the DTMF is going to my first call and not the bridge call.I am using UUID send dtmf using both UUID but both don?t seem to work. I do see the DTMF in wire shark but again looks like its going the wrong direction. Andrew _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch..org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141202/ee17c67a/attachment.html From AWalters at cawst.org Tue Dec 2 23:34:40 2014 From: AWalters at cawst.org (Allison Walters) Date: Tue, 2 Dec 2014 20:34:40 +0000 Subject: [Freeswitch-users] Questions before starting new install In-Reply-To: <547E0965.6050909@mst.edu> References: <20b779910393484881146823fb74258f@BN1PR0101MB0817.prod.exchangelabs.com> <78cb8a4d983743bc94c0f63a732a1c10@BN1PR0101MB0817.prod.exchangelabs.com> <547E0965.6050909@mst.edu> Message-ID: Okay, this half-asleep mistake rectified, I ran it on the right system and got further. However, partway through the make it exited, complaining about various missing pre-reqs. (The beginning of this script claims to check for prereqs but it only seems to check for 4 or so, and I've had to re-run a bunch of other times for lots it apparently doesn't check for). Now after installing a bunch of the things it asked for, I ran into one that apparently I have to basically reset the entire installation process before it will recognize that it's there (libmemcached-dev). I tried deleting the /usr/src/freeswitch dir, which made it re-fetch everything from git. Sadly, it didn't force it to reconfigure, and now I'm getting an error " make: *** No targets specified and no makefile found. Stop." Anyone have any ideas how I can force the entire process to start over? Thanks, and sorry for the questions - this is just getting frustrating. (Removing my own bone-headed mistake from the pile, I've still been at this for 3 hours or so now...) -- Allison Walters IT Coordinator CAWST - Centre for Affordable Water and Sanitation Technology 424 Aviation Road NE, Calgary, Alberta, T2E 8H6, Canada tel: +1-403-243-3285 ext.264 web:?www.cawst.org ? Join us in reaching 20 million people by 2020 with safe drinking water and sanitation.? -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nathan Neulinger Sent: Tuesday, December 02, 2014 11:48 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Questions before starting new install Do you have git installed on your system? Sure looks like it's complaining that you don't have it installed. -- Nathan On 12/02/2014 12:40 PM, Allison Walters wrote: > I tried that version of the script and it says it was written for > 10.04 and 12.04, and it says it doesn't always work outside of that, but it'll "give it a go". And then after a bit, it failed out with a git error: > > installing stable v1.4 of FreeSWITCH > > /usr/bin/time: cannot run /usr/bin/git: No such file or directory > > Command exited with non-zero status 127 > > 0.00user 0.00system 0:00.00elapsed 0%CPU (0avgtext+0avgdata > 1220maxresident)k > > 0inputs+0outputs (0major+27minor)pagefaults 0swaps > > ./install_fusionpbx.sh: line 1114: cd: /usr/src/freeswitch: No such > file or directory > > ./install_fusionpbx.sh: line 1115: /usr/bin/git: No such file or > directory > > GIT ERROR > > I suspect this is because freeswitch moved to stash in July, > apparently. Anyway, I'll see if I can find anyone on the Fusion forum or IRC or somewhere that can help with this part. Thanks! > > -- > > *Allison Walters* > > /IT Coordinator/ > > CAWST - Centre for Affordable Water and Sanitation Technology > > 424 Aviation Road NE, Calgary, Alberta, T2E 8H6, Canada > > tel: +1-403-243-3285 ext.264 > > web: www.cawst.org > > *//* > > */Join us in reaching 20 million people by 2020 with safe drinking > water and sanitation. /* > > *From:*freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of > *Moishe Grunstein > *Sent:* Monday, December 01, 2014 9:37 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Questions before starting new > install > > 1)FusionPBX renames Freeswitch default configs after installation, it > uses and then creates its own XML files (unless you use their XML > handler which writes to db instead). If you then modify XML files > directly, the next time you make a change in the GUI they will overwrite your changes. They have an IRC channel #FusionPBX and they use googlecode for svn and issue tracker.. > > 2)Not sure what scripts you are referring to be dated, FusionPBX > scripts have been last modified this month > http://fusionpbx.googlecode.com/svn/trunk/scripts/install/ubuntu/insta > ll_fusionpbx.sh > > Thanks, > > Moishe Grunstein > > Tornado Computer Systems, Inc. > > 212.400.7650 888.IPPBX.US > *Service Request Email: support at nysolutions.com > * > > Polycom Certified VAR > Microsoft Small Business Specialist, Cisco SMB Select Certified > > cid:image001.jpg at 01C72F94.9EE45D60 > > Computer Networking * Managed Services * IP Video Surveillance * > Network Assessments * Web Solutions * Voice over IP * Disaster > Recovery * Network Security * Site Surveys * CMS > > *From:*freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of > *Allison Walters > *Sent:* Monday, December 1, 2014 8:51 PM > *To:* freeswitch-users at lists.freeswitch.org > > *Subject:* Re: [Freeswitch-users] Questions before starting new > install > > Whoops - accidentally had mail delivery disabled. So I'll respond to Moishe's points with a manual reply/quote: > >> 1) You cannot migrate a working freeswitch to FusionPBX. You would need to recreate all the configs in the FusionPBX web interface. > > I didn't change anything - they were both already installed. Mind > you, I don't know how the person who installed it did it; maybe that's > related to the issues they were seeing? Regardless, the Fusion > interface seems to read all of the correct info, including when things > are changed in the XML files (which is how I was encouraged to do > changes.). Now, I'm confused - Fusion keeps its own copy of the > config files? It's not just a wrapper that modifies the existing > freeswitch ones? (Just going off what I was told.) > >> 2) Ubuntu install is the same as debian https://freeswitch.org/confluence/display/FREESWITCH/Debian if you are going with fusionpbx you would be better off > with one of their scripts. > > I looked at their script, but it didn't seem to have been updated > since 2012, which makes me a bit nervous. I'll take another peek, though. > >> 3) If you are staying with native freeswitch, you can copy over almost all configs to the new server. What version are you migrating from? > > The current server is running 1.3.0. > > Thanks! > > -- > > *Allison Walters* > > /IT Coordinator/ > > CAWST - Centre for Affordable Water and Sanitation Technology > > 424 Aviation Road NE, Calgary, Alberta, T2E 8H6, Canada > > tel: +1-403-243-3285 ext.264 > > web: www.cawst.org > > *//* > > */Join us in reaching 20 million people by 2020 with safe drinking > water and sanitation. /* > > From: *Allison Walters* > > Date: Mon, Dec 1, 2014 at 3:50 PM > Subject: [Freeswitch-users] Questions before starting new install > To: "freeswitch-users at lists.freeswitch.org " > > > > We're wanting to replace our current freeswitch server (it's fairly > old), and I have a few things I'm curious about before I start trying > to build the new one. I inherited the current one and am relatively unfamiliar to all of this, so I appreciate your patience. > > 1.I was told by the consultant that set up the current box that > freeswitch doesn't play well with fusionpbx or really any of the GUI's > - there have been instances (and indeed one happened a couple of months ago while I was here) of the config files getting randomly overwritten with old settings, which supposedly is the fault of trying to run fusion. > Also I was told it doesn't provide nice clean XML, etc. I was > strongly encouraged to go with no GUI. If it were just me, I'd be > okay with that, but we have a couple of other people who might be helping to maintain the server and they're far more comfortable with GUIs. Has anyone else had any of these issues, and/or have any recommendations on the subject? > > 2.I'm trying to install on Ubuntu. When I go to the wiki, I get this page: > https://freeswitch.org/confluence/display/FREESWITCH/Linux , which > just gives an apt-get line for Ubuntu. However, that line just seems > to have pre-reqs and not freeswitch itself. Anyone know where I could > find current Ubuntu instructions? I googled but the most recent pages I could find were from 2012, and most were from 2011. Is that the most up to date? > > 3.Are there any guidelines for migration from an old server running an old version to a new server with a current version? > > Thank you! > > -- > > *Allison Walters* > > /IT Coordinator/ > > CAWST - Centre for Affordable Water and Sanitation Technology > > 424 Aviation Road NE, Calgary, Alberta, T2E 8H6, Canada > > tel: +1-403-243-3285 ext.264 > > web: www.cawst.org > > *//* > > */Join us in reaching 20 million people by 2020 with safe drinking > water and sanitation. /* > > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users 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://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users 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 > -- ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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 ssinyagin at gmail.com Wed Dec 3 02:30:18 2014 From: ssinyagin at gmail.com (Stanislav Sinyagin) Date: Wed, 3 Dec 2014 00:30:18 +0100 Subject: [Freeswitch-users] UniFi phones and FreeSWITCH In-Reply-To: References: Message-ID: Our local reseller expects them only by end of January. But I guess there shouldn't be anything too bad in regards to compatibility :) On Dec 2, 2014 4:59 PM, "Michael Collins" wrote: > Hello FreeSWITCHers! > > Does anybody have any experience with these phones? > http://www.ubnt.com/unifi-voip/uvp/ > > I'm curious to know how they work in a FS environment. > > -MC > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141203/5ea4b837/attachment.html From AHadjiefstathiou at IronFX.com Wed Dec 3 18:11:06 2014 From: AHadjiefstathiou at IronFX.com (Andri Hadjiefstathiou [IronFX Global]) Date: Wed, 3 Dec 2014 15:11:06 +0000 Subject: [Freeswitch-users] Freeswitch Message-ID: I got those errors recently from our freeswitch server [ERR] mod_commands.c:4063 Error connecting [ERR] sofia_glue.c:6464 Error connecting [ERR] sofia_glue.c:6504 Error Opening DB [WARNING] sofia_glue.c:6464 Max handles 50 What does this means? I have load recently the mod_xml_rpc is this relate with our issue? Also how I can check the correct open DB open connections ? Andri Hadjiefstathiou | Network and Security Engineer t +357 25027607 | m +357 96860600 | e AHadjiefstathiou at IronFX.com IronFX Global Limited t +357 25027000 | f +357 25027001 | w www.IronFX.com [cid:image001.jpg at 01D00F1C.1FC46550] [cid:image002.gif at 01D00F1C.1FC46550] [cid:image003.gif at 01D00F1C.1FC46550] [cid:image004.gif at 01D00F1C.1FC46550] [cid:image005.gif at 01D00F1C.1FC46550] [cid:image006.gif at 01D00F1C.1FC46550] London | Hong Kong | Sydney | Shanghai | Shenzhen | Shenyang | Frankfurt | Madrid | Warsaw | Lisbon | Prague | Budapest | Bucharest | Athens | Kiev | Minsk | Moscow | St Petersburg | Mumbai | Jakarta | Manila | Ho Chi Minh City | Kuala Lumpur | Baku | Lagos | Johannesburg | Auckland | Sao Paulo | Buenos Aires | Lima | Santiago | Montevideo | Limassol [cid:image007.jpg at 01D00F1C.1FC46550] ----------------------------------------------------------------------------------------------------------------------- IronFX Global UK Limited is authorised and regulated by the Financial Conduct Authority (FCA no. 585561) IronFX Global (Australia) Pty Limited is authorised and regulated by ASIC (AFSL no. 417482) IronFX Global (South Africa) (Pty) Ltd is authorized by the Financial Services Board (FSP No 45276) IronFX Global NZ Limited is authorised and regulated by FSP (FSPR no. 298966) IronFX Global (Russia) LLC is a member of CRFIN (Membership no. A-8) IronFX Global (Ukraine) LLC is a member of UCRFIN (Membership no. 5) IronFX Global Limited is authorised and regulated by CySEC (Licence no. 125/10) IronFX Global Limited is a Member of Eurex Exchange ----------------------------------------------------------------------------------------------------------------------- This email has been sent from IronFX Global Limited. The information in this email is confidential and may be legally privileged. It is intended solely for the addressee. Access to this email by anyone else is unauthorised. If you are not the intended recipient, any disclosure, copying, distribution or any action taken or omitted to be taken in reliance on it, is prohibited and may be unlawful. Consider the environment before printing this email. -------------- next part -------------- An HTML attachment was scrubbed... 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Name: image004.gif Type: image/gif Size: 590 bytes Desc: image004.gif Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141203/dc6fcbbe/attachment-0007.gif -------------- next part -------------- A non-text attachment was scrubbed... Name: image005.gif Type: image/gif Size: 1054 bytes Desc: image005.gif Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141203/dc6fcbbe/attachment-0008.gif -------------- next part -------------- A non-text attachment was scrubbed... Name: image006.gif Type: image/gif Size: 491 bytes Desc: image006.gif Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141203/dc6fcbbe/attachment-0009.gif -------------- next part -------------- A non-text attachment was scrubbed... Name: image007.jpg Type: image/jpeg Size: 16451 bytes Desc: image007.jpg Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141203/dc6fcbbe/attachment-0003.jpg From AHadjiefstathiou at IronFX.com Wed Dec 3 18:39:56 2014 From: AHadjiefstathiou at IronFX.com (Andri Hadjiefstathiou [IronFX Global]) Date: Wed, 3 Dec 2014 15:39:56 +0000 Subject: [Freeswitch-users] Freeswitch In-Reply-To: References: Message-ID: I got those errors recently from our freeswitch server 2014-11-28 21:19:19.806910 [ERR] sofia_glue.c:6376 Error connecting 2014-11-28 21:19:19.806910 [ERR] sofia_glue.c:6416 Error Opening DB 2014-11-28 21:19:19.806910 [WARNING] sofia_glue.c:6376 Max handles 50 exceeded, blocking.... 2014-11-28 21:19:21.476882 [ERR] sofia_glue.c:6376 Error connecting 2014-11-28 21:19:21.476882 [ERR] sofia_glue.c:6416 Error Opening DB 2014-11-28 21:19:21.476882 [WARNING] sofia_glue.c:6376 Max handles 50 exceeded, blocking.... What does this means? I have load recently the mod_xml_rpc is this relate with our issue? Also how I can check the correct open DB open connections ? Andri Hadjiefstathiou | Network and Security Engineer t +357 25027607 | m +357 96860600 | e AHadjiefstathiou at IronFX.com IronFX Global Limited t +357 25027000 | f +357 25027001 | w www.IronFX.com [cid:image001.jpg at 01D00F20.26E275D0] [cid:image002.gif at 01D00F20.26E275D0] [cid:image003.gif at 01D00F20.26E275D0] [cid:image004.gif at 01D00F20.26E275D0] [cid:image005.gif at 01D00F20.26E275D0] [cid:image006.gif at 01D00F20.26E275D0] London | Hong Kong | Sydney | Shanghai | Shenzhen | Shenyang | Frankfurt | Madrid | Warsaw | Lisbon | Prague | Budapest | Bucharest | Athens | Kiev | Minsk | Moscow | St Petersburg | Mumbai | Jakarta | Manila | Ho Chi Minh City | Kuala Lumpur | Baku | Lagos | Johannesburg | Auckland | Sao Paulo | Buenos Aires | Lima | Santiago | Montevideo | Limassol [cid:image007.jpg at 01D00F20.26E275D0] ----------------------------------------------------------------------------------------------------------------------- IronFX Global UK Limited is authorised and regulated by the Financial Conduct Authority (FCA no. 585561) IronFX Global (Australia) Pty Limited is authorised and regulated by ASIC (AFSL no. 417482) IronFX Global (South Africa) (Pty) Ltd is authorized by the Financial Services Board (FSP No 45276) IronFX Global NZ Limited is authorised and regulated by FSP (FSPR no. 298966) IronFX Global (Russia) LLC is a member of CRFIN (Membership no. A-8) IronFX Global (Ukraine) LLC is a member of UCRFIN (Membership no. 5) IronFX Global Limited is authorised and regulated by CySEC (Licence no. 125/10) IronFX Global Limited is a Member of Eurex Exchange ----------------------------------------------------------------------------------------------------------------------- Consider the environment before printing this email. From: Andri Hadjiefstathiou [IronFX Global] Sent: Wednesday, December 03, 2014 5:11 PM To: 'freeswitch-users at lists.freeswitch.org' Subject: Freeswitch I got those errors recently from our freeswitch server [ERR] mod_commands.c:4063 Error connecting [ERR] sofia_glue.c:6464 Error connecting [ERR] sofia_glue.c:6504 Error Opening DB [WARNING] sofia_glue.c:6464 Max handles 50 What does this means? I have load recently the mod_xml_rpc is this relate with our issue? Also how I can check the correct open DB open connections ? Andri Hadjiefstathiou | Network and Security Engineer t +357 25027607 | m +357 96860600 | e AHadjiefstathiou at IronFX.com IronFX Global Limited t +357 25027000 | f +357 25027001 | w www.IronFX.com [cid:image001.jpg at 01D00F20.26E275D0] [cid:image002.gif at 01D00F20.26E275D0] [cid:image003.gif at 01D00F20.26E275D0] [cid:image004.gif at 01D00F20.26E275D0] [cid:image005.gif at 01D00F20.26E275D0] [cid:image006.gif at 01D00F20.26E275D0] London | Hong Kong | Sydney | Shanghai | Shenzhen | Shenyang | Frankfurt | Madrid | Warsaw | Lisbon | Prague | Budapest | Bucharest | Athens | Kiev | Minsk | Moscow | St Petersburg | Mumbai | Jakarta | Manila | Ho Chi Minh City | Kuala Lumpur | Baku | Lagos | Johannesburg | Auckland | Sao Paulo | Buenos Aires | Lima | Santiago | Montevideo | Limassol [cid:image007.jpg at 01D00F20.26E275D0] ----------------------------------------------------------------------------------------------------------------------- IronFX Global UK Limited is authorised and regulated by the Financial Conduct Authority (FCA no. 585561) IronFX Global (Australia) Pty Limited is authorised and regulated by ASIC (AFSL no. 417482) IronFX Global (South Africa) (Pty) Ltd is authorized by the Financial Services Board (FSP No 45276) IronFX Global NZ Limited is authorised and regulated by FSP (FSPR no. 298966) IronFX Global (Russia) LLC is a member of CRFIN (Membership no. A-8) IronFX Global (Ukraine) LLC is a member of UCRFIN (Membership no. 5) IronFX Global Limited is authorised and regulated by CySEC (Licence no. 125/10) IronFX Global Limited is a Member of Eurex Exchange ----------------------------------------------------------------------------------------------------------------------- This email has been sent from IronFX Global Limited. The information in this email is confidential and may be legally privileged. It is intended solely for the addressee. Access to this email by anyone else is unauthorised. If you are not the intended recipient, any disclosure, copying, distribution or any action taken or omitted to be taken in reliance on it, is prohibited and may be unlawful. Consider the environment before printing this email. -------------- next part -------------- An HTML attachment was scrubbed... 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Name: image007.jpg Type: image/jpeg Size: 16451 bytes Desc: image007.jpg Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141203/35159eb7/attachment-0003.jpg From cmrienzo at gmail.com Wed Dec 3 18:59:51 2014 From: cmrienzo at gmail.com (Christopher Rienzo) Date: Wed, 3 Dec 2014 10:59:51 -0500 Subject: [Freeswitch-users] Freeswitch In-Reply-To: References: Message-ID: Check your sofia profile(s)- it has the database settings in there (dbname, odbc-dsn, db-*) . Then, verify your database is running and can accept connections. On Wed, Dec 3, 2014 at 10:39 AM, Andri Hadjiefstathiou [IronFX Global] < AHadjiefstathiou at ironfx.com> wrote: > I got those errors recently from our freeswitch server > > > > 2014-11-28 21:19:19.806910 [ERR] sofia_glue.c:6376 Error connecting > > 2014-11-28 21:19:19.806910 [ERR] sofia_glue.c:6416 Error Opening DB > > 2014-11-28 21:19:19.806910 [WARNING] sofia_glue.c:6376 Max handles 50 > exceeded, blocking.... > > 2014-11-28 21:19:21.476882 [ERR] sofia_glue.c:6376 Error connecting > > 2014-11-28 21:19:21.476882 [ERR] sofia_glue.c:6416 Error Opening DB > > 2014-11-28 21:19:21.476882 [WARNING] sofia_glue.c:6376 Max handles 50 > exceeded, blocking.... > > > > What does this means? > > > > I have load recently the mod_xml_rpc is this relate with our issue? > > Also how I can check the correct open DB open connections ? > > > > *Andri* *Hadjiefstathiou* | Network and Security Engineer > *t *+357 25027607 | *m *+357 96860600 | *e* AHadjiefstathiou at IronFX.com > > > *IronFX Global Limited t* +357 25027000 | *f* +357 25027001 | *w * > www.IronFX.com > > > > > > > *London | Hong Kong | Sydney | Shanghai | Shenzhen | Shenyang | Frankfurt > | Madrid | Warsaw | Lisbon | Prague | Budapest | Bucharest | Athens | Kiev > | Minsk | Moscow | St Petersburg | Mumbai | Jakarta | Manila | Ho Chi Minh > City | Kuala Lumpur | Baku | Lagos | Johannesburg | Auckland | Sao Paulo | > Buenos Aires | Lima | Santiago | Montevideo | Limassol* > > > ----------------------------------------------------------------------------------------------------------------------- > IronFX Global UK Limited is authorised and regulated by the Financial > Conduct Authority (FCA no. 585561) > IronFX Global (Australia) Pty Limited is authorised and regulated by ASIC > (AFSL no. 417482) > IronFX Global (South Africa) (Pty) Ltd is authorized by the Financial > Services Board (FSP No 45276) > IronFX Global NZ Limited is authorised and regulated by FSP (FSPR no. > 298966) > IronFX Global (Russia) LLC is a member of CRFIN (Membership no. A-8) > IronFX Global (Ukraine) LLC is a member of UCRFIN (Membership no. 5) > IronFX Global Limited is authorised and regulated by CySEC (Licence no. > 125/10) > > IronFX Global Limited is a Member of Eurex Exchange > > ----------------------------------------------------------------------------------------------------------------------- > > *Consider the environment before printing this email.* > > *From:* Andri Hadjiefstathiou [IronFX Global] > *Sent:* Wednesday, December 03, 2014 5:11 PM > *To:* 'freeswitch-users at lists.freeswitch.org' > *Subject:* Freeswitch > > > > I got those errors recently from our freeswitch server > > > > [ERR] mod_commands.c:4063 Error connecting > > [ERR] sofia_glue.c:6464 Error connecting > > [ERR] sofia_glue.c:6504 Error Opening DB > > [WARNING] sofia_glue.c:6464 Max handles 50 > > > > What does this means? > > > > I have load recently the mod_xml_rpc is this relate with our issue? > > Also how I can check the correct open DB open connections ? > > > > > > *Andri* *Hadjiefstathiou* | Network and Security Engineer > *t *+357 25027607 | *m *+357 96860600 | *e* AHadjiefstathiou at IronFX.com > > > *IronFX Global Limited t* +357 25027000 | *f* +357 25027001 | *w * > www.IronFX.com > > > > > > > *London | Hong Kong | Sydney | Shanghai | Shenzhen | Shenyang | Frankfurt > | Madrid | Warsaw | Lisbon | Prague | Budapest | Bucharest | Athens | Kiev > | Minsk | Moscow | St Petersburg | Mumbai | Jakarta | Manila | Ho Chi Minh > City | Kuala Lumpur | Baku | Lagos | Johannesburg | Auckland | Sao Paulo | > Buenos Aires | Lima | Santiago | Montevideo | Limassol* > > > ----------------------------------------------------------------------------------------------------------------------- > IronFX Global UK Limited is authorised and regulated by the Financial > Conduct Authority (FCA no. 585561) > IronFX Global (Australia) Pty Limited is authorised and regulated by ASIC > (AFSL no. 417482) > IronFX Global (South Africa) (Pty) Ltd is authorized by the Financial > Services Board (FSP No 45276) > IronFX Global NZ Limited is authorised and regulated by FSP (FSPR no. > 298966) > IronFX Global (Russia) LLC is a member of CRFIN (Membership no. A-8) > IronFX Global (Ukraine) LLC is a member of UCRFIN (Membership no. 5) > IronFX Global Limited is authorised and regulated by CySEC (Licence no. > 125/10) > > IronFX Global Limited is a Member of Eurex Exchange > > ----------------------------------------------------------------------------------------------------------------------- > > > > > *This email has been sent from IronFX Global Limited. The information in > this email is confidential and may be legally privileged. It is intended > solely for the addressee. Access to this email by anyone else is > unauthorised. If you are not the intended recipient, any disclosure, > copying, distribution or any action taken or omitted to be taken in > reliance on it, is prohibited and may be unlawful.* > > *Consider the environment before printing this email.* > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141203/6562cbac/attachment.html From AHadjiefstathiou at IronFX.com Wed Dec 3 19:03:36 2014 From: AHadjiefstathiou at IronFX.com (Andri Hadjiefstathiou [IronFX Global]) Date: Wed, 3 Dec 2014 16:03:36 +0000 Subject: [Freeswitch-users] Freeswitch In-Reply-To: References: Message-ID: After we restart the server all fine. But I want to know the root of the problem.. why we got this error??? Recently I have enable mod_xml_rpc is this relate with our issue? Andri Hadjiefstathiou | Network and Security Engineer t +357 25027607 | m +357 96860600 | e AHadjiefstathiou at IronFX.com IronFX Global Limited t +357 25027000 | f +357 25027001 | w www.IronFX.com [cid:image001.jpg at 01D00F23.751A8A50] [cid:image002.gif at 01D00F23.751A8A50] [cid:image003.gif at 01D00F23.751A8A50] [cid:image004.gif at 01D00F23.751A8A50] [cid:image005.gif at 01D00F23.751A8A50] [cid:image006.gif at 01D00F23.751A8A50] London | Hong Kong | Sydney | Shanghai | Shenzhen | Shenyang | Frankfurt | Madrid | Warsaw | Lisbon | Prague | Budapest | Bucharest | Athens | Kiev | Minsk | Moscow | St Petersburg | Mumbai | Jakarta | Manila | Ho Chi Minh City | Kuala Lumpur | Baku | Lagos | Johannesburg | Auckland | Sao Paulo | Buenos Aires | Lima | Santiago | Montevideo | Limassol [cid:image007.jpg at 01D00F23.751A8A50] ----------------------------------------------------------------------------------------------------------------------- IronFX Global UK Limited is authorised and regulated by the Financial Conduct Authority (FCA no. 585561) IronFX Global (Australia) Pty Limited is authorised and regulated by ASIC (AFSL no. 417482) IronFX Global (South Africa) (Pty) Ltd is authorized by the Financial Services Board (FSP No 45276) IronFX Global NZ Limited is authorised and regulated by FSP (FSPR no. 298966) IronFX Global (Russia) LLC is a member of CRFIN (Membership no. A-8) IronFX Global (Ukraine) LLC is a member of UCRFIN (Membership no. 5) IronFX Global Limited is authorised and regulated by CySEC (Licence no. 125/10) IronFX Global Limited is a Member of Eurex Exchange ----------------------------------------------------------------------------------------------------------------------- Consider the environment before printing this email. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Christopher Rienzo Sent: Wednesday, December 03, 2014 6:00 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Freeswitch Check your sofia profile(s)- it has the database settings in there (dbname, odbc-dsn, db-*) . Then, verify your database is running and can accept connections. On Wed, Dec 3, 2014 at 10:39 AM, Andri Hadjiefstathiou [IronFX Global] > wrote: I got those errors recently from our freeswitch server 2014-11-28 21:19:19.806910 [ERR] sofia_glue.c:6376 Error connecting 2014-11-28 21:19:19.806910 [ERR] sofia_glue.c:6416 Error Opening DB 2014-11-28 21:19:19.806910 [WARNING] sofia_glue.c:6376 Max handles 50 exceeded, blocking.... 2014-11-28 21:19:21.476882 [ERR] sofia_glue.c:6376 Error connecting 2014-11-28 21:19:21.476882 [ERR] sofia_glue.c:6416 Error Opening DB 2014-11-28 21:19:21.476882 [WARNING] sofia_glue.c:6376 Max handles 50 exceeded, blocking.... What does this means? I have load recently the mod_xml_rpc is this relate with our issue? Also how I can check the correct open DB open connections ? Andri Hadjiefstathiou | Network and Security Engineer t +357 25027607 | m +357 96860600 | e AHadjiefstathiou at IronFX.com IronFX Global Limited t +357 25027000 | f +357 25027001 | w www.IronFX.com London | Hong Kong | Sydney | Shanghai | Shenzhen | Shenyang | Frankfurt | Madrid | Warsaw | Lisbon | Prague | Budapest | Bucharest | Athens | Kiev | Minsk | Moscow | St Petersburg | Mumbai | Jakarta | Manila | Ho Chi Minh City | Kuala Lumpur | Baku | Lagos | Johannesburg | Auckland | Sao Paulo | Buenos Aires | Lima | Santiago | Montevideo | Limassol ----------------------------------------------------------------------------------------------------------------------- IronFX Global UK Limited is authorised and regulated by the Financial Conduct Authority (FCA no. 585561) IronFX Global (Australia) Pty Limited is authorised and regulated by ASIC (AFSL no. 417482) IronFX Global (South Africa) (Pty) Ltd is authorized by the Financial Services Board (FSP No 45276) IronFX Global NZ Limited is authorised and regulated by FSP (FSPR no. 298966) IronFX Global (Russia) LLC is a member of CRFIN (Membership no. A-8) IronFX Global (Ukraine) LLC is a member of UCRFIN (Membership no. 5) IronFX Global Limited is authorised and regulated by CySEC (Licence no. 125/10) IronFX Global Limited is a Member of Eurex Exchange ----------------------------------------------------------------------------------------------------------------------- Consider the environment before printing this email. From: Andri Hadjiefstathiou [IronFX Global] Sent: Wednesday, December 03, 2014 5:11 PM To: 'freeswitch-users at lists.freeswitch.org' Subject: Freeswitch I got those errors recently from our freeswitch server [ERR] mod_commands.c:4063 Error connecting [ERR] sofia_glue.c:6464 Error connecting [ERR] sofia_glue.c:6504 Error Opening DB [WARNING] sofia_glue.c:6464 Max handles 50 What does this means? I have load recently the mod_xml_rpc is this relate with our issue? Also how I can check the correct open DB open connections ? Andri Hadjiefstathiou | Network and Security Engineer t +357 25027607 | m +357 96860600 | e AHadjiefstathiou at IronFX.com IronFX Global Limited t +357 25027000 | f +357 25027001 | w www.IronFX.com London | Hong Kong | Sydney | Shanghai | Shenzhen | Shenyang | Frankfurt | Madrid | Warsaw | Lisbon | Prague | Budapest | Bucharest | Athens | Kiev | Minsk | Moscow | St Petersburg | Mumbai | Jakarta | Manila | Ho Chi Minh City | Kuala Lumpur | Baku | Lagos | Johannesburg | Auckland | Sao Paulo | Buenos Aires | Lima | Santiago | Montevideo | Limassol ----------------------------------------------------------------------------------------------------------------------- IronFX Global UK Limited is authorised and regulated by the Financial Conduct Authority (FCA no. 585561) IronFX Global (Australia) Pty Limited is authorised and regulated by ASIC (AFSL no. 417482) IronFX Global (South Africa) (Pty) Ltd is authorized by the Financial Services Board (FSP No 45276) IronFX Global NZ Limited is authorised and regulated by FSP (FSPR no. 298966) IronFX Global (Russia) LLC is a member of CRFIN (Membership no. A-8) IronFX Global (Ukraine) LLC is a member of UCRFIN (Membership no. 5) IronFX Global Limited is authorised and regulated by CySEC (Licence no. 125/10) IronFX Global Limited is a Member of Eurex Exchange ----------------------------------------------------------------------------------------------------------------------- This email has been sent from IronFX Global Limited. The information in this email is confidential and may be legally privileged. It is intended solely for the addressee. Access to this email by anyone else is unauthorised. If you are not the intended recipient, any disclosure, copying, distribution or any action taken or omitted to be taken in reliance on it, is prohibited and may be unlawful. Consider the environment before printing this email. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141203/bf0f120f/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... 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Name: image007.jpg Type: image/jpeg Size: 16451 bytes Desc: image007.jpg Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141203/bf0f120f/attachment-0003.jpg From cmrienzo at gmail.com Wed Dec 3 19:18:01 2014 From: cmrienzo at gmail.com (Christopher Rienzo) Date: Wed, 3 Dec 2014 11:18:01 -0500 Subject: [Freeswitch-users] Freeswitch In-Reply-To: References: Message-ID: I don't have any ideas for this particular issue other than general db connection troubleshooting tips. Are you using SQLite or odbc w/ mysql or postgresql? Are database handles being released? Is a new long / inefficient query consuming all your connections? Does the issue go away if you disable mod_xml_rpc? Find out these things to troubleshoot your issue. Also- I noticed you are running an old version of FS based on the line numbers in your logs, so your likelihood of getting free support is not good. On Wed, Dec 3, 2014 at 11:03 AM, Andri Hadjiefstathiou [IronFX Global] < AHadjiefstathiou at ironfx.com> wrote: > After we restart the server all fine. > > > > But I want to know the root of the problem.. why we got this error??? > > > > Recently I have enable mod_xml_rpc is this relate with our issue? > > > > *Andri* *Hadjiefstathiou* | Network and Security Engineer > *t *+357 25027607 | *m *+357 96860600 | *e* AHadjiefstathiou at IronFX.com > > > *IronFX Global Limited t* +357 25027000 | *f* +357 25027001 | *w * > www.IronFX.com > > > > > > > *London | Hong Kong | Sydney | Shanghai | Shenzhen | Shenyang | Frankfurt > | Madrid | Warsaw | Lisbon | Prague | Budapest | Bucharest | Athens | Kiev > | Minsk | Moscow | St Petersburg | Mumbai | Jakarta | Manila | Ho Chi Minh > City | Kuala Lumpur | Baku | Lagos | Johannesburg | Auckland | Sao Paulo | > Buenos Aires | Lima | Santiago | Montevideo | Limassol* > > > ----------------------------------------------------------------------------------------------------------------------- > IronFX Global UK Limited is authorised and regulated by the Financial > Conduct Authority (FCA no. 585561) > IronFX Global (Australia) Pty Limited is authorised and regulated by ASIC > (AFSL no. 417482) > IronFX Global (South Africa) (Pty) Ltd is authorized by the Financial > Services Board (FSP No 45276) > IronFX Global NZ Limited is authorised and regulated by FSP (FSPR no. > 298966) > IronFX Global (Russia) LLC is a member of CRFIN (Membership no. A-8) > IronFX Global (Ukraine) LLC is a member of UCRFIN (Membership no. 5) > IronFX Global Limited is authorised and regulated by CySEC (Licence no. > 125/10) > > IronFX Global Limited is a Member of Eurex Exchange > > ----------------------------------------------------------------------------------------------------------------------- > > *Consider the environment before printing this email.* > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Christopher > Rienzo > *Sent:* Wednesday, December 03, 2014 6:00 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Freeswitch > > > > Check your sofia profile(s)- it has the database settings in there > (dbname, odbc-dsn, db-*) . Then, verify your database is running and can > accept connections. > > > > On Wed, Dec 3, 2014 at 10:39 AM, Andri Hadjiefstathiou [IronFX Global] < > AHadjiefstathiou at ironfx.com> wrote: > > I got those errors recently from our freeswitch server > > > > 2014-11-28 21:19:19.806910 [ERR] sofia_glue.c:6376 Error connecting > > 2014-11-28 21:19:19.806910 [ERR] sofia_glue.c:6416 Error Opening DB > > 2014-11-28 21:19:19.806910 [WARNING] sofia_glue.c:6376 Max handles 50 > exceeded, blocking.... > > 2014-11-28 21:19:21.476882 [ERR] sofia_glue.c:6376 Error connecting > > 2014-11-28 21:19:21.476882 [ERR] sofia_glue.c:6416 Error Opening DB > > 2014-11-28 21:19:21.476882 [WARNING] sofia_glue.c:6376 Max handles 50 > exceeded, blocking.... > > > > What does this means? > > > > I have load recently the mod_xml_rpc is this relate with our issue? > > Also how I can check the correct open DB open connections ? > > > > *Andri* *Hadjiefstathiou* | Network and Security Engineer > *t *+357 25027607 | * m *+357 96860600 | *e* > AHadjiefstathiou at IronFX.com > > > *IronFX Global Limited t* +357 25027000 | *f* +357 25027001 | *w * > www.IronFX.com > > > > *London | Hong Kong | Sydney | Shanghai | Shenzhen | Shenyang | Frankfurt > | Madrid | Warsaw | Lisbon | Prague | Budapest | Bucharest | Athens | Kiev > | Minsk | Moscow | St Petersburg | Mumbai | Jakarta | Manila | Ho Chi Minh > City | Kuala Lumpur | Baku | Lagos | Johannesburg | Auckland | Sao Paulo | > Buenos Aires | Lima | Santiago | Montevideo | Limassol* > > > ----------------------------------------------------------------------------------------------------------------------- > > IronFX Global UK Limited is authorised and regulated by the Financial > Conduct Authority (FCA no. 585561) > IronFX Global (Australia) Pty Limited is authorised and regulated by ASIC > (AFSL no. 417482) > IronFX Global (South Africa) (Pty) Ltd is authorized by the Financial > Services Board (FSP No 45276) > IronFX Global NZ Limited is authorised and regulated by FSP (FSPR no. > 298966) > IronFX Global (Russia) LLC is a member of CRFIN (Membership no. A-8) > IronFX Global (Ukraine) LLC is a member of UCRFIN (Membership no. 5) > IronFX Global Limited is authorised and regulated by CySEC (Licence no. > 125/10) > > IronFX Global Limited is a Member of Eurex Exchange > > ----------------------------------------------------------------------------------------------------------------------- > > *Consider the environment before printing this email.* > > *From:* Andri Hadjiefstathiou [IronFX Global] > *Sent:* Wednesday, December 03, 2014 5:11 PM > *To:* '*freeswitch-users at lists.freeswitch.org > *' > *Subject:* Freeswitch > > > > I got those errors recently from our freeswitch server > > > > [ERR] mod_commands.c:4063 Error connecting > > [ERR] sofia_glue.c:6464 Error connecting > > [ERR] sofia_glue.c:6504 Error Opening DB > > [WARNING] sofia_glue.c:6464 Max handles 50 > > > > What does this means? > > > > I have load recently the mod_xml_rpc is this relate with our issue? > > Also how I can check the correct open DB open connections ? > > > > > > *Andri* *Hadjiefstathiou* | Network and Security Engineer > *t **+357 25027607* | *m **+357 96860600* | *e* > *AHadjiefstathiou at IronFX.com* > > > *IronFX Global Limited t* *+357 25027000* | *f* *+357 25027001* | *w **www.IronFX.com > * > > > > *London | Hong Kong | Sydney | Shanghai | Shenzhen | Shenyang | Frankfurt > | Madrid | Warsaw | Lisbon | Prague | Budapest | Bucharest | Athens | Kiev > | Minsk | Moscow | St Petersburg | Mumbai | Jakarta | Manila | Ho Chi Minh > City | Kuala Lumpur | Baku | Lagos | Johannesburg | Auckland | Sao Paulo | > Buenos Aires | Lima | Santiago | Montevideo | Limassol* > > > ----------------------------------------------------------------------------------------------------------------------- > > IronFX Global UK Limited is authorised and regulated by the Financial > Conduct Authority (FCA no. 585561) > IronFX Global (Australia) Pty Limited is authorised and regulated by ASIC > (AFSL no. 417482) > IronFX Global (South Africa) (Pty) Ltd is authorized by the Financial > Services Board (FSP No 45276) > IronFX Global NZ Limited is authorised and regulated by FSP (FSPR no. > 298966) > IronFX Global (Russia) LLC is a member of CRFIN (Membership no. A-8) > IronFX Global (Ukraine) LLC is a member of UCRFIN (Membership no. 5) > IronFX Global Limited is authorised and regulated by CySEC (Licence no. > 125/10) > > IronFX Global Limited is a Member of Eurex Exchange > > ----------------------------------------------------------------------------------------------------------------------- > > > > > *This email has been sent from IronFX Global Limited. The information in > this email is confidential and may be legally privileged. It is intended > solely for the addressee. Access to this email by anyone else is > unauthorised. If you are not the intended recipient, any disclosure, > copying, distribution or any action taken or omitted to be taken in > reliance on it, is prohibited and may be unlawful.* > > *Consider the environment before printing this email.* > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > *consulting at freeswitch.org * > *http://www.freeswitchsolutions.com * > > Official FreeSWITCH Sites > *http://www.freeswitch.org * > *http://confluence.freeswitch.org * > *http://www.cluecon.com * > > FreeSWITCH-users mailing list > *FreeSWITCH-users at lists.freeswitch.org > * > *http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > * > UNSUBSCRIBE:*http://lists.freeswitch.org/mailman/options/freeswitch-users > * > *http://www.freeswitch.org * > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141203/7ce50cdf/attachment-0001.html From AHadjiefstathiou at IronFX.com Wed Dec 3 19:48:45 2014 From: AHadjiefstathiou at IronFX.com (Andri Hadjiefstathiou [IronFX Global]) Date: Wed, 3 Dec 2014 16:48:45 +0000 Subject: [Freeswitch-users] Freeswitch In-Reply-To: References: Message-ID: We have mysql. I have enable mod_xml_rpc and after 3 days the issue appeared. The issue go away when I restart the freeswitch .. but to be sure I have disable mod_xml_rpc. How I can check the below: Are database handles being released? Is a new long / inefficient query consuming all your connections? What do you mean about the below and what do you suggest? Also- I noticed you are running an old version of FS based on the line numbers in your logs, so your likelihood of getting free support is not good. Thanks Andri Hadjiefstathiou | Network and Security Engineer t +357 25027607 | m +357 96860600 | e AHadjiefstathiou at IronFX.com IronFX Global Limited t +357 25027000 | f +357 25027001 | w www.IronFX.com [cid:image001.jpg at 01D00F29.25424440] [cid:image002.gif at 01D00F29.25424440] [cid:image003.gif at 01D00F29.25424440] [cid:image004.gif at 01D00F29.25424440] [cid:image005.gif at 01D00F29.25424440] [cid:image006.gif at 01D00F29.25424440] London | Hong Kong | Sydney | Shanghai | Shenzhen | Shenyang | Frankfurt | Madrid | Warsaw | Lisbon | Prague | Budapest | Bucharest | Athens | Kiev | Minsk | Moscow | St Petersburg | Mumbai | Jakarta | Manila | Ho Chi Minh City | Kuala Lumpur | Baku | Lagos | Johannesburg | Auckland | Sao Paulo | Buenos Aires | Lima | Santiago | Montevideo | Limassol [cid:image007.jpg at 01D00F29.25424440] ----------------------------------------------------------------------------------------------------------------------- IronFX Global UK Limited is authorised and regulated by the Financial Conduct Authority (FCA no. 585561) IronFX Global (Australia) Pty Limited is authorised and regulated by ASIC (AFSL no. 417482) IronFX Global (South Africa) (Pty) Ltd is authorized by the Financial Services Board (FSP No 45276) IronFX Global NZ Limited is authorised and regulated by FSP (FSPR no. 298966) IronFX Global (Russia) LLC is a member of CRFIN (Membership no. A-8) IronFX Global (Ukraine) LLC is a member of UCRFIN (Membership no. 5) IronFX Global Limited is authorised and regulated by CySEC (Licence no. 125/10) IronFX Global Limited is a Member of Eurex Exchange ----------------------------------------------------------------------------------------------------------------------- Consider the environment before printing this email. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Christopher Rienzo Sent: Wednesday, December 03, 2014 6:18 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Freeswitch I don't have any ideas for this particular issue other than general db connection troubleshooting tips. Are you using SQLite or odbc w/ mysql or postgresql? Are database handles being released? Is a new long / inefficient query consuming all your connections? Does the issue go away if you disable mod_xml_rpc? Find out these things to troubleshoot your issue. Also- I noticed you are running an old version of FS based on the line numbers in your logs, so your likelihood of getting free support is not good. On Wed, Dec 3, 2014 at 11:03 AM, Andri Hadjiefstathiou [IronFX Global] > wrote: After we restart the server all fine. But I want to know the root of the problem.. why we got this error??? Recently I have enable mod_xml_rpc is this relate with our issue? Andri Hadjiefstathiou | Network and Security Engineer t +357 25027607 | m +357 96860600 | e AHadjiefstathiou at IronFX.com IronFX Global Limited t +357 25027000 | f +357 25027001 | w www.IronFX.com London | Hong Kong | Sydney | Shanghai | Shenzhen | Shenyang | Frankfurt | Madrid | Warsaw | Lisbon | Prague | Budapest | Bucharest | Athens | Kiev | Minsk | Moscow | St Petersburg | Mumbai | Jakarta | Manila | Ho Chi Minh City | Kuala Lumpur | Baku | Lagos | Johannesburg | Auckland | Sao Paulo | Buenos Aires | Lima | Santiago | Montevideo | Limassol ----------------------------------------------------------------------------------------------------------------------- IronFX Global UK Limited is authorised and regulated by the Financial Conduct Authority (FCA no. 585561) IronFX Global (Australia) Pty Limited is authorised and regulated by ASIC (AFSL no. 417482) IronFX Global (South Africa) (Pty) Ltd is authorized by the Financial Services Board (FSP No 45276) IronFX Global NZ Limited is authorised and regulated by FSP (FSPR no. 298966) IronFX Global (Russia) LLC is a member of CRFIN (Membership no. A-8) IronFX Global (Ukraine) LLC is a member of UCRFIN (Membership no. 5) IronFX Global Limited is authorised and regulated by CySEC (Licence no. 125/10) IronFX Global Limited is a Member of Eurex Exchange ----------------------------------------------------------------------------------------------------------------------- Consider the environment before printing this email. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Christopher Rienzo Sent: Wednesday, December 03, 2014 6:00 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Freeswitch Check your sofia profile(s)- it has the database settings in there (dbname, odbc-dsn, db-*) . Then, verify your database is running and can accept connections. On Wed, Dec 3, 2014 at 10:39 AM, Andri Hadjiefstathiou [IronFX Global] wrote: I got those errors recently from our freeswitch server 2014-11-28 21:19:19.806910 [ERR] sofia_glue.c:6376 Error connecting 2014-11-28 21:19:19.806910 [ERR] sofia_glue.c:6416 Error Opening DB 2014-11-28 21:19:19.806910 [WARNING] sofia_glue.c:6376 Max handles 50 exceeded, blocking.... 2014-11-28 21:19:21.476882 [ERR] sofia_glue.c:6376 Error connecting 2014-11-28 21:19:21.476882 [ERR] sofia_glue.c:6416 Error Opening DB 2014-11-28 21:19:21.476882 [WARNING] sofia_glue.c:6376 Max handles 50 exceeded, blocking.... What does this means? I have load recently the mod_xml_rpc is this relate with our issue? Also how I can check the correct open DB open connections ? Andri Hadjiefstathiou | Network and Security Engineer t +357 25027607 | m +357 96860600 | e AHadjiefstathiou at IronFX.com IronFX Global Limited t +357 25027000 | f +357 25027001 | w www.IronFX.com London | Hong Kong | Sydney | Shanghai | Shenzhen | Shenyang | Frankfurt | Madrid | Warsaw | Lisbon | Prague | Budapest | Bucharest | Athens | Kiev | Minsk | Moscow | St Petersburg | Mumbai | Jakarta | Manila | Ho Chi Minh City | Kuala Lumpur | Baku | Lagos | Johannesburg | Auckland | Sao Paulo | Buenos Aires | Lima | Santiago | Montevideo | Limassol ----------------------------------------------------------------------------------------------------------------------- IronFX Global UK Limited is authorised and regulated by the Financial Conduct Authority (FCA no. 585561) IronFX Global (Australia) Pty Limited is authorised and regulated by ASIC (AFSL no. 417482) IronFX Global (South Africa) (Pty) Ltd is authorized by the Financial Services Board (FSP No 45276) IronFX Global NZ Limited is authorised and regulated by FSP (FSPR no. 298966) IronFX Global (Russia) LLC is a member of CRFIN (Membership no. A-8) IronFX Global (Ukraine) LLC is a member of UCRFIN (Membership no. 5) IronFX Global Limited is authorised and regulated by CySEC (Licence no. 125/10) IronFX Global Limited is a Member of Eurex Exchange ----------------------------------------------------------------------------------------------------------------------- Consider the environment before printing this email. From: Andri Hadjiefstathiou [IronFX Global] Sent: Wednesday, December 03, 2014 5:11 PM To: 'freeswitch-users at lists.freeswitch.org' Subject: Freeswitch I got those errors recently from our freeswitch server [ERR] mod_commands.c:4063 Error connecting [ERR] sofia_glue.c:6464 Error connecting [ERR] sofia_glue.c:6504 Error Opening DB [WARNING] sofia_glue.c:6464 Max handles 50 What does this means? I have load recently the mod_xml_rpc is this relate with our issue? Also how I can check the correct open DB open connections ? Andri Hadjiefstathiou | Network and Security Engineer t +357 25027607 | m +357 96860600 | e AHadjiefstathiou at IronFX.com IronFX Global Limited t +357 25027000 | f +357 25027001 | w www.IronFX.com London | Hong Kong | Sydney | Shanghai | Shenzhen | Shenyang | Frankfurt | Madrid | Warsaw | Lisbon | Prague | Budapest | Bucharest | Athens | Kiev | Minsk | Moscow | St Petersburg | Mumbai | Jakarta | Manila | Ho Chi Minh City | Kuala Lumpur | Baku | Lagos | Johannesburg | Auckland | Sao Paulo | Buenos Aires | Lima | Santiago | Montevideo | Limassol ----------------------------------------------------------------------------------------------------------------------- IronFX Global UK Limited is authorised and regulated by the Financial Conduct Authority (FCA no. 585561) IronFX Global (Australia) Pty Limited is authorised and regulated by ASIC (AFSL no. 417482) IronFX Global (South Africa) (Pty) Ltd is authorized by the Financial Services Board (FSP No 45276) IronFX Global NZ Limited is authorised and regulated by FSP (FSPR no. 298966) IronFX Global (Russia) LLC is a member of CRFIN (Membership no. A-8) IronFX Global (Ukraine) LLC is a member of UCRFIN (Membership no. 5) IronFX Global Limited is authorised and regulated by CySEC (Licence no. 125/10) IronFX Global Limited is a Member of Eurex Exchange ----------------------------------------------------------------------------------------------------------------------- This email has been sent from IronFX Global Limited. The information in this email is confidential and may be legally privileged. It is intended solely for the addressee. Access to this email by anyone else is unauthorised. If you are not the intended recipient, any disclosure, copying, distribution or any action taken or omitted to be taken in reliance on it, is prohibited and may be unlawful. Consider the environment before printing this email. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... 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Name: image007.jpg Type: image/jpeg Size: 16451 bytes Desc: image007.jpg Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141203/251bcd42/attachment-0003.jpg From ahabiba at gmail.com Wed Dec 3 21:36:56 2014 From: ahabiba at gmail.com (Ahmed Habiba) Date: Wed, 3 Dec 2014 21:36:56 +0300 Subject: [Freeswitch-users] FreeSWITCH TLS not able to receive calls In-Reply-To: References: Message-ID: <1662947A-0E07-448C-9BF9-8AF2F99D3FED@gmail.com> Dear Brian, I?m using "zoiper 1.19.7? on android, the issue appears only one I enable SRTP, for TLS only it is working very fine. The issue as shown below it keeps registering till finally zipper restart on mobile. One more point I?m configuring Zoiper without using RPORT for either SRTP or TLS. ?2014-12-01 20:57:24.003230 [NOTICE] switch_ivr_originate.c:3467 Hangup sofia/internal/sip:1000 at 222.248.102.244 :38614 [CS_CONSUME_MEDIA] [NO_ANSWER]" recv 1290 bytes from tls/[222.215.195.234]:2896 at 20:56:54.344450: ------------------------------------------------------------------------ INVITE <>sip:1000 at abc-xyz.com SIP/2.0 v: SIP/2.0/TLS 192.168.8.109:41659;rport;branch=z9hG4bKPjvnvRZjpAXahzIV7S8SRgS5vwKmYI9WFS;alias Max-Forwards: 70 f: < <>sip:1006 at abc-xyz.com >;tag=weeMqzInPcNLPd6s5IX63c8be3DMvGkU t: < <>sip:1000 at abc-xyz.com > m: < <>sip:1006 at 222.215.195.234:2896;transport=TLS;ob > i: sjprJQszC2LHvY6jRr-lm9rau9V1PlYX CSeq: 32737 INVITE Route: < <>sip:abc-xyz.com;transport=tls;lr > k: replaces, 100rel, timer, norefersub x: 1800 Min-SE: 90 User-Agent: CSipSimple_GT-S5830-10/r2450 Proxy-Authorization: Digest username="1006", realm="abc-xyz.com ", nonce="329456ee-7994-11e4-bf79-1182ebca2429", uri=" <>sip:1000 at abc-xyz.com ", response="ab3a0f2fe9207ced62eebd8c9b8c32b4", algorithm=MD5, cnonce="iLa-C.wpnBSeTQjWiXZK3H3PGYhi6EGt", qop=auth, nc=00000001 c: application/sdp l: 469 v=0 o=- 3626452487 3626452487 IN IP4 192.168.8.109 s=pjmedia c=IN IP4 192.168.8.109 t=0 0 m=audio 4000 RTP/SAVP 98 0 8 101 c=IN IP4 192.168.8.109 a=rtcp:4001 IN IP4 192.168.8.109 a=sendrecv a=rtpmap:98 SILK/16000 a=fmtp:98 useinbandfec=0 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:Vci7cNdSFTkkLnqz+qRkqCctPvRT6jIOrNc5KMbz a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:nW3ECq5YHHnrY2q5u7XZE4fvPfMOAWKQ+ehOp7zV ------------------------------------------------------------------------ send 405 bytes to tls/[222.215.195.234]:2896 at 20:56:54.344687: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/TLS 192.168.8.109:41659;rport=2896;branch=z9hG4bKPjvnvRZjpAXahzIV7S8SRgS5vwKmYI9WFS;alias;received=222.215.195.234 f: < <>sip:1006 at abc-xyz.com >;tag=weeMqzInPcNLPd6s5IX63c8be3DMvGkU t: < <>sip:1000 at abc-xyz.com > i: sjprJQszC2LHvY6jRr-lm9rau9V1PlYX CSeq: 32737 INVITE User-Agent: FreeSWITCH-mod_sofia/1.4.13+git~20141103T195300Z~b942d0faa8~64bit Content-Length: 0 send 2133 bytes to tls/[222.248.102.244]:38614 at 20:56:54.360608: ------------------------------------------------------------------------ INVITE <>sip:1000 at 222.248.102.244:38614;rinstance=98c51296f2d531cc;transport=TLS SIP/2.0 Via: SIP/2.0/TLS 176.31.245.17;branch=z9hG4bKjXZvg3cX76Krj Max-Forwards: 69 From: "Extension 1006" < <>sip:1006 at 176.31.245.17 >;tag=15DcKN9tUZeaK To: < <>sip:1000 at 222.248.102.244:38614;rinstance=98c51296f2d531cc;transport=TLS > Call-ID: 0a3e07c5-f437-1232-9bba-d2ab2784dc6a CSeq: 68395843 INVITE Contact: < <>sip:mod_sofia at 176.31.245.17:5061;transport=tls > User-Agent: FreeSWITCH-mod_sofia/1.4.13+git~20141103T195300Z~b942d0faa8~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 1076 X-FS-Support: update_display,send_info Remote-Party-ID: "Extension 1006" < <>sip:1006 at 176.31.245.17 >;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1417445672 1417445673 IN IP4 176.31.245.17 s=FreeSWITCH c=IN IP4 176.31.245.17 t=0 0 m=audio 18142 RTP/SAVP 0 8 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=crypto:1 AEAD_AES_256_GCM_8 inline:6p/uXB7tN5SG9vHP/DMurij6BaZIOKcP4Tle0kjo2lwL1Oa4z9WGyZtk4zs a=crypto:2 AEAD_AES_128_GCM_8 inline:hGeJA6XQV1V35oTu2141o7mD64dFDRbeCbcf6g a=crypto:3 AES_CM_256_HMAC_SHA1_80 inline:XdZwwcl6kMRrWY7Vkrbozf/o1awka9lef0UjQRwvnOnnrbAgzXIm8s3yu7o8DQ a=crypto:4 AES_CM_192_HMAC_SHA1_80 inline:KtM92m0ac2musYPoXLDJb4rUbEMYAwMegPH9RMtzwzPb+0O93e0 a=crypto:5 AES_CM_128_HMAC_SHA1_80 inline:PtOpdIk2LTSCSDVzPaSN6AV2xosBrJUTd4mvKIGh a=crypto:6 AES_CM_256_HMAC_SHA1_32 inline:ZHzWxajRlQp8IuzX5CuFM1SRkL0huC62ukX583+Vg0LBiK9t2NrpP8FNwdkckQ a=crypto:7 AES_CM_192_HMAC_SHA1_32 inline:S9BWrDxGFYpU3P9o3HD0Z97W5jwo8vD+E1sJi821r4/B+cdukTw a=crypto:8 AES_CM_128_HMAC_SHA1_32 inline:l7C0wU29AjF2ZMnCTvks+1ytd/UiXHqN8UIsM6vf a=crypto:9 AES_CM_128_NULL_AUTH inline:tv97QsO7+S9NLTXrD02omyWoyM7I2KaWaimUgq99 a=ptime:20 ------------------------------------------------------------------------ REGISTER <>sip:abc-xyz.com;transport=TLS SIP/2.0 Via: SIP/2.0/TLS 192.168.10.2:41120;branch=z9hG4bK-d8754z-903949acb1d10083-1---d8754z-;rport Max-Forwards: 70 Contact: < <>sip:1000 at 192.168.10.2:41120;rinstance=57d4e16c725a51e6;transport=TLS > To: < <>sip:1000 at abc-xyz.com;transport=TLS > From: < <>sip:1000 at abc-xyz.com;transport=TLS >;tag=4a2eae22 Call-ID: N2RiMjE2ZTQxNzYzMGY3ZmM3ZGJhODM4NTMzZmQ1YTA. CSeq: 1 REGISTER Expires: 60 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Zoiper r27147 Allow-Events: presence, kpml Content-Length: 0 ------------------------------------------------------------------------ send 716 bytes to tls/[222.248.102.244]:38620 at 20:57:01.806433: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/TLS 192.168.10.2:41120;branch=z9hG4bK-d8754z-903949acb1d10083-1---d8754z-;rport=38620;received=222.248.102.244 From: < <>sip:1000 at abc-xyz.com;transport=TLS >;tag=4a2eae22 To: < <>sip:1000 at abc-xyz.com;transport=TLS >;tag=2e74mgtyr84ve Call-ID: N2RiMjE2ZTQxNzYzMGY3ZmM3ZGJhODM4NTMzZmQ1YTA. CSeq: 1 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.4.13+git~20141103T195300Z~b942d0faa8~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces WWW-Authenticate: Digest realm="abc-xyz.com ", nonce="374fdd98-7994-11e4-bf9c-1182ebca2429", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ recv 944 bytes from tls/[222.248.102.244]:38620 at 20:57:01.995662: ------------------------------------------------------------------------ REGISTER <>sip:abc-xyz.com;transport=TLS SIP/2.0 Via: SIP/2.0/TLS 192.168.10.2:41120;branch=z9hG4bK-d8754z-8ca0e782336414eb-1---d8754z-;rport Max-Forwards: 70 Contact: < <>sip:1000 at 192.168.10.2:41120;rinstance=57d4e16c725a51e6;transport=TLS > To: < <>sip:1000 at abc-xyz.com;transport=TLS > From: < <>sip:1000 at abc-xyz.com;transport=TLS >;tag=4a2eae22 Call-ID: N2RiMjE2ZTQxNzYzMGY3ZmM3ZGJhODM4NTMzZmQ1YTA. CSeq: 2 REGISTER Expires: 60 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Zoiper r27147 Authorization: Digest username="1000",realm="abc-xyz.com ",nonce="374fdd98-7994-11e4-bf9c-1182ebca2429",uri=" <>sip:abc-xyz.com;transport=TLS ",response="8d7e108ff98035d37424c32166fe0253",cnonce="107cf281c3b57888a4bfd1a9c3776098",nc=00000001,qop=auth,algorithm=MD5 Allow-Events: presence, kpml Content-Length: 0 ------------------------------------------------------------------------ send 712 bytes to tls/[222.248.102.244]:38620 at 20:57:01.997931: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/TLS 192.168.10.2:41120;branch=z9hG4bK-d8754z-8ca0e782336414eb-1---d8754z-;rport=38620;received=222.248.102.244 From: < <>sip:1000 at abc-xyz.com;transport=TLS >;tag=4a2eae22 To: < <>sip:1000 at abc-xyz.com;transport=TLS >;tag=3Q0XpBB2NHUFa Call-ID: N2RiMjE2ZTQxNzYzMGY3ZmM3ZGJhODM4NTMzZmQ1YTA. CSeq: 2 REGISTER Contact: < <>sip:1000 at 192.168.10.2:41120;rinstance=57d4e16c725a51e6;transport=TLS >;expires=60 Date: Mon, 01 Dec 2014 19:57:01 GMT User-Agent: FreeSWITCH-mod_sofia/1.4.13+git~20141103T195300Z~b942d0faa8~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Content-Length: 0 ------------------------------------------------------------------------ recv 944 bytes from tls/[222.248.102.244]:38620 at 20:57:02.205719: ------------------------------------------------------------------------ REGISTER <>sip:abc-xyz.com;transport=TLS SIP/2.0 Via: SIP/2.0/TLS 222.248.102.244:38620;branch=z9hG4bK-d8754z-59d493abef215091-1---d8754z-;rport Max-Forwards: 70 Contact: < <>sip:1000 at 192.168.10.2:41120;rinstance=57d4e16c725a51e6;transport=TLS >;expires=0 To: < <>sip:1000 at abc-xyz.com;transport=TLS > From: < <>sip:1000 at abc-xyz.com;transport=TLS >;tag=4a2eae22 Call-ID: N2RiMjE2ZTQxNzYzMGY3ZmM3ZGJhODM4NTMzZmQ1YTA. CSeq: 3 REGISTER Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Zoiper r27147 Authorization: Digest username="1000",realm="abc-xyz.com ",nonce="374fdd98-7994-11e4-bf9c-1182ebca2429",uri=" <>sip:abc-xyz.com;transport=TLS ",response="5168bf9afffe9adacef7d898bd6bec9b",cnonce="80108fa519303990cfd2d46639d767ee",nc=00000002,qop=auth,algorithm=MD5 Allow-Events: presence, kpml Content-Length: 0 ------------------------------------------------------------------------ send 598 bytes to tls/[222.248.102.244]:38620 at 20:57:02.207249: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/TLS 222.248.102.244:38620;branch=z9hG4bK-d8754z-59d493abef215091-1---d8754z-;rport=38620 From: < <>sip:1000 at abc-xyz.com;transport=TLS >;tag=4a2eae22 To: < <>sip:1000 at abc-xyz.com;transport=TLS >;tag=40Spr6U5jtH2N Call-ID: N2RiMjE2ZTQxNzYzMGY3ZmM3ZGJhODM4NTMzZmQ1YTA. CSeq: 3 REGISTER Date: Mon, 01 Dec 2014 19:57:02 GMT User-Agent: FreeSWITCH-mod_sofia/1.4.13+git~20141103T195300Z~b942d0faa8~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Content-Length: 0 ------------------------------------------------------------------------ recv 681 bytes from tls/[222.248.102.244]:38620 at 20:57:02.400564: ------------------------------------------------------------------------ REGISTER <>sip:abc-xyz.com;transport=TLS SIP/2.0 Via: SIP/2.0/TLS 222.248.102.244:38620;branch=z9hG4bK-d8754z-d738a1a836608006-1---d8754z-;rport Max-Forwards: 70 Contact: < <>sip:1000 at 222.248.102.244:38620;rinstance=43166fee0aaac1bc;transport=TLS > To: < <>sip:1000 at abc-xyz.com;transport=TLS > From: < <>sip:1000 at abc-xyz.com;transport=TLS >;tag=979e5f46 Call-ID: MTJhNjIzMDM1MDEwMDVjZDRkOGMyYjBlMWU4ZDEzY2Y. CSeq: 1 REGISTER Expires: 60 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Zoiper r27147 Allow-Events: presence, kpml Content-Length: 0 ------------------------------------------------------------------------ send 694 bytes to tls/[222.248.102.244]:38620 at 20:57:02.401167: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/TLS 222.248.102.244:38620;branch=z9hG4bK-d8754z-d738a1a836608006-1---d8754z-;rport=38620 From: < <>sip:1000 at abc-xyz.com;transport=TLS >;tag=979e5f46 To: < <>sip:1000 at abc-xyz.com;transport=TLS >;tag=59jFt1c9F37mH Call-ID: MTJhNjIzMDM1MDEwMDVjZDRkOGMyYjBlMWU4ZDEzY2Y. CSeq: 1 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.4.13+git~20141103T195300Z~b942d0faa8~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces WWW-Authenticate: Digest realm="abc-xyz.com ", nonce="37aaa44e-7994-11e4-bf9d-1182ebca2429", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ recv 950 bytes from tls/[222.248.102.244]:38620 at 20:57:02.655864: ------------------------------------------------------------------------ REGISTER <>sip:abc-xyz.com;transport=TLS SIP/2.0 Via: SIP/2.0/TLS 222.248.102.244:38620;branch=z9hG4bK-d8754z-dfaa7130675d6cfa-1---d8754z-;rport Max-Forwards: 70 Contact: < <>sip:1000 at 222.248.102.244:38620;rinstance=43166fee0aaac1bc;transport=TLS > To: < <>sip:1000 at abc-xyz.com;transport=TLS > From: < <>sip:1000 at abc-xyz.com;transport=TLS >;tag=979e5f46 Call-ID: MTJhNjIzMDM1MDEwMDVjZDRkOGMyYjBlMWU4ZDEzY2Y. CSeq: 2 REGISTER Expires: 60 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Zoiper r27147 Authorization: Digest username="1000",realm="abc-xyz.com ",nonce="37aaa44e-7994-11e4-bf9d-1182ebca2429",uri=" <>sip:abc-xyz.com;transport=TLS ",response="f1667094626fc10ca76a070438497e6f",cnonce="dc790bffe285311f0b903bd515a3e741",nc=00000001,qop=auth,algorithm=MD5 Allow-Events: presence, kpml Content-Length: 0 ------------------------------------------------------------------------ send 693 bytes to tls/[222.248.102.244]:38620 at 20:57:02.657688: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/TLS 222.248.102.244:38620;branch=z9hG4bK-d8754z-dfaa7130675d6cfa-1---d8754z-;rport=38620 From: < <>sip:1000 at abc-xyz.com;transport=TLS >;tag=979e5f46 To: < <>sip:1000 at abc-xyz.com;transport=TLS >;tag=6jc8UvXcDcy7c Call-ID: MTJhNjIzMDM1MDEwMDVjZDRkOGMyYjBlMWU4ZDEzY2Y. CSeq: 2 REGISTER Contact: < <>sip:1000 at 222.248.102.244:38620;rinstance=43166fee0aaac1bc;transport=TLS >;expires=60 Date: Mon, 01 Dec 2014 19:57:02 GMT User-Agent: FreeSWITCH-mod_sofia/1.4.13+git~20141103T195300Z~b942d0faa8~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Content-Length: 0 > > > From: Brian West > > To: FreeSWITCH Users Help > > Date: December 1, 2014 at 5:19:30 PM GMT+3 > Reply-To: FreeSWITCH Users Help > > Subject: Re: [Freeswitch-users] FreeSWITCH TLS not able to receive calls > > > What endpoints are involved? have you looked at 'sofia loglevel all 9' output and see if it gives you a clue? > > On Fri, Nov 28, 2014 at 6:24 PM, Ahmed Habiba > wrote: > Dears, > > I?ve configured FreeSWITCH with the below version with TLS/SRTPas per the recommendation in page ?https://wiki.freeswitch.org/wiki/SIP_TLS ? and it was strait forward, and I was able to connect and make make calls using zoiper, but I was not able to receive any calls after enabling the TLS/SRTP. > > "FreeSWITCH Version 1.4.13+git~20141103T195300Z~b942d0faa8~64bit (git b942d0f 2014-11-03 19:53:00Z 64bit)? > > Your kind feedback will be appreciate. > > Thanks, > > Ahmed Habiba. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -- > Brian West > brian at freeswitch.org > > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378) > iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype: briankwest > -- Thanks and Best Regards, Ahmed Habiba Mob: +20 10 37 82 970 Success: believe (Vision) plus commitment (Action) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141203/254c5089/attachment-0001.html From mike at jerris.com Wed Dec 3 23:31:32 2014 From: mike at jerris.com (Michael Jerris) Date: Wed, 3 Dec 2014 15:31:32 -0500 Subject: [Freeswitch-users] Windows build system Message-ID: Given the recent announcements by Microsoft about the community edition 2013 being available, we are working to migrate the build system towards using that as our primary build. As part of this process we will be very soon dropping support for any version of Visual Studio prior to 2012. If you feel strongly about needing support for these older versions, please speak up now with an offer to maintain these legacy build systems. We are also investigating moving to using chocolatey as a new system to manage dependencies on windows instead of maintaining the build for all our deps ourselves. It is also possible we will drop support for the 2012 build system in the not so distant future. Could the community chime in here as to what their needs are, and what they are willing to do to help support the windows builds so we can determine what we plan to support going forward. Thanks Mike https://chocolatey.org/ http://www.visualstudio.com/en-us/news/vs2013-community-vs.aspx -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141203/cb0960ef/attachment.html From brian at freeswitch.org Wed Dec 3 23:44:31 2014 From: brian at freeswitch.org (Brian West) Date: Wed, 3 Dec 2014 14:44:31 -0600 Subject: [Freeswitch-users] FreeSWITCH TLS not able to receive calls In-Reply-To: <1662947A-0E07-448C-9BF9-8AF2F99D3FED@gmail.com> References: <1662947A-0E07-448C-9BF9-8AF2F99D3FED@gmail.com> Message-ID: I would try limiting the crypto suites on the FS side, Look in vars.xml for details, I know if we send the whole crypt suite list at a polycom on pretty much any firmware the phone goes rum dumb and starts to echo our SDP's back at us. :P On Wed, Dec 3, 2014 at 12:36 PM, Ahmed Habiba wrote: > Dear Brian, > > > I?m using "zoiper 1.19.7? on android, the issue appears only one I enable > SRTP, for TLS only it is working very fine. > > The issue as shown below it keeps registering till finally zipper restart > on mobile. > > One more point I?m configuring Zoiper without using RPORT for either SRTP > or TLS. > > *?2014-12-01 20:57:24.003230 [NOTICE] switch_ivr_originate.c:3467 Hangup > sofia/internal/sip:1000 at 222.248.102.244 <1000 at 222.248.102.244>:38614 > [CS_CONSUME_MEDIA] [NO_ANSWER]"* > > > > recv 1290 bytes from tls/[222.215.195.234]:2896 at 20:56:54.344450: > ------------------------------------------------------------------------ > INVITE sip:1000 at abc-xyz.com SIP/2.0 > v: SIP/2.0/TLS 192.168.8.109:41659 > ;rport;branch=z9hG4bKPjvnvRZjpAXahzIV7S8SRgS5vwKmYI9WFS;alias > Max-Forwards: 70 > f: ;tag=weeMqzInPcNLPd6s5IX63c8be3DMvGkU > t: > m: > i: sjprJQszC2LHvY6jRr-lm9rau9V1PlYX > CSeq: 32737 INVITE > Route: > k: replaces, 100rel, timer, norefersub > x: 1800 > Min-SE: 90 > User-Agent: CSipSimple_GT-S5830-10/r2450 > Proxy-Authorization: Digest username="1006", realm="abc-xyz.com", > nonce="329456ee-7994-11e4-bf79-1182ebca2429", uri="sip:1000 at abc-xyz.com", > response="ab3a0f2fe9207ced62eebd8c9b8c32b4", algorithm=MD5, > cnonce="iLa-C.wpnBSeTQjWiXZK3H3PGYhi6EGt", qop=auth, nc=00000001 > c: application/sdp > l: 469 > > v=0 > o=- 3626452487 3626452487 IN IP4 192.168.8.109 > s=pjmedia > c=IN IP4 192.168.8.109 > t=0 0 > m=audio 4000 RTP/SAVP 98 0 8 101 > c=IN IP4 192.168.8.109 > a=rtcp:4001 IN IP4 192.168.8.109 > a=sendrecv > a=rtpmap:98 SILK/16000 > a=fmtp:98 useinbandfec=0 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=crypto:1 AES_CM_128_HMAC_SHA1_80 > inline:Vci7cNdSFTkkLnqz+qRkqCctPvRT6jIOrNc5KMbz > a=crypto:2 AES_CM_128_HMAC_SHA1_32 > inline:nW3ECq5YHHnrY2q5u7XZE4fvPfMOAWKQ+ehOp7zV > ------------------------------------------------------------------------ > > send 405 bytes to tls/[222.215.195.234]:2896 at 20:56:54.344687: > ------------------------------------------------------------------------ > SIP/2.0 100 Trying > Via: SIP/2.0/TLS 192.168.8.109:41659 > ;rport=2896;branch=z9hG4bKPjvnvRZjpAXahzIV7S8SRgS5vwKmYI9WFS;alias;received=222.215.195.234 > f: ;tag=weeMqzInPcNLPd6s5IX63c8be3DMvGkU > t: > i: sjprJQszC2LHvY6jRr-lm9rau9V1PlYX > CSeq: 32737 INVITE > User-Agent: > FreeSWITCH-mod_sofia/1.4.13+git~20141103T195300Z~b942d0faa8~64bit > Content-Length: 0 > > > > > > > send 2133 bytes to tls/[222.248.102.244]:38614 at 20:56:54.360608: > ------------------------------------------------------------------------ > INVITE > sip:1000 at 222.248.102.244:38614;rinstance=98c51296f2d531cc;transport=TLS > SIP/2.0 > Via: SIP/2.0/TLS 176.31.245.17;branch=z9hG4bKjXZvg3cX76Krj > Max-Forwards: 69 > From: "Extension 1006" ;tag=15DcKN9tUZeaK > To: < > sip:1000 at 222.248.102.244:38614;rinstance=98c51296f2d531cc;transport=TLS> > Call-ID: 0a3e07c5-f437-1232-9bba-d2ab2784dc6a > CSeq: 68395843 INVITE > Contact: > User-Agent: > FreeSWITCH-mod_sofia/1.4.13+git~20141103T195300Z~b942d0faa8~64bit > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, path, replaces > Allow-Events: talk, hold, conference, presence, as-feature-event, > dialog, line-seize, call-info, sla, include-session-description, > presence.winfo, message-summary, refer > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 1076 > X-FS-Support: update_display,send_info > Remote-Party-ID: "Extension 1006" >;party=calling;screen=yes;privacy=off > > v=0 > o=FreeSWITCH 1417445672 1417445673 IN IP4 176.31.245.17 > s=FreeSWITCH > c=IN IP4 176.31.245.17 > t=0 0 > m=audio 18142 RTP/SAVP 0 8 101 13 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=crypto:1 AEAD_AES_256_GCM_8 > inline:6p/uXB7tN5SG9vHP/DMurij6BaZIOKcP4Tle0kjo2lwL1Oa4z9WGyZtk4zs > a=crypto:2 AEAD_AES_128_GCM_8 > inline:hGeJA6XQV1V35oTu2141o7mD64dFDRbeCbcf6g > a=crypto:3 AES_CM_256_HMAC_SHA1_80 > inline:XdZwwcl6kMRrWY7Vkrbozf/o1awka9lef0UjQRwvnOnnrbAgzXIm8s3yu7o8DQ > a=crypto:4 AES_CM_192_HMAC_SHA1_80 > inline:KtM92m0ac2musYPoXLDJb4rUbEMYAwMegPH9RMtzwzPb+0O93e0 > a=crypto:5 AES_CM_128_HMAC_SHA1_80 > inline:PtOpdIk2LTSCSDVzPaSN6AV2xosBrJUTd4mvKIGh > a=crypto:6 AES_CM_256_HMAC_SHA1_32 > inline:ZHzWxajRlQp8IuzX5CuFM1SRkL0huC62ukX583+Vg0LBiK9t2NrpP8FNwdkckQ > a=crypto:7 AES_CM_192_HMAC_SHA1_32 > inline:S9BWrDxGFYpU3P9o3HD0Z97W5jwo8vD+E1sJi821r4/B+cdukTw > a=crypto:8 AES_CM_128_HMAC_SHA1_32 > inline:l7C0wU29AjF2ZMnCTvks+1ytd/UiXHqN8UIsM6vf > a=crypto:9 AES_CM_128_NULL_AUTH > inline:tv97QsO7+S9NLTXrD02omyWoyM7I2KaWaimUgq99 > a=ptime:20 > ------------------------------------------------------------------------ > > > REGISTER sip:abc-xyz.com;transport=TLS SIP/2.0 > Via: SIP/2.0/TLS 192.168.10.2:41120 > ;branch=z9hG4bK-d8754z-903949acb1d10083-1---d8754z-;rport > Max-Forwards: 70 > Contact: < > sip:1000 at 192.168.10.2:41120;rinstance=57d4e16c725a51e6;transport=TLS> > To: > From: ;tag=4a2eae22 > Call-ID: N2RiMjE2ZTQxNzYzMGY3ZmM3ZGJhODM4NTMzZmQ1YTA. > CSeq: 1 REGISTER > Expires: 60 > Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, > SUBSCRIBE > Supported: replaces, norefersub, extended-refer, timer, > X-cisco-serviceuri > User-Agent: Zoiper r27147 > Allow-Events: presence, kpml > Content-Length: 0 > > > > ------------------------------------------------------------------------ > send 716 bytes to tls/[222.248.102.244]:38620 at 20:57:01.806433: > ------------------------------------------------------------------------ > SIP/2.0 401 Unauthorized > Via: SIP/2.0/TLS 192.168.10.2:41120 > ;branch=z9hG4bK-d8754z-903949acb1d10083-1---d8754z-;rport=38620;received=222.248.102.244 > From: ;tag=4a2eae22 > To: ;tag=2e74mgtyr84ve > Call-ID: N2RiMjE2ZTQxNzYzMGY3ZmM3ZGJhODM4NTMzZmQ1YTA. > CSeq: 1 REGISTER > User-Agent: > FreeSWITCH-mod_sofia/1.4.13+git~20141103T195300Z~b942d0faa8~64bit > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, path, replaces > WWW-Authenticate: Digest realm="abc-xyz.com", > nonce="374fdd98-7994-11e4-bf9c-1182ebca2429", algorithm=MD5, qop="auth" > Content-Length: 0 > > > > ------------------------------------------------------------------------ > recv 944 bytes from tls/[222.248.102.244]:38620 at 20:57:01.995662: > ------------------------------------------------------------------------ > REGISTER sip:abc-xyz.com;transport=TLS SIP/2.0 > Via: SIP/2.0/TLS 192.168.10.2:41120 > ;branch=z9hG4bK-d8754z-8ca0e782336414eb-1---d8754z-;rport > Max-Forwards: 70 > Contact: < > sip:1000 at 192.168.10.2:41120;rinstance=57d4e16c725a51e6;transport=TLS> > To: > From: ;tag=4a2eae22 > Call-ID: N2RiMjE2ZTQxNzYzMGY3ZmM3ZGJhODM4NTMzZmQ1YTA. > CSeq: 2 REGISTER > Expires: 60 > Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, > SUBSCRIBE > Supported: replaces, norefersub, extended-refer, timer, > X-cisco-serviceuri > User-Agent: Zoiper r27147 > Authorization: Digest username="1000",realm="abc-xyz.com > ",nonce="374fdd98-7994-11e4-bf9c-1182ebca2429",uri=" > sip:abc-xyz.com;transport=TLS > ",response="8d7e108ff98035d37424c32166fe0253",cnonce="107cf281c3b57888a4bfd1a9c3776098",nc=00000001,qop=auth,algorithm=MD5 > Allow-Events: presence, kpml > Content-Length: 0 > > > > ------------------------------------------------------------------------ > send 712 bytes to tls/[222.248.102.244]:38620 at 20:57:01.997931: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/TLS 192.168.10.2:41120 > ;branch=z9hG4bK-d8754z-8ca0e782336414eb-1---d8754z-;rport=38620;received=222.248.102.244 > From: ;tag=4a2eae22 > To: ;tag=3Q0XpBB2NHUFa > Call-ID: N2RiMjE2ZTQxNzYzMGY3ZmM3ZGJhODM4NTMzZmQ1YTA. > CSeq: 2 REGISTER > Contact: < > sip:1000 at 192.168.10.2:41120;rinstance=57d4e16c725a51e6;transport=TLS > >;expires=60 > Date: Mon, 01 Dec 2014 19:57:01 GMT > User-Agent: > FreeSWITCH-mod_sofia/1.4.13+git~20141103T195300Z~b942d0faa8~64bit > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, path, replaces > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 944 bytes from tls/[222.248.102.244]:38620 at 20:57:02.205719: > ------------------------------------------------------------------------ > REGISTER sip:abc-xyz.com;transport=TLS SIP/2.0 > Via: SIP/2.0/TLS 222.248.102.244:38620 > ;branch=z9hG4bK-d8754z-59d493abef215091-1---d8754z-;rport > Max-Forwards: 70 > Contact: < > sip:1000 at 192.168.10.2:41120;rinstance=57d4e16c725a51e6;transport=TLS > >;expires=0 > To: > From: ;tag=4a2eae22 > Call-ID: N2RiMjE2ZTQxNzYzMGY3ZmM3ZGJhODM4NTMzZmQ1YTA. > CSeq: 3 REGISTER > Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, > SUBSCRIBE > Supported: replaces, norefersub, extended-refer, timer, > X-cisco-serviceuri > User-Agent: Zoiper r27147 > Authorization: Digest username="1000",realm="abc-xyz.com > ",nonce="374fdd98-7994-11e4-bf9c-1182ebca2429",uri=" > sip:abc-xyz.com;transport=TLS > ",response="5168bf9afffe9adacef7d898bd6bec9b",cnonce="80108fa519303990cfd2d46639d767ee",nc=00000002,qop=auth,algorithm=MD5 > Allow-Events: presence, kpml > Content-Length: 0 > > > ------------------------------------------------------------------------ > send 598 bytes to tls/[222.248.102.244]:38620 at 20:57:02.207249: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/TLS 222.248.102.244:38620 > ;branch=z9hG4bK-d8754z-59d493abef215091-1---d8754z-;rport=38620 > From: ;tag=4a2eae22 > To: ;tag=40Spr6U5jtH2N > Call-ID: N2RiMjE2ZTQxNzYzMGY3ZmM3ZGJhODM4NTMzZmQ1YTA. > CSeq: 3 REGISTER > Date: Mon, 01 Dec 2014 19:57:02 GMT > User-Agent: > FreeSWITCH-mod_sofia/1.4.13+git~20141103T195300Z~b942d0faa8~64bit > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, path, replaces > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 681 bytes from tls/[222.248.102.244]:38620 at 20:57:02.400564: > ------------------------------------------------------------------------ > REGISTER sip:abc-xyz.com;transport=TLS SIP/2.0 > Via: SIP/2.0/TLS 222.248.102.244:38620 > ;branch=z9hG4bK-d8754z-d738a1a836608006-1---d8754z-;rport > Max-Forwards: 70 > Contact: < > sip:1000 at 222.248.102.244:38620;rinstance=43166fee0aaac1bc;transport=TLS> > To: > From: ;tag=979e5f46 > Call-ID: MTJhNjIzMDM1MDEwMDVjZDRkOGMyYjBlMWU4ZDEzY2Y. > CSeq: 1 REGISTER > Expires: 60 > Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, > SUBSCRIBE > Supported: replaces, norefersub, extended-refer, timer, > X-cisco-serviceuri > User-Agent: Zoiper r27147 > Allow-Events: presence, kpml > Content-Length: 0 > > > > ------------------------------------------------------------------------ > send 694 bytes to tls/[222.248.102.244]:38620 at 20:57:02.401167: > ------------------------------------------------------------------------ > SIP/2.0 401 Unauthorized > Via: SIP/2.0/TLS 222.248.102.244:38620 > ;branch=z9hG4bK-d8754z-d738a1a836608006-1---d8754z-;rport=38620 > From: ;tag=979e5f46 > To: ;tag=59jFt1c9F37mH > Call-ID: MTJhNjIzMDM1MDEwMDVjZDRkOGMyYjBlMWU4ZDEzY2Y. > CSeq: 1 REGISTER > User-Agent: > FreeSWITCH-mod_sofia/1.4.13+git~20141103T195300Z~b942d0faa8~64bit > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, path, replaces > WWW-Authenticate: Digest realm="abc-xyz.com", > nonce="37aaa44e-7994-11e4-bf9d-1182ebca2429", algorithm=MD5, qop="auth" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 950 bytes from tls/[222.248.102.244]:38620 at 20:57:02.655864: > ------------------------------------------------------------------------ > REGISTER sip:abc-xyz.com;transport=TLS SIP/2.0 > Via: SIP/2.0/TLS 222.248.102.244:38620 > ;branch=z9hG4bK-d8754z-dfaa7130675d6cfa-1---d8754z-;rport > Max-Forwards: 70 > Contact: < > sip:1000 at 222.248.102.244:38620;rinstance=43166fee0aaac1bc;transport=TLS> > To: > From: ;tag=979e5f46 > Call-ID: MTJhNjIzMDM1MDEwMDVjZDRkOGMyYjBlMWU4ZDEzY2Y. > CSeq: 2 REGISTER > Expires: 60 > Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, > SUBSCRIBE > Supported: replaces, norefersub, extended-refer, timer, > X-cisco-serviceuri > User-Agent: Zoiper r27147 > Authorization: Digest username="1000",realm="abc-xyz.com > ",nonce="37aaa44e-7994-11e4-bf9d-1182ebca2429",uri=" > sip:abc-xyz.com;transport=TLS > ",response="f1667094626fc10ca76a070438497e6f",cnonce="dc790bffe285311f0b903bd515a3e741",nc=00000001,qop=auth,algorithm=MD5 > Allow-Events: presence, kpml > Content-Length: 0 > > ------------------------------------------------------------------------ > send 693 bytes to tls/[222.248.102.244]:38620 at 20:57:02.657688: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/TLS 222.248.102.244:38620 > ;branch=z9hG4bK-d8754z-dfaa7130675d6cfa-1---d8754z-;rport=38620 > From: ;tag=979e5f46 > To: ;tag=6jc8UvXcDcy7c > Call-ID: MTJhNjIzMDM1MDEwMDVjZDRkOGMyYjBlMWU4ZDEzY2Y. > CSeq: 2 REGISTER > Contact: < > sip:1000 at 222.248.102.244:38620;rinstance=43166fee0aaac1bc;transport=TLS > >;expires=60 > Date: Mon, 01 Dec 2014 19:57:02 GMT > User-Agent: > FreeSWITCH-mod_sofia/1.4.13+git~20141103T195300Z~b942d0faa8~64bit > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, path, replaces > Content-Length: 0 > > > > > > > *From: *Brian West > *To: *FreeSWITCH Users Help > *Date: *December 1, 2014 at 5:19:30 PM GMT+3 > *Reply-To: *FreeSWITCH Users Help > *Subject: **Re: [Freeswitch-users] FreeSWITCH TLS not able to receive > calls* > > > What endpoints are involved? have you looked at 'sofia loglevel all 9' > output and see if it gives you a clue? > > On Fri, Nov 28, 2014 at 6:24 PM, Ahmed Habiba wrote: > >> Dears, >> >> I?ve configured FreeSWITCH with the below version with TLS/SRTPas per the >> recommendation in page ?https://wiki.freeswitch.org/wiki/SIP_TLS? and it >> was strait forward, and I was able to connect and make make calls using >> zoiper, but I was not able to receive any calls after enabling the TLS/SRTP. >> >> *"FreeSWITCH Version 1.4.13+git~20141103T195300Z~b942d0faa8~64bit (git >> b942d0f 2014-11-03 19:53:00Z 64bit)**?* >> >> Your kind feedback will be appreciate. >> >> Thanks, >> >> Ahmed Habiba. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > > *Brian West* > brian at freeswitch.org > > > *Twitter: @FreeSWITCH , @briankwest* > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest > > > > > > > -- > Thanks and Best Regards, > > Ahmed Habiba > > Mob: +20 10 37 82 970 > > *Success: believe (Vision) plus commitment (Action)* > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141203/9c3bff4f/attachment-0001.html From ahabiba at gmail.com Thu Dec 4 00:31:16 2014 From: ahabiba at gmail.com (Ahmed Habiba) Date: Thu, 4 Dec 2014 00:31:16 +0300 Subject: [Freeswitch-users] FreeSWITCH TLS not able to receive call In-Reply-To: References: Message-ID: you mean the below one, to cut some of it? : > The issue as shown below it keeps registering till finally zipper restart on mobile. > > One more point I?m configuring Zoiper without using RPORT for either SRTP or TLS. > > ?2014-12-01 20:57:24.003230 [NOTICE] switch_ivr_originate.c:3467 Hangup sofia/internal/sip:1000 at 222.248.102.244 :38614 [CS_CONSUME_MEDIA] [NO_ANSWER]" > > > > recv 1290 bytes from tls/[222.215.195.234]:2896 at 20:56:54.344450: > ------------------------------------------------------------------------ > INVITE <>sip:1000 at abc-xyz.com <> SIP/2.0 > v: SIP/2.0/TLS 192.168.8.109:41659;rport;branch=z9hG4bKPjvnvRZjpAXahzIV7S8SRgS5vwKmYI9WFS;alias > Max-Forwards: 70 > f: < <>sip:1006 at abc-xyz.com <>>;tag=weeMqzInPcNLPd6s5IX63c8be3DMvGkU > t: < <>sip:1000 at abc-xyz.com <>> > m: < <>sip:1006 at 222.215.195.234:2896;transport=TLS;ob <>> > i: sjprJQszC2LHvY6jRr-lm9rau9V1PlYX > CSeq: 32737 INVITE > Route: < <>sip:abc-xyz.com;transport=tls;lr <>> > k: replaces, 100rel, timer, norefersub > x: 1800 > Min-SE: 90 > User-Agent: CSipSimple_GT-S5830-10/r2450 > Proxy-Authorization: Digest username="1006", realm="abc-xyz.com ", nonce="329456ee-7994-11e4-bf79-1182ebca2429", uri=" <>sip:1000 at abc-xyz.com <>", response="ab3a0f2fe9207ced62eebd8c9b8c32b4", algorithm=MD5, cnonce="iLa-C.wpnBSeTQjWiXZK3H3PGYhi6EGt", qop=auth, nc=00000001 > c: application/sdp > l: 469 > > v=0 > o=- 3626452487 3626452487 IN IP4 192.168.8.109 > s=pjmedia > c=IN IP4 192.168.8.109 > t=0 0 > m=audio 4000 RTP/SAVP 98 0 8 101 > c=IN IP4 192.168.8.109 > a=rtcp:4001 IN IP4 192.168.8.109 > a=sendrecv > a=rtpmap:98 SILK/16000 > a=fmtp:98 useinbandfec=0 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:Vci7cNdSFTkkLnqz+qRkqCctPvRT6jIOrNc5KMbz > a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:nW3ECq5YHHnrY2q5u7XZE4fvPfMOAWKQ+ehOp7zV > ------------------------------------------------------------------------ > > send 405 bytes to tls/[222.215.195.234]:2896 at 20:56:54.344687: > ------------------------------------------------------------------------ > SIP/2.0 100 Trying > Via: SIP/2.0/TLS 192.168.8.109:41659;rport=2896;branch=z9hG4bKPjvnvRZjpAXahzIV7S8SRgS5vwKmYI9WFS;alias;received=222.215.195.234 > f: < <>sip:1006 at abc-xyz.com <>>;tag=weeMqzInPcNLPd6s5IX63c8be3DMvGkU > t: < <>sip:1000 at abc-xyz.com <>> > i: sjprJQszC2LHvY6jRr-lm9rau9V1PlYX > CSeq: 32737 INVITE > User-Agent: FreeSWITCH-mod_sofia/1.4.13+git~20141103T195300Z~b942d0faa8~64bit > Content-Length: 0 > > > > > > > send 2133 bytes to tls/[222.248.102.244]:38614 at 20:56:54.360608: > ------------------------------------------------------------------------ > INVITE <>sip:1000 at 222.248.102.244:38614;rinstance=98c51296f2d531cc;transport=TLS <> SIP/2.0 > Via: SIP/2.0/TLS 176.31.245.17;branch=z9hG4bKjXZvg3cX76Krj > Max-Forwards: 69 > From: "Extension 1006" < <>sip:1006 at 176.31.245.17 <>>;tag=15DcKN9tUZeaK > To: < <>sip:1000 at 222.248.102.244:38614;rinstance=98c51296f2d531cc;transport=TLS <>> > Call-ID: 0a3e07c5-f437-1232-9bba-d2ab2784dc6a > CSeq: 68395843 INVITE > Contact: < <>sip:mod_sofia at 176.31.245.17:5061;transport=tls <>> > User-Agent: FreeSWITCH-mod_sofia/1.4.13+git~20141103T195300Z~b942d0faa8~64bit > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, path, replaces > Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 1076 > X-FS-Support: update_display,send_info > Remote-Party-ID: "Extension 1006" < <>sip:1006 at 176.31.245.17 <>>;party=calling;screen=yes;privacy=off > > v=0 > o=FreeSWITCH 1417445672 1417445673 IN IP4 176.31.245.17 > s=FreeSWITCH > c=IN IP4 176.31.245.17 > t=0 0 > m=audio 18142 RTP/SAVP 0 8 101 13 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=crypto:1 AEAD_AES_256_GCM_8 inline:6p/uXB7tN5SG9vHP/DMurij6BaZIOKcP4Tle0kjo2lwL1Oa4z9WGyZtk4zs > a=crypto:2 AEAD_AES_128_GCM_8 inline:hGeJA6XQV1V35oTu2141o7mD64dFDRbeCbcf6g > a=crypto:3 AES_CM_256_HMAC_SHA1_80 inline:XdZwwcl6kMRrWY7Vkrbozf/o1awka9lef0UjQRwvnOnnrbAgzXIm8s3yu7o8DQ > a=crypto:4 AES_CM_192_HMAC_SHA1_80 inline:KtM92m0ac2musYPoXLDJb4rUbEMYAwMegPH9RMtzwzPb+0O93e0 > a=crypto:5 AES_CM_128_HMAC_SHA1_80 inline:PtOpdIk2LTSCSDVzPaSN6AV2xosBrJUTd4mvKIGh > a=crypto:6 AES_CM_256_HMAC_SHA1_32 inline:ZHzWxajRlQp8IuzX5CuFM1SRkL0huC62ukX583+Vg0LBiK9t2NrpP8FNwdkckQ > a=crypto:7 AES_CM_192_HMAC_SHA1_32 inline:S9BWrDxGFYpU3P9o3HD0Z97W5jwo8vD+E1sJi821r4/B+cdukTw > a=crypto:8 AES_CM_128_HMAC_SHA1_32 inline:l7C0wU29AjF2ZMnCTvks+1ytd/UiXHqN8UIsM6vf > a=crypto:9 AES_CM_128_NULL_AUTH inline:tv97QsO7+S9NLTXrD02omyWoyM7I2KaWaimUgq99 > a=ptime:20 > ------------------------------------------------------------------------ > > > REGISTER <>sip:abc-xyz.com;transport=TLS <> SIP/2.0 > Via: SIP/2.0/TLS 192.168.10.2:41120;branch=z9hG4bK-d8754z-903949acb1d10083-1---d8754z-;rport > Max-Forwards: 70 > Contact: < <>sip:1000 at 192.168.10.2:41120;rinstance=57d4e16c725a51e6;transport=TLS <>> > To: < <>sip:1000 at abc-xyz.com;transport=TLS <>> > From: < <>sip:1000 at abc-xyz.com;transport=TLS <>>;tag=4a2eae22 > Call-ID: N2RiMjE2ZTQxNzYzMGY3ZmM3ZGJhODM4NTMzZmQ1YTA. > CSeq: 1 REGISTER > Expires: 60 > Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE > Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri > User-Agent: Zoiper r27147 > Allow-Events: presence, kpml > Content-Length: 0 > > > ------------------------------------------------------------------------ > send 716 bytes to tls/[222.248.102.244]:38620 at 20:57:01.806433: > ------------------------------------------------------------------------ > SIP/2.0 401 Unauthorized > Via: SIP/2.0/TLS 192.168.10.2:41120;branch=z9hG4bK-d8754z-903949acb1d10083-1---d8754z-;rport=38620;received=222.248.102.244 > From: < <>sip:1000 at abc-xyz.com;transport=TLS <>>;tag=4a2eae22 > To: < <>sip:1000 at abc-xyz.com;transport=TLS <>>;tag=2e74mgtyr84ve > Call-ID: N2RiMjE2ZTQxNzYzMGY3ZmM3ZGJhODM4NTMzZmQ1YTA. > CSeq: 1 REGISTER > User-Agent: FreeSWITCH-mod_sofia/1.4.13+git~20141103T195300Z~b942d0faa8~64bit > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, path, replaces > WWW-Authenticate: Digest realm="abc-xyz.com ", nonce="374fdd98-7994-11e4-bf9c-1182ebca2429", algorithm=MD5, qop="auth" > Content-Length: 0 > > > ------------------------------------------------------------------------ > recv 944 bytes from tls/[222.248.102.244]:38620 at 20:57:01.995662: > ------------------------------------------------------------------------ > REGISTER <>sip:abc-xyz.com;transport=TLS <> SIP/2.0 > Via: SIP/2.0/TLS 192.168.10.2:41120;branch=z9hG4bK-d8754z-8ca0e782336414eb-1---d8754z-;rport > Max-Forwards: 70 > Contact: < <>sip:1000 at 192.168.10.2:41120;rinstance=57d4e16c725a51e6;transport=TLS <>> > To: < <>sip:1000 at abc-xyz.com;transport=TLS <>> > From: < <>sip:1000 at abc-xyz.com;transport=TLS <>>;tag=4a2eae22 > Call-ID: N2RiMjE2ZTQxNzYzMGY3ZmM3ZGJhODM4NTMzZmQ1YTA. > CSeq: 2 REGISTER > Expires: 60 > Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE > Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri > User-Agent: Zoiper r27147 > Authorization: Digest username="1000",realm="abc-xyz.com ",nonce="374fdd98-7994-11e4-bf9c-1182ebca2429",uri=" <>sip:abc-xyz.com;transport=TLS <>",response="8d7e108ff98035d37424c32166fe0253",cnonce="107cf281c3b57888a4bfd1a9c3776098",nc=00000001,qop=auth,algorithm=MD5 > Allow-Events: presence, kpml > Content-Length: 0 > > > ------------------------------------------------------------------------ > send 712 bytes to tls/[222.248.102.244]:38620 at 20:57:01.997931: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/TLS 192.168.10.2:41120;branch=z9hG4bK-d8754z-8ca0e782336414eb-1---d8754z-;rport=38620;received=222.248.102.244 > From: < <>sip:1000 at abc-xyz.com;transport=TLS <>>;tag=4a2eae22 > To: < <>sip:1000 at abc-xyz.com;transport=TLS <>>;tag=3Q0XpBB2NHUFa > Call-ID: N2RiMjE2ZTQxNzYzMGY3ZmM3ZGJhODM4NTMzZmQ1YTA. > CSeq: 2 REGISTER > Contact: < <>sip:1000 at 192.168.10.2:41120;rinstance=57d4e16c725a51e6;transport=TLS <>>;expires=60 > Date: Mon, 01 Dec 2014 19:57:01 GMT > User-Agent: FreeSWITCH-mod_sofia/1.4.13+git~20141103T195300Z~b942d0faa8~64bit > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, path, replaces > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 944 bytes from tls/[222.248.102.244]:38620 at 20:57:02.205719: > ------------------------------------------------------------------------ > REGISTER <>sip:abc-xyz.com;transport=TLS <> SIP/2.0 > Via: SIP/2.0/TLS 222.248.102.244:38620;branch=z9hG4bK-d8754z-59d493abef215091-1---d8754z-;rport > Max-Forwards: 70 > Contact: < <>sip:1000 at 192.168.10.2:41120;rinstance=57d4e16c725a51e6;transport=TLS <>>;expires=0 > To: < <>sip:1000 at abc-xyz.com;transport=TLS <>> > From: < <>sip:1000 at abc-xyz.com;transport=TLS <>>;tag=4a2eae22 > Call-ID: N2RiMjE2ZTQxNzYzMGY3ZmM3ZGJhODM4NTMzZmQ1YTA. > CSeq: 3 REGISTER > Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE > Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri > User-Agent: Zoiper r27147 > Authorization: Digest username="1000",realm="abc-xyz.com ",nonce="374fdd98-7994-11e4-bf9c-1182ebca2429",uri=" <>sip:abc-xyz.com;transport=TLS <>",response="5168bf9afffe9adacef7d898bd6bec9b",cnonce="80108fa519303990cfd2d46639d767ee",nc=00000002,qop=auth,algorithm=MD5 > Allow-Events: presence, kpml > Content-Length: 0 > > ------------------------------------------------------------------------ > send 598 bytes to tls/[222.248.102.244]:38620 at 20:57:02.207249: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/TLS 222.248.102.244:38620;branch=z9hG4bK-d8754z-59d493abef215091-1---d8754z-;rport=38620 > From: < <>sip:1000 at abc-xyz.com;transport=TLS <>>;tag=4a2eae22 > To: < <>sip:1000 at abc-xyz.com;transport=TLS <>>;tag=40Spr6U5jtH2N > Call-ID: N2RiMjE2ZTQxNzYzMGY3ZmM3ZGJhODM4NTMzZmQ1YTA. > CSeq: 3 REGISTER > Date: Mon, 01 Dec 2014 19:57:02 GMT > User-Agent: FreeSWITCH-mod_sofia/1.4.13+git~20141103T195300Z~b942d0faa8~64bit > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, path, replaces > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 681 bytes from tls/[222.248.102.244]:38620 at 20:57:02.400564: > ------------------------------------------------------------------------ > REGISTER <>sip:abc-xyz.com;transport=TLS <> SIP/2.0 > Via: SIP/2.0/TLS 222.248.102.244:38620;branch=z9hG4bK-d8754z-d738a1a836608006-1---d8754z-;rport > Max-Forwards: 70 > Contact: < <>sip:1000 at 222.248.102.244:38620;rinstance=43166fee0aaac1bc;transport=TLS <>> > To: < <>sip:1000 at abc-xyz.com;transport=TLS <>> > From: < <>sip:1000 at abc-xyz.com;transport=TLS <>>;tag=979e5f46 > Call-ID: MTJhNjIzMDM1MDEwMDVjZDRkOGMyYjBlMWU4ZDEzY2Y. > CSeq: 1 REGISTER > Expires: 60 > Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE > Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri > User-Agent: Zoiper r27147 > Allow-Events: presence, kpml > Content-Length: 0 > > > ------------------------------------------------------------------------ > send 694 bytes to tls/[222.248.102.244]:38620 at 20:57:02.401167: > ------------------------------------------------------------------------ > SIP/2.0 401 Unauthorized > Via: SIP/2.0/TLS 222.248.102.244:38620;branch=z9hG4bK-d8754z-d738a1a836608006-1---d8754z-;rport=38620 > From: < <>sip:1000 at abc-xyz.com;transport=TLS <>>;tag=979e5f46 > To: < <>sip:1000 at abc-xyz.com;transport=TLS <>>;tag=59jFt1c9F37mH > Call-ID: MTJhNjIzMDM1MDEwMDVjZDRkOGMyYjBlMWU4ZDEzY2Y. > CSeq: 1 REGISTER > User-Agent: FreeSWITCH-mod_sofia/1.4.13+git~20141103T195300Z~b942d0faa8~64bit > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, path, replaces > WWW-Authenticate: Digest realm="abc-xyz.com ", nonce="37aaa44e-7994-11e4-bf9d-1182ebca2429", algorithm=MD5, qop="auth" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 950 bytes from tls/[222.248.102.244]:38620 at 20:57:02.655864: > ------------------------------------------------------------------------ > REGISTER <>sip:abc-xyz.com;transport=TLS <> SIP/2.0 > Via: SIP/2.0/TLS 222.248.102.244:38620;branch=z9hG4bK-d8754z-dfaa7130675d6cfa-1---d8754z-;rport > Max-Forwards: 70 > Contact: < <>sip:1000 at 222.248.102.244:38620;rinstance=43166fee0aaac1bc;transport=TLS <>> > To: < <>sip:1000 at abc-xyz.com;transport=TLS <>> > From: < <>sip:1000 at abc-xyz.com;transport=TLS <>>;tag=979e5f46 > Call-ID: MTJhNjIzMDM1MDEwMDVjZDRkOGMyYjBlMWU4ZDEzY2Y. > CSeq: 2 REGISTER > Expires: 60 > Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE > Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri > User-Agent: Zoiper r27147 > Authorization: Digest username="1000",realm="abc-xyz.com ",nonce="37aaa44e-7994-11e4-bf9d-1182ebca2429",uri=" <>sip:abc-xyz.com;transport=TLS <>",response="f1667094626fc10ca76a070438497e6f",cnonce="dc790bffe285311f0b903bd515a3e741",nc=00000001,qop=auth,algorithm=MD5 > Allow-Events: presence, kpml > Content-Length: 0 > > ------------------------------------------------------------------------ > send 693 bytes to tls/[222.248.102.244]:38620 at 20:57:02.657688: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/TLS 222.248.102.244:38620;branch=z9hG4bK-d8754z-dfaa7130675d6cfa-1---d8754z-;rport=38620 > From: < <>sip:1000 at abc-xyz.com;transport=TLS <>>;tag=979e5f46 > To: < <>sip:1000 at abc-xyz.com;transport=TLS <>>;tag=6jc8UvXcDcy7c > Call-ID: MTJhNjIzMDM1MDEwMDVjZDRkOGMyYjBlMWU4ZDEzY2Y. > CSeq: 2 REGISTER > Contact: < <>sip:1000 at 222.248.102.244:38620;rinstance=43166fee0aaac1bc;transport=TLS <>>;expires=60 > Date: Mon, 01 Dec 2014 19:57:02 GMT > User-Agent: FreeSWITCH-mod_sofia/1.4.13+git~20141103T195300Z~b942d0faa8~64bit > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, path, replaces > Content-Length: 0 > > > > >> >> >> From: Brian West > >> To: FreeSWITCH Users Help > >> Date: December 1, 2014 at 5:19:30 PM GMT+3 >> Reply-To: FreeSWITCH Users Help > >> Subject: Re: [Freeswitch-users] FreeSWITCH TLS not able to receive calls >> >> >> What endpoints are involved? have you looked at 'sofia loglevel all 9' output and see if it gives you a clue? >> >> On Fri, Nov 28, 2014 at 6:24 PM, Ahmed Habiba > wrote: >> Dears, >> >> I?ve configured FreeSWITCH with the below version with TLS/SRTPas per the recommendation in page ?https://wiki.freeswitch.org/wiki/SIP_TLS ? and it was strait forward, and I was able to connect and make make calls using zoiper, but I was not able to receive any calls after enabling the TLS/SRTP. >> >> "FreeSWITCH Version 1.4.13+git~20141103T195300Z~b942d0faa8~64bit (git b942d0f 2014-11-03 19:53:00Z 64bit)? >> >> Your kind feedback will be appreciate. >> >> Thanks, >> >> Ahmed Habiba. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> -- >> Brian West >> brian at freeswitch.org >> >> Twitter: @FreeSWITCH , @briankwest >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378) >> iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype: <>briankwest <> >> > > > > > -- > Thanks and Best Regards, > > Ahmed Habiba > > Mob: +20 10 37 82 970 > > Success: believe (Vision) plus commitment (Action) > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -- > Brian West > brian at freeswitch.org > > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378) > iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141204/a056497a/attachment-0001.html From ing.antonyam at gmail.com Thu Dec 4 01:28:22 2014 From: ing.antonyam at gmail.com (Antony Aguirre Morales) Date: Wed, 3 Dec 2014 16:28:22 -0600 Subject: [Freeswitch-users] Interconect 3 box freeswitch. Message-ID: Hi, I would like to configure three freeswitch servers but I need that these servers have the same directory of extensions, is this escenario possible? If this is not possible, there a way where two freeswitch servers can query the same database. For example: (graphic) [10.30.0.0]Freeswitch 1 ---- | [10.20.0.0]Freeswitch 2 ---- | ----- Directory of extention [10.10.0.0]Freeswitch 3 ---- | -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141203/21962e6d/attachment.html From t.mahe at b-and-c.net Thu Dec 4 01:38:26 2014 From: t.mahe at b-and-c.net (=?windows-1252?Q?Tristan_Mah=E9?=) Date: Wed, 03 Dec 2014 14:38:26 -0800 Subject: [Freeswitch-users] Interconect 3 box freeswitch. In-Reply-To: References: Message-ID: <547F90E2.6080503@b-and-c.net> Hi Antony, You may look at xml_curl for these needs. Le 03/12/2014 14:28, Antony Aguirre Morales a ?crit : > Hi, > > I would like to configure three freeswitch servers but I need that these > servers have the same directory of extensions, is this escenario possible? > > If this is not possible, there a way where two freeswitch servers can > query the same database. For example: (graphic) > > [10.30.0.0]Freeswitch 1 ---- | > [10.20.0.0]Freeswitch 2 ---- | ----- Directory of extention > [10.10.0.0]Freeswitch 3 ---- | > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 473 bytes Desc: OpenPGP digital signature Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141203/94d1f7ac/attachment.bin From andretodd at verizon.net Thu Dec 4 05:00:13 2014 From: andretodd at verizon.net (Andre DeMattia) Date: Wed, 03 Dec 2014 21:00:13 -0500 Subject: [Freeswitch-users] emails not coming Message-ID: <0fe501d00f66$0ba778b0$22f66a10$@verizon.net> Hi, I no longer get the FreeSWITCH emails. Can someone check on that for me? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141203/52e42a82/attachment.html From iwada.bassey at gmail.com Thu Dec 4 11:06:56 2014 From: iwada.bassey at gmail.com (Iwada Eja) Date: Thu, 4 Dec 2014 03:06:56 -0500 Subject: [Freeswitch-users] emails not coming In-Reply-To: <0fe501d00f66$0ba778b0$22f66a10$@verizon.net> References: <0fe501d00f66$0ba778b0$22f66a10$@verizon.net> Message-ID: Hi Andre, You still subscribed to the list. Checked your Spam? Probably your Firewall, Content Filter, or Email Security Policy? On Wed, Dec 3, 2014 at 9:00 PM, Andre DeMattia wrote: > Hi, I no longer get the FreeSWITCH emails. Can someone check on that for > me? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 Iwada -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141204/d65e1aef/attachment.html From iwada.bassey at gmail.com Thu Dec 4 11:43:31 2014 From: iwada.bassey at gmail.com (Iwada Eja) Date: Thu, 4 Dec 2014 03:43:31 -0500 Subject: [Freeswitch-users] emails not coming In-Reply-To: References: <0fe501d00f66$0ba778b0$22f66a10$@verizon.net> Message-ID: ... On a Second Look, You have list delivery disabled. You can Log in to your options page and re enable it. On Thu, Dec 4, 2014 at 3:06 AM, Iwada Eja wrote: > Hi Andre, > You still subscribed to the list. Checked your Spam? Probably your > Firewall, Content Filter, or Email Security Policy? > > On Wed, Dec 3, 2014 at 9:00 PM, Andre DeMattia > wrote: > >> Hi, I no longer get the FreeSWITCH emails. Can someone check on that for >> me? >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 > Iwada > -- Kind Regards Iwada -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141204/683cb10d/attachment.html From mbodbg at gmx.net Thu Dec 4 16:24:59 2014 From: mbodbg at gmx.net (mbo) Date: Thu, 4 Dec 2014 14:24:59 +0100 Subject: [Freeswitch-users] Status of chat in mod_verto Message-ID: <350D7621-8242-4C63-A1BB-575E1397D46D@gmx.net> Hello, we are playing around with mod_verto. The audio part is working fine, but what is the status with the chat implementation. On the demo page on the internet (bbb.freeswitch.org), I can see that chat is working at least in a conference. But does chat also work between 2 users logged in via mod_verto? On the demo page I can see that code handling chat is commented out, isn?t it ready yet? We modified the script a bit and we are at least able to see that chat messages are received on the server like: 2014-12-04 12:10:18.228695 [ALERT] mod_verto.c:1284 READ 192.168.0.214:56164 [{ "jsonrpc": "2.0", "method": "verto.info", "params": { "msg": { "to": "1009", "body": "test" }, "sessid": "f7d3c02c-71ac-1e42-9a6b-0d7cc0d355ad" }, "id": 14 }] 2014-12-04 12:10:18.228695 [ALERT] mod_verto.c:594 WRITE 192.168.0.214:56164 [{ "jsonrpc": "2.0", "id": 14, "result": { "message": "SENT", "sessid": "f7d3c02c-71ac-1e42-9a6b-0d7cc0d355ad" } }] but we do not see anything on the destination client. Does anyone else have chat already working? Thanks and Regards Markus From areski at gmail.com Thu Dec 4 21:20:47 2014 From: areski at gmail.com (Areski) Date: Thu, 4 Dec 2014 19:20:47 +0100 Subject: [Freeswitch-users] Doc-Sprint Friday 12 December 2014 Message-ID: Hi everyone, We are planning to organize an other doc sprint on *Friday 12 December at 10am CT*. It will be 4 hours long but you can join for less time. The Doc-sprint will focus on migrating the remaining pages from MediaWiki ( https://wiki.freeswitch.org) to Confluence Wiki ( https://freeswitch.org/confluence). We will use an FS IRC channel during the sprint: *#freeswitch-docs* and will track our work on the spreadsheet: https://docs.google.com/spreadsheets/d/1qsG-kRymvKlNBapnBLw86W130VdbnK6naYapbR_UNds/edit?pli=1#gid=1187898333 During the sprint, please change the URL's "Status" you are working on to "Editing" with your name next to it so we don't duplicate work. Some extra information: - https://freeswitch.org/confluence/display/FREESWITCH/Wiki+Migration - https://freeswitch.org/confluence/display/FREESWITCH/Contributing+Documentation We hope to get a maximum number of people signed up! Peoples confirmed so far: - Italo Rossi (+4) - Areski Belaid So, who is in? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141204/88cbd69a/attachment-0001.html From iwada.bassey at gmail.com Thu Dec 4 23:00:47 2014 From: iwada.bassey at gmail.com (Iwada Eja) Date: Thu, 4 Dec 2014 15:00:47 -0500 Subject: [Freeswitch-users] Doc-Sprint Friday 12 December 2014 In-Reply-To: References: Message-ID: I'm In On Thu, Dec 4, 2014 at 1:20 PM, Areski wrote: > Hi everyone, > > We are planning to organize an other doc sprint on *Friday 12 December at > 10am CT*. > It will be 4 hours long but you can join for less time. > > The Doc-sprint will focus on migrating the remaining pages from MediaWiki ( > https://wiki.freeswitch.org) to Confluence Wiki ( > https://freeswitch.org/confluence). > > We will use an FS IRC channel during the sprint: *#freeswitch-docs* > and will track our work on the spreadsheet: > https://docs.google.com/spreadsheets/d/1qsG-kRymvKlNBapnBLw86W130VdbnK6naYapbR_UNds/edit?pli=1#gid=1187898333 > > During the sprint, please change the URL's "Status" you are working on to > "Editing" with your name next to it so we don't duplicate work. > > Some extra information: > - https://freeswitch.org/confluence/display/FREESWITCH/Wiki+Migration > - > https://freeswitch.org/confluence/display/FREESWITCH/Contributing+Documentation > > We hope to get a maximum number of people signed up! > > Peoples confirmed so far: > - Italo Rossi (+4) > - Areski Belaid > > > So, who is in? > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 Iwada -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141204/a9b9f7b5/attachment.html From brian at freeswitch.org Thu Dec 4 23:22:15 2014 From: brian at freeswitch.org (Brian West) Date: Thu, 4 Dec 2014 14:22:15 -0600 Subject: [Freeswitch-users] FreeSWITCH TLS not able to receive call In-Reply-To: References: Message-ID: Yes. On Wed, Dec 3, 2014 at 3:31 PM, Ahmed Habiba wrote: > you mean the below one, to cut some of it? : > > "rtp_sdes_suites=AEAD_AES_256_GCM_8|AEAD_AES_128_GCM_8|AES_CM_256_HMAC_SHA1_80|AES_CM_192_HMAC_SHA1_80|AES_CM_128_HMAC_SHA1_80|AES_CM > > _256_HMAC_SHA1_32|AES_CM_192_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_32|AES_CM_128_NULL_AUTH? > /> > > > > > > > *From: *Brian West > *To: *FreeSWITCH Users Help > *Date: *December 3, 2014 at 11:44:31 PM GMT+3 > *Reply-To: *FreeSWITCH Users Help > *Subject: **Re: [Freeswitch-users] FreeSWITCH TLS not able to receive > calls* > > > I would try limiting the crypto suites on the FS side, Look in vars.xml > for details, I know if we send the whole crypt suite list at a polycom on > pretty much any firmware the phone goes rum dumb and starts to echo our > SDP's back at us. :P > > On Wed, Dec 3, 2014 at 12:36 PM, Ahmed Habiba wrote: > >> Dear Brian, >> >> >> I?m using "zoiper 1.19.7? on android, the issue appears only one I enable >> SRTP, for TLS only it is working very fine. >> >> The issue as shown below it keeps registering till finally zipper restart >> on mobile. >> >> One more point I?m configuring Zoiper without using RPORT for either SRTP >> or TLS. >> >> *?2014-12-01 20:57:24.003230 [NOTICE] switch_ivr_originate.c:3467 Hangup >> sofia/internal/sip:1000 at 222.248.102.244 <1000 at 222.248.102.244>:38614 >> [CS_CONSUME_MEDIA] [NO_ANSWER]"* >> >> >> >> recv 1290 bytes from tls/[222.215.195.234]:2896 at 20:56:54.344450: >> >> ------------------------------------------------------------------------ >> INVITE sip:1000 at abc-xyz.com SIP/2.0 >> v: SIP/2.0/TLS 192.168.8.109:41659 >> ;rport;branch=z9hG4bKPjvnvRZjpAXahzIV7S8SRgS5vwKmYI9WFS;alias >> Max-Forwards: 70 >> f: ;tag=weeMqzInPcNLPd6s5IX63c8be3DMvGkU >> t: >> m: >> i: sjprJQszC2LHvY6jRr-lm9rau9V1PlYX >> CSeq: 32737 INVITE >> Route: >> k: replaces, 100rel, timer, norefersub >> x: 1800 >> Min-SE: 90 >> User-Agent: CSipSimple_GT-S5830-10/r2450 >> Proxy-Authorization: Digest username="1006", realm="abc-xyz.com", >> nonce="329456ee-7994-11e4-bf79-1182ebca2429", uri="sip:1000 at abc-xyz.com", >> response="ab3a0f2fe9207ced62eebd8c9b8c32b4", algorithm=MD5, >> cnonce="iLa-C.wpnBSeTQjWiXZK3H3PGYhi6EGt", qop=auth, nc=00000001 >> c: application/sdp >> l: 469 >> >> v=0 >> o=- 3626452487 3626452487 IN IP4 192.168.8.109 >> s=pjmedia >> c=IN IP4 192.168.8.109 >> t=0 0 >> m=audio 4000 RTP/SAVP 98 0 8 101 >> c=IN IP4 192.168.8.109 >> a=rtcp:4001 IN IP4 192.168.8.109 >> a=sendrecv >> a=rtpmap:98 SILK/16000 >> a=fmtp:98 useinbandfec=0 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=crypto:1 AES_CM_128_HMAC_SHA1_80 >> inline:Vci7cNdSFTkkLnqz+qRkqCctPvRT6jIOrNc5KMbz >> a=crypto:2 AES_CM_128_HMAC_SHA1_32 >> inline:nW3ECq5YHHnrY2q5u7XZE4fvPfMOAWKQ+ehOp7zV >> >> ------------------------------------------------------------------------ >> >> send 405 bytes to tls/[222.215.195.234]:2896 at 20:56:54.344687: >> >> ------------------------------------------------------------------------ >> SIP/2.0 100 Trying >> Via: SIP/2.0/TLS 192.168.8.109:41659 >> ;rport=2896;branch=z9hG4bKPjvnvRZjpAXahzIV7S8SRgS5vwKmYI9WFS;alias;received=222.215.195.234 >> f: ;tag=weeMqzInPcNLPd6s5IX63c8be3DMvGkU >> t: >> i: sjprJQszC2LHvY6jRr-lm9rau9V1PlYX >> CSeq: 32737 INVITE >> User-Agent: >> FreeSWITCH-mod_sofia/1.4.13+git~20141103T195300Z~b942d0faa8~64bit >> Content-Length: 0 >> >> >> >> >> >> >> send 2133 bytes to tls/[222.248.102.244]:38614 at 20:56:54.360608: >> >> ------------------------------------------------------------------------ >> INVITE >> sip:1000 at 222.248.102.244:38614;rinstance=98c51296f2d531cc;transport=TLS >> SIP/2.0 >> Via: SIP/2.0/TLS 176.31.245.17;branch=z9hG4bKjXZvg3cX76Krj >> Max-Forwards: 69 >> From: "Extension 1006" ;tag=15DcKN9tUZeaK >> To: < >> sip:1000 at 222.248.102.244:38614;rinstance=98c51296f2d531cc;transport=TLS> >> Call-ID: 0a3e07c5-f437-1232-9bba-d2ab2784dc6a >> CSeq: 68395843 INVITE >> Contact: >> User-Agent: >> FreeSWITCH-mod_sofia/1.4.13+git~20141103T195300Z~b942d0faa8~64bit >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, path, replaces >> Allow-Events: talk, hold, conference, presence, as-feature-event, >> dialog, line-seize, call-info, sla, include-session-description, >> presence.winfo, message-summary, refer >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 1076 >> X-FS-Support: update_display,send_info >> Remote-Party-ID: "Extension 1006" > >;party=calling;screen=yes;privacy=off >> >> v=0 >> o=FreeSWITCH 1417445672 1417445673 IN IP4 176.31.245.17 >> s=FreeSWITCH >> c=IN IP4 176.31.245.17 >> t=0 0 >> m=audio 18142 RTP/SAVP 0 8 101 13 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=crypto:1 AEAD_AES_256_GCM_8 >> inline:6p/uXB7tN5SG9vHP/DMurij6BaZIOKcP4Tle0kjo2lwL1Oa4z9WGyZtk4zs >> a=crypto:2 AEAD_AES_128_GCM_8 >> inline:hGeJA6XQV1V35oTu2141o7mD64dFDRbeCbcf6g >> a=crypto:3 AES_CM_256_HMAC_SHA1_80 >> inline:XdZwwcl6kMRrWY7Vkrbozf/o1awka9lef0UjQRwvnOnnrbAgzXIm8s3yu7o8DQ >> a=crypto:4 AES_CM_192_HMAC_SHA1_80 >> inline:KtM92m0ac2musYPoXLDJb4rUbEMYAwMegPH9RMtzwzPb+0O93e0 >> a=crypto:5 AES_CM_128_HMAC_SHA1_80 >> inline:PtOpdIk2LTSCSDVzPaSN6AV2xosBrJUTd4mvKIGh >> a=crypto:6 AES_CM_256_HMAC_SHA1_32 >> inline:ZHzWxajRlQp8IuzX5CuFM1SRkL0huC62ukX583+Vg0LBiK9t2NrpP8FNwdkckQ >> a=crypto:7 AES_CM_192_HMAC_SHA1_32 >> inline:S9BWrDxGFYpU3P9o3HD0Z97W5jwo8vD+E1sJi821r4/B+cdukTw >> a=crypto:8 AES_CM_128_HMAC_SHA1_32 >> inline:l7C0wU29AjF2ZMnCTvks+1ytd/UiXHqN8UIsM6vf >> a=crypto:9 AES_CM_128_NULL_AUTH >> inline:tv97QsO7+S9NLTXrD02omyWoyM7I2KaWaimUgq99 >> a=ptime:20 >> >> ------------------------------------------------------------------------ >> >> >> REGISTER sip:abc-xyz.com;transport=TLS SIP/2.0 >> Via: SIP/2.0/TLS 192.168.10.2:41120 >> ;branch=z9hG4bK-d8754z-903949acb1d10083-1---d8754z-;rport >> Max-Forwards: 70 >> Contact: < >> sip:1000 at 192.168.10.2:41120;rinstance=57d4e16c725a51e6;transport=TLS> >> To: >> From: ;tag=4a2eae22 >> Call-ID: N2RiMjE2ZTQxNzYzMGY3ZmM3ZGJhODM4NTMzZmQ1YTA. >> CSeq: 1 REGISTER >> Expires: 60 >> Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, >> INFO, SUBSCRIBE >> Supported: replaces, norefersub, extended-refer, timer, >> X-cisco-serviceuri >> User-Agent: Zoiper r27147 >> Allow-Events: presence, kpml >> Content-Length: 0 >> >> >> >> ------------------------------------------------------------------------ >> send 716 bytes to tls/[222.248.102.244]:38620 at 20:57:01.806433: >> >> ------------------------------------------------------------------------ >> SIP/2.0 401 Unauthorized >> Via: SIP/2.0/TLS 192.168.10.2:41120 >> ;branch=z9hG4bK-d8754z-903949acb1d10083-1---d8754z-;rport=38620;received=222.248.102.244 >> From: ;tag=4a2eae22 >> To: ;tag=2e74mgtyr84ve >> Call-ID: N2RiMjE2ZTQxNzYzMGY3ZmM3ZGJhODM4NTMzZmQ1YTA. >> CSeq: 1 REGISTER >> User-Agent: >> FreeSWITCH-mod_sofia/1.4.13+git~20141103T195300Z~b942d0faa8~64bit >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, path, replaces >> WWW-Authenticate: Digest realm="abc-xyz.com", >> nonce="374fdd98-7994-11e4-bf9c-1182ebca2429", algorithm=MD5, qop="auth" >> Content-Length: 0 >> >> >> >> ------------------------------------------------------------------------ >> recv 944 bytes from tls/[222.248.102.244]:38620 at 20:57:01.995662: >> >> ------------------------------------------------------------------------ >> REGISTER sip:abc-xyz.com;transport=TLS SIP/2.0 >> Via: SIP/2.0/TLS 192.168.10.2:41120 >> ;branch=z9hG4bK-d8754z-8ca0e782336414eb-1---d8754z-;rport >> Max-Forwards: 70 >> Contact: < >> sip:1000 at 192.168.10.2:41120;rinstance=57d4e16c725a51e6;transport=TLS> >> To: >> From: ;tag=4a2eae22 >> Call-ID: N2RiMjE2ZTQxNzYzMGY3ZmM3ZGJhODM4NTMzZmQ1YTA. >> CSeq: 2 REGISTER >> Expires: 60 >> Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, >> INFO, SUBSCRIBE >> Supported: replaces, norefersub, extended-refer, timer, >> X-cisco-serviceuri >> User-Agent: Zoiper r27147 >> Authorization: Digest username="1000",realm="abc-xyz.com >> ",nonce="374fdd98-7994-11e4-bf9c-1182ebca2429",uri=" >> sip:abc-xyz.com;transport=TLS >> ",response="8d7e108ff98035d37424c32166fe0253",cnonce="107cf281c3b57888a4bfd1a9c3776098",nc=00000001,qop=auth,algorithm=MD5 >> Allow-Events: presence, kpml >> Content-Length: 0 >> >> >> >> ------------------------------------------------------------------------ >> send 712 bytes to tls/[222.248.102.244]:38620 at 20:57:01.997931: >> >> ------------------------------------------------------------------------ >> SIP/2.0 200 OK >> Via: SIP/2.0/TLS 192.168.10.2:41120 >> ;branch=z9hG4bK-d8754z-8ca0e782336414eb-1---d8754z-;rport=38620;received=222.248.102.244 >> From: ;tag=4a2eae22 >> To: ;tag=3Q0XpBB2NHUFa >> Call-ID: N2RiMjE2ZTQxNzYzMGY3ZmM3ZGJhODM4NTMzZmQ1YTA. >> CSeq: 2 REGISTER >> Contact: < >> sip:1000 at 192.168.10.2:41120;rinstance=57d4e16c725a51e6;transport=TLS >> >;expires=60 >> Date: Mon, 01 Dec 2014 19:57:01 GMT >> User-Agent: >> FreeSWITCH-mod_sofia/1.4.13+git~20141103T195300Z~b942d0faa8~64bit >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, path, replaces >> Content-Length: 0 >> >> >> ------------------------------------------------------------------------ >> recv 944 bytes from tls/[222.248.102.244]:38620 at 20:57:02.205719: >> >> ------------------------------------------------------------------------ >> REGISTER sip:abc-xyz.com;transport=TLS SIP/2.0 >> Via: SIP/2.0/TLS 222.248.102.244:38620 >> ;branch=z9hG4bK-d8754z-59d493abef215091-1---d8754z-;rport >> Max-Forwards: 70 >> Contact: < >> sip:1000 at 192.168.10.2:41120;rinstance=57d4e16c725a51e6;transport=TLS >> >;expires=0 >> To: >> From: ;tag=4a2eae22 >> Call-ID: N2RiMjE2ZTQxNzYzMGY3ZmM3ZGJhODM4NTMzZmQ1YTA. >> CSeq: 3 REGISTER >> Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, >> INFO, SUBSCRIBE >> Supported: replaces, norefersub, extended-refer, timer, >> X-cisco-serviceuri >> User-Agent: Zoiper r27147 >> Authorization: Digest username="1000",realm="abc-xyz.com >> ",nonce="374fdd98-7994-11e4-bf9c-1182ebca2429",uri=" >> sip:abc-xyz.com;transport=TLS >> ",response="5168bf9afffe9adacef7d898bd6bec9b",cnonce="80108fa519303990cfd2d46639d767ee",nc=00000002,qop=auth,algorithm=MD5 >> Allow-Events: presence, kpml >> Content-Length: 0 >> >> >> ------------------------------------------------------------------------ >> send 598 bytes to tls/[222.248.102.244]:38620 at 20:57:02.207249: >> >> ------------------------------------------------------------------------ >> SIP/2.0 200 OK >> Via: SIP/2.0/TLS 222.248.102.244:38620 >> ;branch=z9hG4bK-d8754z-59d493abef215091-1---d8754z-;rport=38620 >> From: ;tag=4a2eae22 >> To: ;tag=40Spr6U5jtH2N >> Call-ID: N2RiMjE2ZTQxNzYzMGY3ZmM3ZGJhODM4NTMzZmQ1YTA. >> CSeq: 3 REGISTER >> Date: Mon, 01 Dec 2014 19:57:02 GMT >> User-Agent: >> FreeSWITCH-mod_sofia/1.4.13+git~20141103T195300Z~b942d0faa8~64bit >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, path, replaces >> Content-Length: 0 >> >> >> ------------------------------------------------------------------------ >> recv 681 bytes from tls/[222.248.102.244]:38620 at 20:57:02.400564: >> >> ------------------------------------------------------------------------ >> REGISTER sip:abc-xyz.com;transport=TLS SIP/2.0 >> Via: SIP/2.0/TLS 222.248.102.244:38620 >> ;branch=z9hG4bK-d8754z-d738a1a836608006-1---d8754z-;rport >> Max-Forwards: 70 >> Contact: < >> sip:1000 at 222.248.102.244:38620;rinstance=43166fee0aaac1bc;transport=TLS> >> To: >> From: ;tag=979e5f46 >> Call-ID: MTJhNjIzMDM1MDEwMDVjZDRkOGMyYjBlMWU4ZDEzY2Y. >> CSeq: 1 REGISTER >> Expires: 60 >> Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, >> INFO, SUBSCRIBE >> Supported: replaces, norefersub, extended-refer, timer, >> X-cisco-serviceuri >> User-Agent: Zoiper r27147 >> Allow-Events: presence, kpml >> Content-Length: 0 >> >> >> >> ------------------------------------------------------------------------ >> send 694 bytes to tls/[222.248.102.244]:38620 at 20:57:02.401167: >> >> ------------------------------------------------------------------------ >> SIP/2.0 401 Unauthorized >> Via: SIP/2.0/TLS 222.248.102.244:38620 >> ;branch=z9hG4bK-d8754z-d738a1a836608006-1---d8754z-;rport=38620 >> From: ;tag=979e5f46 >> To: ;tag=59jFt1c9F37mH >> Call-ID: MTJhNjIzMDM1MDEwMDVjZDRkOGMyYjBlMWU4ZDEzY2Y. >> CSeq: 1 REGISTER >> User-Agent: >> FreeSWITCH-mod_sofia/1.4.13+git~20141103T195300Z~b942d0faa8~64bit >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, path, replaces >> WWW-Authenticate: Digest realm="abc-xyz.com", >> nonce="37aaa44e-7994-11e4-bf9d-1182ebca2429", algorithm=MD5, qop="auth" >> Content-Length: 0 >> >> >> ------------------------------------------------------------------------ >> recv 950 bytes from tls/[222.248.102.244]:38620 at 20:57:02.655864: >> >> ------------------------------------------------------------------------ >> REGISTER sip:abc-xyz.com;transport=TLS SIP/2.0 >> Via: SIP/2.0/TLS 222.248.102.244:38620 >> ;branch=z9hG4bK-d8754z-dfaa7130675d6cfa-1---d8754z-;rport >> Max-Forwards: 70 >> Contact: < >> sip:1000 at 222.248.102.244:38620;rinstance=43166fee0aaac1bc;transport=TLS> >> To: >> From: ;tag=979e5f46 >> Call-ID: MTJhNjIzMDM1MDEwMDVjZDRkOGMyYjBlMWU4ZDEzY2Y. >> CSeq: 2 REGISTER >> Expires: 60 >> Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, >> INFO, SUBSCRIBE >> Supported: replaces, norefersub, extended-refer, timer, >> X-cisco-serviceuri >> User-Agent: Zoiper r27147 >> Authorization: Digest username="1000",realm="abc-xyz.com >> ",nonce="37aaa44e-7994-11e4-bf9d-1182ebca2429",uri=" >> sip:abc-xyz.com;transport=TLS >> ",response="f1667094626fc10ca76a070438497e6f",cnonce="dc790bffe285311f0b903bd515a3e741",nc=00000001,qop=auth,algorithm=MD5 >> Allow-Events: presence, kpml >> Content-Length: 0 >> >> >> ------------------------------------------------------------------------ >> send 693 bytes to tls/[222.248.102.244]:38620 at 20:57:02.657688: >> >> ------------------------------------------------------------------------ >> SIP/2.0 200 OK >> Via: SIP/2.0/TLS 222.248.102.244:38620 >> ;branch=z9hG4bK-d8754z-dfaa7130675d6cfa-1---d8754z-;rport=38620 >> From: ;tag=979e5f46 >> To: ;tag=6jc8UvXcDcy7c >> Call-ID: MTJhNjIzMDM1MDEwMDVjZDRkOGMyYjBlMWU4ZDEzY2Y. >> CSeq: 2 REGISTER >> Contact: < >> sip:1000 at 222.248.102.244:38620;rinstance=43166fee0aaac1bc;transport=TLS >> >;expires=60 >> Date: Mon, 01 Dec 2014 19:57:02 GMT >> User-Agent: >> FreeSWITCH-mod_sofia/1.4.13+git~20141103T195300Z~b942d0faa8~64bit >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, path, replaces >> Content-Length: 0 >> >> >> >> >> >> >> *From: *Brian West >> *To: *FreeSWITCH Users Help >> *Date: *December 1, 2014 at 5:19:30 PM GMT+3 >> *Reply-To: *FreeSWITCH Users Help >> *Subject: **Re: [Freeswitch-users] FreeSWITCH TLS not able to receive >> calls* >> >> >> What endpoints are involved? have you looked at 'sofia loglevel all 9' >> output and see if it gives you a clue? >> >> On Fri, Nov 28, 2014 at 6:24 PM, Ahmed Habiba wrote: >> >>> Dears, >>> >>> I?ve configured FreeSWITCH with the below version with TLS/SRTPas per >>> the recommendation in page ?https://wiki.freeswitch.org/wiki/SIP_TLS? >>> and it was strait forward, and I was able to connect and make make calls >>> using zoiper, but I was not able to receive any calls after enabling the >>> TLS/SRTP. >>> >>> *"FreeSWITCH Version 1.4.13+git~20141103T195300Z~b942d0faa8~64bit (git >>> b942d0f 2014-11-03 19:53:00Z 64bit)**?* >>> >>> Your kind feedback will be appreciate. >>> >>> Thanks, >>> >>> Ahmed Habiba. >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> >> *Brian West* >> brian at freeswitch.org >> >> >> *Twitter: @FreeSWITCH , @briankwest* >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >> >> >> >> >> >> >> -- >> Thanks and Best Regards, >> >> Ahmed Habiba >> >> Mob: +20 10 37 82 970 >> >> *Success: believe (Vision) plus commitment (Action)* >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > > *Brian West* > brian at freeswitch.org > > > *Twitter: @FreeSWITCH , @briankwest* > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141204/814b1d9a/attachment-0001.html From anthony.minessale at gmail.com Thu Dec 4 23:23:11 2014 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 4 Dec 2014 14:23:11 -0600 Subject: [Freeswitch-users] Windows build system In-Reply-To: References: Message-ID: *crickets* ? On Wed, Dec 3, 2014 at 2:31 PM, Michael Jerris wrote: > Given the recent announcements by Microsoft about the community edition > 2013 being available, we are working to migrate the build system towards > using that as our primary build. As part of this process we will be very > soon dropping support for any version of Visual Studio prior to 2012. If > you feel strongly about needing support for these older versions, please > speak up now with an offer to maintain these legacy build systems. We are > also investigating moving to using chocolatey as a new system to manage > dependencies on windows instead of maintaining the build for all our deps > ourselves. It is also possible we will drop support for the 2012 build > system in the not so distant future. Could the community chime in here as > to what their needs are, and what they are willing to do to help support > the windows builds so we can determine what we plan to support going > forward. > > Thanks > Mike > > https://chocolatey.org/ > http://www.visualstudio.com/en-us/news/vs2013-community-vs.aspx > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 ? @anthmfs ? @FreeSWITCH ? ? http://freeswitch.org/ ? http://cluecon.com/ ? http://twitter.com/FreeSWITCH ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ * ClueCon Weekly Development Call ? sip:888 at conference.freeswitch.org ? +19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141204/55a012f2/attachment.html From aademattia at comcast.net Fri Dec 5 00:21:08 2014 From: aademattia at comcast.net (Andrew) Date: Thu, 4 Dec 2014 16:21:08 -0500 Subject: [Freeswitch-users] Windows build system In-Reply-To: References: Message-ID: <086001d01008$3c66b090$b53411b0$@comcast.net> Hi, I use Visual Studio prior to 2013. I am sure community edition will still work. I will help for the windows builds anywhere I can. Andrew From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Thursday, December 4, 2014 3:23 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Windows build system *crickets* ? On Wed, Dec 3, 2014 at 2:31 PM, Michael Jerris > wrote: Given the recent announcements by Microsoft about the community edition 2013 being available, we are working to migrate the build system towards using that as our primary build. As part of this process we will be very soon dropping support for any version of Visual Studio prior to 2012. If you feel strongly about needing support for these older versions, please speak up now with an offer to maintain these legacy build systems. We are also investigating moving to using chocolatey as a new system to manage dependencies on windows instead of maintaining the build for all our deps ourselves. It is also possible we will drop support for the 2012 build system in the not so distant future. Could the community chime in here as to what their needs are, and what they are willing to do to help support the windows builds so we can determine what we plan to support going forward. Thanks Mike https://chocolatey.org/ http://www.visualstudio.com/en-us/news/vs2013-community-vs.aspx _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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 ? @anthmfs ? @FreeSWITCH ? ? http://freeswitch.org/ ? http://cluecon.com/ ? http://twitter.com/FreeSWITCH ? irc.freenode.net #freeswitch ? http://freeswitch.org/g+ ClueCon Weekly Development Call ? sip:888 at conference.freeswitch.org ? +19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141204/cd1b57d2/attachment.html From jcabezas at inovax.com.br Fri Dec 5 01:53:16 2014 From: jcabezas at inovax.com.br (Julio Cesar Esteves Cabezas) Date: Thu, 4 Dec 2014 20:53:16 -0200 Subject: [Freeswitch-users] delay_echo with half of time Message-ID: <039001d01015$17b4d8f0$471e8ad0$@inovax.com.br> Hi, The delay I am getting from delay_echo is roughly half of what I pass as parameter in its data. This is the snippet, and the log shows on screen that I am passing what I intend to I am running FreeSWITCH Version 1.5.12b~64bit ( 64bit) (as per command "version" in fs_cli) in a VMware VM with OS=Windows Server 2012. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141204/c5169c78/attachment.html From markus.klenk at googlemail.com Fri Dec 5 13:22:20 2014 From: markus.klenk at googlemail.com (Paul Klenk) Date: Fri, 5 Dec 2014 11:22:20 +0100 Subject: [Freeswitch-users] remove country code from destination_number Message-ID: Hi, I'd like to have the country code "+49" removed from destination_number and changed to "0" I tried: but after correctly executing this extension-statement it reverts to the original version of destination_number From avi at avimarcus.net Fri Dec 5 13:34:03 2014 From: avi at avimarcus.net (Avi Marcus) Date: Fri, 5 Dec 2014 10:34:03 +0000 Subject: [Freeswitch-users] remove country code from destination_number In-Reply-To: References: Message-ID: <0000014a1a037bde-8bdda21d-1ca7-4a75-bbf5-3fccfed8bc30-000000@email.amazonses.com> You can't/shouldn't modify destination_number, it's part of the caller profile. You can either transfer it to 0$1 to have it hit the dialplan again, or bridge to your gateway and use 0$1 as the destination. -Avi On Fri, Dec 5, 2014 at 12:22 PM, Paul Klenk wrote: > Hi, > > I'd like to have the country code "+49" removed from > destination_number and changed to "0" > > I tried: > > > data="destination_number=0$1"/> > > > > but after correctly executing this extension-statement it reverts to > the original version of destination_number > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141205/1f47ae55/attachment.html From ssinyagin at gmail.com Fri Dec 5 14:52:44 2014 From: ssinyagin at gmail.com (Stanislav Sinyagin) Date: Fri, 5 Dec 2014 12:52:44 +0100 Subject: [Freeswitch-users] remove country code from destination_number In-Reply-To: References: Message-ID: you just need to use a different variable to store the modified destination number. here are a couple of examples from my blog: http://txlab.wordpress.com/2014/10/21/using-voxbeam-for-outbound-calls-with-freeswitch/ http://txlab.wordpress.com/2013/08/03/free-us-number-and-caller-id-manipulation/ On Fri, Dec 5, 2014 at 11:22 AM, Paul Klenk wrote: > Hi, > > I'd like to have the country code "+49" removed from > destination_number and changed to "0" > > I tried: > > > > > > > but after correctly executing this extension-statement it reverts to > the original version of destination_number > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 Kamil.Frydryszek at ilim.poznan.pl Fri Dec 5 15:14:40 2014 From: Kamil.Frydryszek at ilim.poznan.pl (Kamil Frydryszek) Date: Fri, 5 Dec 2014 13:14:40 +0100 Subject: [Freeswitch-users] BigBlueButton/FreeSwitch - external SIP account - Registration Failed with status Service Unavailable [503] Message-ID: Hello, I use BigBlueButton, 0.81 and we must enable dial into conference via phone. Several months ago everything was good (working), but now it does not work ( I was working on thuesday for short time). I use external SIP account and there is problem with registration . There is error "2014-12-05 10:38:11.171892 [WARNING] sofia_reg.c:472 Bezposredni Failed Registration [503], setting retry to 60 seconds." I have just tried in different networks, on different servers and with two independent SIP operators. The usual voip client (X-Lite) is working ok with SIP accounts. I have attached a detailed debug log in file freeswitch.log. Please give any suggestions. I am in very difficult situation, because it is my work and I must repair it, but after several days I have no idea. Thank you in advance. Best regards, Kamil ____________________________________________________________________________________________________________________________________ Instytut Logistyki i Magazynowania www.ilim.poznan.pl | office at ilim.poznan.pl ul. Estkowskiego 6 | 61-755 Pozna?, Poland | tel. +48 (61) 850 48 90 | fax +48 (61) 852 63 76 | REGON: 000018603 | NIP: 777-00-20-410 Zarejestrowany pod nr KRS 0000052866 - Wydzia? VIII Gospodarczy Krajowego Rejestru S?dowego Pozna? - Nowe Miasto i Wilda _____________________________________________________________________________________________________________________________________ Instytut Logistyki i Magazynowania z siedzib? w Poznaniu dzia?a na podstawie ustawy z dnia 30 kwietnia 2010 r. o instytutach badawczych (Dz. U. z 2010 r., Nr 96, poz. 618). Zgodnie z art. 24 ust. 1 punkt 5 ustawy o instytutach badawczych oraz ? 4 ust. 2 Statutu Instytutu Logistyki i Magazynowania, do reprezentowania Instytutu Logistyki i Magazynowania uprawniony jest jego dyrektor. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141205/d10c0b99/attachment.html From vittico at gmail.com Fri Dec 5 15:46:15 2014 From: vittico at gmail.com (Victor Medina) Date: Fri, 5 Dec 2014 08:16:15 -0430 Subject: [Freeswitch-users] Building FS with mod_java support Message-ID: Hi all! I?ve been trying to build Freeswitch 1.4 with Java support. It compiles just fine, installation fails with: making install mod_java make[4]: Entering directory `/home/administrador/freeswitch/src/mod/languages/mod_java' make install-am make[5]: Entering directory `/home/administrador/freeswitch/src/mod/languages/mod_java' make[6]: Entering directory `/home/administrador/freeswitch/src/mod/languages/mod_java' make[6]: Nothing to be done for `install-exec-am'. cp freeswitch.jar /opt/CloudVoice-vPBX/fs/scripts cp: cannot stat ?freeswitch.jar?: No such file or directory make[6]: *** [install-data-local] Error 1 make[6]: Leaving directory `/home/administrador/freeswitch/src/mod/languages/mod_java' make[5]: *** [install-am] Error 2 make[5]: Leaving directory `/home/administrador/freeswitch/src/mod/languages/mod_java' make[4]: *** [install] Error 2 make[4]: Leaving directory `/home/administrador/freeswitch/src/mod/languages/mod_java' make[3]: *** [mod_java-install] Error 1 make[3]: Leaving directory `/home/administrador/freeswitch/src/mod' make[2]: *** [install-recursive] Error 1 make[2]: Leaving directory `/home/administrador/freeswitch/src' Can someone help? Sin mas a que hacer referencia, Victor Medina From vittico at gmail.com Fri Dec 5 15:59:01 2014 From: vittico at gmail.com (Victor Medina) Date: Fri, 5 Dec 2014 08:29:01 -0430 Subject: [Freeswitch-users] Building FS with mod_java support In-Reply-To: References: Message-ID: More on the same issue... root at ubuntu:/home/administrador/freeswitch# make mod_java-install make[1]: Entering directory `/home/administrador/freeswitch' /bin/mkdir -p '/opt/CloudVoice-vPBX/fs/lib' /bin/bash /home/administrador/freeswitch/libtool --mode=install /usr/bin/install -c libfreeswitch.la '/opt/CloudVoice-vPBX/fs/lib' libtool: install: /usr/bin/install -c .libs/libfreeswitch.so.1.0.0 /opt/CloudVoice-vPBX/fs/lib/libfreeswitch.so.1.0.0 libtool: install: (cd /opt/CloudVoice-vPBX/fs/lib && { ln -s -f libfreeswitch.so.1.0.0 libfreeswitch.so.1 || { rm -f libfreeswitch.so.1 && ln -s libfreeswitch.so.1.0.0 libfreeswitch.so.1; }; }) libtool: install: (cd /opt/CloudVoice-vPBX/fs/lib && { ln -s -f libfreeswitch.so.1.0.0 libfreeswitch.so || { rm -f libfreeswitch.so && ln -s libfreeswitch.so.1.0.0 libfreeswitch.so; }; }) libtool: install: /usr/bin/install -c .libs/libfreeswitch.lai /opt/CloudVoice-vPBX/fs/lib/libfreeswitch.la libtool: install: /usr/bin/install -c .libs/libfreeswitch.a /opt/CloudVoice-vPBX/fs/lib/libfreeswitch.a libtool: install: chmod 644 /opt/CloudVoice-vPBX/fs/lib/libfreeswitch.a libtool: install: ranlib /opt/CloudVoice-vPBX/fs/lib/libfreeswitch.a libtool: finish: PATH="/opt/CloudVoice-vPBX/jdk/bin/:/usr/local/sbin:/usr/local/bin:/usr/sbin:/usr/bin:/sbin:/bin:/usr/games:/usr/local/games:/sbin" ldconfig -n /opt/CloudVoice-vPBX/fs/lib ---------------------------------------------------------------------- Libraries have been installed in: /opt/CloudVoice-vPBX/fs/lib If you ever happen to want to link against installed libraries in a given directory, LIBDIR, you must either use libtool, and specify the full pathname of the library, or use the `-LLIBDIR' flag during linking and do at least one of the following: - add LIBDIR to the `LD_LIBRARY_PATH' environment variable during execution - add LIBDIR to the `LD_RUN_PATH' environment variable during linking - use the `-Wl,-rpath -Wl,LIBDIR' linker flag - have your system administrator add LIBDIR to `/etc/ld.so.conf' See any operating system documentation about shared libraries for more information, such as the ld(1) and ld.so(8) manual pages. ---------------------------------------------------------------------- make[1]: Leaving directory `/home/administrador/freeswitch' make[1]: Entering directory `/home/administrador/freeswitch/src/mod' making install mod_java make[2]: Entering directory `/home/administrador/freeswitch/src/mod/languages/mod_java' make install-am make[3]: Entering directory `/home/administrador/freeswitch/src/mod/languages/mod_java' CXX mod_java_la-mod_java.lo CXX mod_java_la-freeswitch_java.lo freeswitch_java.cpp: In function 'void setOriginateStateHandler(jobject)': freeswitch_java.cpp:17:85: warning: format not a string literal and no format arguments [-Wformat-security] switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, errorMessage); ^ freeswitch_java.cpp:27:93: warning: format not a string literal and no format arguments [-Wformat-security] switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, errorMessage); ^ CXX mod_java_la-switch_swig_wrap.lo CC mod_java_la-modjava.lo CXXLD mod_java.la make[4]: Entering directory `/home/administrador/freeswitch/src/mod/languages/mod_java' make[4]: Nothing to be done for `install-exec-am'. cp freeswitch.jar /opt/CloudVoice-vPBX/fs/scripts cp: cannot stat ?freeswitch.jar?: No such file or directory make[4]: *** [install-data-local] Error 1 make[4]: Leaving directory `/home/administrador/freeswitch/src/mod/languages/mod_java' make[3]: *** [install-am] Error 2 make[3]: Leaving directory `/home/administrador/freeswitch/src/mod/languages/mod_java' make[2]: *** [install] Error 2 make[2]: Leaving directory `/home/administrador/freeswitch/src/mod/languages/mod_java' make[1]: *** [mod_java-install] Error 1 make[1]: Leaving directory `/home/administrador/freeswitch/src/mod' make: *** [mod_java-install] Error 2 root at ubuntu:/home/administrador/freeswitch# Sin mas a que hacer referencia, Victor Medina On Fri, Dec 5, 2014 at 8:16 AM, Victor Medina wrote: > Hi all! > > I?ve been trying to build Freeswitch 1.4 with Java support. It > compiles just fine, installation fails with: > > making install mod_java > make[4]: Entering directory > `/home/administrador/freeswitch/src/mod/languages/mod_java' > make install-am > make[5]: Entering directory > `/home/administrador/freeswitch/src/mod/languages/mod_java' > make[6]: Entering directory > `/home/administrador/freeswitch/src/mod/languages/mod_java' > make[6]: Nothing to be done for `install-exec-am'. > cp freeswitch.jar /opt/CloudVoice-vPBX/fs/scripts > cp: cannot stat ?freeswitch.jar?: No such file or directory > make[6]: *** [install-data-local] Error 1 > make[6]: Leaving directory > `/home/administrador/freeswitch/src/mod/languages/mod_java' > make[5]: *** [install-am] Error 2 > make[5]: Leaving directory > `/home/administrador/freeswitch/src/mod/languages/mod_java' > make[4]: *** [install] Error 2 > make[4]: Leaving directory > `/home/administrador/freeswitch/src/mod/languages/mod_java' > make[3]: *** [mod_java-install] Error 1 > make[3]: Leaving directory `/home/administrador/freeswitch/src/mod' > make[2]: *** [install-recursive] Error 1 > make[2]: Leaving directory `/home/administrador/freeswitch/src' > > Can someone help? > > Sin mas a que hacer referencia, > > Victor Medina From krice at freeswitch.org Fri Dec 5 18:03:12 2014 From: krice at freeswitch.org (Ken Rice) Date: Fri, 05 Dec 2014 15:03:12 +0000 Subject: [Freeswitch-users] FreeSWITCH Friday FreeForAll Reminder! Message-ID: <5481c930de4d_a5f1bc733449235@ip-10-58-73-253.mail> FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888 at conference.freeswitch.org or see http://ift.tt/1prwIZL for access info! -- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141205/9a63cbb2/attachment-0001.html From acheraime at gmail.com Fri Dec 5 18:14:10 2014 From: acheraime at gmail.com (Adolphe Cher-Aime) Date: Fri, 5 Dec 2014 10:14:10 -0500 Subject: [Freeswitch-users] Doc-Sprint Friday 12 December 2014 In-Reply-To: References: Message-ID: I'm in. -- Adolphe On Thu, Dec 4, 2014 at 3:00 PM, Iwada Eja wrote: > I'm In > > On Thu, Dec 4, 2014 at 1:20 PM, Areski wrote: > >> Hi everyone, >> >> We are planning to organize an other doc sprint on *Friday 12 December >> at 10am CT*. >> It will be 4 hours long but you can join for less time. >> >> The Doc-sprint will focus on migrating the remaining pages from MediaWiki >> (https://wiki.freeswitch.org) to Confluence Wiki ( >> https://freeswitch.org/confluence). >> >> We will use an FS IRC channel during the sprint: *#freeswitch-docs* >> and will track our work on the spreadsheet: >> https://docs.google.com/spreadsheets/d/1qsG-kRymvKlNBapnBLw86W130VdbnK6naYapbR_UNds/edit?pli=1#gid=1187898333 >> >> During the sprint, please change the URL's "Status" you are working on to >> "Editing" with your name next to it so we don't duplicate work. >> >> Some extra information: >> - https://freeswitch.org/confluence/display/FREESWITCH/Wiki+Migration >> - >> https://freeswitch.org/confluence/display/FREESWITCH/Contributing+Documentation >> >> We hope to get a maximum number of people signed up! >> >> Peoples confirmed so far: >> - Italo Rossi (+4) >> - Areski Belaid >> >> >> So, who is in? >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 > Iwada > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141205/9ad9051f/attachment.html From notify.sina at gmail.com Fri Dec 5 19:47:56 2014 From: notify.sina at gmail.com (Notify Me) Date: Fri, 5 Dec 2014 17:47:56 +0100 Subject: [Freeswitch-users] Newbie -- Help Needed Transferring Inbound Caller ID to external SIP Gateway URI In-Reply-To: <0000014a0b2a7f3a-e057e705-e811-4903-b2b9-bbd020abb4a7-000000@email.amazonses.com> References: <0000014a0b2a7f3a-e057e705-e811-4903-b2b9-bbd020abb4a7-000000@email.amazonses.com> Message-ID: Hi Avi! Thanks for the reply, I did figure it out, thanks! This is what I did, I can see from the logs that dials the othersipgw.com URI, and I can see tcpdumps that corresponds to the traffic being sent, and the other side gets it. The other side immediately drops the call and does some processing that returns an SMS to the dialler. I dont know if I can trouble you for more help.. there is no ringing when the transaction happens, and the dialler is unsure what is happening, if the call actually connected. As you might have guessed from the above I am trying to make it ring for a few seconds before the call drops. Is this possible? I would like for the dialler to be able to hear it ring a few times before the call is cut. Any help gratefully accepted. On Tue, Dec 2, 2014 at 2:22 PM, Avi Marcus wrote: > Hi - did you figure this out yet? > > One comment: > -- I > don't know if $1 is available anymore. You might want to just set that as > part of the actual number to route, e.g. add a 1212 prefix and match it > again or set it as a channel variable. You can see in the logs if the $1 is > resolving correctly. > > -Avi > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 Fri Dec 5 22:33:25 2014 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 5 Dec 2014 13:33:25 -0600 Subject: [Freeswitch-users] delay_echo with half of time In-Reply-To: <039001d01015$17b4d8f0$471e8ad0$@inovax.com.br> References: <039001d01015$17b4d8f0$471e8ad0$@inovax.com.br> Message-ID: Report BUGS to JIRA, Not the mailing list. I fixed it for you but I cannot push the commit until you properly report it. On Thu, Dec 4, 2014 at 4:53 PM, Julio Cesar Esteves Cabezas < jcabezas at inovax.com.br> wrote: > Hi, > > The delay I am getting from delay_echo is roughly half > of what I pass as parameter in its data. > > > > This is the snippet, and the log shows on screen that I am > passing what I intend to > > > > > > > > > > > > > > > > > > > > > > I am running FreeSWITCH Version 1.5.12b~64bit ( 64bit) > (as per command ?version? in fs_cli) in a VMware VM with OS=Windows > Server 2012. > > > > Thanks. > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 ? @anthmfs ? @FreeSWITCH ? ? http://freeswitch.org/ ? http://cluecon.com/ ? http://twitter.com/FreeSWITCH ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ * ClueCon Weekly Development Call ? sip:888 at conference.freeswitch.org ? +19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141205/cd649702/attachment.html From aqsyounas at gmail.com Fri Dec 5 22:42:41 2014 From: aqsyounas at gmail.com (Aqs Younas) Date: Sat, 6 Dec 2014 00:42:41 +0500 Subject: [Freeswitch-users] How to reset Freeswitch variable ${read_terminator_used} Message-ID: Hi, All I have a scenario in which next file is played when user press # and previous file when user press *. I want to reset ${read_terminator_used} after user presses either # or *. How can i do so.? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141206/2e6bfd2f/attachment.html From t.mahe at b-and-c.net Sat Dec 6 06:46:37 2014 From: t.mahe at b-and-c.net (=?windows-1252?Q?Tristan_Mah=E9?=) Date: Fri, 05 Dec 2014 19:46:37 -0800 Subject: [Freeswitch-users] Huge CPU LOAD increase between 1.4.12 and 1.4.14 In-Reply-To: <5473a2b195f0d_ea61b9732884134@ip-10-156-208-135.mail> References: <5473a2b195f0d_ea61b9732884134@ip-10-156-208-135.mail> Message-ID: <54827C1D.5000401@b-and-c.net> Hi guys, While trying to find out the root cause of this, I just want to know if someone is also seeing a huge cpu load with freeswitch 1.4.14, which was not present in 1.4.12. OS is debian wheezy, packages are official ones. Same config, same trafic pattern, just an upgrade from 1.4.12 ( no other packages updated ). We were previously seeing load around 0.5, with pikes up to around 2. We now are seeing a constant load of around 2, and pikes up to 20... If someone is also seeing this, that would be interesting to exchange, helping find out what is causing this ! I'll file a jira once I find the root of this problem ( and no, this is production servers, we can't put master there, we need stable ). Best, -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 473 bytes Desc: OpenPGP digital signature Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141205/ab20d807/attachment-0001.bin From shabbirabbasi92 at gmail.com Sat Dec 6 08:31:33 2014 From: shabbirabbasi92 at gmail.com (Shabbir abbasi) Date: Sat, 6 Dec 2014 10:31:33 +0500 Subject: [Freeswitch-users] how to select 2nd row of sql query result in lua Message-ID: if i execute a query in mysql select dialprefix,rateinitial from cc_ratecard where (dialprefix='001' or dialprefix='00123') ORDER BY LENGTH(dialprefix) DESC; result row1 | 001 | 0.01000 | row22 | 00123 | 0.02000 | and this is lua code local rates = {} local count = ""; assert(dbh:query(query, function(qrow) for key, val in pairs(qrow) do rates[key] = val freeswitch.consoleLog("INFO"," rates :"..key.. "..="..val.." \n") end count = qrow.count; end)) questios is 1 how to select 2nd row from result ? 2 how to count total rows ? 3 how to print all 2 rows in console ? any suggestios wellcome -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141206/9f181130/attachment.html From steveayre at gmail.com Sat Dec 6 13:17:40 2014 From: steveayre at gmail.com (Steven Ayre) Date: Sat, 6 Dec 2014 10:17:40 +0000 Subject: [Freeswitch-users] how to select 2nd row of sql query result in lua In-Reply-To: References: Message-ID: The function you pass to dbh:query is a callback. It is run for every row in the result set, in order. 1) Will be qrow on 2nd invocation 2) 'count = count + 1' on each invocation 3) Print on all invocations. You should be seeing that already, so perhaps the 2nd row isn't selected by your SQL query? On 6 December 2014 at 05:31, Shabbir abbasi wrote: > if i execute a query in mysql select dialprefix,rateinitial from > cc_ratecard where (dialprefix='001' or dialprefix='00123') ORDER BY > LENGTH(dialprefix) DESC; > > result > row1 | 001 | 0.01000 | > row22 | 00123 | 0.02000 | > > and this is lua code > > local rates = {} > local count = ""; > > assert(dbh:query(query, function(qrow) > for key, val in pairs(qrow) do > rates[key] = val > freeswitch.consoleLog("INFO"," rates :"..key.. "..="..val.." \n") > end > count = qrow.count; > end)) > > questios is > 1 how to select 2nd row from result ? > 2 how to count total rows ? > 3 how to print all 2 rows in console ? > > any suggestios wellcome > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141206/1ae0aee1/attachment.html From notify.sina at gmail.com Sun Dec 7 14:47:24 2014 From: notify.sina at gmail.com (Notify Me) Date: Sun, 7 Dec 2014 12:47:24 +0100 Subject: [Freeswitch-users] Newbie -- Help Needed Transferring Inbound Caller ID to external SIP Gateway URI In-Reply-To: <0000014a22d99893-8fb2f862-1a14-4cb2-b363-8e924d56012d-000000@email.amazonses.com> References: <0000014a0b2a7f3a-e057e705-e811-4903-b2b9-bbd020abb4a7-000000@email.amazonses.com> <0000014a22d99893-8fb2f862-1a14-4cb2-b363-8e924d56012d-000000@email.amazonses.com> Message-ID: The delay is very short, they send a100 Trying and then a 480 temporarily Unavailable back, as I can see in tcpdump. After they send the fail and its acknowledged, they end it and send an SMS back. This is a full dump 12:41:27.169928 IP (tos 0x0, ttl 64, id 34092, offset 0, flags [none], proto UDP (17), length 1146) 10.22.0.252.5080 > othersipgw.ip.address.5060: SIP, length: 1118 INVITE sip:sina.nowahala at sip.othersipgw.com;+234diallednumber SIP/2.0 Via: SIP/2.0/UDP my.public.ip.address:5080;rport;branch=z9hG4bK2S6KjD1jXFj1Q Max-Forwards: 5 From: "234diallednumber" ;tag=04tDF6pXQU4Ng To: Call-ID: d1ef2fdb-f8a8-1232-80a6-525400ecad09 CSeq: 68640179 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20141120T035109Z~79de78a0fb~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 247 X-FS-Support: update_display,send_info Remote-Party-ID: "234diallednumber" ;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1417930197 1417930198 IN IP4 my.public.ip.address s=FreeSWITCH c=IN IP4 my.public.ip.address t=0 0 m=audio 22290 RTP/AVP 8 0 101 13 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 12:41:27.385419 IP (tos 0x0, ttl 53, id 5736, offset 0, flags [none], proto UDP (17), length 384) othersipgw.ip.address.5060 > 10.22.0.252.5080: SIP, length: 356 SIP/2.0 100 Trying Via: SIP/2.0/UDP my.public.ip.address:5080;rport=5080;branch=z9hG4bK2S6KjD1jXFj1Q From: "234diallednumber" ;tag=04tDF6pXQU4Ng To: Call-ID: d1ef2fdb-f8a8-1232-80a6-525400ecad09 CSeq: 68640179 INVITE User-Agent: FreeSWITCH-mod_sofia/1.2.14 Content-Length: 0 12:41:27.424422 IP (tos 0x0, ttl 53, id 5737, offset 0, flags [none], proto UDP (17), length 1015) othersipgw.ip.address.5060 > 10.22.0.252.5080: SIP, length: 987 SIP/2.0 480 Temporarily Unavailable Via: SIP/2.0/UDP my.public.ip.address:5080;rport=5080;branch=z9hG4bK2S6KjD1jXFj1Q Max-Forwards: 4 From: "234diallednumber" ;tag=04tDF6pXQU4Ng To: ;tag=8c8DZK1aK9mKN Call-ID: d1ef2fdb-f8a8-1232-80a6-525400ecad09 CSeq: 68640179 INVITE User-Agent: FreeSWITCH-mod_sofia/1.2.14 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=16;text="NORMAL_CLEARING" Content-Length: 0 X-FS-Display-Name: sina.nowahala X-FS-Display-Number: sip:sina.nowahala at sip.othersipgw.com Remote-Party-ID: "sina.nowahala" ;party=calling;privacy=off;screen=no 12:41:27.424789 IP (tos 0x0, ttl 64, id 34093, offset 0, flags [none], proto UDP (17), length 412) 10.22.0.252.5080 > othersipgw.ip.address.5060: SIP, length: 384 ACK sip:sina.nowahala at sip.othersipgw.com;+234diallednumber SIP/2.0 Via: SIP/2.0/UDP my.public.ip.address:5080;rport;branch=z9hG4bK2S6KjD1jXFj1Q Max-Forwards: 5 From: "234diallednumber" ;tag=04tDF6pXQU4Ng To: ;tag=8c8DZK1aK9mKN Call-ID: d1ef2fdb-f8a8-1232-80a6-525400ecad09 CSeq: 68640179 ACK Content-Length: 0 So you recommend a transfer on fail wav to be played? I have been tearing my hair out trying to play sounds before the call is transferred to them, so that the dialler hears something. On Sun, Dec 7, 2014 at 4:44 AM, Avi Marcus wrote: > How long is the delay? I assume they return a normal_clearing or fail 4XX? > If the delay is short and they have a "real" status return, then.. for > success, you can just queue a "success.wav" to play after. > And for fails, you can use > https://wiki.freeswitch.org/wiki/Variable_transfer_on_fail and transfer to > an extension that plays an error message. > > This SMS sending sounds like a normal API call.. which if it was, could be > done with curl or inside a lua/js script. > > -Avi > > On Fri, Dec 5, 2014 at 6:47 PM, Notify Me wrote: >> >> Hi Avi! >> >> Thanks for the reply, I did figure it out, thanks! This is what I did, >> >> >> >> >> > require-nested="true"> >> >> > data="caller_id_number=+234${effective_caller_id_number}"/> >> >> > data="transfer_ringback=file_string://$${hold_music}"/> >> > data="transfer_ringback=local_stream://connecting"/> >> > >> data="sofia/external/sip:user.name at sip.othersipgw.com;${effective_caller_id_number}"/> >> >> >> >> >> I can see from the logs that dials the othersipgw.com URI, and I can >> see tcpdumps that corresponds to the traffic being sent, and the >> other side gets it. The other side immediately drops the call and does >> some processing that returns an SMS to the dialler. >> >> I dont know if I can trouble you for more help.. there is no ringing >> when the transaction happens, and the dialler is unsure what is >> happening, if the call actually connected. As you might have guessed >> from the above I am trying to make it ring for a few seconds before >> the call drops. Is this possible? I would like for the dialler to be >> able to hear it ring a few times before the call is cut. >> >> Any help gratefully accepted. >> >> On Tue, Dec 2, 2014 at 2:22 PM, Avi Marcus wrote: >> > Hi - did you figure this out yet? >> > >> > One comment: >> > >> > -- I >> > don't know if $1 is available anymore. You might want to just set that >> > as >> > part of the actual number to route, e.g. add a 1212 prefix and match it >> > again or set it as a channel variable. You can see in the logs if the $1 >> > is >> > resolving correctly. >> > >> > -Avi >> > >> > >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://confluence.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at 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 luis.daniel.lucio at gmail.com Sun Dec 7 17:49:51 2014 From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz) Date: Sun, 7 Dec 2014 09:49:51 -0500 Subject: [Freeswitch-users] Recommendation for Billing for Hosted PBX Platform In-Reply-To: References: <519A0253.3060009@gmail.com> Message-ID: Maybe a little late, But yes, Deon, you are right, FusionPBX is now near to be the best frontend for Freeswitch. You can do almost anything with it. Gladly, after some years of using FusionPBX I decide to develop a native Billing application for FusionPBX with Freeswitch. It integrates with FusionPBX in many ways, not an external instance of freeswitch, but inside the FusionPBX itself. If you are interested, you can go to my website www.okay.com.mx/en or contact me (gtalk prefered) 2013-05-20 12:43 GMT-04:00 Brian Foster : > Jbilling uses CDRs and events to do mediation. So using your guide, you > should have an idea of what info is required from freeswitch. If you can > give us some examples of what jbilling is asking for and we can give you > ideas. Please start a new thread specifically for your jbilling issues. > > Thanks, > > - BDF > > On May 20, 2013 10:02 AM, "Cal Leeming [Simplicity Media Ltd]" > wrote: >> >> Take a look at chargify.com - they are extremely good. >> >> Cal >> >> On Mon, May 20, 2013 at 12:00 PM, Deon Vermeulen >> wrote: >>> >>> Hi, >>> >>> >>> I've been looking at the Kazoo as a very viable option as a HostedPBX >>> platform, but don't find any information with regards to Billing. >>> >>> I've also been tracking the thread about recommended GUI where 2600Hz are >>> in the process for updated the code in git for Blue.box. >>> This also looks like a possibility for a HostedPBX platform, but then >>> again I have no idea about the Billing software to use for this. >>> >>> FusionPBX is a top contender if no the number 1 for a HostedPBX platform >>> but again information for Billing is lacking. >>> >>> I would like to know what the feel out there is for Billing on a >>> HostedPBX Platform, i.e. Cloud. >>> >>> If would be great if there is an opensource Billing system. >>> >>> I've purchased the jBilling Telecom guide, but no where can I find >>> information about integrating with FreeSWITCH nor implementing it for a >>> HostedPBX platform in the Cloud. >>> >>> >>> Kind Regards >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at 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 at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE: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 >> 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE: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 > 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 at 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 luis.daniel.lucio at gmail.com Sun Dec 7 17:58:04 2014 From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz) Date: Sun, 7 Dec 2014 09:58:04 -0500 Subject: [Freeswitch-users] Billing software In-Reply-To: <518CA7DD.8020802@gmail.com> References: <518CA7DD.8020802@gmail.com> Message-ID: Maybe a little late, Thanks everyone, but I am really against to install an extra freeswitch instance for the billing. It makes very complex the call flow, troubleshoting and in most cases FAX wont work. I did try the ASTPP before, horrible, hard to configure. I consider myself good on freeswitch, i know to code on perl, and a good sysadmin. But after watching how complex is to get this to work I abort. So I decide to code my own billing and gladly, I can tell you that if you have all information needed (such as carrier rates and about 15 mins to get an explanation on how it works, you can have it working in less than an hour). I started to study FusionPBX code, and after some months I have a working nice FusionPBX LCR+Billing native application. It fully integrates in the Fusion, not out side, so you are able to bill for anything, incoming, internal (ext to ext, ext to conference, ext to ivr), or outgoing calls and it is able to get money by Paypal, Stripe (a good gw for CreditCards) and offline. Currently, I have just release version 1.0.2 If someone is interested, you can go to my website www.okay.com.mx/en or contact me (gtalk prefered) LD 2013-05-10 3:55 GMT-04:00 Deon Vermeulen : > Hi Luis > > We just bought Commercial services from ASTPP. > When it comes to comparing price, features this is the BEST solution out > there at the moment. > > This is a very active project and you can contact Samir Doshi, Project > Maintainer & Developer, directly for more information wrt your technical > questions. > > samir at astpp.org > > > Kind Regards > Deon > > Luis Daniel Lucio Quiroz > May 9, 2013 11:46 PM > > > Deon, thank you. Can you talk me about your exerience and how fast they > are on fixing bugs. I have a really good or bad luck to find bugs. > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > Deon Vermeulen > May 5, 2013 4:52 PM > Check out ASTPP > > > Kind Regards > Deon Vermeulen > > Sent from my iPhone > > Luis Daniel Lucio Quiroz > May 5, 2013 3:36 PM > What other options for FS compatible software other than vBilling do you > recommend me? > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE: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 > 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141207/9c3dc698/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: compose-unknown-contact.jpg Type: image/jpeg Size: 770 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141207/9c3dc698/attachment-0002.jpg -------------- next part -------------- A non-text attachment was scrubbed... Name: postbox-contact.jpg Type: image/jpeg Size: 1143 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141207/9c3dc698/attachment-0003.jpg From avi at avimarcus.net Sun Dec 7 18:12:22 2014 From: avi at avimarcus.net (Avi Marcus) Date: Sun, 7 Dec 2014 15:12:22 +0000 Subject: [Freeswitch-users] Recommendation for Billing for Hosted PBX Platform In-Reply-To: References: <519A0253.3060009@gmail.com> Message-ID: <0000014a254f0100-ddeb3eb5-c532-4c5e-9507-1e199a208bbf-000000@email.amazonses.com> BTW - Stripe has recurring billing and the ability to create add-ons for the monthly bills, via API. They don't charge anything extra for these features beyond the normal credit card processing fees. -Avi On Mon, May 20, 2013 at 4:54 PM, Cal Leeming [Simplicity Media Ltd] < cal.leeming at simplicitymedialtd.co.uk> wrote: > Take a look at chargify.com - they are extremely good. > > Cal > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141207/edb47607/attachment.html From andrew at cassidywebservices.co.uk Sun Dec 7 18:12:53 2014 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Sun, 7 Dec 2014 15:12:53 +0000 Subject: [Freeswitch-users] remove country code from destination_number In-Reply-To: References: Message-ID: mod_translate On 5 December 2014 at 11:52, Stanislav Sinyagin wrote: > you just need to use a different variable to store the modified > destination number. > here are a couple of examples from my blog: > > > http://txlab.wordpress.com/2014/10/21/using-voxbeam-for-outbound-calls-with-freeswitch/ > > http://txlab.wordpress.com/2013/08/03/free-us-number-and-caller-id-manipulation/ > > > > > On Fri, Dec 5, 2014 at 11:22 AM, Paul Klenk > wrote: > > Hi, > > > > I'd like to have the country code "+49" removed from > > destination_number and changed to "0" > > > > I tried: > > > > > > data="destination_number=0$1"/> > > > > > > > > but after correctly executing this extension-statement it reverts to > > the original version of destination_number > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://confluence.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20141207/5538f1ba/attachment.html From andrew.keil at visytel.com Mon Dec 8 05:17:45 2014 From: andrew.keil at visytel.com (Andrew Keil) Date: Mon, 8 Dec 2014 02:17:45 +0000 Subject: [Freeswitch-users] Re- LUA audio playback on Windows 64bit not working? Message-ID: To FreeSWITCH users, I am just starting to use Lua inside FreeSWITCH and I am having difficulty with playing back audio to the caller. It seems the Media is not ready or available even though I have answered the caller using session:answer(). It also seems that session:ready() always returns false after session:answer(). Very strange. Version: FreeSWITCH 1.5.12b(64bit) on Windows 8.1 Pro (installed from freeswitch.msi) - default installation except for conf\vars.xml (default_password changed) conf\dialplan\default\00_ASKTestLua.xml: scripts\ASKTestLua.lua: -- Test Lua application -- answer the call session:answer(); session:sleep(1000); prompt = "ivr/ivr-welcome_to_freeswitch.wav"; freeswitch.consoleLog("INFO", string.format("prompt: %s\n", prompt)); session:execute("playback",prompt); --session:streamFile(prompt); freeswitch.consoleLog("INFO", "Prompt played\n") -- hangup session:hangup(); Logs inside FS_CLI: 2014-12-08 12:44:15.002800 [DEBUG] switch_core_session.c:2725 Application playback Requires media! pre_answering channel sofia/internal/1000 at 192.168.15.74 EXECUTE sofia/internal/1000 at 192.168.15.74 playback(ivr/ivr-welcome_to_freeswitch.wav) vs. conf\dialplan\default\00_ASKTestLua.xml: Which works!! However this is not using Lua. I have tried using "session:streamFile(prompt);" instead of "session:execute("playback",prompt);" and this also does not work. I would appreciate any insight into why this is happening. Kind Regards, Andrew Keil -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141208/1c5a91e2/attachment-0001.html From andrew.keil at visytel.com Mon Dec 8 09:12:59 2014 From: andrew.keil at visytel.com (Andrew Keil) Date: Mon, 8 Dec 2014 06:12:59 +0000 Subject: [Freeswitch-users] Re- LUA audio playback on Windows 64bit not working? (FOUND & FIXED) Message-ID: To all, Found the issue below: Removed the inline="true" from the xml dialplan and everything works as expected. Now: Not sure why I put it there in the first place. Sorry about that. Andrew From: Andrew Keil Sent: Monday, 8 December 2014 1:18 PM To: 'freeswitch-users at lists.freeswitch.org' Subject: Re- LUA audio playback on Windows 64bit not working? To FreeSWITCH users, I am just starting to use Lua inside FreeSWITCH and I am having difficulty with playing back audio to the caller. It seems the Media is not ready or available even though I have answered the caller using session:answer(). It also seems that session:ready() always returns false after session:answer(). Very strange. Version: FreeSWITCH 1.5.12b(64bit) on Windows 8.1 Pro (installed from freeswitch.msi) - default installation except for conf\vars.xml (default_password changed) conf\dialplan\default\00_ASKTestLua.xml: scripts\ASKTestLua.lua: -- Test Lua application -- answer the call session:answer(); session:sleep(1000); prompt = "ivr/ivr-welcome_to_freeswitch.wav"; freeswitch.consoleLog("INFO", string.format("prompt: %s\n", prompt)); session:execute("playback",prompt); --session:streamFile(prompt); freeswitch.consoleLog("INFO", "Prompt played\n") -- hangup session:hangup(); Logs inside FS_CLI: 2014-12-08 12:44:15.002800 [DEBUG] switch_core_session.c:2725 Application playback Requires media! pre_answering channel sofia/internal/1000 at 192.168.15.74 EXECUTE sofia/internal/1000 at 192.168.15.74 playback(ivr/ivr-welcome_to_freeswitch.wav) vs. conf\dialplan\default\00_ASKTestLua.xml: Which works!! However this is not using Lua. I have tried using "session:streamFile(prompt);" instead of "session:execute("playback",prompt);" and this also does not work. I would appreciate any insight into why this is happening. Kind Regards, Andrew Keil -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141208/7e6b68d3/attachment-0001.html From bordmi at rarus.ru Mon Dec 8 10:14:54 2014 From: bordmi at rarus.ru (=?UTF-8?B?0JHQvtGA0LjRgdC+0LIsINCU0LzQuNGC0YDQuNC5?=) Date: Mon, 8 Dec 2014 11:14:54 +0400 Subject: [Freeswitch-users] Re- LUA audio playback on Windows 64bit not working? (FOUND & FIXED) In-Reply-To: References: Message-ID: In the hunting state of dialplan treatment FS doesn`t permit to work with media, only in execution state. 2014-12-08 9:12 GMT+03:00 Andrew Keil : > To all, > > > > Found the issue below: > > > > Removed the inline=?true? from the xml dialplan and everything works as > expected. > > > > > > > > Now: > > > > > > > > Not sure why I put it there in the first place. > > > > Sorry about that. > > > > Andrew > > > > *From:* Andrew Keil > *Sent:* Monday, 8 December 2014 1:18 PM > *To:* 'freeswitch-users at lists.freeswitch.org' > *Subject:* Re- LUA audio playback on Windows 64bit not working? > > > > To FreeSWITCH users, > > > > I am just starting to use Lua inside FreeSWITCH and I am having difficulty > with playing back audio to the caller. It seems the Media is not ready or > available even though I have answered the caller using session:answer(). > > It also seems that session:ready() always returns false after > session:answer(). Very strange. > > > > Version: FreeSWITCH 1.5.12b(64bit) on Windows 8.1 Pro (installed from > freeswitch.msi) ? default installation except for conf\vars.xml > (default_password changed) > > > > *conf\dialplan\default\00_ASKTestLua.xml:* > > > > > > > > > > > > > > > > data="ASKTestLua.lua $2"/> > > > > > > > > > > *scripts\ASKTestLua.lua:* > > -- Test Lua application > > > > -- answer the call > > session:answer(); > > > > session:sleep(1000); > > > > prompt = "ivr/ivr-welcome_to_freeswitch.wav"; > > freeswitch.consoleLog("INFO", string.format("prompt: %s\n", prompt)); > > > > session:execute("playback",prompt); > > --session:streamFile(prompt); > > > > freeswitch.consoleLog("INFO", "Prompt played\n") > > > > -- hangup > > session:hangup(); > > > > > > Logs inside FS_CLI: 2014-12-08 12:44:15.002800 [DEBUG] > switch_core_session.c:2725 Application playback Requires media! > pre_answering channel sofia/internal/1000 at 192.168.15.74 > > EXECUTE sofia/internal/1000 at 192.168.15.74 > playback(ivr/ivr-welcome_to_freeswitch.wav) > > > > vs. > > > > conf\dialplan\default\00_ASKTestLua.xml: > > > > > > > > > > > > > > > > > > > > > > > > > > Which works!! However this is not using Lua. > > > > I have tried using ?session:streamFile(prompt);? instead of > ?session:execute("playback",prompt);? and this also does not work. > > > > I would appreciate any insight into why this is happening. > > > > Kind Regards, > > > > Andrew Keil > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- with best regrds, Dmitriy Borisov -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141208/1c17db53/attachment.html From bordmi at rarus.ru Mon Dec 8 10:17:12 2014 From: bordmi at rarus.ru (=?UTF-8?B?0JHQvtGA0LjRgdC+0LIsINCU0LzQuNGC0YDQuNC5?=) Date: Mon, 8 Dec 2014 11:17:12 +0400 Subject: [Freeswitch-users] Re- LUA audio playback on Windows 64bit not working? (FOUND & FIXED) In-Reply-To: References: Message-ID: But "inline" directive force FS to run script on hunting phase My yesturday`s sunday have died with this issue... :( 2014-12-08 10:14 GMT+03:00 ???????, ??????? : > In the hunting state of dialplan treatment FS doesn`t permit to work with > media, only in execution state. > > 2014-12-08 9:12 GMT+03:00 Andrew Keil : > >> To all, >> >> >> >> Found the issue below: >> >> >> >> Removed the inline=?true? from the xml dialplan and everything works as >> expected. >> >> >> >> >> >> >> >> Now: >> >> >> >> >> >> >> >> Not sure why I put it there in the first place. >> >> >> >> Sorry about that. >> >> >> >> Andrew >> >> >> >> *From:* Andrew Keil >> *Sent:* Monday, 8 December 2014 1:18 PM >> *To:* 'freeswitch-users at lists.freeswitch.org' >> *Subject:* Re- LUA audio playback on Windows 64bit not working? >> >> >> >> To FreeSWITCH users, >> >> >> >> I am just starting to use Lua inside FreeSWITCH and I am having >> difficulty with playing back audio to the caller. It seems the Media is >> not ready or available even though I have answered the caller using >> session:answer(). >> >> It also seems that session:ready() always returns false after >> session:answer(). Very strange. >> >> >> >> Version: FreeSWITCH 1.5.12b(64bit) on Windows 8.1 Pro (installed from >> freeswitch.msi) ? default installation except for conf\vars.xml >> (default_password changed) >> >> >> >> *conf\dialplan\default\00_ASKTestLua.xml:* >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> > data="ASKTestLua.lua $2"/> >> >> >> >> >> >> >> >> >> >> *scripts\ASKTestLua.lua:* >> >> -- Test Lua application >> >> >> >> -- answer the call >> >> session:answer(); >> >> >> >> session:sleep(1000); >> >> >> >> prompt = "ivr/ivr-welcome_to_freeswitch.wav"; >> >> freeswitch.consoleLog("INFO", string.format("prompt: %s\n", prompt)); >> >> >> >> session:execute("playback",prompt); >> >> --session:streamFile(prompt); >> >> >> >> freeswitch.consoleLog("INFO", "Prompt played\n") >> >> >> >> -- hangup >> >> session:hangup(); >> >> >> >> >> >> Logs inside FS_CLI: 2014-12-08 12:44:15.002800 [DEBUG] >> switch_core_session.c:2725 Application playback Requires media! >> pre_answering channel sofia/internal/1000 at 192.168.15.74 >> >> EXECUTE sofia/internal/1000 at 192.168.15.74 >> playback(ivr/ivr-welcome_to_freeswitch.wav) >> >> >> >> vs. >> >> >> >> conf\dialplan\default\00_ASKTestLua.xml: >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> Which works!! However this is not using Lua. >> >> >> >> I have tried using ?session:streamFile(prompt);? instead of >> ?session:execute("playback",prompt);? and this also does not work. >> >> >> >> I would appreciate any insight into why this is happening. >> >> >> >> Kind Regards, >> >> >> >> Andrew Keil >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > with best regrds, > Dmitriy Borisov > -- with best regrds, Dmitriy Borisov -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141208/bd11594e/attachment-0001.html From mbike2000ru at yahoo.com Mon Dec 8 10:28:41 2014 From: mbike2000ru at yahoo.com (Dmitry) Date: Mon, 8 Dec 2014 07:28:41 +0000 (UTC) Subject: [Freeswitch-users] outgoing speech path is broken sometimes in 1.4.14 Message-ID: <1071887578.4011778.1418023721454.JavaMail.yahoo@jws100167.mail.ne1.yahoo.com> Hi I installed FS 1.4.14 from git Now I test outgoing calls with Radius authorization. The problem that I encountered is that sometimes there is no RTP from Freeswitch. I see it in tcpdump dump. Iptables ?-no restrictions. I also tested when there is no mod_rad_auth loaded (I commented it out in autoload_configs/modules.conf.xml). - the same result And when I put the call on hold - then when I return - the call path is normal. Does anyone encountered the same behaviour? The version?FreeSWITCH Version 1.4.14+git~20141119T221113Z~ca1d990cfc~64bit (git ca1d990 2014-11-19 22:11:13Z 64bit) P.S. I am new to FS. I came from Asterisk world. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141208/d792c49c/attachment.html From den.pavlovsky at gmail.com Mon Dec 8 10:40:40 2014 From: den.pavlovsky at gmail.com (Den Pavlovsky) Date: Mon, 8 Dec 2014 13:40:40 +0600 Subject: [Freeswitch-users] outgoing speech path is broken sometimes in 1.4.14 Message-ID: Hi I installed FS 1.4.14 from git Now I test outgoing calls with Radius authorization. The problem that I encountered is that sometimes there is no RTP from Freeswitch. I see it in tcpdump dump. Iptables -no restrictions. I also tested when there is no mod_rad_auth loaded (I commented it out in autoload_configs/modules.conf.xml). - the same result And when I put the call on hold - then when I return - the call path is normal. Any help on how to debug this issue is appreciated. Does anyone encountered the same behaviour? The version FreeSWITCH Version 1.4.14+git~20141119T221113Z~ca1d990cfc~64bit (git ca1d990 2014-11-19 22:11:13Z 64bit) P.S. I am new to FS. I came from Asterisk world. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141208/9a8b6240/attachment.html From danb.lists at gmail.com Mon Dec 8 11:04:43 2014 From: danb.lists at gmail.com (DanB) Date: Mon, 08 Dec 2014 09:04:43 +0100 Subject: [Freeswitch-users] Recommendation for Billing for Hosted PBX Platform In-Reply-To: References: Message-ID: <54855B9B.8010409@gmail.com> Hi Deon, If you are looking for a billing engine (which I find it more suitable than complete billing system for hosted environments, same approach as FreeSWITCH +/or Kazoo - it does not enforce any business rules), take a look on CGRateS. It is opensource, very fast (12k+ requests per second), real-time, scalable (fire up more instances as you need them), feature-rich (multi-tenancy, supports advanced concepts as multiple calls out of same account, multiple balances inside one account, derived charging - eg: reseller chaining, grouped rating - charge using various billing increments within same call), API driven (full set of APIs available in both JSON and GOB) and production ready (runs already at MVNOs, ISPs, wholesale suppliers sites live). Happy to help you more! DanB From notify.sina at gmail.com Mon Dec 8 11:55:02 2014 From: notify.sina at gmail.com (Notify Me) Date: Mon, 8 Dec 2014 09:55:02 +0100 Subject: [Freeswitch-users] Trouble Playing Audio to external dialler after bridged call to external SIP URI is hung up Message-ID: Hi! I am still a Freeswitch Newbie, and I am trying to learn by doing. I have installed and using version 1.5.15b+git~20141120T035109Z~79de78a0fb~64bit on a 64-bit CentOS 6.6 kvm node. I have been able to successfully transfer calls for an external dialler, through a SIP trunk, to call an external SIP URI. This works and I can see it in the logs as very successful. The external SIP URI is supposed to process the caller_id, and send the dialler an SMS. The problem is the SIP URI being dialled ends the call very quickly and the external dialler has a busy tone and the call is dropped, or other signals that do not indicate that the call was successful and may dial again several times, if the SMS does not get delivered in time. The SIP URI sends back a 100 Trying and a 480 Temporarily Unavailable response as you can see below from a dump of the sequence. I have tried to introduce ivr sounds before and after the user connects and the bridge is dropped but is also not working. I can see that the wav files are being called from the logs but I dont hear anything at all. Please can anyone guide me? my dialplan: public: default: TCP DUMP 09:31:58.808760 IP (tos 0x0, ttl 64, id 34090, offset 0, flags [none], proto UDP (17), length 1146) my.natted.priv.ip.address.5080 > othersipgw.ip.address.5060: SIP, length: 1118 INVITE sip:user.name at sip.othersipgw.com;+234diallednumber SIP/2.0 Via: SIP/2.0/UDP my.public.ip.address:5080;rport;branch=z9hG4bK8eDcDvZpBm4yr Max-Forwards: 5 From: "0diallednumber" ;tag=6pKBvpXc32yeD To: Call-ID: 840abf24-f957-1232-2ca9-525400ecad09 CSeq: 68677695 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20141120T035109Z~79de78a0fb~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 247 X-FS-Support: update_display,send_info Remote-Party-ID: "0diallednumber" ;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1418006836 1418006837 IN IP4 my.public.ip.address s=FreeSWITCH c=IN IP4 my.public.ip.address t=0 0 m=audio 20682 RTP/AVP 8 0 101 13 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 09:31:59.080097 IP (tos 0x0, ttl 53, id 5734, offset 0, flags [none], proto UDP (17), length 384) othersipgw.ip.address.5060 > my.natted.priv.ip.address.5080: SIP, length: 356 SIP/2.0 100 Trying Via: SIP/2.0/UDP my.public.ip.address:5080;rport=5080;branch=z9hG4bK8eDcDvZpBm4yr From: "0diallednumber" ;tag=6pKBvpXc32yeD To: Call-ID: 840abf24-f957-1232-2ca9-525400ecad09 CSeq: 68677695 INVITE User-Agent: FreeSWITCH-mod_sofia/1.2.14 Content-Length: 0 09:31:59.145629 IP (tos 0x0, ttl 53, id 5735, offset 0, flags [none], proto UDP (17), length 1015) othersipgw.ip.address.5060 > my.natted.priv.ip.address.5080: SIP, length: 987 SIP/2.0 480 Temporarily Unavailable Via: SIP/2.0/UDP my.public.ip.address:5080;rport=5080;branch=z9hG4bK8eDcDvZpBm4yr Max-Forwards: 4 From: "0diallednumber" ;tag=6pKBvpXc32yeD To: ;tag=Ur840N1DvjvpH Call-ID: 840abf24-f957-1232-2ca9-525400ecad09 CSeq: 68677695 INVITE User-Agent: FreeSWITCH-mod_sofia/1.2.14 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=16;text="NORMAL_CLEARING" Content-Length: 0 X-FS-Display-Name: user.name X-FS-Display-Number: sip:user.name at sip.othersipgw.com Remote-Party-ID: "user.name" ;party=calling;privacy=off;screen=no 09:31:59.146980 IP (tos 0x0, ttl 64, id 34091, offset 0, flags [none], proto UDP (17), length 412) my.natted.priv.ip.address.5080 > othersipgw.ip.address.5060: SIP, length: 384 ACK sip:user.name at sip.othersipgw.com;+234diallednumber SIP/2.0 Via: SIP/2.0/UDP my.public.ip.address:5080;rport;branch=z9hG4bK8eDcDvZpBm4yr Max-Forwards: 5 From: "0diallednumber" ;tag=6pKBvpXc32yeD To: ;tag=Ur840N1DvjvpH Call-ID: 840abf24-f957-1232-2ca9-525400ecad09 CSeq: 68677695 ACK Content-Length: 0 From iwada.bassey at gmail.com Mon Dec 8 12:15:53 2014 From: iwada.bassey at gmail.com (Iwada Eja) Date: Mon, 8 Dec 2014 04:15:53 -0500 Subject: [Freeswitch-users] Recommendation for Billing for Hosted PBX Platform In-Reply-To: <54855B9B.8010409@gmail.com> References: <54855B9B.8010409@gmail.com> Message-ID: +1 for CGRateS. Been using it in Production with no hassles On Mon, Dec 8, 2014 at 3:04 AM, DanB wrote: > Hi Deon, > > If you are looking for a billing engine (which I find it more suitable > than complete billing system for hosted environments, same approach as > FreeSWITCH +/or Kazoo - it does not enforce any business rules), take a > look on CGRateS. > It is opensource, very fast (12k+ requests per second), real-time, > scalable (fire up more instances as you need them), feature-rich > (multi-tenancy, supports advanced concepts as multiple calls out of same > account, multiple balances inside one account, derived charging - eg: > reseller chaining, grouped rating - charge using various billing > increments within same call), API driven (full set of APIs available in > both JSON and GOB) and production ready (runs already at MVNOs, ISPs, > wholesale suppliers sites live). > > Happy to help you more! > DanB > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 Iwada -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141208/3d3199a7/attachment-0001.html From mkvonarx at gmail.com Mon Dec 8 13:29:33 2014 From: mkvonarx at gmail.com (Markus von Arx) Date: Mon, 8 Dec 2014 11:29:33 +0100 Subject: [Freeswitch-users] Windows build system In-Reply-To: References: Message-ID: We are fine with your Visual Studio plans. We only use VS2012 at the moment and actually would like to migrate to VS2013. Not so sure about the use of chocolatey though. Are you planning to use it to download libraries required by the build or for installing software/tools required for the build? Both look a bit problematic to me, because chocolatey (as far as I understand it) always acts globally on the target machine, meaning that it installs the libs/tools/software not locally inside the build directory but globally on the machine. I wouldn't like that at all. I don't want a build tool/process to install anything outside the build directory. If you manage to use chocolatey to only work in the build directory that's fine for me. But I'd strongly vote against any use of chocolatey to install libraries, tools or any software globally or outside the build directory. I wouldn't like a build tool/process/system installing anything on my machine for me automatically. Kind of like calling apt-get on a Linux machine from a build script. I think that is a no-go. Also, chocolatey is not so good detecting installed software that was not installed by chocolatey itself and would often try to re-install software that is already there. I'd vote against using chocolatey in the FreeSWITCH build if you ask me. Wouldn't nuget be the more natural choice anyway to install modules/libraries in Visual Studio? Just my opinion. Markus 2014-12-03 21:31 GMT+01:00 Michael Jerris : > Given the recent announcements by Microsoft about the community edition > 2013 being available, we are working to migrate the build system towards > using that as our primary build. As part of this process we will be very > soon dropping support for any version of Visual Studio prior to 2012. If > you feel strongly about needing support for these older versions, please > speak up now with an offer to maintain these legacy build systems. We are > also investigating moving to using chocolatey as a new system to manage > dependencies on windows instead of maintaining the build for all our deps > ourselves. It is also possible we will drop support for the 2012 build > system in the not so distant future. Could the community chime in here as > to what their needs are, and what they are willing to do to help support > the windows builds so we can determine what we plan to support going > forward. > > Thanks > Mike > > https://chocolatey.org/ > http://www.visualstudio.com/en-us/news/vs2013-community-vs.aspx > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141208/1d01c3d4/attachment.html From den.pavlovsky at gmail.com Mon Dec 8 14:28:45 2014 From: den.pavlovsky at gmail.com (Den Pavlovsky) Date: Mon, 8 Dec 2014 17:28:45 +0600 Subject: [Freeswitch-users] outgoing speech path is broken sometimes in 1.4.14 In-Reply-To: References: Message-ID: I found what caused such behaviour: I have 2 VLANs coming to my box. When I commented out the second VLAN - I have encountered no problems since then. I didn't quite get what xml files to change so as to work with 2 VLANs. 2014-12-08 12:40 GMT+05:00 Den Pavlovsky : > Hi > > I installed FS 1.4.14 from git > > Now I test outgoing calls with Radius authorization. > > The problem that I encountered is that sometimes there is no RTP from > Freeswitch. I see it in tcpdump dump. > > Iptables -no restrictions. > > I also tested when there is no mod_rad_auth loaded (I commented it out in > autoload_configs/modules.conf.xml). - the same result > > And when I put the call on hold - then when I return - the call path is > normal. > > Any help on how to debug this issue is appreciated. > > Does anyone encountered the same behaviour? The version FreeSWITCH Version > 1.4.14+git~20141119T221113Z~ca1d990cfc~64bit (git ca1d990 2014-11-19 > 22:11:13Z 64bit) > > P.S. I am new to FS. I came from Asterisk world. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141208/faddef45/attachment.html From avi at avimarcus.net Mon Dec 8 16:21:10 2014 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 8 Dec 2014 13:21:10 +0000 Subject: [Freeswitch-users] Newbie -- Help Needed Transferring Inbound Caller ID to external SIP Gateway URI In-Reply-To: References: <0000014a0b2a7f3a-e057e705-e811-4903-b2b9-bbd020abb4a7-000000@email.amazonses.com> <0000014a22d99893-8fb2f862-1a14-4cb2-b363-8e924d56012d-000000@email.amazonses.com> Message-ID: <0000014a2a0f8f54-5ad8d656-a43c-424b-aa41-8a315705cea0-000000@email.amazonses.com> Since there's only a 100 trying and no 18X, you won't get a ringing tone. If it's short, just play a message after. If not, you can use lua/js and run this second call as a bgapi call and try to capture the response. -Avi On Sun, Dec 7, 2014 at 1:47 PM, Notify Me wrote: > The delay is very short, they send a100 Trying and then a 480 temporarily > Unavailable back, as I can see in tcpdump. After they send the fail > and its acknowledged, they end it and send an SMS back. > This is a full dump > > 12:41:27.169928 IP (tos 0x0, ttl 64, id 34092, offset 0, flags [none], > proto UDP (17), length 1146) > 10.22.0.252.5080 > othersipgw.ip.address.5060: SIP, length: 1118 > INVITE sip:sina.nowahala at sip.othersipgw.com;+234diallednumber > SIP/2.0 > Via: SIP/2.0/UDP > my.public.ip.address:5080;rport;branch=z9hG4bK2S6KjD1jXFj1Q > Max-Forwards: 5 > From: "234diallednumber" > ;tag=04tDF6pXQU4Ng > To: > Call-ID: d1ef2fdb-f8a8-1232-80a6-525400ecad09 > CSeq: 68640179 INVITE > Contact: > User-Agent: > FreeSWITCH-mod_sofia/1.5.15b+git~20141120T035109Z~79de78a0fb~64bit > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, > UPDATE, REGISTER, REFER, NOTIFY > Supported: timer, path, replaces > Allow-Events: talk, hold, conference, refer > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 247 > X-FS-Support: update_display,send_info > Remote-Party-ID: "234diallednumber" > >;party=calling;screen=yes;privacy=off > > v=0 > o=FreeSWITCH 1417930197 1417930198 IN IP4 my.public.ip.address > s=FreeSWITCH > c=IN IP4 my.public.ip.address > t=0 0 > m=audio 22290 RTP/AVP 8 0 101 13 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > > 12:41:27.385419 IP (tos 0x0, ttl 53, id 5736, offset 0, flags [none], > proto UDP (17), length 384) > othersipgw.ip.address.5060 > 10.22.0.252.5080: SIP, length: 356 > SIP/2.0 100 Trying > Via: SIP/2.0/UDP > my.public.ip.address:5080;rport=5080;branch=z9hG4bK2S6KjD1jXFj1Q > From: "234diallednumber" > ;tag=04tDF6pXQU4Ng > To: > Call-ID: d1ef2fdb-f8a8-1232-80a6-525400ecad09 > CSeq: 68640179 INVITE > User-Agent: FreeSWITCH-mod_sofia/1.2.14 > Content-Length: 0 > > > 12:41:27.424422 IP (tos 0x0, ttl 53, id 5737, offset 0, flags [none], > proto UDP (17), length 1015) > othersipgw.ip.address.5060 > 10.22.0.252.5080: SIP, length: 987 > SIP/2.0 480 Temporarily Unavailable > Via: SIP/2.0/UDP > my.public.ip.address:5080;rport=5080;branch=z9hG4bK2S6KjD1jXFj1Q > Max-Forwards: 4 > From: "234diallednumber" > ;tag=04tDF6pXQU4Ng > To: ;+234diallednumber>;tag=8c8DZK1aK9mKN > Call-ID: d1ef2fdb-f8a8-1232-80a6-525400ecad09 > CSeq: 68640179 INVITE > User-Agent: FreeSWITCH-mod_sofia/1.2.14 > 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=16;text="NORMAL_CLEARING" > Content-Length: 0 > X-FS-Display-Name: sina.nowahala > X-FS-Display-Number: sip:sina.nowahala at sip.othersipgw.com > Remote-Party-ID: "sina.nowahala" > ;party=calling;privacy=off;screen=no > > > 12:41:27.424789 IP (tos 0x0, ttl 64, id 34093, offset 0, flags [none], > proto UDP (17), length 412) > 10.22.0.252.5080 > othersipgw.ip.address.5060: SIP, length: 384 > ACK sip:sina.nowahala at sip.othersipgw.com;+234diallednumber SIP/2.0 > Via: SIP/2.0/UDP > my.public.ip.address:5080;rport;branch=z9hG4bK2S6KjD1jXFj1Q > Max-Forwards: 5 > From: "234diallednumber" > ;tag=04tDF6pXQU4Ng > To: ;+234diallednumber>;tag=8c8DZK1aK9mKN > Call-ID: d1ef2fdb-f8a8-1232-80a6-525400ecad09 > CSeq: 68640179 ACK > Content-Length: 0 > > > So you recommend a transfer on fail wav to be played? I have been > tearing my hair out trying to play sounds before the call is > transferred to them, so that the dialler hears something. > > On Sun, Dec 7, 2014 at 4:44 AM, Avi Marcus wrote: > > How long is the delay? I assume they return a normal_clearing or fail > 4XX? > > If the delay is short and they have a "real" status return, then.. for > > success, you can just queue a "success.wav" to play after. > > And for fails, you can use > > https://wiki.freeswitch.org/wiki/Variable_transfer_on_fail and transfer > to > > an extension that plays an error message. > > > > This SMS sending sounds like a normal API call.. which if it was, could > be > > done with curl or inside a lua/js script. > > > > -Avi > > > > On Fri, Dec 5, 2014 at 6:47 PM, Notify Me wrote: > >> > >> Hi Avi! > >> > >> Thanks for the reply, I did figure it out, thanks! This is what I did, > >> > >> > >> > >> > >> >> require-nested="true"> > >> data="effective_caller_id_number=+234$1"/> > >> >> data="caller_id_number=+234${effective_caller_id_number}"/> > >> > >> >> data="transfer_ringback=file_string://$${hold_music}"/> > >> >> data="transfer_ringback=local_stream://connecting"/> > >> >> > >> data="sofia/external/sip:user.name at sip.othersipgw.com > ;${effective_caller_id_number}"/> > >> > >> > >> > >> > >> I can see from the logs that dials the othersipgw.com URI, and I can > >> see tcpdumps that corresponds to the traffic being sent, and the > >> other side gets it. The other side immediately drops the call and does > >> some processing that returns an SMS to the dialler. > >> > >> I dont know if I can trouble you for more help.. there is no ringing > >> when the transaction happens, and the dialler is unsure what is > >> happening, if the call actually connected. As you might have guessed > >> from the above I am trying to make it ring for a few seconds before > >> the call drops. Is this possible? I would like for the dialler to be > >> able to hear it ring a few times before the call is cut. > >> > >> Any help gratefully accepted. > >> > >> On Tue, Dec 2, 2014 at 2:22 PM, Avi Marcus wrote: > >> > Hi - did you figure this out yet? > >> > > >> > One comment: > >> > > >> > -- I > >> > don't know if $1 is available anymore. You might want to just set that > >> > as > >> > part of the actual number to route, e.g. add a 1212 prefix and match > it > >> > again or set it as a channel variable. You can see in the logs if the > $1 > >> > is > >> > resolving correctly. > >> > > >> > -Avi > >> > > >> > > >> > > _________________________________________________________________________ > >> > Professional FreeSWITCH Consulting Services: > >> > consulting at freeswitch.org > >> > http://www.freeswitchsolutions.com > >> > > >> > Official FreeSWITCH Sites > >> > http://www.freeswitch.org > >> > http://confluence.freeswitch.org > >> > http://www.cluecon.com > >> > > >> > FreeSWITCH-users mailing list > >> > FreeSWITCH-users at lists.freeswitch.org > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > http://www.freeswitch.org > > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141208/a90e65f3/attachment-0001.html From notify.sina at gmail.com Mon Dec 8 17:07:57 2014 From: notify.sina at gmail.com (Sina Owolabi) Date: Mon, 8 Dec 2014 15:07:57 +0100 Subject: [Freeswitch-users] Newbie -- Help Needed Transferring Inbound Caller ID to external SIP Gateway URI In-Reply-To: <0000014a2a0f8f9d-1038c04d-e533-483b-8322-bd27ace4ba19-000000@email.amazonses.com> References: <0000014a0b2a7f3a-e057e705-e811-4903-b2b9-bbd020abb4a7-000000@email.amazonses.com> <0000014a22d99893-8fb2f862-1a14-4cb2-b363-8e924d56012d-000000@email.amazonses.com> <0000014a2a0f8f9d-1038c04d-e533-483b-8322-bd27ace4ba19-000000@email.amazonses.com> Message-ID: I've tried to play a message after but I cannot hear it. Please can you tell me if this is correct? Thanks again for all your help: On Mon, Dec 8, 2014 at 2:21 PM, Avi Marcus wrote: > Since there's only a 100 trying and no 18X, you won't get a ringing tone. > If it's short, just play a message after. If not, you can use lua/js and run > this second call as a bgapi call and try to capture the response. > -Avi > > > On Sun, Dec 7, 2014 at 1:47 PM, Notify Me wrote: >> >> The delay is very short, they send a100 Trying and then a 480 temporarily >> Unavailable back, as I can see in tcpdump. After they send the fail >> and its acknowledged, they end it and send an SMS back. >> This is a full dump >> >> 12:41:27.169928 IP (tos 0x0, ttl 64, id 34092, offset 0, flags [none], >> proto UDP (17), length 1146) >> 10.22.0.252.5080 > othersipgw.ip.address.5060: SIP, length: 1118 >> INVITE sip:sina.nowahala at sip.othersipgw.com;+234diallednumber >> SIP/2.0 >> Via: SIP/2.0/UDP >> my.public.ip.address:5080;rport;branch=z9hG4bK2S6KjD1jXFj1Q >> Max-Forwards: 5 >> From: "234diallednumber" >> ;tag=04tDF6pXQU4Ng >> To: >> Call-ID: d1ef2fdb-f8a8-1232-80a6-525400ecad09 >> CSeq: 68640179 INVITE >> Contact: >> User-Agent: >> FreeSWITCH-mod_sofia/1.5.15b+git~20141120T035109Z~79de78a0fb~64bit >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, >> UPDATE, REGISTER, REFER, NOTIFY >> Supported: timer, path, replaces >> Allow-Events: talk, hold, conference, refer >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 247 >> X-FS-Support: update_display,send_info >> Remote-Party-ID: "234diallednumber" >> >> ;party=calling;screen=yes;privacy=off >> >> v=0 >> o=FreeSWITCH 1417930197 1417930198 IN IP4 my.public.ip.address >> s=FreeSWITCH >> c=IN IP4 my.public.ip.address >> t=0 0 >> m=audio 22290 RTP/AVP 8 0 101 13 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=ptime:20 >> >> 12:41:27.385419 IP (tos 0x0, ttl 53, id 5736, offset 0, flags [none], >> proto UDP (17), length 384) >> othersipgw.ip.address.5060 > 10.22.0.252.5080: SIP, length: 356 >> SIP/2.0 100 Trying >> Via: SIP/2.0/UDP >> my.public.ip.address:5080;rport=5080;branch=z9hG4bK2S6KjD1jXFj1Q >> From: "234diallednumber" >> ;tag=04tDF6pXQU4Ng >> To: >> Call-ID: d1ef2fdb-f8a8-1232-80a6-525400ecad09 >> CSeq: 68640179 INVITE >> User-Agent: FreeSWITCH-mod_sofia/1.2.14 >> Content-Length: 0 >> >> >> 12:41:27.424422 IP (tos 0x0, ttl 53, id 5737, offset 0, flags [none], >> proto UDP (17), length 1015) >> othersipgw.ip.address.5060 > 10.22.0.252.5080: SIP, length: 987 >> SIP/2.0 480 Temporarily Unavailable >> Via: SIP/2.0/UDP >> my.public.ip.address:5080;rport=5080;branch=z9hG4bK2S6KjD1jXFj1Q >> Max-Forwards: 4 >> From: "234diallednumber" >> ;tag=04tDF6pXQU4Ng >> To: >> ;tag=8c8DZK1aK9mKN >> Call-ID: d1ef2fdb-f8a8-1232-80a6-525400ecad09 >> CSeq: 68640179 INVITE >> User-Agent: FreeSWITCH-mod_sofia/1.2.14 >> 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=16;text="NORMAL_CLEARING" >> Content-Length: 0 >> X-FS-Display-Name: sina.nowahala >> X-FS-Display-Number: sip:sina.nowahala at sip.othersipgw.com >> Remote-Party-ID: "sina.nowahala" >> ;party=calling;privacy=off;screen=no >> >> >> 12:41:27.424789 IP (tos 0x0, ttl 64, id 34093, offset 0, flags [none], >> proto UDP (17), length 412) >> 10.22.0.252.5080 > othersipgw.ip.address.5060: SIP, length: 384 >> ACK sip:sina.nowahala at sip.othersipgw.com;+234diallednumber SIP/2.0 >> Via: SIP/2.0/UDP >> my.public.ip.address:5080;rport;branch=z9hG4bK2S6KjD1jXFj1Q >> Max-Forwards: 5 >> From: "234diallednumber" >> ;tag=04tDF6pXQU4Ng >> To: >> ;tag=8c8DZK1aK9mKN >> Call-ID: d1ef2fdb-f8a8-1232-80a6-525400ecad09 >> CSeq: 68640179 ACK >> Content-Length: 0 >> >> >> So you recommend a transfer on fail wav to be played? I have been >> tearing my hair out trying to play sounds before the call is >> transferred to them, so that the dialler hears something. >> >> On Sun, Dec 7, 2014 at 4:44 AM, Avi Marcus wrote: >> > How long is the delay? I assume they return a normal_clearing or fail >> > 4XX? >> > If the delay is short and they have a "real" status return, then.. for >> > success, you can just queue a "success.wav" to play after. >> > And for fails, you can use >> > https://wiki.freeswitch.org/wiki/Variable_transfer_on_fail and transfer >> > to >> > an extension that plays an error message. >> > >> > This SMS sending sounds like a normal API call.. which if it was, could >> > be >> > done with curl or inside a lua/js script. >> > >> > -Avi >> > >> > On Fri, Dec 5, 2014 at 6:47 PM, Notify Me wrote: >> >> >> >> Hi Avi! >> >> >> >> Thanks for the reply, I did figure it out, thanks! This is what I did, >> >> >> >> >> >> >> >> > >> expression="^0123450(\d{2})$"/> >> >> > >> require-nested="true"> >> >> > >> data="effective_caller_id_number=+234$1"/> >> >> > >> data="caller_id_number=+234${effective_caller_id_number}"/> >> >> >> >> > >> data="transfer_ringback=file_string://$${hold_music}"/> >> >> > >> data="transfer_ringback=local_stream://connecting"/> >> >> > >> >> >> >> >> data="sofia/external/sip:user.name at sip.othersipgw.com;${effective_caller_id_number}"/> >> >> >> >> >> >> >> >> >> >> I can see from the logs that dials the othersipgw.com URI, and I can >> >> see tcpdumps that corresponds to the traffic being sent, and the >> >> other side gets it. The other side immediately drops the call and does >> >> some processing that returns an SMS to the dialler. >> >> >> >> I dont know if I can trouble you for more help.. there is no ringing >> >> when the transaction happens, and the dialler is unsure what is >> >> happening, if the call actually connected. As you might have guessed >> >> from the above I am trying to make it ring for a few seconds before >> >> the call drops. Is this possible? I would like for the dialler to be >> >> able to hear it ring a few times before the call is cut. >> >> >> >> Any help gratefully accepted. >> >> >> >> On Tue, Dec 2, 2014 at 2:22 PM, Avi Marcus wrote: >> >> > Hi - did you figure this out yet? >> >> > >> >> > One comment: >> >> > > >> > data="sofia/gateway/othersipgw/+234$1"/> >> >> > -- I >> >> > don't know if $1 is available anymore. You might want to just set >> >> > that >> >> > as >> >> > part of the actual number to route, e.g. add a 1212 prefix and match >> >> > it >> >> > again or set it as a channel variable. You can see in the logs if the >> >> > $1 >> >> > is >> >> > resolving correctly. >> >> > >> >> > -Avi >> >> > >> >> > >> >> > >> >> > _________________________________________________________________________ >> >> > Professional FreeSWITCH Consulting Services: >> >> > consulting at freeswitch.org >> >> > http://www.freeswitchsolutions.com >> >> > >> >> > Official FreeSWITCH Sites >> >> > http://www.freeswitch.org >> >> > http://confluence.freeswitch.org >> >> > http://www.cluecon.com >> >> > >> >> > FreeSWITCH-users mailing list >> >> > FreeSWITCH-users at 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 Mon Dec 8 17:20:01 2014 From: vitaliy.davudov at vts24.ru (Vitaly) Date: Mon, 08 Dec 2014 17:20:01 +0300 Subject: [Freeswitch-users] Log-auth-failures question Message-ID: <5485B391.5050506@vts24.ru> Hi, list! In sip-profiles: "log-auth-failures" - write log entries ( Warning ) on authentication failures ( Registration & Invite ) in /usr/local/freeswitch/log/freeswitch.log files. Is it possible to write these entries in other file of in other directory? Thanks in advance. -- Best regards, Vitaly. From flokrrr at gmail.com Mon Dec 8 17:31:40 2014 From: flokrrr at gmail.com (Florent Krieg) Date: Mon, 8 Dec 2014 15:31:40 +0100 Subject: [Freeswitch-users] Log-auth-failures question In-Reply-To: <5485B391.5050506@vts24.ru> References: <5485B391.5050506@vts24.ru> Message-ID: Hi Vitaly, You can configure the logfile path in the config file (logfile.conf): But it will change the path of the file for all the logs, not only auth failures! Florent 2014-12-08 15:20 GMT+01:00 Vitaly : > Hi, list! > > In sip-profiles: "log-auth-failures" - write log entries ( Warning ) on > authentication failures ( Registration & Invite ) in > /usr/local/freeswitch/log/freeswitch.log files. Is it possible to write > these entries in other file of in other directory? > > Thanks in advance. > > -- > Best regards, > Vitaly. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141208/977173f5/attachment.html From steveayre at gmail.com Mon Dec 8 18:37:36 2014 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 8 Dec 2014 15:37:36 +0000 Subject: [Freeswitch-users] Log-auth-failures question In-Reply-To: References: <5485B391.5050506@vts24.ru> Message-ID: If you're wanting to do to implement fail2ban and don't want large log files you can also look at https://wiki.freeswitch.org/wiki/Mod_fail2ban On 8 December 2014 at 14:31, Florent Krieg wrote: > Hi Vitaly, > > You can configure the logfile path in the config file (logfile.conf): > > > But it will change the path of the file for all the logs, not only auth > failures! > > Florent > > 2014-12-08 15:20 GMT+01:00 Vitaly : > >> Hi, list! >> >> In sip-profiles: "log-auth-failures" - write log entries ( Warning ) on >> authentication failures ( Registration & Invite ) in >> /usr/local/freeswitch/log/freeswitch.log files. Is it possible to write >> these entries in other file of in other directory? >> >> Thanks in advance. >> >> -- >> Best regards, >> Vitaly. >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141208/cb234b60/attachment.html From aqsyounas at gmail.com Mon Dec 8 20:06:05 2014 From: aqsyounas at gmail.com (aqs younas) Date: Mon, 8 Dec 2014 22:06:05 +0500 Subject: [Freeswitch-users] Building FS with mod_java support In-Reply-To: References: Message-ID: <5485da7c.475cc20a.5edd.6ff2@mx.google.com> Uncomment mod_java from modules.conf in source directory and do there compile and install instead making it compile from jave directory. -----Original Message----- From: "Victor Medina" Sent: ?12/?5/?2014 5:47 PM To: "FreeSWITCH Users Help" Subject: [Freeswitch-users] Building FS with mod_java support Hi all! I?ve been trying to build Freeswitch 1.4 with Java support. It compiles just fine, installation fails with: making install mod_java make[4]: Entering directory `/home/administrador/freeswitch/src/mod/languages/mod_java' make install-am make[5]: Entering directory `/home/administrador/freeswitch/src/mod/languages/mod_java' make[6]: Entering directory `/home/administrador/freeswitch/src/mod/languages/mod_java' make[6]: Nothing to be done for `install-exec-am'. cp freeswitch.jar /opt/CloudVoice-vPBX/fs/scripts cp: cannot stat ?freeswitch.jar?: No such file or directory make[6]: *** [install-data-local] Error 1 make[6]: Leaving directory `/home/administrador/freeswitch/src/mod/languages/mod_java' make[5]: *** [install-am] Error 2 make[5]: Leaving directory `/home/administrador/freeswitch/src/mod/languages/mod_java' make[4]: *** [install] Error 2 make[4]: Leaving directory `/home/administrador/freeswitch/src/mod/languages/mod_java' make[3]: *** [mod_java-install] Error 1 make[3]: Leaving directory `/home/administrador/freeswitch/src/mod' make[2]: *** [install-recursive] Error 1 make[2]: Leaving directory `/home/administrador/freeswitch/src' Can someone help? Sin mas a que hacer referencia, Victor Medina _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141208/a99d254a/attachment-0001.html From mike at jerris.com Mon Dec 8 20:15:47 2014 From: mike at jerris.com (Michael Jerris) Date: Mon, 8 Dec 2014 12:15:47 -0500 Subject: [Freeswitch-users] Billing software In-Reply-To: References: <518CA7DD.8020802@gmail.com> Message-ID: <3792C5C9-2B4D-4E9C-BA65-B857F7D9893A@jerris.com> Is this something that would be useful to integrate directly into Fusion? > On Dec 7, 2014, at 9:58 AM, Luis Daniel Lucio Quiroz wrote: > > Maybe a little late, > > Thanks everyone, but I am really against to install an extra freeswitch instance for the billing. It makes very complex the call flow, troubleshoting and in most cases FAX wont work. > > I did try the ASTPP before, horrible, hard to configure. I consider myself good on freeswitch, i know to code on perl, and a good sysadmin. But after watching how complex is to get this to work I abort. > > So I decide to code my own billing and gladly, I can tell you that if you have all information needed (such as carrier rates and about 15 mins to get an explanation on how it works, you can have it working in less than an hour). I started to study FusionPBX code, and after some months I have a working nice FusionPBX LCR+Billing native application. It fully integrates in the Fusion, not out side, so you are able to bill for anything, incoming, internal (ext to ext, ext to conference, ext to ivr), or outgoing calls and it is able to get money by Paypal, Stripe (a good gw for CreditCards) and offline. Currently, I have just release version 1.0.2 > > If someone is interested, you can go to my website www.okay.com.mx/en or contact me (gtalk prefered) > > LD > > 2013-05-10 3:55 GMT-04:00 Deon Vermeulen >: > Hi Luis > > We just bought Commercial services from ASTPP. > When it comes to comparing price, features this is the BEST solution out there at the moment. > > This is a very active project and you can contact Samir Doshi, Project Maintainer & Developer, directly for more information wrt your technical questions. > > samir at astpp.org > > > Kind Regards > Deon -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141208/e78cf14d/attachment.html From mike at jerris.com Mon Dec 8 20:22:02 2014 From: mike at jerris.com (Michael Jerris) Date: Mon, 8 Dec 2014 12:22:02 -0500 Subject: [Freeswitch-users] Windows build system In-Reply-To: References: Message-ID: I'm a bit out of date on what these tools provide and need someone to get me up to speed on what is available. Can you help me understand what the options are for maintaining c libraries in a way that is cleaner than needing to stuff them all into our build. > On Dec 8, 2014, at 5:29 AM, Markus von Arx wrote: > > We are fine with your Visual Studio plans. We only use VS2012 at the moment and actually would like to migrate to VS2013. > > Not so sure about the use of chocolatey though. Are you planning to use it to download libraries required by the build or for installing software/tools required for the build? Both look a bit problematic to me, because chocolatey (as far as I understand it) always acts globally on the target machine, meaning that it installs the libs/tools/software not locally inside the build directory but globally on the machine. I wouldn't like that at all. I don't want a build tool/process to install anything outside the build directory. If you manage to use chocolatey to only work in the build directory that's fine for me. But I'd strongly vote against any use of chocolatey to install libraries, tools or any software globally or outside the build directory. I wouldn't like a build tool/process/system installing anything on my machine for me automatically. Kind of like calling apt-get on a Linux machine from a build script. I think that is a no-go. Also, chocolatey is not so good detecting installed software that was not installed by chocolatey itself and would often try to re-install software that is already there. I'd vote against using chocolatey in the FreeSWITCH build if you ask me. Wouldn't nuget be the more natural choice anyway to install modules/libraries in Visual Studio? Just my opinion. > > Markus > > > 2014-12-03 21:31 GMT+01:00 Michael Jerris >: > Given the recent announcements by Microsoft about the community edition 2013 being available, we are working to migrate the build system towards using that as our primary build. As part of this process we will be very soon dropping support for any version of Visual Studio prior to 2012. If you feel strongly about needing support for these older versions, please speak up now with an offer to maintain these legacy build systems. We are also investigating moving to using chocolatey as a new system to manage dependencies on windows instead of maintaining the build for all our deps ourselves. It is also possible we will drop support for the 2012 build system in the not so distant future. Could the community chime in here as to what their needs are, and what they are willing to do to help support the windows builds so we can determine what we plan to support going forward. > > Thanks > Mike > > https://chocolatey.org/ > http://www.visualstudio.com/en-us/news/vs2013-community-vs.aspx -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141208/c859a51e/attachment.html From luis.daniel.lucio at gmail.com Mon Dec 8 20:28:41 2014 From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz) Date: Mon, 8 Dec 2014 12:28:41 -0500 Subject: [Freeswitch-users] Billing software In-Reply-To: <3792C5C9-2B4D-4E9C-BA65-B857F7D9893A@jerris.com> References: <518CA7DD.8020802@gmail.com> <3792C5C9-2B4D-4E9C-BA65-B857F7D9893A@jerris.com> Message-ID: Michael, indeed it is. Its a native application that fully integrates into the FusionPBX 2014-12-08 12:15 GMT-05:00 Michael Jerris : > Is this something that would be useful to integrate directly into Fusion? > > On Dec 7, 2014, at 9:58 AM, Luis Daniel Lucio Quiroz > wrote: > > Maybe a little late, > > Thanks everyone, but I am really against to install an extra freeswitch > instance for the billing. It makes very complex the call flow, > troubleshoting and in most cases FAX wont work. > > I did try the ASTPP before, horrible, hard to configure. I consider myself > good on freeswitch, i know to code on perl, and a good sysadmin. But after > watching how complex is to get this to work I abort. > > So I decide to code my own billing and gladly, I can tell you that if you > have all information needed (such as carrier rates and about 15 mins to get > an explanation on how it works, you can have it working in less than an > hour). I started to study FusionPBX code, and after some months I have a > working nice FusionPBX LCR+Billing native application. It fully integrates > in the Fusion, not out side, so you are able to bill for anything, incoming, > internal (ext to ext, ext to conference, ext to ivr), or outgoing calls and > it is able to get money by Paypal, Stripe (a good gw for CreditCards) and > offline. Currently, I have just release version 1.0.2 > > If someone is interested, you can go to my website www.okay.com.mx/en or > contact me (gtalk prefered) > > LD > > 2013-05-10 3:55 GMT-04:00 Deon Vermeulen : >> >> Hi Luis >> >> We just bought Commercial services from ASTPP. >> When it comes to comparing price, features this is the BEST solution out >> there at the moment. >> >> This is a very active project and you can contact Samir Doshi, Project >> Maintainer & Developer, directly for more information wrt your technical >> questions. >> >> samir at astpp.org >> >> >> Kind Regards >> Deon > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 jim at homeqwik.com Mon Dec 8 23:52:32 2014 From: jim at homeqwik.com (Jim Hughes) Date: Mon, 8 Dec 2014 13:52:32 -0700 Subject: [Freeswitch-users] Logging Time After Transfer Message-ID: We would like our Call Detail Records from FreeSwitch to log the time after a call is transferred. I read entries in the forums that if we ?enable legs? we might get that info, but it didn?t seem to work. We got lots more information, but not the time after transfer. For example: An agent receives a call and talks to somebody for 1 minute, then transfers the call to a 3rd party and that call continues for 5 minutes. We?d like the Call Detail Record to contain both pieces of information: Time Before Transfer: 1 Minute Time After Transfer: 5 Minutes Has somebody already done this so we don?t have to re-invent the wheel? Thanks, Jim. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141208/a9d66545/attachment-0001.html From t.mahe at b-and-c.net Tue Dec 9 01:23:17 2014 From: t.mahe at b-and-c.net (=?windows-1252?Q?Tristan_Mah=E9?=) Date: Mon, 08 Dec 2014 14:23:17 -0800 Subject: [Freeswitch-users] Huge CPU LOAD increase between 1.4.12 and 1.4.14 In-Reply-To: <54827C1D.5000401@b-and-c.net> References: <5473a2b195f0d_ea61b9732884134@ip-10-156-208-135.mail> <54827C1D.5000401@b-and-c.net> Message-ID: <548624D5.5050902@b-and-c.net> Hi everyone, Just a little followup, this is caused by in a sip_profile. Happens on baremetal servers as well as Xen guest. Still investigating on the exact root of this behaviour change, but at least now we know what is causing the trouble. Best, Tristan. Le 05/12/2014 19:46, Tristan Mah? a ?crit : > Hi guys, > > While trying to find out the root cause of this, I just want to know if > someone is also seeing a huge cpu load with freeswitch 1.4.14, which was > not present in 1.4.12. > > OS is debian wheezy, packages are official ones. > > Same config, same trafic pattern, just an upgrade from 1.4.12 ( no other > packages updated ). > > We were previously seeing load around 0.5, with pikes up to around 2. > > We now are seeing a constant load of around 2, and pikes up to 20... > > If someone is also seeing this, that would be interesting to exchange, > helping find out what is causing this ! > > I'll file a jira once I find the root of this problem ( and no, this is > production servers, we can't put master there, we need stable ). > > Best, > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 473 bytes Desc: OpenPGP digital signature Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141208/80841994/attachment.bin From kbdfck at gmail.com Tue Dec 9 01:33:26 2014 From: kbdfck at gmail.com (Dmitry Sytchev) Date: Tue, 9 Dec 2014 02:33:26 +0400 Subject: [Freeswitch-users] Huge CPU LOAD increase between 1.4.12 and 1.4.14 In-Reply-To: <548624D5.5050902@b-and-c.net> References: <5473a2b195f0d_ea61b9732884134@ip-10-156-208-135.mail> <54827C1D.5000401@b-and-c.net> <548624D5.5050902@b-and-c.net> Message-ID: So, have you solved your problem? Did you change rtp-timer-name? After we moved to 64-bit system, 1.4 branch shows huge LA spikes up to 35 on 4-core (or 4 single core virtual processor) with only 200-300 concurrent channels and very low CPS (1-5). On the same call load LA of 1.2 server is significantly lower. Setting 1.4 64bit to normal priority with -np helps to lower LA, but not completely solves a problem. It shows very big values in "ni" top fields (20-30%), but changing nice value of FS process to zero or positive value doesn't help. 2014-12-09 2:23 GMT+04:00 Tristan Mah? : > Hi everyone, > > Just a little followup, this is caused by value="soft"/> in a sip_profile. Happens on baremetal servers as well as > Xen guest. > > Still investigating on the exact root of this behaviour change, but at > least now we know what is causing the trouble. > > Best, > > Tristan. > > Le 05/12/2014 19:46, Tristan Mah? a ?crit : > > Hi guys, > > > > While trying to find out the root cause of this, I just want to know if > > someone is also seeing a huge cpu load with freeswitch 1.4.14, which was > > not present in 1.4.12. > > > > OS is debian wheezy, packages are official ones. > > > > Same config, same trafic pattern, just an upgrade from 1.4.12 ( no other > > packages updated ). > > > > We were previously seeing load around 0.5, with pikes up to around 2. > > > > We now are seeing a constant load of around 2, and pikes up to 20... > > > > If someone is also seeing this, that would be interesting to exchange, > > helping find out what is causing this ! > > > > I'll file a jira once I find the root of this problem ( and no, this is > > production servers, we can't put master there, we need stable ). > > > > Best, > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://confluence.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20141209/03ac3dde/attachment.html From t.mahe at b-and-c.net Tue Dec 9 01:48:42 2014 From: t.mahe at b-and-c.net (=?windows-1252?Q?Tristan_Mah=E9?=) Date: Mon, 08 Dec 2014 14:48:42 -0800 Subject: [Freeswitch-users] Huge CPU LOAD increase between 1.4.12 and 1.4.14 In-Reply-To: References: <5473a2b195f0d_ea61b9732884134@ip-10-156-208-135.mail> <54827C1D.5000401@b-and-c.net> <548624D5.5050902@b-and-c.net> Message-ID: <54862ACA.6000506@b-and-c.net> Still on it, but I have some servers deployed without rtp-timer-name set to soft ( blocking read is something I'd like to avoid, but needs to find out what changed in FS code and also find out the impact on our users ), and so far they have load avg divided by 10 with the same amount of calls and the same trafic as those with rtp-timer-name to soft. Had no issue with 1.4.12 as said previously, but 1.4.14 has some interesting fixes that justify staying in production. Le 08/12/2014 14:33, Dmitry Sytchev a ?crit : > So, have you solved your problem? Did you change rtp-timer-name? > After we moved to 64-bit system, 1.4 branch shows huge LA spikes up to > 35 on 4-core (or 4 single core virtual processor) with only 200-300 > concurrent channels and very low CPS (1-5). On the same call load LA of > 1.2 server is significantly lower. > Setting 1.4 64bit to normal priority with -np helps to lower LA, but not > completely solves a problem. > It shows very big values in "ni" top fields (20-30%), but changing nice > value of FS process to zero or positive value doesn't help. > > > > > 2014-12-09 2:23 GMT+04:00 Tristan Mah? >: > > Hi everyone, > > Just a little followup, this is caused by value="soft"/> in a sip_profile. Happens on baremetal servers as well as > Xen guest. > > Still investigating on the exact root of this behaviour change, but at > least now we know what is causing the trouble. > > Best, > > Tristan. > > Le 05/12/2014 19:46, Tristan Mah? a ?crit : > > Hi guys, > > > > While trying to find out the root cause of this, I just want to > know if > > someone is also seeing a huge cpu load with freeswitch 1.4.14, > which was > > not present in 1.4.12. > > > > OS is debian wheezy, packages are official ones. > > > > Same config, same trafic pattern, just an upgrade from 1.4.12 ( no > other > > packages updated ). > > > > We were previously seeing load around 0.5, with pikes up to around 2. > > > > We now are seeing a constant load of around 2, and pikes up to 20... > > > > If someone is also seeing this, that would be interesting to exchange, > > helping find out what is causing this ! > > > > I'll file a jira once I find the root of this problem ( and no, > this is > > production servers, we can't put master there, we need stable ). > > > > Best, > > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://confluence.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 473 bytes Desc: OpenPGP digital signature Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141208/59624dcc/attachment.bin From andrew.keil at visytel.com Tue Dec 9 02:00:39 2014 From: andrew.keil at visytel.com (Andrew Keil) Date: Mon, 8 Dec 2014 23:00:39 +0000 Subject: [Freeswitch-users] Re - FreeSWITCH Lua CUSTOM events interrupting session:streamFile playback - is this possible? Message-ID: To FreeSWITCH users, After progressing with Lua inside FreeSWITCH I have a question that I cannot source an answer for within the current FreeSWITCH documentation. I have a requirement for an external application to process an event (via the event socket layer) then return the results back to the application while an audio file is played back in a loop to the caller. >From the current documentation this method allows for the event to be sent then the return event is "consumed" using a polling approach. However the caller is listening to silence. function poll() -- create event and listener local event = freeswitch.Event("CUSTOM", "ping::running?") local con = freeswitch.EventConsumer("CUSTOM", "ping::running!") -- add text ad libitum event:addHeader("hi", "there") -- fire event event:fire() -- and wait for reply but not very long local retevent = con:pop(1, 5000) if retevent then print("reply received") freeswitch.consoleLog("DEBUG", string.format("reply received: %s\n",retevent:getHeader("Result"))) return true end print("no reply") freeswitch.consoleLog("DEBUG", "no reply\n") return false end Some questions: 1) Is there a way to playback audio (eg. session:streamFile(...)) while this takes place, since currently session:streamFile(...) seems to be a blocking function (ie. Finishes when the audio file is played back completely)? 2) Is there a way to use session:setInputCallback(...) to handle an external CUSTOM event being returned (since this would then be able to interrupt the session:streamFile(...) just like it does for DTMF or speech recognition)? If you have a preferred approach to solving this then I am open to your suggestions. I appreciate any assistance that you can give. Kind Regards, Andrew Keil -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141208/b64d6937/attachment-0001.html From anthony.minessale at gmail.com Tue Dec 9 02:02:41 2014 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 8 Dec 2014 17:02:41 -0600 Subject: [Freeswitch-users] Huge CPU LOAD increase between 1.4.12 and 1.4.14 In-Reply-To: <54862ACA.6000506@b-and-c.net> References: <5473a2b195f0d_ea61b9732884134@ip-10-156-208-135.mail> <54827C1D.5000401@b-and-c.net> <548624D5.5050902@b-and-c.net> <54862ACA.6000506@b-and-c.net> Message-ID: This should have been in JIRA... Drop the mentality that you have to make a case first to open a JIRA. The actual process should be, if its any question about how FS works, if you are right or wrong about it, file it as a JIRA. If its wrong we can just close it........ If you want to send one email on the list to get some attention to your jira for discussion its ok but really you want to have the entire thing preserved there. I pushed a patch (4bcf1d888a4629993923527c26d4dbc6dae5d470) that might be related to your problem but now I have to wing it and use email where I get 200 emails a day and there is no tracking at all in email when it comes to issues so you put more resources on me to try to remember to look for your reply when we have a perfectly valid issue tracker system in place..... If master works better, good, if not, try git checkout 878a04715ae801ccd9587249240f5a6e4f16dd0e src/switch_time.c and make install_core All of this should be transposed back to JIRA by one of you..... On Mon, Dec 8, 2014 at 4:48 PM, Tristan Mah? wrote: > Still on it, but I have some servers deployed without rtp-timer-name set > to soft ( blocking read is something I'd like to avoid, but needs to > find out what changed in FS code and also find out the impact on our > users ), and so far they have load avg divided by 10 with the same > amount of calls and the same trafic as those with rtp-timer-name to soft. > > Had no issue with 1.4.12 as said previously, but 1.4.14 has some > interesting fixes that justify staying in production. > > Le 08/12/2014 14:33, Dmitry Sytchev a ?crit : > > So, have you solved your problem? Did you change rtp-timer-name? > > After we moved to 64-bit system, 1.4 branch shows huge LA spikes up to > > 35 on 4-core (or 4 single core virtual processor) with only 200-300 > > concurrent channels and very low CPS (1-5). On the same call load LA of > > 1.2 server is significantly lower. > > Setting 1.4 64bit to normal priority with -np helps to lower LA, but not > > completely solves a problem. > > It shows very big values in "ni" top fields (20-30%), but changing nice > > value of FS process to zero or positive value doesn't help. > > > > > > > > > > 2014-12-09 2:23 GMT+04:00 Tristan Mah? > >: > > > > Hi everyone, > > > > Just a little followup, this is caused by name="rtp-timer-name" > > value="soft"/> in a sip_profile. Happens on baremetal servers as > well as > > Xen guest. > > > > Still investigating on the exact root of this behaviour change, but > at > > least now we know what is causing the trouble. > > > > Best, > > > > Tristan. > > > > Le 05/12/2014 19:46, Tristan Mah? a ?crit : > > > Hi guys, > > > > > > While trying to find out the root cause of this, I just want to > > know if > > > someone is also seeing a huge cpu load with freeswitch 1.4.14, > > which was > > > not present in 1.4.12. > > > > > > OS is debian wheezy, packages are official ones. > > > > > > Same config, same trafic pattern, just an upgrade from 1.4.12 ( no > > other > > > packages updated ). > > > > > > We were previously seeing load around 0.5, with pikes up to around > 2. > > > > > > We now are seeing a constant load of around 2, and pikes up to > 20... > > > > > > If someone is also seeing this, that would be interesting to > exchange, > > > helping find out what is causing this ! > > > > > > I'll file a jira once I find the root of this problem ( and no, > > this is > > > production servers, we can't put master there, we need stable ). > > > > > > Best, > > > > > > > > > > > > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org > > > http://www.freeswitchsolutions.com > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://confluence.freeswitch.org > > > http://www.cluecon.com > > > > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://confluence.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://confluence.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 ? @anthmfs ? @FreeSWITCH ? ? http://freeswitch.org/ ? http://cluecon.com/ ? http://twitter.com/FreeSWITCH ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ * ClueCon Weekly Development Call ? sip:888 at conference.freeswitch.org ? +19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141208/b9c54c62/attachment.html From dujinfang at gmail.com Tue Dec 9 02:15:45 2014 From: dujinfang at gmail.com (Seven Du) Date: Tue, 9 Dec 2014 07:15:45 +0800 Subject: [Freeswitch-users] Re - FreeSWITCH Lua CUSTOM events interrupting session:streamFile playback - is this possible? In-Reply-To: References: Message-ID: I believe the playback_terminitors chan var works, or you could try set an input callback and return ?break? when dtmf detected. -- Seven Du http://about.me/dujinfang http://www.dujinfang.com http://www.freeswitch.org.cn Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Tuesday, December 9, 2014 at 7:00 AM, Andrew Keil wrote: > To FreeSWITCH users, > > After progressing with Lua inside FreeSWITCH I have a question that I cannot source an answer for within the current FreeSWITCH documentation. > > I have a requirement for an external application to process an event (via the event socket layer) then return the results back to the application while an audio file is played back in a loop to the caller. > > From the current documentation this method allows for the event to be sent then the return event is ?consumed? using a polling approach. However the caller is listening to silence. > > function poll() > -- create event and listener > local event = freeswitch.Event("CUSTOM", "ping::running?") > local con = freeswitch.EventConsumer("CUSTOM", "ping::running!") > > -- add text ad libitum > event:addHeader("hi", "there") > -- fire event > event:fire() > -- and wait for reply but not very long > local retevent = con:pop(1, 5000) > if retevent then > print("reply received") > freeswitch.consoleLog("DEBUG", string.format("reply received: %s\n",retevent:getHeader("Result"))) > return true > end > print("no reply") > freeswitch.consoleLog("DEBUG", "no reply\n") > return false > end > > Some questions: > 1) Is there a way to playback audio (eg. session:streamFile(?)) while this takes place, since currently session:streamFile(?) seems to be a blocking function (ie. Finishes when the audio file is played back completely)? > 2) Is there a way to use session:setInputCallback(?) to handle an external CUSTOM event being returned (since this would then be able to interrupt the session:streamFile(?) just like it does for DTMF or speech recognition)? > > If you have a preferred approach to solving this then I am open to your suggestions. > > I appreciate any assistance that you can give. > > Kind Regards, > > Andrew Keil > > > > _________________________________________________________________________ > 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://confluence.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/20141209/ff09e0ae/attachment-0001.html From krice at freeswitch.org Tue Dec 9 02:25:33 2014 From: krice at freeswitch.org (Ken Rice) Date: Mon, 08 Dec 2014 23:25:33 +0000 Subject: [Freeswitch-users] FreeSWITCH Week in Review (Master Branch) November 30th-December 6th Message-ID: <5486336d44e1e_a5711c53302017d@ip-10-69-141-60.mail> New Post on freeswitch.org from kathleen check it out at http://ift.tt/1AWUUpw FreeSWITCH Week in Review (Master Branch) November 30th-December 6th Hello, again. This week in the FreeSWITCH master branch we had 27 commits. The features this week are: timezone support for mod_say_{de,es,ja,nl,th,zh},? and the addition of a drop_dtmf_masking_tone channel variable which, if set, would replace all DTMF tones by the single tone specified in the variable. New features that were added: e55aee1 FS-7025 Add drop_dtmf_masking_tone channel_variable [Jira: http://ift.tt/1yJc1Mn] a8c5a0c FS-7048 Add timezone support to mod_say_{de,es,ja,nl,th,zh} Improvements in cross platform build supports: dc9e904 FS-7025 fix compiler warning introduced from e55aee14 [Jira: http://ift.tt/1yJc1Mn] b69c93e FS-7030 More work toward fixing FS build on Windows Visual Studio 2012 [Jira: http://ift.tt/1AWUUFW] db66cdb Fix mrcp libraries to build correctly c327455 FS-7030 More work toward getting FS to build on Windows Visual Studio 2012 [Jira: http://ift.tt/1AWUUFW] b341ff7 FS-7046: fix data types and casting on some vars to silence windows build warnings in mod_verto [Jira: http://ift.tt/1yJc4rg] 7ce5171 FS-7046 follow up on type change in mod_verto [Jira: http://ift.tt/1yJc4rg] The following bugs were squashed: 46adbec FS-7030 #comment [unimrcp] restore visual studio 2010/2012 project files added by FS project [Jira: http://ift.tt/1AWUUFW] bad5dc3 FS-7037 Fix for T38 fax break started by commit 5bbef7f1e50 [Jira: http://ift.tt/1AWUXBF] 72c3df5 FS-6891 FS-6713 #comment revert 1b612fecb6e8db11da9b15c5522b87e7b642423d [Jira: http://ift.tt/1yJc22F] 2a7b022 FS-6980 #resolve don?t crash when using native recording on recordstop the redo [Jira: http://ift.tt/11E7i0U] 35ba6a3 FS-6766 Fix verto caller ringback missing on conference bridge in mod_verto [Jira: http://ift.tt/1AWUUG2] e8cf9c7 FS-7045 Guarantee that dialed call can be joined when answered event is sent in mod_rayo [Jira: http://ift.tt/1yJc4rj] 4be6290 FS-7052 Moving jb queue swap operation out of the debug block. [Jira: http://ift.tt/1AWUXS0] 843e495 FS-7051 Preserve the annexb=no/yes status in mod_sangoma_codec [Jira: http://ift.tt/1yJc4rl] 158c1f2 FS-7002 Fix for recorded audio being choppy when diferent ptimes present and record session starts on bleg [Jira: http://ift.tt/1AWUXS3] The complete list of commits can be found here:2014_11_30-2014_12_7 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141208/8ad49840/attachment.html From t.mahe at b-and-c.net Tue Dec 9 02:28:33 2014 From: t.mahe at b-and-c.net (=?UTF-8?B?VHJpc3RhbiBNYWjDqQ==?=) Date: Mon, 08 Dec 2014 15:28:33 -0800 Subject: [Freeswitch-users] Huge CPU LOAD increase between 1.4.12 and 1.4.14 In-Reply-To: References: <5473a2b195f0d_ea61b9732884134@ip-10-156-208-135.mail> <54827C1D.5000401@b-and-c.net> <548624D5.5050902@b-and-c.net> <54862ACA.6000506@b-and-c.net> Message-ID: <54863421.1080809@b-and-c.net> thanks Anthony, FS-7066 created with the content of the email. Will try this patch on our test server with a little flow of trafic as soon as I have the chance ! Le 08/12/2014 15:02, Anthony Minessale a ?crit : > This should have been in JIRA... Drop the mentality that you have to > make a case first to open a JIRA. > The actual process should be, if its any question about how FS works, if > you are right or wrong about it, file it as a JIRA. If its wrong we can > just close it........ > > If you want to send one email on the list to get some attention to your > jira for discussion its ok but really you want to have the entire thing > preserved there. > > > I pushed a patch (4bcf1d888a4629993923527c26d4dbc6dae5d470) that might > be related to your problem but now I have to wing it and use email where > I get 200 emails a day and there is no tracking at all in email when it > comes to issues so you put more resources on me to try to remember to > look for your reply when we have a perfectly valid issue tracker system > in place..... > > > If master works better, good, if not, try > > git checkout 878a04715ae801ccd9587249240f5a6e4f16dd0e src/switch_time.c > and make install_core > > All of this should be transposed back to JIRA by one of you..... > > > > > On Mon, Dec 8, 2014 at 4:48 PM, Tristan Mah? > wrote: > > Still on it, but I have some servers deployed without rtp-timer-name set > to soft ( blocking read is something I'd like to avoid, but needs to > find out what changed in FS code and also find out the impact on our > users ), and so far they have load avg divided by 10 with the same > amount of calls and the same trafic as those with rtp-timer-name to > soft. > > Had no issue with 1.4.12 as said previously, but 1.4.14 has some > interesting fixes that justify staying in production. > > Le 08/12/2014 14:33, Dmitry Sytchev a ?crit : > > So, have you solved your problem? Did you change rtp-timer-name? > > After we moved to 64-bit system, 1.4 branch shows huge LA spikes up to > > 35 on 4-core (or 4 single core virtual processor) with only 200-300 > > concurrent channels and very low CPS (1-5). On the same call load LA of > > 1.2 server is significantly lower. > > Setting 1.4 64bit to normal priority with -np helps to lower LA, but not > > completely solves a problem. > > It shows very big values in "ni" top fields (20-30%), but changing nice > > value of FS process to zero or positive value doesn't help. > > > > > > > > > > 2014-12-09 2:23 GMT+04:00 Tristan Mah? > > >>: > > > > Hi everyone, > > > > Just a little followup, this is caused by name="rtp-timer-name" > > value="soft"/> in a sip_profile. Happens on baremetal servers > as well as > > Xen guest. > > > > Still investigating on the exact root of this behaviour > change, but at > > least now we know what is causing the trouble. > > > > Best, > > > > Tristan. > > > > Le 05/12/2014 19:46, Tristan Mah? a ?crit : > > > Hi guys, > > > > > > While trying to find out the root cause of this, I just want to > > know if > > > someone is also seeing a huge cpu load with freeswitch 1.4.14, > > which was > > > not present in 1.4.12. > > > > > > OS is debian wheezy, packages are official ones. > > > > > > Same config, same trafic pattern, just an upgrade from > 1.4.12 ( no > > other > > > packages updated ). > > > > > > We were previously seeing load around 0.5, with pikes up to > around 2. > > > > > > We now are seeing a constant load of around 2, and pikes up > to 20... > > > > > > If someone is also seeing this, that would be interesting to > exchange, > > > helping find out what is causing this ! > > > > > > I'll file a jira once I find the root of this problem ( and no, > > this is > > > production servers, we can't put master there, we need stable ). > > > > > > Best, > > > > > > > > > > > > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org > > > > > http://www.freeswitchsolutions.com > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://confluence.freeswitch.org > > > http://www.cluecon.com > > > > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > > > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://confluence.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://confluence.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 ? @anthmfs ? @FreeSWITCH ? > > ? http://freeswitch.org/ ? http://cluecon.com/ ? > http://twitter.com/FreeSWITCH > ? irc.freenode.net #freeswitch ? > _http://freeswitch.org/g+_ > > ClueCon Weekly Development Call > ? sip:888 at conference.freeswitch.org > ? +19193869900 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 473 bytes Desc: OpenPGP digital signature Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141208/579c5891/attachment.bin From umairedu at gmail.com Tue Dec 9 13:31:41 2014 From: umairedu at gmail.com (Umair Khan) Date: Tue, 9 Dec 2014 15:31:41 +0500 Subject: [Freeswitch-users] Billing software In-Reply-To: References: <518CA7DD.8020802@gmail.com> Message-ID: Hey Guys, Checkout Hyperbilling.com for freeswitch billing. Thanks & Regards On Sun, Dec 7, 2014 at 7:58 PM, Luis Daniel Lucio Quiroz < luis.daniel.lucio at gmail.com> wrote: > Maybe a little late, > > Thanks everyone, but I am really against to install an extra freeswitch > instance for the billing. It makes very complex the call flow, > troubleshoting and in most cases FAX wont work. > > I did try the ASTPP before, horrible, hard to configure. I consider myself > good on freeswitch, i know to code on perl, and a good sysadmin. But after > watching how complex is to get this to work I abort. > > So I decide to code my own billing and gladly, I can tell you that if you > have all information needed (such as carrier rates and about 15 mins to get > an explanation on how it works, you can have it working in less than an > hour). I started to study FusionPBX code, and after some months I have a > working nice FusionPBX LCR+Billing native application. It fully integrates > in the Fusion, not out side, so you are able to bill for anything, > incoming, internal (ext to ext, ext to conference, ext to ivr), or outgoing > calls and it is able to get money by Paypal, Stripe (a good gw for > CreditCards) and offline. Currently, I have just release version 1.0.2 > > If someone is interested, you can go to my website www.okay.com.mx/en or > contact me (gtalk prefered) > > LD > > 2013-05-10 3:55 GMT-04:00 Deon Vermeulen : > >> Hi Luis >> >> We just bought Commercial services from ASTPP. >> When it comes to comparing price, features this is the BEST solution out >> there at the moment. >> >> This is a very active project and you can contact Samir Doshi, Project >> Maintainer & Developer, directly for more information wrt your technical >> questions. >> >> samir at astpp.org >> >> >> Kind Regards >> Deon >> >> Luis Daniel Lucio Quiroz >> May 9, 2013 11:46 PM >> >> >> Deon, thank you. Can you talk me about your exerience and how fast they >> are on fixing bugs. I have a really good or bad luck to find bugs. >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> Deon Vermeulen >> May 5, 2013 4:52 PM >> Check out ASTPP >> >> >> Kind Regards >> Deon Vermeulen >> >> Sent from my iPhone >> >> Luis Daniel Lucio Quiroz >> May 5, 2013 3:36 PM >> What other options for FS compatible software other than vBilling do you >> recommend me? >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE: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 >> 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141209/5c01f25e/attachment-0001.html From akhilgarg7 at gmail.com Tue Dec 9 16:28:19 2014 From: akhilgarg7 at gmail.com (akhil garg) Date: Tue, 9 Dec 2014 18:58:19 +0530 Subject: [Freeswitch-users] fs_cli hangs Message-ID: freeswitch is running on 192.168.1.1 by command freeswitch. file "freeswitch/autoload_configs/event_socket.conf.xml" has been edited running "fs_cli -H 192.168.1.1 -d 7" from other machine e.g. 192.168.1.2 on lan is successfull. running "fs_cli -H 192.168.1.1 -d 7" from same machine i.e. 192.168.1.1 hangs. Regards, Akhil -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141209/cdbfd846/attachment.html From akhilgarg7 at gmail.com Tue Dec 9 16:28:49 2014 From: akhilgarg7 at gmail.com (akhil garg) Date: Tue, 9 Dec 2014 18:58:49 +0530 Subject: [Freeswitch-users] fs_cli hangs In-Reply-To: References: Message-ID: I am using freeswitch 1.5.6b On Tue, Dec 9, 2014 at 6:58 PM, akhil garg wrote: > freeswitch is running on 192.168.1.1 by command freeswitch. > > file "freeswitch/autoload_configs/event_socket.conf.xml" has been edited > > > > running "fs_cli -H 192.168.1.1 -d 7" from other machine e.g. 192.168.1.2 > on lan is successfull. > > running "fs_cli -H 192.168.1.1 -d 7" from same machine i.e. 192.168.1.1 > hangs. > > > Regards, > Akhil > -- regards, akhil -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141209/be75c270/attachment.html From vitaliy.davudov at vts24.ru Tue Dec 9 16:46:05 2014 From: vitaliy.davudov at vts24.ru (Vitaly) Date: Tue, 09 Dec 2014 16:46:05 +0300 Subject: [Freeswitch-users] Log-auth-failures question In-Reply-To: References: <5485B391.5050506@vts24.ru> Message-ID: <5486FD1D.4080900@vts24.ru> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141209/8de1001c/attachment.html From blefko5361 at gmail.com Tue Dec 9 18:19:32 2014 From: blefko5361 at gmail.com (Bruce Lefko) Date: Tue, 9 Dec 2014 09:19:32 -0600 Subject: [Freeswitch-users] packaging FS debs using existing source tree Message-ID: I'd like to try making some changes to FS and then use the debian/util.sh to create deb packages, however it looks like that script re-checkouts the source to create the packages. Is there a way to make that script use the existing source tree that's already been checked out? Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141209/d3ff9a21/attachment.html From leo.noordergraaf at deanconnect.nl Tue Dec 9 18:34:42 2014 From: leo.noordergraaf at deanconnect.nl (Leo Noordergraaf [Dean Connect]) Date: Tue, 9 Dec 2014 15:34:42 +0000 Subject: [Freeswitch-users] packaging FS debs using existing source tree In-Reply-To: References: Message-ID: <1418139238.27339.3.camel@debian> I solved this by checking in my changes in a branch. Once everything is checked in, debian/util.sh works like a charm. Leo On Tue, 2014-12-09 at 09:19 -0600, Bruce Lefko wrote: > I'd like to try making some changes to FS and then use the > debian/util.sh to create deb packages, however it looks like that > script re-checkouts the source to create the packages. > > > Is there a way to make that script use the existing source tree that's > already been checked out? > > > Thanks! From blefko5361 at gmail.com Tue Dec 9 18:37:32 2014 From: blefko5361 at gmail.com (Bruce Lefko) Date: Tue, 9 Dec 2014 09:37:32 -0600 Subject: [Freeswitch-users] packaging FS debs using existing source tree In-Reply-To: <1418139238.27339.3.camel@debian> References: <1418139238.27339.3.camel@debian> Message-ID: So just to confirm, as long as I check in my changes locally (on the machine where I'm building), I can just run debian/util.sh and it'll pick up my changes? I don't have to push my changes out to stash or anything right? Thanks! On Tue, Dec 9, 2014 at 9:34 AM, Leo Noordergraaf [Dean Connect] < leo.noordergraaf at deanconnect.nl> wrote: > I solved this by checking in my changes in a branch. > Once everything is checked in, debian/util.sh works like a charm. > Leo > > On Tue, 2014-12-09 at 09:19 -0600, Bruce Lefko wrote: > > I'd like to try making some changes to FS and then use the > > debian/util.sh to create deb packages, however it looks like that > > script re-checkouts the source to create the packages. > > > > > > Is there a way to make that script use the existing source tree that's > > already been checked out? > > > > > > Thanks! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141209/2d72f1c2/attachment.html From leo.noordergraaf at deanconnect.nl Tue Dec 9 18:54:43 2014 From: leo.noordergraaf at deanconnect.nl (Leo Noordergraaf [Dean Connect]) Date: Tue, 9 Dec 2014 15:54:43 +0000 Subject: [Freeswitch-users] packaging FS debs using existing source tree In-Reply-To: References: <1418139238.27339.3.camel@debian> Message-ID: <1418140438.27339.12.camel@debian> That's right, you don't have to push anything. Later you can do 'git pull' and your changes are automatically merged. Actually, I didn't even use a separate branch for the check-in. Leo On Tue, 2014-12-09 at 09:37 -0600, Bruce Lefko wrote: > So just to confirm, as long as I check in my changes locally (on the machine where I'm building), I can just run debian/util.sh and it'll pick up my changes? I don't have to push my changes out to stash or anything right? > > Thanks! > > On Tue, Dec 9, 2014 at 9:34 AM, Leo Noordergraaf [Dean Connect] wrote: > > > I solved this by checking in my changes in a branch. > Once everything is checked in, debian/util.sh works like a charm. > Leo > > > On Tue, 2014-12-09 at 09:19 -0600, Bruce Lefko wrote: > > I'd like to try making some changes to FS and then use the > > debian/util.sh to create deb packages, however it looks like that > > script re-checkouts the source to create the packages. > > > > > > Is there a way to make that script use the existing source tree that's > > already been checked out? > > > > > > Thanks! > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 luis.daniel.lucio at gmail.com Tue Dec 9 19:15:12 2014 From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz) Date: Tue, 9 Dec 2014 11:15:12 -0500 Subject: [Freeswitch-users] Billing software In-Reply-To: References: <518CA7DD.8020802@gmail.com> Message-ID: 2014-12-09 5:31 GMT-05:00 Umair Khan : > Hyperbilling.com wow http://www.hyperbilling.com/pricing/ that is expensive!! Mine is 49.99 monthly, the current call limit is set by your hardware not a license -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141209/94acd3a2/attachment.html From brian at freeswitch.org Tue Dec 9 21:09:01 2014 From: brian at freeswitch.org (Brian West) Date: Tue, 9 Dec 2014 12:09:01 -0600 Subject: [Freeswitch-users] fs_cli hangs In-Reply-To: References: Message-ID: on debian? On Tue, Dec 9, 2014 at 7:28 AM, akhil garg wrote: > freeswitch is running on 192.168.1.1 by command freeswitch. > > file "freeswitch/autoload_configs/event_socket.conf.xml" has been edited > > > > running "fs_cli -H 192.168.1.1 -d 7" from other machine e.g. 192.168.1.2 > on lan is successfull. > > running "fs_cli -H 192.168.1.1 -d 7" from same machine i.e. 192.168.1.1 > hangs. > > > Regards, > Akhil > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141209/a66f2b6b/attachment.html From steveayre at gmail.com Wed Dec 10 00:28:06 2014 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 9 Dec 2014 21:28:06 +0000 Subject: [Freeswitch-users] fs_cli hangs In-Reply-To: References: Message-ID: Do you have a firewall protecting the port? If so perhaps it's blocking the local connection based on the IP you connect from. Does it eventually time out, eg after 30-60 seconds? Or hang forever? On 9 December 2014 at 13:28, akhil garg wrote: > freeswitch is running on 192.168.1.1 by command freeswitch. > > file "freeswitch/autoload_configs/event_socket.conf.xml" has been edited > > > > running "fs_cli -H 192.168.1.1 -d 7" from other machine e.g. 192.168.1.2 > on lan is successfull. > > running "fs_cli -H 192.168.1.1 -d 7" from same machine i.e. 192.168.1.1 > hangs. > > > Regards, > Akhil > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141209/7ddc9ee2/attachment.html From joelewhite at gmail.com Wed Dec 10 00:47:21 2014 From: joelewhite at gmail.com (Joel White) Date: Tue, 9 Dec 2014 16:47:21 -0500 Subject: [Freeswitch-users] Caller ID In-Reply-To: References: Message-ID: Sorry for the delay in response. I have gone over the config with a fine tooth comb, it matches another config of a server in which the caller id works fine. What I am seeing however is that in this system the variable is not exported to the dialplan. I may be missing something, and most likely I am. I do have a question though. Is there a way to see what variables are defined for a particular user in the FreeSWITCH console? On Thu, Nov 13, 2014 at 1:56 PM, Stanislav Sinyagin wrote: > Probably your ITSP does not allow you to set the caller ID? Did you run a > SIP packet trace to see what caller ID is sent out? > On Nov 13, 2014 4:45 PM, "Joel White" wrote: > >> I have several installations of FreeSWITCH. I have managed to get one to >> dynamically create the user directory from PostgreSQL and it properly sets >> the outbound caller id. I have another system, very similar in how it was >> configured. The second installation also dynamically generates the user >> directory from PostgreSQL as well. I am however having an issue with >> setting the caller id on this other system. I have went through with a >> fine tooth comb and am having a hard time locating the discrepancy between >> the two systems. >> >> Has anyone had an issue similar to this? >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141209/4cc30887/attachment.html From joelewhite at gmail.com Wed Dec 10 00:59:14 2014 From: joelewhite at gmail.com (Joel White) Date: Tue, 9 Dec 2014 16:59:14 -0500 Subject: [Freeswitch-users] Caller ID In-Reply-To: References: Message-ID: Here is some output of the Lua script on the server not pushing caller id 2014-12-09 16:54:22.819251 [NOTICE] switch_cpp.cpp:1328 Debug from gen_dir_user_xml.lua, generated XML:
And some output from the Lua script on the server with CID functioning 2014-12-09 21:50:55.458996 [NOTICE] switch_cpp.cpp:1328 Debug from gen_dir_user_xml.lua, generated XML:
Of course I removed any identifiable information, but it looks like the CID is being set. What am I missing here that is not allowing for the variable to be passed to the dialplan? On Tue, Dec 9, 2014 at 4:47 PM, Joel White wrote: > Sorry for the delay in response. I have gone over the config with a fine > tooth comb, it matches another config of a server in which the caller id > works fine. What I am seeing however is that in this system the variable > is not exported to the dialplan. I may be missing something, and most > likely I am. I do have a question though. Is there a way to see what > variables are defined for a particular user in the FreeSWITCH console? > > > > On Thu, Nov 13, 2014 at 1:56 PM, Stanislav Sinyagin > wrote: > >> Probably your ITSP does not allow you to set the caller ID? Did you run a >> SIP packet trace to see what caller ID is sent out? >> On Nov 13, 2014 4:45 PM, "Joel White" wrote: >> >>> I have several installations of FreeSWITCH. I have managed to get one >>> to dynamically create the user directory from PostgreSQL and it properly >>> sets the outbound caller id. I have another system, very similar in how it >>> was configured. The second installation also dynamically generates the >>> user directory from PostgreSQL as well. I am however having an issue with >>> setting the caller id on this other system. I have went through with a >>> fine tooth comb and am having a hard time locating the discrepancy between >>> the two systems. >>> >>> Has anyone had an issue similar to this? >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141209/e24b4ddd/attachment-0001.html From joelewhite at gmail.com Wed Dec 10 01:02:56 2014 From: joelewhite at gmail.com (Joel White) Date: Tue, 9 Dec 2014 17:02:56 -0500 Subject: [Freeswitch-users] Caller ID In-Reply-To: References: Message-ID: This is what I get when the dialplan executes EXECUTE sofia/internal/26342 at voip.net set(effective_caller_id_number=) 2014-12-09 17:00:24.839237 [DEBUG] mod_dptools.c:1435 sofia/internal/ 26342 at voip.net SET [effective_caller_id_number]=[UNDEF] Kinda strange and I could not find a discrepancy between the dialplan configuration of the working server vs the non-working server On Tue, Dec 9, 2014 at 4:59 PM, Joel White wrote: > Here is some output of the Lua script on the server not pushing caller id > > 2014-12-09 16:54:22.819251 [NOTICE] switch_cpp.cpp:1328 Debug from > gen_dir_user_xml.lua, generated XML: > > >
> > > > > > > > > > > > > > > > > > >
> > > > And some output from the Lua script on the server with CID functioning > > 2014-12-09 21:50:55.458996 [NOTICE] switch_cpp.cpp:1328 Debug from > gen_dir_user_xml.lua, generated XML: > > >
> > > > > > > > > > > > > > > > >
> > > > Of course I removed any identifiable information, but it looks like the > CID is being set. What am I missing here that is not allowing for the > variable to be passed to the dialplan? > > On Tue, Dec 9, 2014 at 4:47 PM, Joel White wrote: > >> Sorry for the delay in response. I have gone over the config with a fine >> tooth comb, it matches another config of a server in which the caller id >> works fine. What I am seeing however is that in this system the variable >> is not exported to the dialplan. I may be missing something, and most >> likely I am. I do have a question though. Is there a way to see what >> variables are defined for a particular user in the FreeSWITCH console? >> >> >> >> On Thu, Nov 13, 2014 at 1:56 PM, Stanislav Sinyagin >> wrote: >> >>> Probably your ITSP does not allow you to set the caller ID? Did you run >>> a SIP packet trace to see what caller ID is sent out? >>> On Nov 13, 2014 4:45 PM, "Joel White" wrote: >>> >>>> I have several installations of FreeSWITCH. I have managed to get one >>>> to dynamically create the user directory from PostgreSQL and it properly >>>> sets the outbound caller id. I have another system, very similar in how it >>>> was configured. The second installation also dynamically generates the >>>> user directory from PostgreSQL as well. I am however having an issue with >>>> setting the caller id on this other system. I have went through with a >>>> fine tooth comb and am having a hard time locating the discrepancy between >>>> the two systems. >>>> >>>> Has anyone had an issue similar to this? >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141209/f6dae8c7/attachment.html From joelewhite at gmail.com Wed Dec 10 01:06:29 2014 From: joelewhite at gmail.com (Joel White) Date: Tue, 9 Dec 2014 17:06:29 -0500 Subject: [Freeswitch-users] Caller ID In-Reply-To: References: Message-ID: Here is the version running on the server that works properly FreeSWITCH Version 1.4.13+git~20141103T195300Z~b942d0faa8~64bit (git b942d0f 2014-11-03 19:53:00Z 64bit) And the version of the server having issue with CID FreeSWITCH Version 1.4.13+git~20141103T195300Z~b942d0faa8~64bit (git b942d0f 2014-11-03 19:53:00Z 64bit) On Tue, Dec 9, 2014 at 5:02 PM, Joel White wrote: > This is what I get when the dialplan executes > > EXECUTE sofia/internal/26342 at voip.net > set(effective_caller_id_number=) > 2014-12-09 17:00:24.839237 [DEBUG] mod_dptools.c:1435 sofia/internal/ > 26342 at voip.net SET [effective_caller_id_number]=[UNDEF] > > > Kinda strange and I could not find a discrepancy between the dialplan > configuration of the working server vs the non-working server > > > > On Tue, Dec 9, 2014 at 4:59 PM, Joel White wrote: > >> Here is some output of the Lua script on the server not pushing caller id >> >> 2014-12-09 16:54:22.819251 [NOTICE] switch_cpp.cpp:1328 Debug from >> gen_dir_user_xml.lua, generated XML: >> >> >>
>> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >>
>> >> >> >> And some output from the Lua script on the server with CID functioning >> >> 2014-12-09 21:50:55.458996 [NOTICE] switch_cpp.cpp:1328 Debug from >> gen_dir_user_xml.lua, generated XML: >> >> >>
>> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >>
>> >> >> >> Of course I removed any identifiable information, but it looks like the >> CID is being set. What am I missing here that is not allowing for the >> variable to be passed to the dialplan? >> >> On Tue, Dec 9, 2014 at 4:47 PM, Joel White wrote: >> >>> Sorry for the delay in response. I have gone over the config with a >>> fine tooth comb, it matches another config of a server in which the caller >>> id works fine. What I am seeing however is that in this system the >>> variable is not exported to the dialplan. I may be missing something, and >>> most likely I am. I do have a question though. Is there a way to see what >>> variables are defined for a particular user in the FreeSWITCH console? >>> >>> >>> >>> On Thu, Nov 13, 2014 at 1:56 PM, Stanislav Sinyagin >> > wrote: >>> >>>> Probably your ITSP does not allow you to set the caller ID? Did you run >>>> a SIP packet trace to see what caller ID is sent out? >>>> On Nov 13, 2014 4:45 PM, "Joel White" wrote: >>>> >>>>> I have several installations of FreeSWITCH. I have managed to get one >>>>> to dynamically create the user directory from PostgreSQL and it properly >>>>> sets the outbound caller id. I have another system, very similar in how it >>>>> was configured. The second installation also dynamically generates the >>>>> user directory from PostgreSQL as well. I am however having an issue with >>>>> setting the caller id on this other system. I have went through with a >>>>> fine tooth comb and am having a hard time locating the discrepancy between >>>>> the two systems. >>>>> >>>>> Has anyone had an issue similar to this? >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141209/aae1f2a2/attachment-0001.html From joseph at onsip.com Wed Dec 10 01:30:49 2014 From: joseph at onsip.com (Joseph Frazier) Date: Tue, 9 Dec 2014 17:30:49 -0500 Subject: [Freeswitch-users] Incomplete UPDATEs after Attended Transfer In-Reply-To: References: Message-ID: Hi everyone, I'm trying to do attended transfers on my FS 1.5.14 box and have the endpoints receive UPDATE requests indicating that the identity of the other leg has changed. I installed the v1.5.14 tag from source and uncommented the following line of sip_profiles/internal.xml: Other than the above line, my configuration is completely vanilla. I have three extensions registered: 1000, 1007, 1006. My scenario is: - 1000 calls 1007 - 1000 calls 1006 - 1000 attended transfers 1007 to 1006 Here is my fs_cli with the following debugging options: sofia global siptrace on > sofia loglevel all 9 > sofia tracelevel alert > console loglevel debug > fsctl debug_level 10 https://gist.github.com/joseph-onsip/de90a66afdf37994c334 After receiving the REFER, FS sends an UPDATE to the refer target, but not to the referee. Is there a configuration option I can set in order to have an UPDATE sent to the referee as well? Thanks, Joseph -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141209/8a9baf83/attachment.html From krice at freeswitch.org Wed Dec 10 02:40:20 2014 From: krice at freeswitch.org (Ken Rice) Date: Tue, 09 Dec 2014 23:40:20 +0000 Subject: [Freeswitch-users] Happy holidays! Message-ID: <548788643371_3cb969133813668@ip-10-136-113-197.mail> New Post on freeswitch.org from kathleen check it out at http://ift.tt/1sgteqg Happy holidays! Hello, FreeSWITCH community! As I am sure you are all aware, the holiday season is upon us! The developers have been busy moving FreeSWITCH in some exciting new directions and I know some of you are very excited about some of the results. If you are feeling particularly appreciative, I?ve provided a few links to the wish-lists of the developers. These lists will make it easy to spread the holiday cheer and show just how much you are enjoying FreeSWITCH. Happy holidays! Anthony Minessale (anthm) ? http://ift.tt/1IwrlQE Mike Jerris (MikeJ) ? http://ift.tt/1Iwrkwa Brian West (bkw_) ? http://ift.tt/1uje6sN Ken Rice (SwK) ? http://ift.tt/1IwrlQI William King (quentusrex) ? http://ift.tt/1IwrkMo ? -Kathleen -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141209/55848ffb/attachment.html From brian at freeswitch.org Wed Dec 10 02:54:22 2014 From: brian at freeswitch.org (Brian West) Date: Tue, 9 Dec 2014 17:54:22 -0600 Subject: [Freeswitch-users] Incomplete UPDATEs after Attended Transfer In-Reply-To: References: Message-ID: What device is the Referee? On Tue, Dec 9, 2014 at 4:30 PM, Joseph Frazier wrote: > Hi everyone, > > I'm trying to do attended transfers on my FS 1.5.14 box and have the > endpoints receive UPDATE requests indicating that the identity of the other > leg has changed. I installed the v1.5.14 tag from source and uncommented > the following line of sip_profiles/internal.xml: > > > > Other than the above line, my configuration is completely vanilla. > > I have three extensions registered: 1000, 1007, 1006. My scenario is: > > - 1000 calls 1007 > - 1000 calls 1006 > - 1000 attended transfers 1007 to 1006 > > Here is my fs_cli with the following debugging options: > > sofia global siptrace on >> sofia loglevel all 9 >> sofia tracelevel alert >> console loglevel debug >> fsctl debug_level 10 > > > https://gist.github.com/joseph-onsip/de90a66afdf37994c334 > > After receiving the REFER, FS sends an UPDATE to the refer target, but not > to the referee. Is there a configuration option I can set in order to have > an UPDATE sent to the referee as well? > > Thanks, > Joseph > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141209/4bcb4e23/attachment.html From joseph at onsip.com Wed Dec 10 05:13:25 2014 From: joseph at onsip.com (Joseph Frazier) Date: Tue, 9 Dec 2014 21:13:25 -0500 Subject: [Freeswitch-users] Incomplete UPDATEs after Attended Transfer In-Reply-To: References: Message-ID: The Referee is a "Grandstream GXV3275 1.0.2.15". For context, here's where it OKs the INVITE from the first call: https://gist.github.com/joseph-onsip/de90a66afdf37994c334#file-gistfile1-txt-L1813-L1838 Thanks, Joseph On Tue, Dec 9, 2014 at 6:54 PM, Brian West wrote: > What device is the Referee? > > On Tue, Dec 9, 2014 at 4:30 PM, Joseph Frazier wrote: > >> Hi everyone, >> >> I'm trying to do attended transfers on my FS 1.5.14 box and have the >> endpoints receive UPDATE requests indicating that the identity of the other >> leg has changed. I installed the v1.5.14 tag from source and uncommented >> the following line of sip_profiles/internal.xml: >> >> >> >> Other than the above line, my configuration is completely vanilla. >> >> I have three extensions registered: 1000, 1007, 1006. My scenario is: >> >> - 1000 calls 1007 >> - 1000 calls 1006 >> - 1000 attended transfers 1007 to 1006 >> >> Here is my fs_cli with the following debugging options: >> >> sofia global siptrace on >>> sofia loglevel all 9 >>> sofia tracelevel alert >>> console loglevel debug >>> fsctl debug_level 10 >> >> >> https://gist.github.com/joseph-onsip/de90a66afdf37994c334 >> >> After receiving the REFER, FS sends an UPDATE to the refer target, but >> not to the referee. Is there a configuration option I can set in order to >> have an UPDATE sent to the referee as well? >> >> Thanks, >> Joseph >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > > *Brian West* > brian at freeswitch.org > > > *Twitter: @FreeSWITCH , @briankwest* > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141209/aa52b06a/attachment-0001.html From avi at avimarcus.net Wed Dec 10 08:50:53 2014 From: avi at avimarcus.net (Avi Marcus) Date: Wed, 10 Dec 2014 05:50:53 +0000 Subject: [Freeswitch-users] Newbie -- Help Needed Transferring Inbound Caller ID to external SIP Gateway URI In-Reply-To: References: <0000014a0b2a7f3a-e057e705-e811-4903-b2b9-bbd020abb4a7-000000@email.amazonses.com> <0000014a22d99893-8fb2f862-1a14-4cb2-b363-8e924d56012d-000000@email.amazonses.com> <0000014a2a0f8f9d-1038c04d-e533-483b-8322-bd27ace4ba19-000000@email.amazonses.com> Message-ID: <0000014a32c0097e-f9c55def-289f-48c9-bd1f-00e755c51f28-000000@email.amazonses.com> I would try something "simpler". This will always play the file, though, no matter the response from the othersipgw. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141210/ee0ecfbd/attachment.html From akhilgarg7 at gmail.com Wed Dec 10 09:16:56 2014 From: akhilgarg7 at gmail.com (akhil garg) Date: Wed, 10 Dec 2014 11:46:56 +0530 Subject: [Freeswitch-users] fs_cli hangs Message-ID: It hangs forever. command "netstat -anlp | grep 8021" gives the following output. tcp 0 0 127.0.0.1:34893 127.0.0.1:8021 ESTABLISHED 2020/fs_cli tcp 0 54 127.0.0.1:8021 127.0.0.1:34893 ESTABLISHED 1862/freeswitch -- regards, akhil -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141210/290d96c4/attachment.html From avi at avimarcus.net Wed Dec 10 14:13:02 2014 From: avi at avimarcus.net (Avi Marcus) Date: Wed, 10 Dec 2014 11:13:02 +0000 Subject: [Freeswitch-users] Newbie -- Help Needed Transferring Inbound Caller ID to external SIP Gateway URI In-Reply-To: References: <0000014a0b2a7f3a-e057e705-e811-4903-b2b9-bbd020abb4a7-000000@email.amazonses.com> <0000014a22d99893-8fb2f862-1a14-4cb2-b363-8e924d56012d-000000@email.amazonses.com> <0000014a2a0f8f9d-1038c04d-e533-483b-8322-bd27ace4ba19-000000@email.amazonses.com> <0000014a32c00982-5991e950-58e0-477c-bcab-ead784e5d4d6-000000@email.amazonses.com> Message-ID: <0000014a33e6f7d3-05962ccb-d0e7-424d-aafc-a4dcd666d72f-000000@email.amazonses.com> Your log still shows: Action set(exec_after_bridge_arg=1213) ... and if they set a 180, then you can set the ringback. -Avi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141210/78fffa22/attachment.html From italorossib at gmail.com Wed Dec 10 14:47:30 2014 From: italorossib at gmail.com (=?UTF-8?B?w410YWxvIFJvc3Np?=) Date: Wed, 10 Dec 2014 08:47:30 -0300 Subject: [Freeswitch-users] Doc-Sprint Friday 12 December 2014 In-Reply-To: References: Message-ID: Confirmed so far: - Italo Rossi (+4) - Areski Belaid - Iwada Eja - Adolphe Cher-Aime - Bote Who else? On Fri, Dec 5, 2014 at 12:14 PM, Adolphe Cher-Aime wrote: > I'm in. > > > -- > Adolphe > > On Thu, Dec 4, 2014 at 3:00 PM, Iwada Eja wrote: > >> I'm In >> >> On Thu, Dec 4, 2014 at 1:20 PM, Areski wrote: >> >>> Hi everyone, >>> >>> We are planning to organize an other doc sprint on *Friday 12 December >>> at 10am CT*. >>> It will be 4 hours long but you can join for less time. >>> >>> The Doc-sprint will focus on migrating the remaining pages from >>> MediaWiki (https://wiki.freeswitch.org) to Confluence Wiki ( >>> https://freeswitch.org/confluence). >>> >>> We will use an FS IRC channel during the sprint: *#freeswitch-docs* >>> and will track our work on the spreadsheet: >>> https://docs.google.com/spreadsheets/d/1qsG-kRymvKlNBapnBLw86W130VdbnK6naYapbR_UNds/edit?pli=1#gid=1187898333 >>> >>> During the sprint, please change the URL's "Status" you are working on >>> to "Editing" with your name next to it so we don't duplicate work. >>> >>> Some extra information: >>> - https://freeswitch.org/confluence/display/FREESWITCH/Wiki+Migration >>> - >>> https://freeswitch.org/confluence/display/FREESWITCH/Contributing+Documentation >>> >>> We hope to get a maximum number of people signed up! >>> >>> Peoples confirmed so far: >>> - Italo Rossi (+4) >>> - Areski Belaid >>> >>> >>> So, who is in? >>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 >> Iwada >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- ?talo Rossi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141210/53111ce8/attachment.html From liam at maxumdata.com Wed Dec 10 12:20:29 2014 From: liam at maxumdata.com (Liam Farr) Date: Wed, 10 Dec 2014 22:20:29 +1300 Subject: [Freeswitch-users] Mod perl using $session->setVariable inline Message-ID: Hi, I need to set a variable using a perl script inline. Normally I would use $session->setVariable('some_var',$my_value); However I cant find any options to execute this inline / as an inline action. (I need to evaluate the variable later in the hunting phase of the dialplan hence need it to be an inline action). Is there a way to do this? Or is there another way to achieve the same result, e.g. hacking $session->execute("set","some_var=$my_value") to run as an inline action? Any help would be much appreciated, I've scoured the wiki, confluence and git repo and come up empty. -- Kind Regards Liam Farr -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141210/085aa3b5/attachment-0001.html From notify.sina at gmail.com Wed Dec 10 13:05:20 2014 From: notify.sina at gmail.com (Sina Owolabi) Date: Wed, 10 Dec 2014 11:05:20 +0100 Subject: [Freeswitch-users] Newbie -- Help Needed Transferring Inbound Caller ID to external SIP Gateway URI In-Reply-To: <0000014a32c00982-5991e950-58e0-477c-bcab-ead784e5d4d6-000000@email.amazonses.com> References: <0000014a0b2a7f3a-e057e705-e811-4903-b2b9-bbd020abb4a7-000000@email.amazonses.com> <0000014a22d99893-8fb2f862-1a14-4cb2-b363-8e924d56012d-000000@email.amazonses.com> <0000014a2a0f8f9d-1038c04d-e533-483b-8322-bd27ace4ba19-000000@email.amazonses.com> <0000014a32c00982-5991e950-58e0-477c-bcab-ead784e5d4d6-000000@email.amazonses.com> Message-ID: It doesn't seem to, though I do hear some kind of a tiny pause between when the othersipgw is dialled. I hope being NATed is not a problem! They are now sending a "180 Ringing" before they send a "480 unavailable" in their reply and a single ring: Here's a tcp dump of a call: 11:00:33.241307 IP (tos 0x0, ttl 64, id 34094, offset 0, flags [none], proto UDP (17), length 1146) 10.22.0.252.5080 > othersipgw.ip.address.5060: SIP, length: 1118 INVITE sip:user.name at sip.othersipgw.com;+234diallednumber SIP/2.0 Via: SIP/2.0/UDP my.public.addr.ess:5080;rport;branch=z9hG4bKZBN9c87mpSHDa Max-Forwards: 5 From: "0diallednumber" ;tag=eF54Q5UjvXQZB To: Call-ID: 38c0157b-faf6-1232-f39d-525400ecad09 CSeq: 68766752 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20141120T035109Z~79de78a0fb~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 247 X-FS-Support: update_display,send_info Remote-Party-ID: "0diallednumber" ;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1418182171 1418182172 IN IP4 my.public.addr.ess s=FreeSWITCH c=IN IP4 my.public.addr.ess t=0 0 m=audio 23462 RTP/AVP 8 0 101 13 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 11:00:33.497529 IP (tos 0x0, ttl 53, id 5741, offset 0, flags [none], proto UDP (17), length 1038) othersipgw.ip.address.5060 > 10.22.0.252.5080: SIP, length: 1010 SIP/2.0 180 Ringing Via: SIP/2.0/UDP my.public.addr.ess:5080;rport=5080;branch=z9hG4bKZBN9c87mpSHDa From: "0diallednumber" ;tag=eF54Q5UjvXQZB To: ;tag=tt76SS77SZvjr Call-ID: 38c0157b-faf6-1232-f39d-525400ecad09 CSeq: 68766752 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.2.14 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 X-FS-Display-Name: user.name X-FS-Display-Number: sip:user.name at sip.othersipgw.com X-FS-Support: update_display,send_info Remote-Party-ID: "user.name" ;party=calling;privacy=off;screen=no 11:00:36.552573 IP (tos 0x0, ttl 53, id 5742, offset 0, flags [none], proto UDP (17), length 990) othersipgw.ip.address.5060 > 10.22.0.252.5080: SIP, length: 962 SIP/2.0 480 Temporarily Unavailable Via: SIP/2.0/UDP my.public.addr.ess:5080;rport=5080;branch=z9hG4bKZBN9c87mpSHDa Max-Forwards: 4 From: "0diallednumber" ;tag=eF54Q5UjvXQZB To: ;tag=tt76SS77SZvjr Call-ID: 38c0157b-faf6-1232-f39d-525400ecad09 CSeq: 68766752 INVITE User-Agent: FreeSWITCH-mod_sofia/1.2.14 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=16;text="NORMAL_CLEARING" Content-Length: 0 X-FS-Display-Name: user.name X-FS-Display-Number: sip:user.name at sip.othersipgw.com Remote-Party-ID: "user.name" ;party=calling;privacy=off;screen=no 11:00:36.552866 IP (tos 0x0, ttl 64, id 34095, offset 0, flags [none], proto UDP (17), length 412) 10.22.0.252.5080 > othersipgw.ip.address.5060: SIP, length: 384 ACK sip:user.name at sip.othersipgw.com;+234diallednumber SIP/2.0 Via: SIP/2.0/UDP my.public.addr.ess:5080;rport;branch=z9hG4bKZBN9c87mpSHDa Max-Forwards: 5 From: "0diallednumber" ;tag=eF54Q5UjvXQZB To: ;tag=tt76SS77SZvjr Call-ID: 38c0157b-faf6-1232-f39d-525400ecad09 CSeq: 68766752 ACK Content-Length: 0 Here's a log of a call after deleting and using your suggestions: 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [NOTICE] switch_channel.c:1055 New Channel sofia/external/0diallednumber at first.sipgw.net [1517b746-8051-11e4-909f-2d01d6dac59f] 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_session.c:1053 Send signal sofia/external/0diallednumber at first.sipgw.net [BREAK] 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_session.c:1053 Send signal sofia/external/0diallednumber at first.sipgw.net [BREAK] 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_state_machine.c:472 (sofia/external/0diallednumber at first.sipgw.net) Running State Change CS_NEW 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] sofia.c:8834 sofia/external/0diallednumber at first.sipgw.net receiving invite from 62.173.32.89:5060 version: 1.5.15b git 79de78a 2014-11-20 03:51:09Z 64bit 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] sofia.c:6614 Channel sofia/external/0diallednumber at first.sipgw.net entering state [received][100] 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] sofia.c:6624 Remote SDP: 1517b746-8051-11e4-909f-2d01d6dac59f v=0 1517b746-8051-11e4-909f-2d01d6dac59f o=CGPLeg006300 1254596129 627298065 IN IP4 first.sipgw.public.addresss 1517b746-8051-11e4-909f-2d01d6dac59f s=Cisco SDP 0 1517b746-8051-11e4-909f-2d01d6dac59f c=IN IP4 first.sipgw.public.addresss 1517b746-8051-11e4-909f-2d01d6dac59f t=0 0 1517b746-8051-11e4-909f-2d01d6dac59f o=CGPLeg006300 1254596129 627298065 IN IP4 first.sipgw.public.addresss 1517b746-8051-11e4-909f-2d01d6dac59f s=Cisco SDP 0 1517b746-8051-11e4-909f-2d01d6dac59f c=IN IP4 first.sipgw.public.addresss 1517b746-8051-11e4-909f-2d01d6dac59f t=0 0 1517b746-8051-11e4-909f-2d01d6dac59f a=mediagateway:first.sipgw.net:2852669:10.51.2.3 1517b746-8051-11e4-909f-2d01d6dac59f m=audio 60008 RTP/AVP 18 8 0 101 100 1517b746-8051-11e4-909f-2d01d6dac59f c=IN IP4 first.sipgw.public.addresss 1517b746-8051-11e4-909f-2d01d6dac59f a=rtpmap:18 G729/8000 1517b746-8051-11e4-909f-2d01d6dac59f m=audio 60008 RTP/AVP 18 8 0 101 100 1517b746-8051-11e4-909f-2d01d6dac59f c=IN IP4 first.sipgw.public.addresss 1517b746-8051-11e4-909f-2d01d6dac59f a=rtpmap:18 G729/8000 1517b746-8051-11e4-909f-2d01d6dac59f a=rtpmap:8 PCMA/8000 1517b746-8051-11e4-909f-2d01d6dac59f a=rtpmap:0 PCMU/8000 1517b746-8051-11e4-909f-2d01d6dac59f a=rtpmap:101 telephone-event/8000 1517b746-8051-11e4-909f-2d01d6dac59f a=fmtp:101 0-15 1517b746-8051-11e4-909f-2d01d6dac59f a=rtpmap:0 PCMU/8000 1517b746-8051-11e4-909f-2d01d6dac59f a=rtpmap:101 telephone-event/8000 1517b746-8051-11e4-909f-2d01d6dac59f a=fmtp:101 0-15 1517b746-8051-11e4-909f-2d01d6dac59f a=rtpmap:100 X-NSE/8000 1517b746-8051-11e4-909f-2d01d6dac59f a=fmtp:100 200-202 1517b746-8051-11e4-909f-2d01d6dac59f a=rtcpping:T:181652:18165278 1517b746-8051-11e4-909f-2d01d6dac59f a=fmtp:100 200-202 1517b746-8051-11e4-909f-2d01d6dac59f a=rtcpping:T:181652:18165278 1517b746-8051-11e4-909f-2d01d6dac59f a=X-cap: 1 audio RTP/AVP 100 1517b746-8051-11e4-909f-2d01d6dac59f a=X-cap: 2 image udptl t38 1517b746-8051-11e4-909f-2d01d6dac59f a=X-cpar: a=rtpmap:100 X-NSE/8000 1517b746-8051-11e4-909f-2d01d6dac59f a=X-cap: 2 image udptl t38 1517b746-8051-11e4-909f-2d01d6dac59f a=X-cpar: a=rtpmap:100 X-NSE/8000 1517b746-8051-11e4-909f-2d01d6dac59f a=X-cpar: a=fmtp:100 200-202 1517b746-8051-11e4-909f-2d01d6dac59f a=X-sqn:0 1517b746-8051-11e4-909f-2d01d6dac59f 1517b746-8051-11e4-909f-2d01d6dac59f a=X-sqn:0 1517b746-8051-11e4-909f-2d01d6dac59f 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] sofia.c:6890 (sofia/external/0diallednumber at first.sipgw.net) State Change CS_NEW -> CS_INIT 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] sofia.c:6890 (sofia/external/0diallednumber at first.sipgw.net) State Change CS_NEW -> CS_INIT 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_session.c:1388 Send signal sofia/external/0diallednumber at first.sipgw.net [BREAK] 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_state_machine.c:491 (sofia/external/0diallednumber at first.sipgw.net) State NEW 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_state_machine.c:472 (sofia/external/0diallednumber at first.sipgw.net) Running State Change CS_INIT 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_state_machine.c:512 (sofia/external/0diallednumber at first.sipgw.net) State INIT 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] mod_sofia.c:87 sofia/external/0diallednumber at first.sipgw.net SOFIA INIT 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_state_machine.c:40 sofia/external/0diallednumber at first.sipgw.net Standard INIT 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_state_machine.c:40 sofia/external/0diallednumber at first.sipgw.net Standard INIT 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_state_machine.c:48 (sofia/external/0diallednumber at first.sipgw.net) State Change CS_INIT -> CS_ROUTING 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_sessio1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_session.c:1388 Send signal sofia/external/0diallednumber at first.sipgw.net [BREAK] n.c:1388 Send signal sofia/external/0diallednumber at first.sipgw.net [BREAK] 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_state_machine.c:512 (sofia/external/0diallednumber at first.sipgw.net) State INIT going to sleep 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_state_machine.c:512 (sofia/external/0diallednumber at first.sipgw.net) State INIT going to sleep 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_state_machine.c:472 (sofia/external/0diallednumber at first.sipgw.net) Running State Change CS_ROUTING 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_channel.c:2184 (sofia/external/0diallednumber at first.sipgw.net) Callstate Change DOWN -> RINGING 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_state_machine.c:528 (sofia/external/0diallednumber at first.sipgw.net) State ROUTING 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] mod_sofia.c:123 sofia/external/0diallednumber at first.sipgw.net SOFIA ROUTING 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_state_machine.c:166 sofia/external/0diallednumber at first.sipgw.net Standard ROUTING 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [INFO] mod_dialplan_xml.c:635 Processing 0diallednumber <0diallednumber>->016311084 in context public 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [public->unloop] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [public->unloop] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [public->outside_call] continue=true 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [public->outside_call] continue=true 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Absolute Condition [outside_call] 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Action set(outside_call=true) 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Action export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [public->call_debug] continue=true 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [public->public_extensions] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [public->public_extensions] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [public_extensions] destination_number(016311084) =~ /^(10[01][0-9])$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [public_extensions] destination_number(016311084) =~ /^(10[01][0-9])$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [public->public_did] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [public_did] destination_number(016311084) =~ /^(5551212)$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [public->ipnx-inbound] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (PASS) [ipnx-inbound] destination_number(016311084) =~ /^0163110(\d{2})$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (PASS) [ipnx-inbound] caller_id_number(0diallednumber) =~ /^0(\d{10})$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Action set(effective_caller_id_number=+234diallednumber) 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Action transfer(1212 XML default) 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_state_machine.c:216 (sofia/external/0diallednumber at first.sipgw.net) State Change CS_ROUTING -> CS_EXECUTE 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_session.c:1388 Send signal sofia/external/0diallednumber at first.sipgw.net [BREAK] 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_state_machine.c:528 (sofia/external/0diallednumber at first.sipgw.net) State ROUTING going to sleep 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_state_machine.c:472 (sofia/external/0diallednumber at first.sipgw.net) Running State Change CS_EXECUTE 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_state_machine.c:535 (sofia/external/0diallednumber at first.sipgw.net) State EXECUTE 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] mod_sofia.c:178 sofia/external/0diallednumber at first.sipgw.net SOFIA EXECUTE 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_state_machine.c:258 sofia/external/0diallednumber at first.sipgw.net Standard EXECUTE 1517b746-8051-11e4-909f-2d01d6dac59f EXECUTE sofia/external/0diallednumber at first.sipgw.net set(outside_call=true) 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] mod_dptools.c:1435 sofia/external/0diallednumber at first.sipgw.net SET [outside_call]=[true] 1517b746-8051-11e4-909f-2d01d6dac59f EXECUTE sofia/external/0diallednumber at first.sipgw.net export(RFC2822_DATE=Wed, 10 Dec 2014 10:44:06 +0100) 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_channel.c:1247 EXPORT (export_vars) [RFC2822_DATE]=[Wed, 10 Dec 2014 10:44:06 +0100] 1517b746-8051-11e4-909f-2d01d6dac59f EXECUTE sofia/external/0diallednumber at first.sipgw.net set(effective_caller_id_number=+234diallednumber) 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] mod_dptools.c:1435 sofia/external/0diallednumber at first.sipgw.net SET [effective_caller_id_number]=[+234diallednumber] 1517b746-8051-11e4-909f-2d01d6dac59f EXECUTE sofia/external/0diallednumber at first.sipgw.net transfer(1212 XML default) 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_ivr.c:1847 (sofia/external/0diallednumber at first.sipgw.net) State Change CS_EXECUTE -> CS_ROUTING 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_session.c:1388 Send signal sofia/external/0diallednumber at first.sipgw.net [BREAK] 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_session.c:908 Send signal sofia/external/0diallednumber at first.sipgw.net [BREAK] 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [NOTICE] switch_ivr.c:1854 Transfer sofia/external/0diallednumber at first.sipgw.net to XML[1212 at default] 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_state_machine.c:535 (sofia/external/0diallednumber at first.sipgw.net) State EXECUTE going to sleep 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_state_machine.c:472 (sofia/external/0diallednumber at first.sipgw.net) Running State Change CS_ROUTING 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_state_machine.c:528 (sofia/external/0diallednumber at first.sipgw.net) State ROUTING 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] mod_sofia.c:123 sofia/external/0diallednumber at first.sipgw.net SOFIA ROUTING 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_state_machine.c:166 sofia/external/0diallednumber at first.sipgw.net Standard ROUTING 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [INFO] mod_dialplan_xml.c:635 Processing 0diallednumber <0diallednumber>->1212 in context default 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->unloop] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->tod_example] continue=true 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Date/Time Match (PASS) [tod_example] break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Action set(open=true) 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->holiday_example] continue=true 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Date/TimeMatch (FAIL) [holiday_example] break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->global-intercept] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [global-intercept] destination_number(1212) =~ /^886$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->group-intercept] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [group-intercept] destination_number(1212) =~ /^\*8$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->intercept-ext] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [intercept-ext] destination_number(1212) =~ /^\*\*(\d+)$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->redial] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [redial] destination_number(1212) =~ /^(redial|870)$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->global] continue=true 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [global] ${default_password}(N0w4h4l4) =~ /^1234$/ break=never 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [global] ${rtp_has_crypto}() =~ /^(AEAD_AES_256_GCM_8|AEAD_AES_128_GCM_8|AES_CM_256_HMAC_SHA1_80|AES_CM_192_HMAC_SHA1_80|AES_CM_128_HMAC_SHA1_80|AES_CM_256_HMAC_SHA1_32|AES_CM_192_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_32|AES_CM_128_NULL_AUTH)$/ break=never 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (PASS) [global] ${endpoint_disposition}(DELAYED NEGOTIATION) =~ /^(DELAYED NEGOTIATION)/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [global] ${switch_r_sdp}(v=0 1517b746-8051-11e4-909f-2d01d6dac59f o=CGPLeg006300 1254596129 627298065 IN IP4 first.sipgw.public.addresss 1517b746-8051-11e4-909f-2d01d6dac59f s=Cisco SDP 0 1517b746-8051-11e4-909f-2d01d6dac59f c=IN IP4 first.sipgw.public.addresss 1517b746-8051-11e4-909f-2d01d6dac59f t=0 0 1517b746-8051-11e4-909f-2d01d6dac59f a=mediagateway:first.sipgw.net:2852669:10.51.2.3 1517b746-8051-11e4-909f-2d01d6dac59f m=audio 60008 RTP/AVP 18 8 0 101 100 1517b746-8051-11e4-909f-2d01d6dac59f c=IN IP4 first.sipgw.public.addresss 1517b746-8051-11e4-909f-2d01d6dac59f a=rtpmap:18 G729/8000 1517b746-8051-11e4-909f-2d01d6dac59f a=rtpmap:8 PCMA/8000 1517b746-8051-11e4-909f-2d01d6dac59f a=rtpmap:0 PCMU/8000 1517b746-8051-11e4-909f-2d01d6dac59f a=rtpmap:101 telephone-event/8000 1517b746-8051-11e4-909f-2d01d6dac59f a=fmtp:101 0-15 1517b746-8051-11e4-909f-2d01d6dac59f a=rtpmap:100 X-NSE/8000 1517b746-8051-11e4-909f-2d01d6dac59f a=fmtp:100 200-202 1517b746-8051-11e4-909f-2d01d6dac59f a=rtcpping:T:181652:18165278 1517b746-8051-11e4-909f-2d01d6dac59f a=X-cap: 1 audio RTP/AVP 100 1517b746-8051-11e4-909f-2d01d6dac59f a=X-cap: 2 image udptl t38 1517b746-8051-11e4-909f-2d01d6dac59f a=X-cpar: a=rtpmap:100 X-NSE/8000 1517b746-8051-11e4-909f-2d01d6dac59f a=X-cpar: a=fmtp:100 200-202 1517b746-8051-11e4-909f-2d01d6dac59f a=X-sqn:0 1517b746-8051-11e4-909f-2d01d6dac59f ) =~ /(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)/ break=never 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Absolute Condition [global] 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Action hash(insert/${domain_name}-last_dial/global/${uuid}) 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Action export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->snom-demo-2] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [snom-demo-2] destination_number(1212) =~ /^9001$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->snom-demo-1] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [snom-demo-1] destination_number(1212) =~ /^9000$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->eavesdrop] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [eavesdrop] destination_number(1212) =~ /^88(\d{4})$|^\*0(.*)$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->eavesdrop] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [eavesdrop] destination_number(1212) =~ /^779$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->call_return] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [call_return] destination_number(1212) =~ /^\*69$|^869$|^lcr$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->del-group] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [del-group] destination_number(1212) =~ /^80(\d{2})$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->add-group] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [add-group] destination_number(1212) =~ /^81(\d{2})$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->call-group-simo] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [call-group-simo] destination_number(1212) =~ /^82(\d{2})$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->call-group-order] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [call-group-order] destination_number(1212) =~ /^83(\d{2})$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->extension-intercom] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [extension-intercom] destination_number(1212) =~ /^8(10[01][0-9])$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->Local_Extension] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [Local_Extension] destination_number(1212) =~ /^(10[01][0-9])$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->Local_Extension_Skinny] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [Local_Extension_Skinny] destination_number(1212) =~ /^(11[01][0-9])$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->group_dial_sales] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [group_dial_sales] destination_number(1212) =~ /^2000$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->group_dial_support] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [group_dial_support] destination_number(1212) =~ /^2001$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->group_dial_billing] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [group_dial_billing] destination_number(1212) =~ /^2002$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->operator] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [operator] destination_number(1212) =~ /^(operator|0)$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->vmain] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [vmain] destination_number(1212) =~ /^vmain$|^4000$|^\*98$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->sip_uri] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [sip_uri] destination_number(1212) =~ /^sip:(.*)$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->nb_conferences] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [nb_conferences] destination_number(1212) =~ /^(30\d{2})$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->wb_conferences] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [wb_conferences] destination_number(1212) =~ /^(31\d{2})$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->uwb_conferences] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [uwb_conferences] destination_number(1212) =~ /^(32\d{2})$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->cdquality_conferences] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [cdquality_conferences] destination_number(1212) =~ /^(33\d{2})$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->freeswitch_public_conf_via_sip] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [freeswitch_public_conf_via_sip] destination_number(1212) =~ /^9(888|8888|1616|3232)$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->mad_boss_intercom] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [mad_boss_intercom] destination_number(1212) =~ /^0911$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->mad_boss_intercom] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [mad_boss_intercom] destination_number(1212) =~ /^0912$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->mad_boss] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [mad_boss] destination_number(1212) =~ /^0913$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->ivr_demo] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [ivr_demo] destination_number(1212) =~ /^5000$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->dynamic_conference] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [dynamic_conference] destination_number(1212) =~ /^5001$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->rtp_multicast_page] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [rtp_multicast_page] destination_number(1212) =~ /^pagegroup$|^7243$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->park] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [park] destination_number(1212) =~ /^5900$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->unpark] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [unpark] destination_number(1212) =~ /^5901$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->valet_park] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [valet_park] destination_number(1212) =~ /^(6000)$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->valet_park] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [valet_park] destination_number(1212) =~ /^((?!6000)60\d{2})$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->park] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (PASS) [park] source(mod_sofia) =~ /mod_sofia/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [park] destination_number(1212) =~ /park\+(\d+)/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->unpark] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (PASS) [unpark] source(mod_sofia) =~ /mod_sofia/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [unpark] destination_number(1212) =~ /^parking$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->park] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (PASS) [park] source(mod_sofia) =~ /mod_sofia/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [park] destination_number(1212) =~ /callpark/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->unpark] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (PASS) [unpark] source(mod_sofia) =~ /mod_sofia/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [unpark] destination_number(1212) =~ /pickup/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->wait] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [wait] destination_number(1212) =~ /^wait$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->fax_receive] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [fax_receive] destination_number(1212) =~ /^9178$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->fax_transmit] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [fax_transmit] destination_number(1212) =~ /^9179$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->ringback_180] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [ringback_180] destination_number(1212) =~ /^9180$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->ringback_183_uk_ring] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [ringback_183_uk_ring] destination_number(1212) =~ /^9181$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->ringback_183_music_ring] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [ringback_183_music_ring] destination_number(1212) =~ /^9182$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->ringback_post_answer_uk_ring] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [ringback_post_answer_uk_ring] destination_number(1212) =~ /^9183$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->ringback_post_answer_music] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [ringback_post_answer_music] destination_number(1212) =~ /^9184$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->ClueCon] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [ClueCon] destination_number(1212) =~ /^9191$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->show_info] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [show_info] destination_number(1212) =~ /^9192$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [public_did] destination_number(016311084) =~ /^(5551212)$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [public->ipnx-inbound] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (PASS) [ipnx-inbound] destination_number(016311084) =~ /^0163110(\d{2})$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (PASS) [ipnx-inbound] caller_id_number(0diallednumber) =~ /^0(\d{10})$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Action set(effective_caller_id_number=+234diallednumber) 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Action transfer(1212 XML default) 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_state_machine.c:216 (sofia/external/0diallednumber at first.sipgw.net) State Change CS_ROUTING -> CS_EXECUTE 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_session.c:1388 Send signal sofia/external/0diallednumber at first.sipgw.net [BREAK] 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_state_machine.c:528 (sofia/external/0diallednumber at first.sipgw.net) State ROUTING going to sleep 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_state_machine.c:472 (sofia/external/0diallednumber at first.sipgw.net) Running State Change CS_EXECUTE 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_state_machine.c:535 (sofia/external/0diallednumber at first.sipgw.net) State EXECUTE 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] mod_sofia.c:178 sofia/external/0diallednumber at first.sipgw.net SOFIA EXECUTE 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_state_machine.c:258 sofia/external/0diallednumber at first.sipgw.net Standard EXECUTE 1517b746-8051-11e4-909f-2d01d6dac59f EXECUTE sofia/external/0diallednumber at first.sipgw.net set(outside_call=true) 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] mod_dptools.c:1435 sofia/external/0diallednumber at first.sipgw.net SET [outside_call]=[true] 1517b746-8051-11e4-909f-2d01d6dac59f EXECUTE sofia/external/0diallednumber at first.sipgw.net export(RFC2822_DATE=Wed, 10 Dec 2014 10:44:06 +0100) 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_channel.c:1247 EXPORT (export_vars) [RFC2822_DATE]=[Wed, 10 Dec 2014 10:44:06 +0100] 1517b746-8051-11e4-909f-2d01d6dac59f EXECUTE sofia/external/0diallednumber at first.sipgw.net set(effective_caller_id_number=+234diallednumber) 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] mod_dptools.c:1435 sofia/external/0diallednumber at first.sipgw.net SET [effective_caller_id_number]=[+234diallednumber] 1517b746-8051-11e4-909f-2d01d6dac59f EXECUTE sofia/external/0diallednumber at first.sipgw.net transfer(1212 XML default) 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_ivr.c:1847 (sofia/external/0diallednumber at first.sipgw.net) State Change CS_EXECUTE -> CS_ROUTING 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_session.c:1388 Send signal sofia/external/0diallednumber at first.sipgw.net [BREAK] 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_session.c:908 Send signal sofia/external/0diallednumber at first.sipgw.net [BREAK] 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [NOTICE] switch_ivr.c:1854 Transfer sofia/external/0diallednumber at first.sipgw.net to XML[1212 at default] 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_state_machine.c:535 (sofia/external/0diallednumber at first.sipgw.net) State EXECUTE going to sleep 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_state_machine.c:472 (sofia/external/0diallednumber at first.sipgw.net) Running State Change CS_ROUTING 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_state_machine.c:528 (sofia/external/0diallednumber at first.sipgw.net) State ROUTING 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] mod_sofia.c:123 sofia/external/0diallednumber at first.sipgw.net SOFIA ROUTING 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_state_machine.c:166 sofia/external/0diallednumber at first.sipgw.net Standard ROUTING 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [INFO] mod_dialplan_xml.c:635 Processing 0diallednumber <0diallednumber>->1212 in context default 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->unloop] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->tod_example] continue=true 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Date/Time Match (PASS) [tod_example] break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Action set(open=true) 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->holiday_example] continue=true 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Date/TimeMatch (FAIL) [holiday_example] break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->global-intercept] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [global-intercept] destination_number(1212) =~ /^886$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->group-intercept] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [group-intercept] destination_number(1212) =~ /^\*8$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->intercept-ext] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [intercept-ext] destination_number(1212) =~ /^\*\*(\d+)$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->redial] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [redial] destination_number(1212) =~ /^(redial|870)$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->global] continue=true 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [global] ${default_password}(N0w4h4l4) =~ /^1234$/ break=never 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [global] ${rtp_has_crypto}() =~ /^(AEAD_AES_256_GCM_8|AEAD_AES_128_GCM_8|AES_CM_256_HMAC_SHA1_80|AES_CM_192_HMAC_SHA1_80|AES_CM_128_HMAC_SHA1_80|AES_CM_256_HMAC_SHA1_32|AES_CM_192_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_32|AES_CM_128_NULL_AUTH)$/ break=never 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (PASS) [global] ${endpoint_disposition}(DELAYED NEGOTIATION) =~ /^(DELAYED NEGOTIATION)/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [global] ${switch_r_sdp}(v=0 1517b746-8051-11e4-909f-2d01d6dac59f o=CGPLeg006300 1254596129 627298065 IN IP4 first.sipgw.public.addresss 1517b746-8051-11e4-909f-2d01d6dac59f s=Cisco SDP 0 1517b746-8051-11e4-909f-2d01d6dac59f c=IN IP4 first.sipgw.public.addresss 1517b746-8051-11e4-909f-2d01d6dac59f t=0 0 1517b746-8051-11e4-909f-2d01d6dac59f a=mediagateway:first.sipgw.net:2852669:10.51.2.3 1517b746-8051-11e4-909f-2d01d6dac59f m=audio 60008 RTP/AVP 18 8 0 101 100 1517b746-8051-11e4-909f-2d01d6dac59f c=IN IP4 first.sipgw.public.addresss 1517b746-8051-11e4-909f-2d01d6dac59f a=rtpmap:18 G729/8000 1517b746-8051-11e4-909f-2d01d6dac59f a=rtpmap:8 PCMA/8000 1517b746-8051-11e4-909f-2d01d6dac59f a=rtpmap:0 PCMU/8000 1517b746-8051-11e4-909f-2d01d6dac59f a=rtpmap:101 telephone-event/8000 1517b746-8051-11e4-909f-2d01d6dac59f a=fmtp:101 0-15 1517b746-8051-11e4-909f-2d01d6dac59f a=rtpmap:100 X-NSE/8000 1517b746-8051-11e4-909f-2d01d6dac59f a=fmtp:100 200-202 1517b746-8051-11e4-909f-2d01d6dac59f a=rtcpping:T:181652:18165278 1517b746-8051-11e4-909f-2d01d6dac59f a=X-cap: 1 audio RTP/AVP 100 1517b746-8051-11e4-909f-2d01d6dac59f a=X-cap: 2 image udptl t38 1517b746-8051-11e4-909f-2d01d6dac59f a=X-cpar: a=rtpmap:100 X-NSE/8000 1517b746-8051-11e4-909f-2d01d6dac59f a=X-cpar: a=fmtp:100 200-202 1517b746-8051-11e4-909f-2d01d6dac59f a=X-sqn:0 1517b746-8051-11e4-909f-2d01d6dac59f ) =~ /(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)/ break=never 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Absolute Condition [global] 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Action hash(insert/${domain_name}-last_dial/global/${uuid}) 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Action export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->snom-demo-2] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [snom-demo-2] destination_number(1212) =~ /^9001$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->snom-demo-1] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [snom-demo-1] destination_number(1212) =~ /^9000$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->eavesdrop] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [eavesdrop] destination_number(1212) =~ /^88(\d{4})$|^\*0(.*)$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->eavesdrop] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [eavesdrop] destination_number(1212) =~ /^779$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->call_return] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [call_return] destination_number(1212) =~ /^\*69$|^869$|^lcr$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->del-group] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [del-group] destination_number(1212) =~ /^80(\d{2})$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->add-group] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [add-group] destination_number(1212) =~ /^81(\d{2})$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->call-group-simo] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [call-group-simo] destination_number(1212) =~ /^82(\d{2})$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->call-group-order] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [call-group-order] destination_number(1212) =~ /^83(\d{2})$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->extension-intercom] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [extension-intercom] destination_number(1212) =~ /^8(10[01][0-9])$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->Local_Extension] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [Local_Extension] destination_number(1212) =~ /^(10[01][0-9])$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->Local_Extension_Skinny] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [Local_Extension_Skinny] destination_number(1212) =~ /^(11[01][0-9])$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->group_dial_sales] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [group_dial_sales] destination_number(1212) =~ /^2000$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->group_dial_support] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [group_dial_support] destination_number(1212) =~ /^2001$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->group_dial_billing] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [group_dial_billing] destination_number(1212) =~ /^2002$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->operator] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [operator] destination_number(1212) =~ /^(operator|0)$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->vmain] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [vmain] destination_number(1212) =~ /^vmain$|^4000$|^\*98$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->sip_uri] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [sip_uri] destination_number(1212) =~ /^sip:(.*)$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->nb_conferences] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [nb_conferences] destination_number(1212) =~ /^(30\d{2})$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->wb_conferences] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [wb_conferences] destination_number(1212) =~ /^(31\d{2})$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->uwb_conferences] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [uwb_conferences] destination_number(1212) =~ /^(32\d{2})$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->cdquality_conferences] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [cdquality_conferences] destination_number(1212) =~ /^(33\d{2})$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->freeswitch_public_conf_via_sip] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [freeswitch_public_conf_via_sip] destination_number(1212) =~ /^9(888|8888|1616|3232)$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->mad_boss_intercom] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [mad_boss_intercom] destination_number(1212) =~ /^0911$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->mad_boss_intercom] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [mad_boss_intercom] destination_number(1212) =~ /^0912$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->mad_boss] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [mad_boss] destination_number(1212) =~ /^0913$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->ivr_demo] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [ivr_demo] destination_number(1212) =~ /^5000$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->dynamic_conference] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [dynamic_conference] destination_number(1212) =~ /^5001$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->rtp_multicast_page] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [rtp_multicast_page] destination_number(1212) =~ /^pagegroup$|^7243$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->park] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [park] destination_number(1212) =~ /^5900$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->unpark] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [unpark] destination_number(1212) =~ /^5901$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->valet_park] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [valet_park] destination_number(1212) =~ /^(6000)$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->valet_park] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [valet_park] destination_number(1212) =~ /^((?!6000)60\d{2})$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->park] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (PASS) [park] source(mod_sofia) =~ /mod_sofia/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [park] destination_number(1212) =~ /park\+(\d+)/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->unpark] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (PASS) [unpark] source(mod_sofia) =~ /mod_sofia/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [unpark] destination_number(1212) =~ /^parking$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->park] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (PASS) [park] source(mod_sofia) =~ /mod_sofia/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [park] destination_number(1212) =~ /callpark/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->unpark] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (PASS) [unpark] source(mod_sofia) =~ /mod_sofia/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [unpark] destination_number(1212) =~ /pickup/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->wait] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [wait] destination_number(1212) =~ /^wait$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->fax_receive] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [fax_receive] destination_number(1212) =~ /^9178$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->fax_transmit] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [fax_transmit] destination_number(1212) =~ /^9179$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->ringback_180] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [ringback_180] destination_number(1212) =~ /^9180$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->ringback_183_uk_ring] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [ringback_183_uk_ring] destination_number(1212) =~ /^9181$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->ringback_183_music_ring] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [ringback_183_music_ring] destination_number(1212) =~ /^9182$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->ringback_post_answer_uk_ring] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [ringback_post_answer_uk_ring] destination_number(1212) =~ /^9183$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->ringback_post_answer_music] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [ringback_post_answer_music] destination_number(1212) =~ /^9184$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->ClueCon] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [ClueCon] destination_number(1212) =~ /^9191$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->show_info] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [show_info] destination_number(1212) =~ /^9192$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->video_record] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [video_record] destination_number(1212) =~ /^9193$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->video_playback] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [video_playback] destination_number(1212) =~ /^9194$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->delay_echo] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [delay_echo] destination_number(1212) =~ /^9195$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->echo] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [echo] destination_number(1212) =~ /^9196$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->milliwatt] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [milliwatt] destination_number(1212) =~ /^9197$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->tone_stream] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [tone_stream] destination_number(1212) =~ /^9198$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->zrtp_enrollement] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [zrtp_enrollement] destination_number(1212) =~ /^9787$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->hold_music] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [hold_music] destination_number(1212) =~ /^9664$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->laugh break] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [laugh break] destination_number(1212) =~ /^9386$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->1212] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (PASS) [1212] destination_number(1212) =~ /1212/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Action set(hangup_after_bridge=false) 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Action bridge(sofia/external/sip:user.name at sip.othersipgw.com;${effective_caller_id_number}) 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Action set(exec_after_bridge_arg=1213) 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_state_machine.c:216 (sofia/external/0diallednumber at first.sipgw.net) State Change CS_ROUTING -> CS_EXECUTE 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_session.c:1388 Send signal sofia/external/0diallednumber at first.sipgw.net [BREAK] 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_state_machine.c:528 (sofia/external/0diallednumber at first.sipgw.net) State ROUTING going to sleep 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_state_machine.c:472 (sofia/external/0diallednumber at first.sipgw.net) Running State Change CS_EXECUTE 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_state_machine.c:535 (sofia/external/0diallednumber at first.sipgw.net) State EXECUTE 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] mod_sofia.c:178 sofia/external/0diallednumber at first.sipgw.net SOFIA EXECUTE 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_state_machine.c:258 sofia/external/0diallednumber at first.sipgw.net Standard EXECUTE 1517b746-8051-11e4-909f-2d01d6dac59f EXECUTE sofia/external/0diallednumber at first.sipgw.net set(open=true) 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] mod_dptools.c:1435 sofia/external/0diallednumber at first.sipgw.net SET [open]=[true] 1517b746-8051-11e4-909f-2d01d6dac59f EXECUTE sofia/external/0diallednumber at first.sipgw.net hash(insert/10.22.0.252-spymap/0diallednumber/1517b746-8051-11e4-909f-2d01d6dac59f) 1517b746-8051-11e4-909f-2d01d6dac59f EXECUTE sofia/external/0diallednumber at first.sipgw.net hash(insert/10.22.0.252-last_dial/0diallednumber/1212) 1517b746-8051-11e4-909f-2d01d6dac59f EXECUTE sofia/external/0diallednumber at first.sipgw.net hash(insert/10.22.0.252-last_dial/global/1517b746-8051-11e4-909f-2d01d6dac59f) 1517b746-8051-11e4-909f-2d01d6dac59f EXECUTE sofia/external/0diallednumber at first.sipgw.net export(RFC2822_DATE=Wed, 10 Dec 2014 10:44:06 +0100) 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_channel.c:1247 EXPORT (export_vars) [RFC2822_DATE]=[Wed, 10 Dec 2014 10:44:06 +0100] 1517b746-8051-11e4-909f-2d01d6dac59f EXECUTE sofia/external/0diallednumber at first.sipgw.net set(hangup_after_bridge=false) 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] mod_dptools.c:1435 sofia/external/0diallednumber at first.sipgw.net SET [hangup_after_bridge]=[false] 1517b746-8051-11e4-909f-2d01d6dac59f EXECUTE sofia/external/0diallednumber at first.sipgw.net bridge(sofia/external/sip:user.name at sip.othersipgw.com;+234diallednumber) 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_channel.c:1201 sofia/external/0diallednumber at first.sipgw.net EXPORTING[export_vars] [RFC2822_DATE]=[Wed, 10 Dec 2014 10:44:06 +0100] to event 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_channel.c:1201 sofia/external/0diallednumber at first.sipgw.net EXPORTING[export_vars] [RFC2822_DATE]=[Wed, 10 Dec 2014 10:44:06 +0100] to event 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_ivr_originate.c:2079 Parsing global variables 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285 [NOTICE] switch_channel.c:1055 New Channel sofia/external/sip:user.name at sip.othersipgw.com [15187302-8051-11e4-90b0-2d01d6dac59f] 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] mod_sofia.c:4615 (sofia/external/sip:user.name at sip.othersipgw.com) State Change CS_NEW -> CS_INIT 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_session.c:1388 Send signal sofia/external/sip:user.name at sip.othersipgw.com [BREAK] 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] mod_sofia.c:4685 [zrtp_passthru] Setting a-leg inherit_codec=true 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] mod_sofia.c:4688 [zrtp_passthru] Setting b-leg absolute_codec_string=PCMA at 8000h@20i at 64000b,PCMU at 8000h@20i at 64000b 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_state_machine.c:472 (sofia/external/sip:user.name at sip.othersipgw.com) Running State Change CS_INIT 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_state_machine.c:512 (sofia/external/sip:user.name at sip.othersipgw.com) State INIT 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] mod_sofia.c:87 sofia/external/sip:user.name at sip.othersipgw.com SOFIA INIT 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] sofia_glue.c:1232 sofia/external/sip:user.name at sip.othersipgw.com sending invite version: 1.5.15b git 79de78a 2014-11-20 03:51:09Z 64bit 15187302-8051-11e4-90b0-2d01d6dac59f Local SDP: 15187302-8051-11e4-90b0-2d01d6dac59f v=0 15187302-8051-11e4-90b0-2d01d6dac59f o=FreeSWITCH 1418187966 1418187967 IN IP4 my.public.addr.ess 15187302-8051-11e4-90b0-2d01d6dac59f s=FreeSWITCH 15187302-8051-11e4-90b0-2d01d6dac59f c=IN IP4 my.public.addr.ess 15187302-8051-11e4-90b0-2d01d6dac59f t=0 0 15187302-8051-11e4-90b0-2d01d6dac59f m=audio 16680 RTP/AVP 8 0 101 13 15187302-8051-11e4-90b0-2d01d6dac59f a=rtpmap:8 PCMA/8000 15187302-8051-11e4-90b0-2d01d6dac59f a=rtpmap:0 PCMU/8000 15187302-8051-11e4-90b0-2d01d6dac59f a=rtpmap:101 telephone-event/8000 15187302-8051-11e4-90b0-2d01d6dac59f a=fmtp:101 0-16 15187302-8051-11e4-90b0-2d01d6dac59f a=ptime:20 15187302-8051-11e4-90b0-2d01d6dac59f a=sendrecv 15187302-8051-11e4-90b0-2d01d6dac59f 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_state_machine.c:40 sofia/external/sip:user.name at sip.othersipgw.com Standard INIT 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_state_machine.c:48 (sofia/external/sip:user.name at sip.othersipgw.com) State Change CS_INIT -> CS_ROUTING 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_session.c:1388 Send signal sofia/external/sip:user.name at sip.othersipgw.com [BREAK] 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_state_machine.c:512 (sofia/external/sip:user.name at sip.othersipgw.com) State INIT going to sleep 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_state_machine.c:472 (sofia/external/sip:user.name at sip.othersipgw.com) Running State Change CS_ROUTING 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_state_machine.c:528 (sofia/external/sip:user.name at sip.othersipgw.com) State ROUTING 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] mod_sofia.c:123 sofia/external/sip:user.name at sip.othersipgw.com SOFIA ROUTING 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_ivr_originate.c:67 (sofia/external/sip:user.name at sip.othersipgw.com) State Change CS_ROUTING -> CS_CONSUME_MEDIA 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_session.c:1388 Send signal sofia/external/sip:user.name at sip.othersipgw.com [BREAK] 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_state_machine.c:528 (sofia/external/sip:user.name at sip.othersipgw.com) State ROUTING going to sleep 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_state_machine.c:472 (sofia/external/sip:user.name at sip.othersipgw.com) Running State Change CS_CONSUME_MEDIA 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_state_machine.c:547 (sofia/external/sip:user.name at sip.othersipgw.com) State CONSUME_MEDIA 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_state_machine.c:547 (sofia/external/sip:user.name at sip.othersipgw.com) State CONSUME_MEDIA going to sleep 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_session.c:1053 Send signal sofia/external/sip:user.name at sip.othersipgw.com [BREAK] 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] sofia.c:6614 Channel sofia/external/sip:user.name at sip.othersipgw.com entering state [calling][0] 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->video_record] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [video_record] destination_number(1212) =~ /^9193$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->video_playback] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [video_playback] destination_number(1212) =~ /^9194$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->delay_echo] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [delay_echo] destination_number(1212) =~ /^9195$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->echo] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [echo] destination_number(1212) =~ /^9196$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->milliwatt] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [milliwatt] destination_number(1212) =~ /^9197$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->tone_stream] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [tone_stream] destination_number(1212) =~ /^9198$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->zrtp_enrollement] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [zrtp_enrollement] destination_number(1212) =~ /^9787$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->hold_music] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [hold_music] destination_number(1212) =~ /^9664$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->laugh break] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [laugh break] destination_number(1212) =~ /^9386$/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net parsing [default->1212] continue=false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Regex (PASS) [1212] destination_number(1212) =~ /1212/ break=on-false 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Action set(hangup_after_bridge=false) 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Action bridge(sofia/external/sip:user.name at sip.othersipgw.com;${effective_caller_id_number}) 1517b746-8051-11e4-909f-2d01d6dac59f Dialplan: sofia/external/0diallednumber at first.sipgw.net Action set(exec_after_bridge_arg=1213) 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_state_machine.c:216 (sofia/external/0diallednumber at first.sipgw.net) State Change CS_ROUTING -> CS_EXECUTE 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_session.c:1388 Send signal sofia/external/0diallednumber at first.sipgw.net [BREAK] 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_state_machine.c:528 (sofia/external/0diallednumber at first.sipgw.net) State ROUTING going to sleep 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_state_machine.c:472 (sofia/external/0diallednumber at first.sipgw.net) Running State Change CS_EXECUTE 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_state_machine.c:535 (sofia/external/0diallednumber at first.sipgw.net) State EXECUTE 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] mod_sofia.c:178 sofia/external/0diallednumber at first.sipgw.net SOFIA EXECUTE 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_state_machine.c:258 sofia/external/0diallednumber at first.sipgw.net Standard EXECUTE 1517b746-8051-11e4-909f-2d01d6dac59f EXECUTE sofia/external/0diallednumber at first.sipgw.net set(open=true) 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] mod_dptools.c:1435 sofia/external/0diallednumber at first.sipgw.net SET [open]=[true] 1517b746-8051-11e4-909f-2d01d6dac59f EXECUTE sofia/external/0diallednumber at first.sipgw.net hash(insert/10.22.0.252-spymap/0diallednumber/1517b746-8051-11e4-909f-2d01d6dac59f) 1517b746-8051-11e4-909f-2d01d6dac59f EXECUTE sofia/external/0diallednumber at first.sipgw.net hash(insert/10.22.0.252-last_dial/0diallednumber/1212) 1517b746-8051-11e4-909f-2d01d6dac59f EXECUTE sofia/external/0diallednumber at first.sipgw.net hash(insert/10.22.0.252-last_dial/global/1517b746-8051-11e4-909f-2d01d6dac59f) 1517b746-8051-11e4-909f-2d01d6dac59f EXECUTE sofia/external/0diallednumber at first.sipgw.net export(RFC2822_DATE=Wed, 10 Dec 2014 10:44:06 +0100) 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_channel.c:1247 EXPORT (export_vars) [RFC2822_DATE]=[Wed, 10 Dec 2014 10:44:06 +0100] 1517b746-8051-11e4-909f-2d01d6dac59f EXECUTE sofia/external/0diallednumber at first.sipgw.net set(hangup_after_bridge=false) 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] mod_dptools.c:1435 sofia/external/0diallednumber at first.sipgw.net SET [hangup_after_bridge]=[false] 1517b746-8051-11e4-909f-2d01d6dac59f EXECUTE sofia/external/0diallednumber at first.sipgw.net bridge(sofia/external/sip:user.name at sip.othersipgw.com;+234diallednumber) 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_channel.c:1201 sofia/external/0diallednumber at first.sipgw.net EXPORTING[export_vars] [RFC2822_DATE]=[Wed, 10 Dec 2014 10:44:06 +0100] to event 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_channel.c:1201 sofia/external/0diallednumber at first.sipgw.net EXPORTING[export_vars] [RFC2822_DATE]=[Wed, 10 Dec 2014 10:44:06 +0100] to event 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_ivr_originate.c:2079 Parsing global variables 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285 [NOTICE] switch_channel.c:1055 New Channel sofia/external/sip:user.name at sip.othersipgw.com [15187302-8051-11e4-90b0-2d01d6dac59f] 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] mod_sofia.c:4615 (sofia/external/sip:user.name at sip.othersipgw.com) State Change CS_NEW -> CS_INIT 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_session.c:1388 Send signal sofia/external/sip:user.name at sip.othersipgw.com [BREAK] 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] mod_sofia.c:4685 [zrtp_passthru] Setting a-leg inherit_codec=true 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] mod_sofia.c:4688 [zrtp_passthru] Setting b-leg absolute_codec_string=PCMA at 8000h@20i at 64000b,PCMU at 8000h@20i at 64000b 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_state_machine.c:472 (sofia/external/sip:user.name at sip.othersipgw.com) Running State Change CS_INIT 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_state_machine.c:512 (sofia/external/sip:user.name at sip.othersipgw.com) State INIT 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] mod_sofia.c:87 sofia/external/sip:user.name at sip.othersipgw.com SOFIA INIT 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] sofia_glue.c:1232 sofia/external/sip:user.name at sip.othersipgw.com sending invite version: 1.5.15b git 79de78a 2014-11-20 03:51:09Z 64bit 15187302-8051-11e4-90b0-2d01d6dac59f Local SDP: 15187302-8051-11e4-90b0-2d01d6dac59f v=0 15187302-8051-11e4-90b0-2d01d6dac59f o=FreeSWITCH 1418187966 1418187967 IN IP4 my.public.addr.ess 15187302-8051-11e4-90b0-2d01d6dac59f s=FreeSWITCH 15187302-8051-11e4-90b0-2d01d6dac59f c=IN IP4 my.public.addr.ess 15187302-8051-11e4-90b0-2d01d6dac59f t=0 0 15187302-8051-11e4-90b0-2d01d6dac59f m=audio 16680 RTP/AVP 8 0 101 13 15187302-8051-11e4-90b0-2d01d6dac59f a=rtpmap:8 PCMA/8000 15187302-8051-11e4-90b0-2d01d6dac59f a=rtpmap:0 PCMU/8000 15187302-8051-11e4-90b0-2d01d6dac59f a=rtpmap:101 telephone-event/8000 15187302-8051-11e4-90b0-2d01d6dac59f a=fmtp:101 0-16 15187302-8051-11e4-90b0-2d01d6dac59f a=ptime:20 15187302-8051-11e4-90b0-2d01d6dac59f a=sendrecv 15187302-8051-11e4-90b0-2d01d6dac59f 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_state_machine.c:40 sofia/external/sip:user.name at sip.othersipgw.com Standard INIT 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_state_machine.c:48 (sofia/external/sip:user.name at sip.othersipgw.com) State Change CS_INIT -> CS_ROUTING 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_session.c:1388 Send signal sofia/external/sip:user.name at sip.othersipgw.com [BREAK] 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_state_machine.c:512 (sofia/external/sip:user.name at sip.othersipgw.com) State INIT going to sleep 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_state_machine.c:472 (sofia/external/sip:user.name at sip.othersipgw.com) Running State Change CS_ROUTING 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_state_machine.c:528 (sofia/external/sip:user.name at sip.othersipgw.com) State ROUTING 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] mod_sofia.c:123 sofia/external/sip:user.name at sip.othersipgw.com SOFIA ROUTING 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_ivr_originate.c:67 (sofia/external/sip:user.name at sip.othersipgw.com) State Change CS_ROUTING -> CS_CONSUME_MEDIA 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_session.c:1388 Send signal sofia/external/sip:user.name at sip.othersipgw.com [BREAK] 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_state_machine.c:528 (sofia/external/sip:user.name at sip.othersipgw.com) State ROUTING going to sleep 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_state_machine.c:472 (sofia/external/sip:user.name at sip.othersipgw.com) Running State Change CS_CONSUME_MEDIA 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_state_machine.c:547 (sofia/external/sip:user.name at sip.othersipgw.com) State CONSUME_MEDIA 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_state_machine.c:547 (sofia/external/sip:user.name at sip.othersipgw.com) State CONSUME_MEDIA going to sleep 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] switch_core_session.c:1053 Send signal sofia/external/sip:user.name at sip.othersipgw.com [BREAK] 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285 [DEBUG] sofia.c:6614 Channel sofia/external/sip:user.name at sip.othersipgw.com entering state [calling][0] 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.965294 [DEBUG] switch_core_session.c:1053 Send signal sofia/external/sip:user.name at sip.othersipgw.com [BREAK] 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.965294 [DEBUG] switch_core_session.c:1053 Send signal sofia/external/sip:user.name at sip.othersipgw.com [BREAK] 2014-12-10 10:44:06.965294 [INFO] sofia.c:1203 sofia/external/sip:user.name at sip.othersipgw.com Update Callee ID to "user.name" 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.965294 [DEBUG] sofia.c:6614 Channel sofia/external/sip:user.name at sip.othersipgw.com entering state [proceeding][180] 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.965294 [NOTICE] sofia.c:6716 Ring-Ready sofia/external/sip:user.name at sip.othersipgw.com! 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.965294 [DEBUG] switch_channel.c:3277 (sofia/external/sip:user.name at sip.othersipgw.com) Callstate Change DOWN -> RINGING 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.965294 [NOTICE] mod_sofia.c:2086 Ring-Ready sofia/external/0diallednumber at first.sipgw.net! 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.965294 [DEBUG] switch_core_session.c:1053 Send signal sofia/external/0diallednumber at first.sipgw.net [BREAK] 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.965294 [DEBUG] sofia.c:6614 Channel sofia/external/0diallednumber at first.sipgw.net entering state [early][180] 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.965294 [DEBUG] switch_core_session.c:908 Send signal sofia/external/0diallednumber at first.sipgw.net [BREAK] 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.965294 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/external/0diallednumber at first.sipgw.net! 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:09.965334 [DEBUG] switch_core_session.c:1053 Send signal sofia/external/sip:user.name at sip.othersipgw.com [BREAK] 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:09.965334 [DEBUG] switch_core_session.c:1053 Send signal sofia/external/sip:user.name at sip.othersipgw.com [BREAK] 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:09.965334 [DEBUG] switch_core_session.c:1053 Send signal sofia/external/sip:user.name at sip.othersipgw.com [BREAK] 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:09.965334 [DEBUG] sofia.c:5840 Remote Reason: 16 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:09.965334 [DEBUG] sofia.c:6614 Channel sofia/external/sip:user.name at sip.othersipgw.com entering state [terminated][480] 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:09.965334 [NOTICE] sofia.c:7530 Hangup sofia/external/sip:user.name at sip.othersipgw.com [CS_CONSUME_MEDIA] [NORMAL_CLEARING] 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:09.965334 [DEBUG] switch_channel.c:3222 Send signal sofia/external/sip:user.name at sip.othersipgw.com [KILL] 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:09.965334 [DEBUG] switch_core_session.c:1388 Send signal sofia/external/sip:user.name at sip.othersipgw.com [BREAK] 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:09.965334 [DEBUG] switch_core_state_machine.c:472 (sofia/external/sip:user.name at sip.othersipgw.com) Running State Change CS_HANGUP 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:09.965334 [DEBUG] switch_core_state_machine.c:735 (sofia/external/sip:user.name at sip.othersipgw.com) Callstate Change RINGING -> HANGUP 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:09.965334 [DEBUG] switch_core_state_machine.c:737 (sofia/external/sip:user.name at sip.othersipgw.com) State HANGUP 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:09.965334 [DEBUG] mod_sofia.c:413 Channel sofia/external/sip:user.name at sip.othersipgw.com hanging up, cause: NORMAL_CLEARING 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:09.965334 [DEBUG] switch_core_state_machine.c:60 sofia/external/sip:user.name at sip.othersipgw.com Standard HANGUP, cause: NORMAL_CLEARING 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:09.965334 [DEBUG] switch_core_state_machine.c:737 (sofia/external/sip:user.name at sip.othersipgw.com) State HANGUP going to sleep 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:09.965334 [DEBUG] switch_core_state_machine.c:504 (sofia/external/sip:user.name at sip.othersipgw.com) State Change CS_HANGUP -> CS_REPORTING 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:09.965334 [DEBUG] switch_core_session.c:1388 Send signal sofia/external/sip:user.name at sip.othersipgw.com [BREAK] 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:09.965334 [DEBUG] switch_core_state_machine.c:472 (sofia/external/sip:user.name at sip.othersipgw.com) Running State Change CS_REPORTING 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:09.965334 [DEBUG] switch_core_state_machine.c:823 (sofia/external/sip:user.name at sip.othersipgw.com) State REPORTING 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:09.965334 [DEBUG] switch_core_state_machine.c:104 sofia/external/sip:user.name at sip.othersipgw.com Standard REPORTING, cause: NORMAL_CLEARING 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:09.965334 [DEBUG] switch_core_state_machine.c:823 (sofia/external/sip:user.name at sip.othersipgw.com) State REPORTING going to sleep 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:09.965334 [DEBUG] switch_core_state_machine.c:498 (sofia/external/sip:user.name at sip.othersipgw.com) State Change CS_REPORTING -> CS_DESTROY 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:09.965334 [DEBUG] switch_core_session.c:1388 Send signal sofia/external/sip:user.name at sip.othersipgw.com [BREAK] 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:09.965334 [DEBUG] switch_core_session.c:1615 Session 232 (sofia/external/sip:user.name at sip.othersipgw.com) Locked, Waiting on external entities 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.965334 [DEBUG] switch_ivr_originate.c:3698 Originate Resulted in Error Cause: 16 [NORMAL_CLEARING] 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:09.965334 [NOTICE] switch_core_session.c:1633 Session 232 (sofia/external/sip:user.name at sip.othersipgw.com) Ended 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:09.965334 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/sip:user.name at sip.othersipgw.com [CS_DESTROY] 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.965334 [INFO] mod_dptools.c:3234 Originate Failed. Cause: NORMAL_CLEARING 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.965334 [NOTICE] switch_channel.c:4724 Hangup sofia/external/0diallednumber at first.sipgw.net [CS_EXECUTE] [NORMAL_CLEARING] 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.965334 [DEBUG] switch_channel.c:3222 Send signal sofia/external/0diallednumber at first.sipgw.net [KILL] 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.965334 [DEBUG] switch_core_session.c:1388 Send signal sofia/external/0diallednumber at first.sipgw.net [BREAK] 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.965334 [DEBUG] switch_core_session.c:2893 sofia/external/0diallednumber at first.sipgw.net skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.965334 [DEBUG] switch_core_state_machine.c:535 (sofia/external/0diallednumber at first.sipgw.net) State EXECUTE going to sleep 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.965334 [DEBUG] switch_core_state_machine.c:472 (sofia/external/0diallednumber at first.sipgw.net) Running State Change CS_HANGUP 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.965334 [DEBUG] switch_core_state_machine.c:472 (sofia/external/0diallednumber at first.sipgw.net) Running State Change CS_HANGUP 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:09.965334 [DEBUG] switch_core_state_machine.c:626 (sofia/external/sip:user.name at sip.othersipgw.com) Running State Change CS_DESTROY 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:09.965334 [DEBUG] switch_core_state_machine.c:636 (sofia/external/sip:user.name at sip.othersipgw.com) State DESTROY 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:09.965334 [DEBUG] mod_sofia.c:323 sofia/external/sip:user.name at sip.othersipgw.com SOFIA DESTROY 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:09.965334 [DEBUG] switch_core_state_machine.c:111 sofia/external/sip:user.name at sip.othersipgw.com Standard DESTROY 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:09.965334 [DEBUG] switch_core_state_machine.c:636 (sofia/external/sip:user.name at sip.othersipgw.com) State DESTROY going to sleep 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:09.965334 [DEBUG] switch_core_state_machine.c:111 sofia/external/sip:user.name at sip.othersipgw.com Standard DESTROY 15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:09.965334 [DEBUG] switch_core_state_machine.c:636 (sofia/external/sip:user.name at sip.othersipgw.com) State DESTROY going to sleep 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.965334 [DEBUG] switch_core_state_machine.c:735 (sofia/external/0diallednumber at first.sipgw.net) Callstate Change RINGING -> HANGUP 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.965334 [DEBUG] switch_core_state_machine.c:737 (sofia/external/0diallednumber at first.sipgw.net) State HANGUP 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.965334 [DEBUG] mod_sofia.c:407 sofia/external/0diallednumber at first.sipgw.net Overriding SIP cause 480 with 480 from the other leg 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.965334 [DEBUG] mod_sofia.c:413 Channel sofia/external/0diallednumber at first.sipgw.net hanging up, cause: NORMAL_CLEARING 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.985315 [DEBUG] mod_sofia.c:549 Responding to INVITE with: 480 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.985315 [DEBUG] switch_core_state_machine.c:60 sofia/external/0diallednumber at first.sipgw.net Standard HANGUP, cause: NORMAL_CLEARING 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.985315 [DEBUG] switch_core_state_machine.c:737 (sofia/external/0diallednumber at first.sipgw.net) State HANGUP going to sleep 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.985315 [DEBUG] switch_core_state_machine.c:504 (sofia/external/0diallednumber at first.sipgw.net) State Change CS_HANGUP -> CS_REPORTING 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.985315 [DEBUG] switch_core_session.c:1388 Send signal sofia/external/0diallednumber at first.sipgw.net [BREAK] 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.985315 [DEBUG] switch_core_state_machine.c:472 (sofia/external/0diallednumber at first.sipgw.net) Running State Change CS_REPORTING 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.985315 [DEBUG] switch_core_state_machine.c:823 (sofia/external/0diallednumber at first.sipgw.net) State REPORTING 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.985315 [DEBUG] switch_core_state_machine.c:104 sofia/external/0diallednumber at first.sipgw.net Standard REPORTING, cause: NORMAL_CLEARING 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.985315 [DEBUG] switch_core_state_machine.c:737 (sofia/external/0diallednumber at first.sipgw.net) State HANGUP going to sleep 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.985315 [DEBUG] switch_core_state_machine.c:504 (sofia/external/0diallednumber at first.sipgw.net) State Change CS_HANGUP -> CS_REPORTING 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.985315 [DEBUG] switch_core_session.c:1388 Send signal sofia/external/0diallednumber at first.sipgw.net [BREAK] 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.985315 [DEBUG] switch_core_state_machine.c:472 (sofia/external/0diallednumber at first.sipgw.net) Running State Change CS_REPORTING 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.985315 [DEBUG] switch_core_state_machine.c:823 (sofia/external/0diallednumber at first.sipgw.net) State REPORTING 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.985315 [DEBUG] switch_core_state_machine.c:104 sofia/external/0diallednumber at first.sipgw.net Standard REPORTING, cause: NORMAL_CLEARING 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.985315 [DEBUG] switch_core_state_machine.c:823 (sofia/external/0diallednumber at first.sipgw.net) State REPORTING going to sleep 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.985315 [DEBUG] switch_core_state_machine.c:498 (sofia/external/0diallednumber at first.sipgw.net) State Change CS_REPORTING -> CS_DESTROY 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.985315 [DEBUG] switch_core_session.c:1388 Send signal sofia/external/0diallednumber at first.sipgw.net [BREAK] 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.985315 [DEBUG] switch_core_session.c:1615 Session 231 (sofia/external/0diallednumber at first.sipgw.net) Locked, Waiting on external entities 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.985315 [DEBUG] switch_core_session.c:1388 Send signal sofia/external/0diallednumber at first.sipgw.net [BREAK] 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.985315 [DEBUG] switch_core_session.c:1615 Session 231 (sofia/external/0diallednumber at first.sipgw.net) Locked, Waiting on external entities 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.985315 [NOTICE] switch_core_session.c:1633 Session 231 (sofia/external/0diallednumber at first.sipgw.net) Ended 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.985315 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/0diallednumber at first.sipgw.net [CS_DESTROY] 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.985315 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/0diallednumber at first.sipgw.net [CS_DESTROY] 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.985315 [DEBUG] switch_core_state_machine.c:626 (sofia/external/0diallednumber at first.sipgw.net) Running State Change CS_DESTROY 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.985315 [DEBUG] switch_core_state_machine.c:626 (sofia/external/0diallednumber at first.sipgw.net) Running State Change CS_DESTROY 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.985315 [DEBUG] switch_core_state_machine.c:636 (sofia/external/0diallednumber at first.sipgw.net) State DESTROY 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.985315 [DEBUG] mod_sofia.c:323 sofia/external/0diallednumber at first.sipgw.net SOFIA DESTROY 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.985315 [DEBUG] switch_core_state_machine.c:111 sofia/external/0diallednumber at first.sipgw.net Standard DESTROY 1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.985315 [DEBUG] switch_core_state_machine.c:636 (sofia/external/0diallednumber at first.sipgw.net) State DESTROY going to sleep On Wed, Dec 10, 2014 at 6:50 AM, Avi Marcus wrote: > I would try something "simpler". This will always play the file, though, no > matter the response from the othersipgw. > > > > > data="sofia/external/sip:user.name at sip.othersipgw.ip.address.com;${effective_caller_id_number}"/> > > > > > > > > > > > From jobindcruz at gmail.com Wed Dec 10 15:13:40 2014 From: jobindcruz at gmail.com (jobin dcruz) Date: Wed, 10 Dec 2014 17:43:40 +0530 Subject: [Freeswitch-users] Call Conference Time out issue Message-ID: Hi, I had set call conference hang up time,but it is working after 4 seconds as per attendees count. This is dial plan example Ex: i had set one min(60 sec) hangup time,but call conference ended with 64 sec -- Jobin D'cruz (LAMP Developer) Email: jobindcruz at gmail.com | lampdeveloper.jobindcruz at gmail.com Php MySql Linux Apache Ajax FFmpeg FFmpeg-php Mplayer Zend Framework Codeigniter CakePhp Wordpress Drupal Joomla Html5 Css3 jQuery YUI MooTools Smarty Firefox Extension -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141210/451017d8/attachment.html From fs at voice2net.ca Wed Dec 10 15:52:41 2014 From: fs at voice2net.ca (Darcy Primrose) Date: Wed, 10 Dec 2014 07:52:41 -0500 Subject: [Freeswitch-users] cisco spa50xG and music on hold References: <0000014a0b2a7f3a-e057e705-e811-4903-b2b9-bbd020abb4a7-000000@email.amazonses.com><0000014a22d99893-8fb2f862-1a14-4cb2-b363-8e924d56012d-000000@email.amazonses.com><0000014a2a0f8f9d-1038c04d-e533-483b-8322-bd27ace4ba19-000000@email.amazonses.com><0000014a32c00982-5991e950-58e0-477c-bcab-ead784e5d4d6-000000@email.amazonses.com> Message-ID: <45748814B5CE4A8C9CC11B597D06F498@DARCY> We have a problem with music on hold on the 504g phones, wonder if anyone has this working. When an inbound call to a 504g is placed on hold, the stream is played towards to cisco504g instead of the calling party who is now on hold. In the Cisco 504g we have the moh server set blank. Also, if we park the call to the 504g, retrieve it, then place it on hold, then the music on hold works. Any one have any thoughts or ideas on this. Darcy Primrose Voice2Net From vipkilla at gmail.com Wed Dec 10 16:00:14 2014 From: vipkilla at gmail.com (Vik Killa) Date: Wed, 10 Dec 2014 08:00:14 -0500 Subject: [Freeswitch-users] cisco spa50xG and music on hold In-Reply-To: <45748814B5CE4A8C9CC11B597D06F498@DARCY> References: <0000014a0b2a7f3a-e057e705-e811-4903-b2b9-bbd020abb4a7-000000@email.amazonses.com> <0000014a22d99893-8fb2f862-1a14-4cb2-b363-8e924d56012d-000000@email.amazonses.com> <0000014a2a0f8f9d-1038c04d-e533-483b-8322-bd27ace4ba19-000000@email.amazonses.com> <0000014a32c00982-5991e950-58e0-477c-bcab-ead784e5d4d6-000000@email.amazonses.com> <45748814B5CE4A8C9CC11B597D06F498@DARCY> Message-ID: Hi Darcy, I know there are some issues with MOH right now. Please see https://freeswitch.org/jira/browse/FS-7015 I had to revert to commit 39be8777608f3eed13738a50e03c0969b366414b which doesn't seem to have the MOH issues. Thanks /V On Wed, Dec 10, 2014 at 7:52 AM, Darcy Primrose wrote: > We have a problem with music on hold on the 504g phones, wonder if anyone > has this working. When an inbound call to a 504g is placed on hold, the > stream is played towards to cisco504g instead of the calling party who is > now on hold. In the Cisco 504g we have the moh server set blank. Also, if > we park the call to the 504g, retrieve it, then place it on hold, then the > music on hold works. > > Any one have any thoughts or ideas on this. > > Darcy Primrose > Voice2Net > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141210/abb12f6c/attachment.html From fs at voice2net.ca Wed Dec 10 16:23:29 2014 From: fs at voice2net.ca (Darcy Primrose) Date: Wed, 10 Dec 2014 08:23:29 -0500 Subject: [Freeswitch-users] cisco spa50xG and music on hold References: <0000014a0b2a7f3a-e057e705-e811-4903-b2b9-bbd020abb4a7-000000@email.amazonses.com><0000014a22d99893-8fb2f862-1a14-4cb2-b363-8e924d56012d-000000@email.amazonses.com><0000014a2a0f8f9d-1038c04d-e533-483b-8322-bd27ace4ba19-000000@email.amazonses.com><0000014a32c00982-5991e950-58e0-477c-bcab-ead784e5d4d6-000000@email.amazonses.com><45748814B5CE4A8C9CC11B597D06F498@DARCY> Message-ID: <98566ABE87D844A7BA99E0ED4C433571@DARCY> This happens on all Versions of Freeswitch I have running with the Cisco spa50xG phones, I have tried it on a version that is relatively stable from 2012 and on the latest version I just download from GIT, All other phones work perfectly, that is Snom, Grandstream and Yealink. Darcy ----- Original Message ----- From: Vik Killa To: FreeSWITCH Users Help Sent: Wednesday, December 10, 2014 8:00 AM Subject: Re: [Freeswitch-users] cisco spa50xG and music on hold Hi Darcy, I know there are some issues with MOH right now. Please see https://freeswitch.org/jira/browse/FS-7015 I had to revert to commit 39be8777608f3eed13738a50e03c0969b366414b which doesn't seem to have the MOH issues. Thanks /V On Wed, Dec 10, 2014 at 7:52 AM, Darcy Primrose wrote: We have a problem with music on hold on the 504g phones, wonder if anyone has this working. When an inbound call to a 504g is placed on hold, the stream is played towards to cisco504g instead of the calling party who is now on hold. In the Cisco 504g we have the moh server set blank. Also, if we park the call to the 504g, retrieve it, then place it on hold, then the music on hold works. Any one have any thoughts or ideas on this. Darcy Primrose Voice2Net _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ------------------------------------------------------------------------------ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141210/9986a206/attachment.html From miha at softnet.si Wed Dec 10 16:52:58 2014 From: miha at softnet.si (Miha) Date: Wed, 10 Dec 2014 14:52:58 +0100 Subject: [Freeswitch-users] Listening on two different ports Message-ID: <5488503A.1080502@softnet.si> Hi, which config should be made that FS will listen on two different ports (for exp 5060 and 5070) and also sending on two different ports. I have created two external profiles, so I must create also to different internal profiles? br miha From bote_radio at botecomm.com Wed Dec 10 17:31:18 2014 From: bote_radio at botecomm.com (Bote Man) Date: Wed, 10 Dec 2014 09:31:18 -0500 Subject: [Freeswitch-users] Listening on two different ports In-Reply-To: <5488503A.1080502@softnet.si> References: <5488503A.1080502@softnet.si> Message-ID: <038301d01485$f70e74b0$e52b5e10$@com> One profile listens on one port. The names "external" and "internal" are arbitrary, just be careful as to which dialplan you hand off inbound calls so that they are processed with minimal permissions necessary. Bote > -----Original Message----- > From: Miha > Sent: Wednesday, 10 December, 2014 08:53 > Subject: [Freeswitch-users] Listening on two different ports > > Hi, > > which config should be made that FS will listen on two different ports > (for exp 5060 and 5070) and also sending on two different ports. > > I have created two external profiles, so I must create also to > different internal profiles? > > br > miha > > From vipkilla at gmail.com Wed Dec 10 17:33:06 2014 From: vipkilla at gmail.com (Vik Killa) Date: Wed, 10 Dec 2014 09:33:06 -0500 Subject: [Freeswitch-users] cisco spa50xG and music on hold In-Reply-To: <98566ABE87D844A7BA99E0ED4C433571@DARCY> References: <0000014a0b2a7f3a-e057e705-e811-4903-b2b9-bbd020abb4a7-000000@email.amazonses.com> <0000014a22d99893-8fb2f862-1a14-4cb2-b363-8e924d56012d-000000@email.amazonses.com> <0000014a2a0f8f9d-1038c04d-e533-483b-8322-bd27ace4ba19-000000@email.amazonses.com> <0000014a32c00982-5991e950-58e0-477c-bcab-ead784e5d4d6-000000@email.amazonses.com> <45748814B5CE4A8C9CC11B597D06F498@DARCY> <98566ABE87D844A7BA99E0ED4C433571@DARCY> Message-ID: I have two SPA50X setup with FS and I'm not having this issue. On Wed, Dec 10, 2014 at 8:23 AM, Darcy Primrose wrote: > This happens on all Versions of Freeswitch I have running with the Cisco > spa50xG phones, I have tried it on a version that is relatively stable from > 2012 and on the latest version I just download from GIT, All other phones > work perfectly, that is Snom, Grandstream and Yealink. > > Darcy > > ----- Original Message ----- > *From:* Vik Killa > *To:* FreeSWITCH Users Help > *Sent:* Wednesday, December 10, 2014 8:00 AM > *Subject:* Re: [Freeswitch-users] cisco spa50xG and music on hold > > Hi Darcy, > I know there are some issues with MOH right now. > Please see https://freeswitch.org/jira/browse/FS-7015 > I had to revert to commit 39be8777608f3eed13738a50e03c0969b366414b which > doesn't seem to have the MOH issues. > Thanks > /V > > On Wed, Dec 10, 2014 at 7:52 AM, Darcy Primrose wrote: > >> We have a problem with music on hold on the 504g phones, wonder if anyone >> has this working. When an inbound call to a 504g is placed on hold, the >> stream is played towards to cisco504g instead of the calling party who is >> now on hold. In the Cisco 504g we have the moh server set blank. Also, >> if >> we park the call to the 504g, retrieve it, then place it on hold, then the >> music on hold works. >> >> Any one have any thoughts or ideas on this. >> >> Darcy Primrose >> Voice2Net >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > ------------------------------ > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141210/c8eed102/attachment-0001.html From aqsyounas at gmail.com Wed Dec 10 17:39:14 2014 From: aqsyounas at gmail.com (Aqs Younas) Date: Wed, 10 Dec 2014 19:39:14 +0500 Subject: [Freeswitch-users] error: Frame size too big: 12121 Message-ID: Hi, All I am new to free-switch. Sometimes i encounter this error while playing a stream with mod_shout. These are the logs that i see while running freeswitch without background. [parse.c:1026] error: Frame size too big: 12121 [parse.c:1026] error: Frame size too big: 12121 [parse.c:1026] error: Frame size too big: 12121 [parse.c:1026] error: Frame size too big: 12121 [parse.c:1026] error: Frame size too big: 12121 [parse.c:1026] error: Frame size too big: 12121 2014-12-10 14:30:38.149601 [ERR] mod_shout.c:692 Error opening streaming205.radionomy.com/Aurora-de-Siempre?group=0&countrycode=SV&similarprocess=0&crashed=0 (data stream timeout) Any Help would be much appreciated. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141210/05b36c69/attachment.html From bote_radio at botecomm.com Wed Dec 10 17:42:04 2014 From: bote_radio at botecomm.com (Bote Man) Date: Wed, 10 Dec 2014 09:42:04 -0500 Subject: [Freeswitch-users] cisco spa50xG and music on hold In-Reply-To: <98566ABE87D844A7BA99E0ED4C433571@DARCY> References: <0000014a0b2a7f3a-e057e705-e811-4903-b2b9-bbd020abb4a7-000000@email.amazonses.com><0000014a22d99893-8fb2f862-1a14-4cb2-b363-8e924d56012d-000000@email.amazonses.com><0000014a2a0f8f9d-1038c04d-e533-483b-8322-bd27ace4ba19-000000@email.amazonses.com><0000014a32c00982-5991e950-58e0-477c-bcab-ead784e5d4d6-000000@email.amazonses.com><45748814B5CE4A8C9CC11B597D06F498@DARCY> <98566ABE87D844A7BA99E0ED4C433571@DARCY> Message-ID: <038401d01487$7794b800$66be2800$@com> Can you compare the signaling sent by the Cisco phones against the signaling sent by the others like the Grandstream? It sounds like the Cisco is sending a message that is confusing FS. Although I have read comments here that indicate that Cisco phones do lots of strange things :- ) Bote From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Darcy Primrose Sent: Wednesday, 10 December, 2014 08:23 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] cisco spa50xG and music on hold This happens on all Versions of Freeswitch I have running with the Cisco spa50xG phones, I have tried it on a version that is relatively stable from 2012 and on the latest version I just download from GIT, All other phones work perfectly, that is Snom, Grandstream and Yealink. Darcy ----- Original Message ----- From: Vik Killa To: FreeSWITCH Users Help Sent: Wednesday, December 10, 2014 8:00 AM Subject: Re: [Freeswitch-users] cisco spa50xG and music on hold Hi Darcy, I know there are some issues with MOH right now. Please see https://freeswitch.org/jira/browse/FS-7015 I had to revert to commit 39be8777608f3eed13738a50e03c0969b366414b which doesn't seem to have the MOH issues. Thanks /V On Wed, Dec 10, 2014 at 7:52 AM, Darcy Primrose wrote: We have a problem with music on hold on the 504g phones, wonder if anyone has this working. When an inbound call to a 504g is placed on hold, the stream is played towards to cisco504g instead of the calling party who is now on hold. In the Cisco 504g we have the moh server set blank. Also, if we park the call to the 504g, retrieve it, then place it on hold, then the music on hold works. Any one have any thoughts or ideas on this. Darcy Primrose Voice2Net _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _____ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141210/619446fa/attachment.html From bilaln018 at gmail.com Wed Dec 10 18:09:55 2014 From: bilaln018 at gmail.com (Bilal Abbasi) Date: Wed, 10 Dec 2014 20:09:55 +0500 Subject: [Freeswitch-users] [Issue][CDR with mod_cdr_mongodb] Message-ID: Hi All, I am newbie in Freeswitch, i have successfully configured mongodb with freeswitch. The issue is that there is no schema provided for mongodb collection CDR. So when ever i put a test call it logs too much information ,so what should i do if i want limited variables to log in that collecction? Please help Regards Bilal Abbasi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141210/289840e3/attachment.html From fs at voice2net.ca Wed Dec 10 19:02:15 2014 From: fs at voice2net.ca (Darcy Primrose) Date: Wed, 10 Dec 2014 11:02:15 -0500 Subject: [Freeswitch-users] cisco spa50xG and music on hold References: <0000014a0b2a7f3a-e057e705-e811-4903-b2b9-bbd020abb4a7-000000@email.amazonses.com><0000014a22d99893-8fb2f862-1a14-4cb2-b363-8e924d56012d-000000@email.amazonses.com><0000014a2a0f8f9d-1038c04d-e533-483b-8322-bd27ace4ba19-000000@email.amazonses.com><0000014a32c00982-5991e950-58e0-477c-bcab-ead784e5d4d6-000000@email.amazonses.com><45748814B5CE4A8C9CC11B597D06F498@DARCY> <98566ABE87D844A7BA99E0ED4C433571@DARCY> <038401d01487$7794b800$66be2800$@com> Message-ID: The Cisco does the sendonly invite to mod_sofia at domain where as the other products send the sendonly invite to 1001 at domain, that seems to be the main difference. I will poke at it to see why. I am also running the lastest version of cisco software, perhaps they did a recent patch that caused this. Darcy ----- Original Message ----- From: Bote Man To: 'FreeSWITCH Users Help' Sent: Wednesday, December 10, 2014 9:42 AM Subject: Re: [Freeswitch-users] cisco spa50xG and music on hold Can you compare the signaling sent by the Cisco phones against the signaling sent by the others like the Grandstream? It sounds like the Cisco is sending a message that is confusing FS. Although I have read comments here that indicate that Cisco phones do lots of strange things :- ) Bote From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Darcy Primrose Sent: Wednesday, 10 December, 2014 08:23 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] cisco spa50xG and music on hold This happens on all Versions of Freeswitch I have running with the Cisco spa50xG phones, I have tried it on a version that is relatively stable from 2012 and on the latest version I just download from GIT, All other phones work perfectly, that is Snom, Grandstream and Yealink. Darcy ----- Original Message ----- From: Vik Killa To: FreeSWITCH Users Help Sent: Wednesday, December 10, 2014 8:00 AM Subject: Re: [Freeswitch-users] cisco spa50xG and music on hold Hi Darcy, I know there are some issues with MOH right now. Please see https://freeswitch.org/jira/browse/FS-7015 I had to revert to commit 39be8777608f3eed13738a50e03c0969b366414b which doesn't seem to have the MOH issues. Thanks /V On Wed, Dec 10, 2014 at 7:52 AM, Darcy Primrose wrote: We have a problem with music on hold on the 504g phones, wonder if anyone has this working. When an inbound call to a 504g is placed on hold, the stream is played towards to cisco504g instead of the calling party who is now on hold. In the Cisco 504g we have the moh server set blank. Also, if we park the call to the 504g, retrieve it, then place it on hold, then the music on hold works. Any one have any thoughts or ideas on this. Darcy Primrose Voice2Net _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ---------------------------------------------------------------------------- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ------------------------------------------------------------------------------ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141210/ec21d133/attachment-0001.html From telishisheer at gmail.com Wed Dec 10 19:42:37 2014 From: telishisheer at gmail.com (Shisheer Teli) Date: Wed, 10 Dec 2014 22:12:37 +0530 Subject: [Freeswitch-users] Ldap alise and freeswitch integration problem In-Reply-To: References: Message-ID: from my experiments and test i found that freeswitch sending password in plain text. Test 1: when i set ldap password in md5 , in freeswitch cli i see hash password and authentication failed. Test 2: when i set ldap password in plain text, in freeswitch cli i can see plain text password and authentication success. On Tue, Dec 2, 2014 at 9:55 AM, Shisheer Teli wrote: > Hi, > > I am able to bind with any alise on ldap server except userPassword (MD5) > alise. > > when i bind password with userPassword , authentication fails. > > any MD5 authentication configuration in freeswitch? > > Configuration file: > > > > > > > > > > > > > > > > > > > > > -- > Regards, > Shisheer T > Phone: +91-022 2278 2763 > > -- Regards, Shisheer Teli Phone: +91-022 2278 2519 / 2121 shisheer at tifr.res.in -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141210/49651d90/attachment.html From hexade at hotmail.com Wed Dec 10 19:45:40 2014 From: hexade at hotmail.com (Adelia C.) Date: Wed, 10 Dec 2014 11:45:40 -0500 Subject: [Freeswitch-users] Twilio SIP trunk config - outbound call rejected In-Reply-To: References: Message-ID: Any idea why this INVITE is rejected with 400 Invalid Phone Number by Twilio? The first INVITE asks for Proxy Authentication, that seems to pass and this is rejected. Thank you! INVITE sip:5101112222 at mycompany.pstn.twilio.com SIP/2.0 Via: SIP/2.0/UDP 10.50.1.116:5080;rport;branch=z9hG4bKa8QvU1B6QmeKc Max-Forwards: 69 From: "unknown" ;tag=6UetZr6rDZgHF To: Call-ID: 1ffda8b7-faaf-1232-8380-a9a65d1e96b1 CSeq: 68751485 INVITE Contact: Expires: 3600 User-Agent: FreeSWITCH-mod_sofia/1.5.12b~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Proxy-Authorization: Digest username="4152221111", realm="sip.twilio.com", nonce="f722ae99d0a818bc422baa5d8", cnonce="IA1gJvqvEg6mmXR6WsQ", opaque="e89d2328e07da020d75a", algorithm=MD5, uri="sip:5101112222 at mycompany.pstn.twilio.com", response="e65d3dea5b1e3b35306cfa121ec2ae4d", qop=auth, nc=00000001 Content-Type: application/sdp Content-Disposition: session Content-Length: 268 X-FS-Support: update_display,send_info Remote-Party-ID: "unknown" ;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1418153731 1418153732 IN IP4 10.50.1.116 s=FreeSWITCH c=IN IP4 10.50.1.116 t=0 0 m=audio 21366 RTP/AVP 0 8 3 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 -- SIP/2.0 400 Invalid phone number To: ;tag=08602519_6772d868_5eef74bb-957c-486e-b261-2a2e63704d73 Via: SIP/2.0/UDP 10.50.1.116:5080;received=10.50.1.116;rport=5080;branch=z9hG4bKa8QvU1B6QmeKc CSeq: 68751485 INVITE Call-ID: 1ffda8b7-faaf-1232-8380-a9a65d1e96b1 From: "unknown" ;tag=6UetZr6rDZgHF Contact: Content-Length: 0 -- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141210/e2b73c1e/attachment.html From mike at jerris.com Wed Dec 10 20:28:07 2014 From: mike at jerris.com (Michael Jerris) Date: Wed, 10 Dec 2014 12:28:07 -0500 Subject: [Freeswitch-users] Mod perl using $session->setVariable inline In-Reply-To: References: Message-ID: <38660852-0B8E-43DD-929A-0D3C8CB2DD5B@jerris.com> If you are in a perl script using session->setVariable, how are you still in the dialplan? This doesn't make sense to me. There should be no need for an inline if you are already in the script. > On Dec 10, 2014, at 4:20 AM, Liam Farr wrote: > > Hi, > > I need to set a variable using a perl script inline. > > Normally I would use $session->setVariable('some_var',$my_value); > > However I cant find any options to execute this inline / as an inline action. (I need to evaluate the variable later in the hunting phase of the dialplan hence need it to be an inline action). > > Is there a way to do this? > > Or is there another way to achieve the same result, e.g. hacking $session->execute("set","some_var=$my_value") to run as an inline action? > > Any help would be much appreciated, I've scoured the wiki, confluence and git repo and come up empty. > > -- > Kind Regards > > > Liam Farr From wampir990 at gmail.com Wed Dec 10 20:48:19 2014 From: wampir990 at gmail.com (Jacekalex) Date: Wed, 10 Dec 2014 18:48:19 +0100 Subject: [Freeswitch-users] Mod_sofia build error. Message-ID: <54888763.1020107@gmail.com> Hi OS Gentoo Linux x86_64 gcc (Gentoo Hardened 4.8.3 p1.1, pie-0.5.9) 4.8.3 The previous system: gcc (Gentoo Hardened 4.7.3-r1 p1.3, pie-0.5.5) 4.7.3 Freeswitch versions - all 1.4.x Mod_sofia - sofia-sip error: ... Making all in tport mke[9]: Entering directory '/home/fabryka/local/src/freeswitch/freeswitch481/libs/sofia-sip/libsofia-sip-ua/tport' LTake[8]: Entering directory '/home/fabryka/local/src/freeswitch/freeswitch481/libs/sofia-sip/libsofia-sip-ua/tport' make all-am maCOMPILE tport_type_sctp.lo tport_type_sctp.c: In function ?tport_sctp_init_socket?: tport_type_sctp.c:206:10: error: variable ?initmsg? has initializer but incomplete type struct sctp_initmsg initmsg = { 0 }; ^ tport_type_sctp.c:206:10: error: excess elements in struct initializer [-Werror] tport_type_sctp.c:206:10: error: (near initialization for ?initmsg?) [-Werror] tport_type_sctp.c:206:23: error: storage size of ?initmsg? isn?t known struct sctp_initmsg initmsg = { 0 }; ^ tport_type_sctp.c:211:36: error: ?SCTP_INITMSG? undeclared (first use in this function) if (setsockopt(socket, SOL_SCTP, SCTP_INITMSG, &initmsg, sizeof initmsg) < 0) ^ tport_type_sctp.c:211:36: note: each undeclared identifier is reported only once for each function it appears in tport_type_sctp.c:206:23: error: unused variable ?initmsg? [-Werror=unused-variable] struct sctp_initmsg initmsg = { 0 }; ^ cc1: all warnings being treated as errors Makefile:1452: recipe for target 'tport_type_sctp.lo' failed make[9]: *** [tport_type_sctp.lo] Error 1 make[9]: Leaving directory '/home/fabryka/local/src/freeswitch/freeswitch481/libs/sofia-sip/libsofia-sip-ua/tport' Makefile:736: recipe for target 'all' failed make[8]: *** [all] Error 2 make[8]: Leaving directory '/home/fabryka/local/src/freeswitch/freeswitch481/libs/sofia-sip/libsofia-sip-ua/tport' Makefile:547: recipe for target 'all-recursive' failed make[7]: *** [all-recursive] Error 1 make[7]: Leaving directory '/home/fabryka/local/src/freeswitch/freeswitch481/libs/sofia-sip/libsofia-sip-ua' Makefile:490: recipe for target 'all-recursive' failed make[6]: *** [all-recursive] Error 1 make[6]: Leaving directory '/home/fabryka/local/src/freeswitch/freeswitch481/libs/sofia-sip' Makefile:410: recipe for target 'all' failed make[5]: *** [all] Error 2 make[5]: Leaving directory '/home/fabryka/local/src/freeswitch/freeswitch481/libs/sofia-sip' Makefile:906: recipe for target '/usr/local/src/freeswitch/freeswitch481/libs/sofia-sip/libsofia-sip-ua/libsofia-sip-ua.la' failed make[4]: *** [/usr/local/src/freeswitch/freeswitch481/libs/sofia-sip/libsofia-sip-ua/libsofia-sip-ua.la] Error 2 make[4]: Leaving directory '/home/fabryka/local/src/freeswitch/freeswitch481/src/mod/endpoints/mod_sofia' Makefile:577: recipe for target 'mod_sofia-all' failed make[3]: *** [mod_sofia-all] Error 1 make[3]: Leaving directory '/home/fabryka/local/src/freeswitch/freeswitch481/src/mod' Makefile:491: recipe for target 'all-recursive' failed make[2]: *** [all-recursive] Error 1 make[2]: Leaving directory '/home/fabryka/local/src/freeswitch/freeswitch481/src' Makefile:2487: recipe for target 'all-recursive' failed make[1]: *** [all-recursive] Error 1 make[1]: Leaving directory '/home/fabryka/local/src/freeswitch/freeswitch481' Makefile:969: recipe for target 'all' failed make: *** [all] Error 2 Error amazes me because sofia-sip library from portage ebuild installed without a problem: qlist -UCvq sofia-sip net-libs/sofia-sip-1.12.11 ssl static-libs http://data.gpo.zugaina.org/gentoo/net-libs/sofia-sip/ The latest version of FreeSWITCH, I have been able to install without a problem are: FreeSWITCH version: 1.5.14b+git~20140725T212415Z~c411f8c7a9~64bit (git c411f8c 2014-07-25 21:24:15Z 64bit) Where can I find the source FreeSWITCH-1.5.x, or alternatively, how to improve this error? Cheers -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 213 bytes Desc: OpenPGP digital signature Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141210/1880ac48/attachment.bin From liam at maxumdata.com Wed Dec 10 20:56:33 2014 From: liam at maxumdata.com (Liam Farr) Date: Thu, 11 Dec 2014 06:56:33 +1300 Subject: [Freeswitch-users] Mod perl using $session->setVariable inline In-Reply-To: <38660852-0B8E-43DD-929A-0D3C8CB2DD5B@jerris.com> References: <38660852-0B8E-43DD-929A-0D3C8CB2DD5B@jerris.com> Message-ID: Heres an expanded example; Extension runs the perl script as part of the public dialplan first; Script sets some variables; $my_value = "123"; $session->setVariable('some_var',$my_value); Use that variable later in another extension in the public dialplan as a condition selector; However this will fail, because the perl script "$session->setVariable" run earlier is not inline, so you cant make use of it in the condition () in the second extension that exists in a later part of the public dialplan. Hence I need a way to set inline variables from my perl script. Cheers Liam Farr On 11 December 2014 at 06:28, Michael Jerris wrote: > If you are in a perl script using session->setVariable, how are you still > in the dialplan? This doesn't make sense to me. There should be no need > for an inline if you are already in the script. > > > On Dec 10, 2014, at 4:20 AM, Liam Farr wrote: > > > > Hi, > > > > I need to set a variable using a perl script inline. > > > > Normally I would use $session->setVariable('some_var',$my_value); > > > > However I cant find any options to execute this inline / as an inline > action. (I need to evaluate the variable later in the hunting phase of the > dialplan hence need it to be an inline action). > > > > Is there a way to do this? > > > > Or is there another way to achieve the same result, e.g. hacking > $session->execute("set","some_var=$my_value") to run as an inline action? > > > > Any help would be much appreciated, I've scoured the wiki, confluence > and git repo and come up empty. > > > > -- > > Kind Regards > > > > > > Liam Farr > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141211/72257774/attachment-0001.html From mike at jerris.com Wed Dec 10 20:58:19 2014 From: mike at jerris.com (Michael Jerris) Date: Wed, 10 Dec 2014 12:58:19 -0500 Subject: [Freeswitch-users] Mod_sofia build error. In-Reply-To: <54888763.1020107@gmail.com> References: <54888763.1020107@gmail.com> Message-ID: <422D5329-EA7F-4D72-8D2D-BD9EAF8D7F32@jerris.com> This could be related to bugs in gentoo headers related to sctp or due to issues in the autoconf detection for these. We would be happy to take patches to correct this issue that don't break other distro/os if you can provide a patch. > On Dec 10, 2014, at 12:48 PM, Jacekalex wrote: > > > Hi > > OS Gentoo Linux x86_64 > gcc (Gentoo Hardened 4.8.3 p1.1, pie-0.5.9) 4.8.3 > > The previous system: > gcc (Gentoo Hardened 4.7.3-r1 p1.3, pie-0.5.5) 4.7.3 > > Freeswitch versions - all 1.4.x > > Mod_sofia - sofia-sip error: > ... > Making all in tport > mke[9]: Entering directory > '/home/fabryka/local/src/freeswitch/freeswitch481/libs/sofia-sip/libsofia-sip-ua/tport' > LTake[8]: Entering directory > '/home/fabryka/local/src/freeswitch/freeswitch481/libs/sofia-sip/libsofia-sip-ua/tport' > make all-am > maCOMPILE tport_type_sctp.lo > tport_type_sctp.c: In function ?tport_sctp_init_socket?: > tport_type_sctp.c:206:10: error: variable ?initmsg? has initializer but > incomplete type > struct sctp_initmsg initmsg = { 0 }; > ^ > tport_type_sctp.c:206:10: error: excess elements in struct initializer > [-Werror] > tport_type_sctp.c:206:10: error: (near initialization for ?initmsg?) > [-Werror] > tport_type_sctp.c:206:23: error: storage size of ?initmsg? isn?t known > struct sctp_initmsg initmsg = { 0 }; > ^ > tport_type_sctp.c:211:36: error: ?SCTP_INITMSG? undeclared (first use in > this function) > if (setsockopt(socket, SOL_SCTP, SCTP_INITMSG, &initmsg, sizeof > initmsg) < 0) > ^ > tport_type_sctp.c:211:36: note: each undeclared identifier is reported > only once for each function it appears in > tport_type_sctp.c:206:23: error: unused variable ?initmsg? > [-Werror=unused-variable] > struct sctp_initmsg initmsg = { 0 }; > ^ > cc1: all warnings being treated as errors > Makefile:1452: recipe for target 'tport_type_sctp.lo' failed > make[9]: *** [tport_type_sctp.lo] Error 1 > make[9]: Leaving directory > '/home/fabryka/local/src/freeswitch/freeswitch481/libs/sofia-sip/libsofia-sip-ua/tport' > Makefile:736: recipe for target 'all' failed > make[8]: *** [all] Error 2 > make[8]: Leaving directory > '/home/fabryka/local/src/freeswitch/freeswitch481/libs/sofia-sip/libsofia-sip-ua/tport' > Makefile:547: recipe for target 'all-recursive' failed > make[7]: *** [all-recursive] Error 1 > make[7]: Leaving directory > '/home/fabryka/local/src/freeswitch/freeswitch481/libs/sofia-sip/libsofia-sip-ua' > Makefile:490: recipe for target 'all-recursive' failed > make[6]: *** [all-recursive] Error 1 > make[6]: Leaving directory > '/home/fabryka/local/src/freeswitch/freeswitch481/libs/sofia-sip' > Makefile:410: recipe for target 'all' failed > make[5]: *** [all] Error 2 > make[5]: Leaving directory > '/home/fabryka/local/src/freeswitch/freeswitch481/libs/sofia-sip' > Makefile:906: recipe for target > '/usr/local/src/freeswitch/freeswitch481/libs/sofia-sip/libsofia-sip-ua/libsofia-sip-ua.la' > failed > make[4]: *** > [/usr/local/src/freeswitch/freeswitch481/libs/sofia-sip/libsofia-sip-ua/libsofia-sip-ua.la] > Error 2 > make[4]: Leaving directory > '/home/fabryka/local/src/freeswitch/freeswitch481/src/mod/endpoints/mod_sofia' > Makefile:577: recipe for target 'mod_sofia-all' failed > make[3]: *** [mod_sofia-all] Error 1 > make[3]: Leaving directory > '/home/fabryka/local/src/freeswitch/freeswitch481/src/mod' > Makefile:491: recipe for target 'all-recursive' failed > make[2]: *** [all-recursive] Error 1 > make[2]: Leaving directory > '/home/fabryka/local/src/freeswitch/freeswitch481/src' > Makefile:2487: recipe for target 'all-recursive' failed > make[1]: *** [all-recursive] Error 1 > make[1]: Leaving directory > '/home/fabryka/local/src/freeswitch/freeswitch481' > Makefile:969: recipe for target 'all' failed > make: *** [all] Error 2 > > > Error amazes me because sofia-sip library from portage ebuild > installed without a problem: > qlist -UCvq sofia-sip > net-libs/sofia-sip-1.12.11 ssl static-libs > http://data.gpo.zugaina.org/gentoo/net-libs/sofia-sip/ > > > The latest version of FreeSWITCH, I have been able to install without a > problem are: > FreeSWITCH version: 1.5.14b+git~20140725T212415Z~c411f8c7a9~64bit (git > c411f8c 2014-07-25 21:24:15Z 64bit) > > > Where can I find the source FreeSWITCH-1.5.x, or alternatively, how to > improve this error? > > > Cheers From mike at jerris.com Wed Dec 10 21:01:55 2014 From: mike at jerris.com (Michael Jerris) Date: Wed, 10 Dec 2014 13:01:55 -0500 Subject: [Freeswitch-users] Mod perl using $session->setVariable inline In-Reply-To: References: <38660852-0B8E-43DD-929A-0D3C8CB2DD5B@jerris.com> Message-ID: That perl script will not run at all until the dialplan parsing is complete. You would need to run the perl script inline (which I am not sure we allow) or run the perl script, then transfer to re-enter the dialplan after it is run. > On Dec 10, 2014, at 12:56 PM, Liam Farr wrote: > > Heres an expanded example; > > > > > > Extension runs the perl script as part of the public dialplan first; > > > > > > > > > > > > > > > > > > Script sets some variables; > > > > $my_value = "123"; > > $session->setVariable('some_var',$my_value); > > > > > > Use that variable later in another extension in the public dialplan as a condition selector; > > > > > > > > > > > > > > > > > > > > > > > > However this will fail, because the perl script "$session->setVariable" run earlier is not inline, so you cant make use of it in the condition () in the second extension that exists in a later part of the public dialplan. Hence I need a way to set inline variables from my perl script. > > > > > > Cheers > > > > Liam Farr > > > > > On 11 December 2014 at 06:28, Michael Jerris > wrote: > If you are in a perl script using session->setVariable, how are you still in the dialplan? This doesn't make sense to me. There should be no need for an inline if you are already in the script. > > > On Dec 10, 2014, at 4:20 AM, Liam Farr > wrote: > > > > Hi, > > > > I need to set a variable using a perl script inline. > > > > Normally I would use $session->setVariable('some_var',$my_value); > > > > However I cant find any options to execute this inline / as an inline action. (I need to evaluate the variable later in the hunting phase of the dialplan hence need it to be an inline action). > > > > Is there a way to do this? > > > > Or is there another way to achieve the same result, e.g. hacking $session->execute("set","some_var=$my_value") to run as an inline action? > > > > Any help would be much appreciated, I've scoured the wiki, confluence and git repo and come up empty. > > > > -- > > Kind Regards > > > > > > Liam Farr > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141210/168eba1f/attachment.html From brian at freeswitch.org Wed Dec 10 21:10:30 2014 From: brian at freeswitch.org (Brian West) Date: Wed, 10 Dec 2014 12:10:30 -0600 Subject: [Freeswitch-users] Mod_sofia build error. In-Reply-To: <422D5329-EA7F-4D72-8D2D-BD9EAF8D7F32@jerris.com> References: <54888763.1020107@gmail.com> <422D5329-EA7F-4D72-8D2D-BD9EAF8D7F32@jerris.com> Message-ID: Oh and file a JIRA! :P On Wed, Dec 10, 2014 at 11:58 AM, Michael Jerris wrote: > This could be related to bugs in gentoo headers related to sctp or due to > issues in the autoconf detection for these. We would be happy to take > patches to correct this issue that don't break other distro/os if you can > provide a patch. > > > > On Dec 10, 2014, at 12:48 PM, Jacekalex wrote: > > > > > > Hi > > > > OS Gentoo Linux x86_64 > > gcc (Gentoo Hardened 4.8.3 p1.1, pie-0.5.9) 4.8.3 > > > > The previous system: > > gcc (Gentoo Hardened 4.7.3-r1 p1.3, pie-0.5.5) 4.7.3 > > > > Freeswitch versions - all 1.4.x > > > > Mod_sofia - sofia-sip error: > > ... > > Making all in tport > > mke[9]: Entering directory > > > '/home/fabryka/local/src/freeswitch/freeswitch481/libs/sofia-sip/libsofia-sip-ua/tport' > > LTake[8]: Entering directory > > > '/home/fabryka/local/src/freeswitch/freeswitch481/libs/sofia-sip/libsofia-sip-ua/tport' > > make all-am > > maCOMPILE tport_type_sctp.lo > > tport_type_sctp.c: In function ?tport_sctp_init_socket?: > > tport_type_sctp.c:206:10: error: variable ?initmsg? has initializer but > > incomplete type > > struct sctp_initmsg initmsg = { 0 }; > > ^ > > tport_type_sctp.c:206:10: error: excess elements in struct initializer > > [-Werror] > > tport_type_sctp.c:206:10: error: (near initialization for ?initmsg?) > > [-Werror] > > tport_type_sctp.c:206:23: error: storage size of ?initmsg? isn?t known > > struct sctp_initmsg initmsg = { 0 }; > > ^ > > tport_type_sctp.c:211:36: error: ?SCTP_INITMSG? undeclared (first use in > > this function) > > if (setsockopt(socket, SOL_SCTP, SCTP_INITMSG, &initmsg, sizeof > > initmsg) < 0) > > ^ > > tport_type_sctp.c:211:36: note: each undeclared identifier is reported > > only once for each function it appears in > > tport_type_sctp.c:206:23: error: unused variable ?initmsg? > > [-Werror=unused-variable] > > struct sctp_initmsg initmsg = { 0 }; > > ^ > > cc1: all warnings being treated as errors > > Makefile:1452: recipe for target 'tport_type_sctp.lo' failed > > make[9]: *** [tport_type_sctp.lo] Error 1 > > make[9]: Leaving directory > > > '/home/fabryka/local/src/freeswitch/freeswitch481/libs/sofia-sip/libsofia-sip-ua/tport' > > Makefile:736: recipe for target 'all' failed > > make[8]: *** [all] Error 2 > > make[8]: Leaving directory > > > '/home/fabryka/local/src/freeswitch/freeswitch481/libs/sofia-sip/libsofia-sip-ua/tport' > > Makefile:547: recipe for target 'all-recursive' failed > > make[7]: *** [all-recursive] Error 1 > > make[7]: Leaving directory > > > '/home/fabryka/local/src/freeswitch/freeswitch481/libs/sofia-sip/libsofia-sip-ua' > > Makefile:490: recipe for target 'all-recursive' failed > > make[6]: *** [all-recursive] Error 1 > > make[6]: Leaving directory > > '/home/fabryka/local/src/freeswitch/freeswitch481/libs/sofia-sip' > > Makefile:410: recipe for target 'all' failed > > make[5]: *** [all] Error 2 > > make[5]: Leaving directory > > '/home/fabryka/local/src/freeswitch/freeswitch481/libs/sofia-sip' > > Makefile:906: recipe for target > > '/usr/local/src/freeswitch/freeswitch481/libs/sofia-sip/libsofia-sip-ua/ > libsofia-sip-ua.la' > > failed > > make[4]: *** > > [/usr/local/src/freeswitch/freeswitch481/libs/sofia-sip/libsofia-sip-ua/ > libsofia-sip-ua.la] > > Error 2 > > make[4]: Leaving directory > > > '/home/fabryka/local/src/freeswitch/freeswitch481/src/mod/endpoints/mod_sofia' > > Makefile:577: recipe for target 'mod_sofia-all' failed > > make[3]: *** [mod_sofia-all] Error 1 > > make[3]: Leaving directory > > '/home/fabryka/local/src/freeswitch/freeswitch481/src/mod' > > Makefile:491: recipe for target 'all-recursive' failed > > make[2]: *** [all-recursive] Error 1 > > make[2]: Leaving directory > > '/home/fabryka/local/src/freeswitch/freeswitch481/src' > > Makefile:2487: recipe for target 'all-recursive' failed > > make[1]: *** [all-recursive] Error 1 > > make[1]: Leaving directory > > '/home/fabryka/local/src/freeswitch/freeswitch481' > > Makefile:969: recipe for target 'all' failed > > make: *** [all] Error 2 > > > > > > Error amazes me because sofia-sip library from portage ebuild > > installed without a problem: > > qlist -UCvq sofia-sip > > net-libs/sofia-sip-1.12.11 ssl static-libs > > http://data.gpo.zugaina.org/gentoo/net-libs/sofia-sip/ > > > > > > The latest version of FreeSWITCH, I have been able to install without a > > problem are: > > FreeSWITCH version: 1.5.14b+git~20140725T212415Z~c411f8c7a9~64bit (git > > c411f8c 2014-07-25 21:24:15Z 64bit) > > > > > > Where can I find the source FreeSWITCH-1.5.x, or alternatively, how to > > improve this error? > > > > > > Cheers > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141210/151aca6c/attachment-0001.html From aqsyounas at gmail.com Wed Dec 10 21:11:06 2014 From: aqsyounas at gmail.com (Aqs Younas) Date: Wed, 10 Dec 2014 23:11:06 +0500 Subject: [Freeswitch-users] Core Dumped Assertion `channel != ((void *)0)' failed Message-ID: Hi,All I am new to freeswitch. Freeswitch crashed out after playing a url with mod_shout. In freeswitch logs i have this. [parse.c:710] error: decode header failed before first valid one, going to read again freeswitch: src/switch_channel.c:909: switch_channel_get_variable_dup: Assertion `channel != ((void *)0)' failed. What can i do to prevent my freeswitch from crashing? Any help would be much appreciated. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141210/c8bcfeab/attachment.html From mike at jerris.com Wed Dec 10 21:14:14 2014 From: mike at jerris.com (Michael Jerris) Date: Wed, 10 Dec 2014 13:14:14 -0500 Subject: [Freeswitch-users] Core Dumped Assertion `channel != ((void *)0)' failed In-Reply-To: References: Message-ID: <86D27DE1-128F-42CC-917E-02C02BD2A16B@jerris.com> https://freeswitch.org/confluence/display/FREESWITCH/Reporting+Bugs+to+JIRA > On Dec 10, 2014, at 1:11 PM, Aqs Younas wrote: > > Hi,All > > I am new to freeswitch. Freeswitch crashed out after playing a url with mod_shout. > > In freeswitch logs i have this. > > [parse.c:710] error: decode header failed before first valid one, going to read again > freeswitch: src/switch_channel.c:909: switch_channel_get_variable_dup: Assertion `channel != ((void *)0)' failed. > > What can i do to prevent my freeswitch from crashing? > > Any help would be much appreciated. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141210/dfab76f1/attachment.html From joelewhite at gmail.com Wed Dec 10 22:27:50 2014 From: joelewhite at gmail.com (Joel White) Date: Wed, 10 Dec 2014 14:27:50 -0500 Subject: [Freeswitch-users] Caller ID - Not Defined when executing dialplan, although Lua script is pulling CID from Database Message-ID: I have gone over the config with a fine tooth comb, it matches another config of a server in which the caller id works fine. What I am seeing however is that in this system the variable is not exported to the dialplan. I may be missing something, and most likely I am. I do have a question though. Is there a way to see what variables are defined for a particular user in the FreeSWITCH console? Here is some output of the Lua script on the server not pushing caller id 2014-12-09 16:54:22.819251 [NOTICE] switch_cpp.cpp:1328 Debug from gen_dir_user_xml.lua, generated XML:
And some output from the Lua script on the server with CID functioning 2014-12-09 21:50:55.458996 [NOTICE] switch_cpp.cpp:1328 Debug from gen_dir_user_xml.lua, generated XML:
Of course I removed any identifiable information, but it looks like the CID is being set. What am I missing here that is not allowing for the variable to be passed to the dialplan? This is what I get when the dialplan executes EXECUTE sofia/internal/26342 at voip.net set(effective_caller_id_number=) 2014-12-09 17:00:24.839237 [DEBUG] mod_dptools.c:1435 sofia/internal/ 26342 at voip.net SET [effective_caller_id_number]=[UNDEF] Kinda strange and I could not find a discrepancy between the dialplan configuration of the working server vs the non-working server Here is the version running on the server that works properly FreeSWITCH Version 1.4.13+git~20141103T195300Z~b942d0faa8~64bit (git b942d0f 2014-11-03 19:53:00Z 64bit) And the version of the server having issue with CID FreeSWITCH Version 1.4.13+git~20141103T195300Z~ b942d0faa8~64bit (git b942d0f 2014-11-03 19:53:00Z 64bit) I used diff and compared both servers conf directory recursively. I could not find a discrepancy in the files aside from Switch name, etc. What am I missing? Could there be anything that that would overwrite the CID variable after it is set by Lua (Generating the user profile)? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141210/32322c04/attachment.html From ahabiba at gmail.com Wed Dec 10 22:58:55 2014 From: ahabiba at gmail.com (Ahmed Habiba) Date: Wed, 10 Dec 2014 22:58:55 +0300 Subject: [Freeswitch-users] SIP Header Manipulation Message-ID: <29A45828-2427-492A-B18E-9260B4880745@gmail.com> Dears, kindly i have a case as below, now I need in case of B number is not registered to modify the sip header ?TO? parameter and add prefix to the A Number, do you have any idea how can I do [A Number] [SIP GW][FreeSWitch][B Number] Your help will be appreciated. Thanks, Ahmed Habiba. From ahabiba at gmail.com Wed Dec 10 23:02:38 2014 From: ahabiba at gmail.com (Ahmed Habiba) Date: Wed, 10 Dec 2014 23:02:38 +0300 Subject: [Freeswitch-users] Twilio SIP trunk config - outbound call rejected Message-ID: <46F54EC6-E3D0-42C0-9AC0-71B48CDE3FA7@gmail.com> Mainly in Twilio if you are using sip trunk you should always call with international code prefix like you should put +1 before number for USA. it looks like the number you are using in local format ? 5101112222 ? it should "+xxx 510111 " From: Adelia C. > To: "freeswitch-users at lists.freeswitch.org " > Date: December 10, 2014 at 7:45:40 PM GMT+3 Reply-To: FreeSWITCH Users Help > Subject: [Freeswitch-users] Twilio SIP trunk config - outbound call rejected Any idea why this INVITE is rejected with 400 Invalid Phone Number by Twilio? The first INVITE asks for Proxy Authentication, that seems to pass and this is rejected. Thank you! INVITE sip:5101112222 at mycompany.pstn.twilio.com SIP/2.0 Via: SIP/2.0/UDP 10.50.1.116:5080;rport;branch=z9hG4bKa8QvU1B6QmeKc Max-Forwards: 69 From: "unknown" >;tag=6UetZr6rDZgHF To: > Call-ID: 1ffda8b7-faaf-1232-8380-a9a65d1e96b1 CSeq: 68751485 INVITE Contact: > Expires: 3600 User-Agent: FreeSWITCH-mod_sofia/1.5.12b~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Proxy-Authorization: Digest username="4152221111", realm="sip.twilio.com ", nonce="f722ae99d0a818bc422baa5d8", cnonce="IA1gJvqvEg6mmXR6WsQ", opaque="e89d2328e07da020d75a", algorithm=MD5, uri="sip:5101112222 at mycompany.pstn.twilio.com ", response="e65d3dea5b1e3b35306cfa121ec2ae4d", qop=auth, nc=00000001 Content-Type: application/sdp Content-Disposition: session Content-Length: 268 X-FS-Support: update_display,send_info Remote-Party-ID: "unknown" >;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1418153731 1418153732 IN IP4 10.50.1.116 s=FreeSWITCH c=IN IP4 10.50.1.116 t=0 0 m=audio 21366 RTP/AVP 0 8 3 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 -- SIP/2.0 400 Invalid phone number To: >;tag=08602519_6772d868_5eef74bb-957c-486e-b261-2a2e63704d73 Via: SIP/2.0/UDP 10.50.1.116:5080;received=10.50.1.116;rport=5080;branch=z9hG4bKa8QvU1B6QmeKc CSeq: 68751485 INVITE Call-ID: 1ffda8b7-faaf-1232-8380-a9a65d1e96b1 From: "unknown" >;tag=6UetZr6rDZgHF Contact: > Content-Length: 0 -- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141210/e73dad38/attachment-0001.html From ssinyagin at gmail.com Thu Dec 11 03:47:10 2014 From: ssinyagin at gmail.com (Stanislav Sinyagin) Date: Thu, 11 Dec 2014 01:47:10 +0100 Subject: [Freeswitch-users] Mod perl using $session->setVariable inline In-Reply-To: References: <38660852-0B8E-43DD-929A-0D3C8CB2DD5B@jerris.com> Message-ID: I'm using mod_perl quite intensively, and I do all the logic and bridging from within the Perl script. It works fine, and you don't really need to return the control back to the dialplan. The only concern is, that if you do "bridge" from the script, your perl script stays in memory and finishes only when the call ends. It should be harmless for a modestly loaded server, and for a massive service you would anyway do much better with an external script talking via ESL. On Wed, Dec 10, 2014 at 6:56 PM, Liam Farr wrote: > Heres an expanded example; > > > > Extension runs the perl script as part of the public dialplan first; > > > > > > > > > > > > > > > Script sets some variables; > > > $my_value = "123"; > > $session->setVariable('some_var',$my_value); > > > > Use that variable later in another extension in the public dialplan as a > condition selector; > > > > > > > > > > > > data="sofia/internal/${destination_number}@1.2.3.4:5060" /> > > > > > > > > However this will fail, because the perl script "$session->setVariable" run > earlier is not inline, so you cant make use of it in the condition > () in the second extension > that exists in a later part of the public dialplan. Hence I need a way to > set inline variables from my perl script. > > > > Cheers > > > Liam Farr > > > > On 11 December 2014 at 06:28, Michael Jerris wrote: >> >> If you are in a perl script using session->setVariable, how are you still >> in the dialplan? This doesn't make sense to me. There should be no need >> for an inline if you are already in the script. >> >> > On Dec 10, 2014, at 4:20 AM, Liam Farr wrote: >> > >> > Hi, >> > >> > I need to set a variable using a perl script inline. >> > >> > Normally I would use $session->setVariable('some_var',$my_value); >> > >> > However I cant find any options to execute this inline / as an inline >> > action. (I need to evaluate the variable later in the hunting phase of the >> > dialplan hence need it to be an inline action). >> > >> > Is there a way to do this? >> > >> > Or is there another way to achieve the same result, e.g. hacking >> > $session->execute("set","some_var=$my_value") to run as an inline action? >> > >> > Any help would be much appreciated, I've scoured the wiki, confluence >> > and git repo and come up empty. >> > >> > -- >> > Kind Regards >> > >> > >> > Liam Farr >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 mike at jerris.com Thu Dec 11 04:27:57 2014 From: mike at jerris.com (Michael Jerris) Date: Wed, 10 Dec 2014 20:27:57 -0500 Subject: [Freeswitch-users] Mod perl using $session->setVariable inline In-Reply-To: References: <38660852-0B8E-43DD-929A-0D3C8CB2DD5B@jerris.com> Message-ID: <2A1755A5-FD0D-45A7-B67E-C2C5F1AC5E29@jerris.com> The issue here is he is expecting the value to be set later in the dialplan. That won't work as he's doing. Another option for both of you is to handle the logic in perl, then set vars from perl for what to bridge to, then have a bridge line in the dp fter perl that uses those vars (not a condition). > On Dec 10, 2014, at 7:47 PM, Stanislav Sinyagin wrote: > > I'm using mod_perl quite intensively, and I do all the logic and > bridging from within the Perl script. It works fine, and you don't > really need to return the control back to the dialplan. > > The only concern is, that if you do "bridge" from the script, your > perl script stays in memory and finishes only when the call ends. It > should be harmless for a modestly loaded server, and for a massive > service you would anyway do much better with an external script > talking via ESL. > > > > > > On Wed, Dec 10, 2014 at 6:56 PM, Liam Farr wrote: >> Heres an expanded example; >> >> >> >> Extension runs the perl script as part of the public dialplan first; >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> Script sets some variables; >> >> >> $my_value = "123"; >> >> $session->setVariable('some_var',$my_value); >> >> >> >> Use that variable later in another extension in the public dialplan as a >> condition selector; >> >> >> >> >> >> >> >> >> >> >> >> > data="sofia/internal/${destination_number}@1.2.3.4:5060" /> >> >> >> >> >> >> >> >> However this will fail, because the perl script "$session->setVariable" run >> earlier is not inline, so you cant make use of it in the condition >> () in the second extension >> that exists in a later part of the public dialplan. Hence I need a way to >> set inline variables from my perl script. >> >> >> >> Cheers >> >> >> Liam Farr >> >> >> >> On 11 December 2014 at 06:28, Michael Jerris wrote: >>> >>> If you are in a perl script using session->setVariable, how are you still >>> in the dialplan? This doesn't make sense to me. There should be no need >>> for an inline if you are already in the script. >>> >>>> On Dec 10, 2014, at 4:20 AM, Liam Farr wrote: >>>> >>>> Hi, >>>> >>>> I need to set a variable using a perl script inline. >>>> >>>> Normally I would use $session->setVariable('some_var',$my_value); >>>> >>>> However I cant find any options to execute this inline / as an inline >>>> action. (I need to evaluate the variable later in the hunting phase of the >>>> dialplan hence need it to be an inline action). >>>> >>>> Is there a way to do this? >>>> >>>> Or is there another way to achieve the same result, e.g. hacking >>>> $session->execute("set","some_var=$my_value") to run as an inline action? >>>> >>>> Any help would be much appreciated, I've scoured the wiki, confluence >>>> and git repo and come up empty. >>>> >>>> -- >>>> Kind Regards >>>> >>>> >>>> Liam Farr >>> From andrew.keil at visytel.com Thu Dec 11 07:11:15 2014 From: andrew.keil at visytel.com (Andrew Keil) Date: Thu, 11 Dec 2014 04:11:15 +0000 Subject: [Freeswitch-users] Re - FreeSWITCH Lua CUSTOM events interrupting session:streamFile playback - is this possible? In-Reply-To: References: Message-ID: Seven Du, Thanks for your response. I eventually found a way by sending via the Event Socket Layer: sendmsg call-command: execute execute-app-name: break Then sending the CUSTOM message and using the con:pop(1,5000) approach to pop the CUSTOM message of the event queue. This approach solves the problem, although I wish I could send an event via ESL that will trigger the setInputCallback(?) function. However I have not found a way to do that as yet. Andrew From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Seven Du Sent: Tuesday, 9 December 2014 10:16 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Re - FreeSWITCH Lua CUSTOM events interrupting session:streamFile playback - is this possible? I believe the playback_terminitors chan var works, or you could try set an input callback and return ?break? when dtmf detected. -- Seven Du http://about.me/dujinfang http://www.dujinfang.com http://www.freeswitch.org.cn Sent with Sparrow On Tuesday, December 9, 2014 at 7:00 AM, Andrew Keil wrote: To FreeSWITCH users, After progressing with Lua inside FreeSWITCH I have a question that I cannot source an answer for within the current FreeSWITCH documentation. I have a requirement for an external application to process an event (via the event socket layer) then return the results back to the application while an audio file is played back in a loop to the caller. From the current documentation this method allows for the event to be sent then the return event is ?consumed? using a polling approach. However the caller is listening to silence. function poll() -- create event and listener local event = freeswitch.Event("CUSTOM", "ping::running?") local con = freeswitch.EventConsumer("CUSTOM", "ping::running!") -- add text ad libitum event:addHeader("hi", "there") -- fire event event:fire() -- and wait for reply but not very long local retevent = con:pop(1, 5000) if retevent then print("reply received") freeswitch.consoleLog("DEBUG", string.format("reply received: %s\n",retevent:getHeader("Result"))) return true end print("no reply") freeswitch.consoleLog("DEBUG", "no reply\n") return false end Some questions: Is there a way to playback audio (eg. session:streamFile(?)) while this takes place, since currently session:streamFile(?) seems to be a blocking function (ie. Finishes when the audio file is played back completely)? Is there a way to use session:setInputCallback(?) to handle an external CUSTOM event being returned (since this would then be able to interrupt the session:streamFile(?) just like it does for DTMF or speech recognition)? If you have a preferred approach to solving this then I am open to your suggestions. I appreciate any assistance that you can give. Kind Regards, Andrew Keil _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141211/7ef0658b/attachment-0001.html From akhilgarg7 at gmail.com Thu Dec 11 12:00:09 2014 From: akhilgarg7 at gmail.com (akhil garg) Date: Thu, 11 Dec 2014 14:30:09 +0530 Subject: [Freeswitch-users] fs_cli hangs Message-ID: running "fs_cli -H 127.0.0.1 -P 8021 -d 7" gives different outputs but no success. ------------------------------------------------------------------------------------------------------------------------------------------------ OUTPUT 1: ------------------------------------------------------------------------------------------------------------------------------------------------ [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is /root/.fs_cli_conf. [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is /etc/fs_cli.conf. [DEBUG] fs_cli.c:1438 main() profile default does not exist using builtin profile [DEBUG] fs_cli.c:1468 main() Using profile internal [127.0.0.1] ------------------------------------------------------------------------------------------------------------------------------------------------ ------------------------------------------------------------------------------------------------------------------------------------------------ OUTPUT 2: ------------------------------------------------------------------------------------------------------------------------------------------------ [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is /root/.fs_cli_conf. [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is /etc/fs_cli.conf. [DEBUG] fs_cli.c:1438 main() profile default does not exist using builtin profile [DEBUG] fs_cli.c:1468 main() Using profile internal [127.0.0.1] [DEBUG] esl.c:1265 esl_recv_event() RECV HEADER [Content-Type] = [auth/request] [DEBUG] esl.c:1437 esl_recv_event() RECV MESSAGE Event-Name: SOCKET_DATA Content-Type: auth/request [DEBUG] esl.c:1465 esl_send() SEND auth ClueCon ------------------------------------------------------------------------------------------------------------------------------------------------ ------------------------------------------------------------------------------------------------------------------------------------------------ OUTPUT 3: ------------------------------------------------------------------------------------------------------------------------------------------------ [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is /root/.fs_cli_conf. [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is /etc/fs_cli.conf. [DEBUG] fs_cli.c:1438 main() profile default does not exist using builtin profile [DEBUG] fs_cli.c:1468 main() Using profile internal [127.0.0.1] [DEBUG] esl.c:1265 esl_recv_event() RECV HEADER [Content-Type] = [auth/request] [DEBUG] esl.c:1437 esl_recv_event() RECV MESSAGE Event-Name: SOCKET_DATA Content-Type: auth/request [DEBUG] esl.c:1465 esl_send() SEND auth ClueCon [DEBUG] esl.c:1265 esl_recv_event() RECV HEADER [Content-Type] = [command/reply] [DEBUG] esl.c:1265 esl_recv_event() RECV HEADER [Reply-Text] = [+OK accepted] [DEBUG] esl.c:1437 esl_recv_event() RECV MESSAGE Event-Name: SOCKET_DATA Content-Type: command/reply Reply-Text: +OK accepted [DEBUG] esl.c:1465 esl_send() SEND log ------------------------------------------------------------------------------------------------------------------------------------------------ -- regards, akhil -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141211/9dba4b04/attachment.html From mbsip at gazeta.pl Thu Dec 11 16:45:17 2014 From: mbsip at gazeta.pl (Maciej Bylica) Date: Thu, 11 Dec 2014 14:45:17 +0100 Subject: [Freeswitch-users] Invalid codec CN is tearing down the call request Message-ID: Hello, I am running FreeSWITCH Version 1.5.14b+git~20140917T231120Z~8f85b5204c~64bit, proxy media mode and heaving problem with some call requests setup. Here is one of them: 2014-12-11 13:38:15.034612 [NOTICE] switch_channel.c:1055 New Channel sofia/outside_1/20049112223344 [] 2014-12-11 13:38:15.034612 [DEBUG] mod_sofia.c:4579 (sofia/outside_1/20049112223344) State Change CS_NEW -> CS_INIT 2014-12-11 13:38:15.034612 [DEBUG] switch_core_session.c:1388 Send signal sofia/outside_1/20049112223344 [BREAK] 2014-12-11 13:38:15.054610 [DEBUG] switch_core_state_machine.c:472 (sofia/outside_1/20049112223344) Running State Change CS_INIT 2014-12-11 13:38:15.054610 [DEBUG] switch_core_state_machine.c:512 (sofia/outside_1/20049112223344) State INIT 2014-12-11 13:38:15.054610 [DEBUG] mod_sofia.c:87 sofia/outside_1/20049112223344 SOFIA INIT 2014-12-11 13:38:15.054610 [DEBUG] switch_core_media.c:7510 sofia/outside_1/20049112223344 Patched SDP --- v=0 o=CiscoSystemsSIP-GW-UserAgent 6016 7716 IN IP4 10.10.10.226 s=SIP Call c=IN IP4 10.10.10.12 t=0 0 m=audio 24782 RTP/AVP 18 19 c=IN IP4 10.10.10.12 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:19 CN/8000 +++ v=0 o=FreeSWITCH 0255539876 0255539877 IN IP4 10.10.10.166 s=FreeSWITCH c=IN IP4 10.10.10.166 t=0 0 m=audio 19690 RTP/AVP 18 19 c=IN IP4 10.10.10.166 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:19 CN/8000 2014-12-11 13:38:15.054610 [DEBUG] sofia_glue.c:1228 sofia/outside_1/20049112223344 sending invite version: 1.5.14b git 8f85b52 2014-09-17 23:11:20Z 64bit Local SDP: v=0 o=FreeSWITCH 0255539876 0255539877 IN IP4 10.10.10.166 s=FreeSWITCH c=IN IP4 10.10.10.166 t=0 0 m=audio 19690 RTP/AVP 18 19 c=IN IP4 10.10.10.166 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:19 CN/8000 sending out SDP exactly as shown above. The other side is responsing with 100, then 180 Ringing wo/SDP then 183 w/SDP 2014-12-11 14:29:12.214608 [DEBUG] sofia.c:6423 Channel sofia/outside_1/20049112223344 entering state [proceeding][183] 2014-12-11 14:29:12.214608 [DEBUG] sofia.c:6433 Remote SDP: v=0 o=Dialogic_SDP 11471558 0 IN IP4 10.10.10.218 s=Dialogic-SIP c=IN IP4 10.10.10.198 t=0 0 m=audio 10024 RTP/AVP 19 18 a=rtpmap:19 CN/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=silenceSupp:off - - - - a=ptime:20 2014-12-11 14:29:12.214608 [DEBUG] switch_core_media.c:7510 sofia/outside_1/20049112223344 Patched SDP --- v=0 o=FreeSWITCH 0351957506 0351957507 IN IP4 10.10.10.166 s=FreeSWITCH c=IN IP4 10.10.10.166 t=0 0 m=audio 29492 RTP/AVP 18 19 c=IN IP4 10.10.10.166 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:19 CN/8000 +++ v=0 o=FreeSWITCH 0351957506 0351957508 IN IP4 10.10.10.166 s=FreeSWITCH c=IN IP4 10.10.10.166 t=0 0 m=audio 29492 RTP/AVP 18 19 c=IN IP4 10.10.10.166 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:19 CN/8000 2014-12-11 14:29:12.214608 [ERR] switch_core_codec.c:651 *Invalid codec CN!* 2014-12-11 14:29:12.214608 [ERR] switch_core_media.c:2294 *Can't load codec?* 2014-12-11 14:29:12.214608 [NOTICE] switch_core_media.c:2295 Hangup sofia/outside_1/20049112223344 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] 2014-12-11 14:29:12.214608 [DEBUG] switch_channel.c:3222 Send signal sofia/outside_1/20049112223344 [KILL] and Freeswitch is terminating the call by using CANCEL. My modules.conf.xml config part looks like following: Is it a problem with CN definition? Could you please lead me where the problem is located? Thanks in advance. Mac. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141211/c5387d53/attachment.html From blasterjr at gmail.com Thu Dec 11 18:26:02 2014 From: blasterjr at gmail.com (Chris Tunbridge) Date: Thu, 11 Dec 2014 08:26:02 -0700 Subject: [Freeswitch-users] Caller ID - Not Defined when executing dialplan, although Lua script is pulling CID from Database In-Reply-To: References: Message-ID: The variable outbound_caller_id_number and outbound_caller_id_name are not related to the caller id on outbound calls. On your outbound dial plan you need to set something like the following This will cause the system to pull the settings from the users profile and use it for the outgoing call. On Wed, Dec 10, 2014 at 12:27 PM, Joel White wrote: > I have gone over the config with a fine tooth comb, it matches another > config of a server in which the caller id works fine. What I am seeing > however is that in this system the variable is not exported to the > dialplan. I may be missing something, and most likely I am. I do have a > question though. Is there a way to see what variables are defined for a > particular user in the FreeSWITCH console? > > > Here is some output of the Lua script on the server not pushing caller id > > 2014-12-09 16:54:22.819251 [NOTICE] switch_cpp.cpp:1328 Debug from > gen_dir_user_xml.lua, generated XML: > > >
> > > > > > > > > > > > > > > > > > >
> > > > And some output from the Lua script on the server with CID functioning > > 2014-12-09 21:50:55.458996 [NOTICE] switch_cpp.cpp:1328 Debug from > gen_dir_user_xml.lua, generated XML: > > >
> > > > > > > > > > > > > > > > >
> > > > Of course I removed any identifiable information, but it looks like the > CID is being set. What am I missing here that is not allowing for the > variable to be passed to the dialplan? > > > This is what I get when the dialplan executes > > EXECUTE sofia/internal/26342 at voip.net > set(effective_caller_id_number=) > 2014-12-09 17:00:24.839237 [DEBUG] mod_dptools.c:1435 sofia/internal/ > 26342 at voip.net SET [effective_caller_id_number]=[UNDEF] > > > Kinda strange and I could not find a discrepancy between the dialplan > configuration of the working server vs the non-working server > > > > Here is the version running on the server that works properly > > FreeSWITCH Version 1.4.13+git~20141103T195300Z~b942d0faa8~64bit (git > b942d0f 2014-11-03 19:53:00Z 64bit) > > And the version of the server having issue with CID > > FreeSWITCH Version 1.4.13+git~20141103T195300Z~ > b942d0faa8~64bit (git b942d0f 2014-11-03 19:53:00Z 64bit) > > > > I used diff and compared both servers conf directory recursively. I could > not find a discrepancy in the files aside from Switch name, etc. > > > What am I missing? > > Could there be anything that that would overwrite the CID variable after > it is set by Lua (Generating the user profile)? > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141211/2425dc39/attachment-0001.html From mike at jerris.com Thu Dec 11 18:32:52 2014 From: mike at jerris.com (Michael Jerris) Date: Thu, 11 Dec 2014 10:32:52 -0500 Subject: [Freeswitch-users] Caller ID - Not Defined when executing dialplan, although Lua script is pulling CID from Database In-Reply-To: References: Message-ID: you should be looking at the origination_caller_id_* vars, not effective > On Dec 11, 2014, at 10:26 AM, Chris Tunbridge wrote: > > The variable outbound_caller_id_number and outbound_caller_id_name are not related to the caller id on outbound calls. > > On your outbound dial plan you need to set something like the following > > > > > This will cause the system to pull the settings from the users profile and use it for the outgoing call. > > On Wed, Dec 10, 2014 at 12:27 PM, Joel White > wrote: > I have gone over the config with a fine tooth comb, it matches another config of a server in which the caller id works fine. What I am seeing however is that in this system the variable is not exported to the dialplan. I may be missing something, and most likely I am. I do have a question though. Is there a way to see what variables are defined for a particular user in the FreeSWITCH console? > > > Here is some output of the Lua script on the server not pushing caller id > > 2014-12-09 16:54:22.819251 [NOTICE] switch_cpp.cpp:1328 Debug from gen_dir_user_xml.lua, generated XML: > > >
> > > > > > > > > > > > > > > > > > >
> > > > And some output from the Lua script on the server with CID functioning > > 2014-12-09 21:50:55.458996 [NOTICE] switch_cpp.cpp:1328 Debug from gen_dir_user_xml.lua, generated XML: > > >
> > > > > > > > > > > > > > > > >
> > > > Of course I removed any identifiable information, but it looks like the CID is being set. What am I missing here that is not allowing for the variable to be passed to the dialplan? > > > This is what I get when the dialplan executes > > EXECUTE sofia/internal/26342 at voip.net > set(effective_caller_id_number=) > 2014-12-09 17:00:24.839237 [DEBUG] mod_dptools.c:1435 sofia/internal/26342 at voip.net SET [effective_caller_id_number]=[UNDEF] > > > Kinda strange and I could not find a discrepancy between the dialplan configuration of the working server vs the non-working server > > > > Here is the version running on the server that works properly > > FreeSWITCH Version 1.4.13+git~20141103T195300Z~b942d0faa8~64bit (git b942d0f 2014-11-03 19:53:00Z 64bit) > > And the version of the server having issue with CID > > FreeSWITCH Version 1.4.13+git~20141103T195300Z~ > b942d0faa8~64bit (git b942d0f 2014-11-03 19:53:00Z 64bit) > > > > I used diff and compared both servers conf directory recursively. I could not find a discrepancy in the files aside from Switch name, etc. > > > What am I missing? > > Could there be anything that that would overwrite the CID variable after it is set by Lua (Generating the user profile)? > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141211/b8140adb/attachment.html From mike at jerris.com Thu Dec 11 18:34:48 2014 From: mike at jerris.com (Michael Jerris) Date: Thu, 11 Dec 2014 10:34:48 -0500 Subject: [Freeswitch-users] Caller ID - Not Defined when executing dialplan, although Lua script is pulling CID from Database In-Reply-To: References: Message-ID: <9312DD37-1999-4B29-A2C0-BC8429C6B866@jerris.com> you should be looking at the origination_caller_id_* vars, not effective > On Dec 11, 2014, at 10:26 AM, Chris Tunbridge > wrote: > > The variable outbound_caller_id_number and outbound_caller_id_name are not related to the caller id on outbound calls. > > On your outbound dial plan you need to set something like the following > > > > > This will cause the system to pull the settings from the users profile and use it for the outgoing call. > > On Wed, Dec 10, 2014 at 12:27 PM, Joel White > wrote: > I have gone over the config with a fine tooth comb, it matches another config of a server in which the caller id works fine. What I am seeing however is that in this system the variable is not exported to the dialplan. I may be missing something, and most likely I am. I do have a question though. Is there a way to see what variables are defined for a particular user in the FreeSWITCH console? > > > Here is some output of the Lua script on the server not pushing caller id > > 2014-12-09 16:54:22.819251 [NOTICE] switch_cpp.cpp:1328 Debug from gen_dir_user_xml.lua, generated XML: > > >
> > > > > > > > > > > > > > > > > > >
> > > > And some output from the Lua script on the server with CID functioning > > 2014-12-09 21:50:55.458996 [NOTICE] switch_cpp.cpp:1328 Debug from gen_dir_user_xml.lua, generated XML: > > >
> > > > > > > > > > > > > > > > >
> > > > Of course I removed any identifiable information, but it looks like the CID is being set. What am I missing here that is not allowing for the variable to be passed to the dialplan? > > > This is what I get when the dialplan executes > > EXECUTE sofia/internal/26342 at voip.net > set(effective_caller_id_number=) > 2014-12-09 17:00:24.839237 [DEBUG] mod_dptools.c:1435 sofia/internal/26342 at voip.net SET [effective_caller_id_number]=[UNDEF] > > > Kinda strange and I could not find a discrepancy between the dialplan configuration of the working server vs the non-working server > > > > Here is the version running on the server that works properly > > FreeSWITCH Version 1.4.13+git~20141103T195300Z~b942d0faa8~64bit (git b942d0f 2014-11-03 19:53:00Z 64bit) > > And the version of the server having issue with CID > > FreeSWITCH Version 1.4.13+git~20141103T195300Z~ > b942d0faa8~64bit (git b942d0f 2014-11-03 19:53:00Z 64bit) > > > > I used diff and compared both servers conf directory recursively. I could not find a discrepancy in the files aside from Switch name, etc. > > > What am I missing? > > Could there be anything that that would overwrite the CID variable after it is set by Lua (Generating the user profile)? > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141211/a033879d/attachment-0001.html From steveayre at gmail.com Thu Dec 11 18:53:36 2014 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 11 Dec 2014 15:53:36 +0000 Subject: [Freeswitch-users] Invalid codec CN is tearing down the call request In-Reply-To: References: Message-ID: Your version is 3 months old. Can you reproduce this on the latest git head? On 11 December 2014 at 13:45, Maciej Bylica wrote: > > Hello, > > I am running FreeSWITCH Version > 1.5.14b+git~20140917T231120Z~8f85b5204c~64bit, proxy media mode and heaving > problem with some call requests setup. > Here is one of them: > > 2014-12-11 13:38:15.034612 [NOTICE] switch_channel.c:1055 New Channel > sofia/outside_1/20049112223344 [] > 2014-12-11 13:38:15.034612 [DEBUG] mod_sofia.c:4579 > (sofia/outside_1/20049112223344) State Change CS_NEW -> CS_INIT > 2014-12-11 13:38:15.034612 [DEBUG] switch_core_session.c:1388 Send signal > sofia/outside_1/20049112223344 [BREAK] > 2014-12-11 13:38:15.054610 [DEBUG] switch_core_state_machine.c:472 > (sofia/outside_1/20049112223344) Running State Change CS_INIT > 2014-12-11 13:38:15.054610 [DEBUG] switch_core_state_machine.c:512 > (sofia/outside_1/20049112223344) State INIT > 2014-12-11 13:38:15.054610 [DEBUG] mod_sofia.c:87 > sofia/outside_1/20049112223344 SOFIA INIT > 2014-12-11 13:38:15.054610 [DEBUG] switch_core_media.c:7510 > sofia/outside_1/20049112223344 Patched SDP > --- > v=0 > o=CiscoSystemsSIP-GW-UserAgent 6016 7716 IN IP4 10.10.10.226 > s=SIP Call > c=IN IP4 10.10.10.12 > t=0 0 > m=audio 24782 RTP/AVP 18 19 > c=IN IP4 10.10.10.12 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=yes > a=rtpmap:19 CN/8000 > > +++ > v=0 > o=FreeSWITCH 0255539876 0255539877 IN IP4 10.10.10.166 > s=FreeSWITCH > c=IN IP4 10.10.10.166 > t=0 0 > m=audio 19690 RTP/AVP 18 19 > c=IN IP4 10.10.10.166 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=yes > a=rtpmap:19 CN/8000 > > 2014-12-11 13:38:15.054610 [DEBUG] sofia_glue.c:1228 > sofia/outside_1/20049112223344 sending invite version: 1.5.14b git 8f85b52 > 2014-09-17 23:11:20Z 64bit > Local SDP: > v=0 > o=FreeSWITCH 0255539876 0255539877 IN IP4 10.10.10.166 > s=FreeSWITCH > c=IN IP4 10.10.10.166 > t=0 0 > m=audio 19690 RTP/AVP 18 19 > c=IN IP4 10.10.10.166 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=yes > a=rtpmap:19 CN/8000 > > sending out SDP exactly as shown above. > The other side is responsing with 100, then 180 Ringing wo/SDP then 183 > w/SDP > > 2014-12-11 14:29:12.214608 [DEBUG] sofia.c:6423 Channel > sofia/outside_1/20049112223344 entering state [proceeding][183] > 2014-12-11 14:29:12.214608 [DEBUG] sofia.c:6433 Remote SDP: > v=0 > o=Dialogic_SDP 11471558 0 IN IP4 10.10.10.218 > s=Dialogic-SIP > c=IN IP4 10.10.10.198 > t=0 0 > m=audio 10024 RTP/AVP 19 18 > a=rtpmap:19 CN/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=yes > a=silenceSupp:off - - - - > a=ptime:20 > > 2014-12-11 14:29:12.214608 [DEBUG] switch_core_media.c:7510 > sofia/outside_1/20049112223344 Patched SDP > --- > v=0 > o=FreeSWITCH 0351957506 0351957507 IN IP4 10.10.10.166 > s=FreeSWITCH > c=IN IP4 10.10.10.166 > t=0 0 > m=audio 29492 RTP/AVP 18 19 > c=IN IP4 10.10.10.166 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=yes > a=rtpmap:19 CN/8000 > > +++ > v=0 > o=FreeSWITCH 0351957506 0351957508 IN IP4 10.10.10.166 > s=FreeSWITCH > c=IN IP4 10.10.10.166 > t=0 0 > m=audio 29492 RTP/AVP 18 19 > c=IN IP4 10.10.10.166 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=yes > a=rtpmap:19 CN/8000 > > 2014-12-11 14:29:12.214608 [ERR] switch_core_codec.c:651 *Invalid codec > CN!* > 2014-12-11 14:29:12.214608 [ERR] switch_core_media.c:2294 *Can't load > codec?* > 2014-12-11 14:29:12.214608 [NOTICE] switch_core_media.c:2295 Hangup > sofia/outside_1/20049112223344 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] > 2014-12-11 14:29:12.214608 [DEBUG] switch_channel.c:3222 Send signal > sofia/outside_1/20049112223344 [KILL] > > and Freeswitch is terminating the call by using CANCEL. > > My modules.conf.xml config part looks like following: > > > > > > > > > > > > > > > Is it a problem with CN definition? > Could you please lead me where the problem is located? > > Thanks in advance. > Mac. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141211/b731dd7f/attachment.html From krice at freeswitch.org Thu Dec 11 19:12:00 2014 From: krice at freeswitch.org (Ken Rice) Date: Thu, 11 Dec 2014 10:12:00 -0600 Subject: [Freeswitch-users] Invalid codec CN is tearing down the call request In-Reply-To: Message-ID: CN is comfort noise... Your cisco is sending it, and freeswitch is passing it thru because you have told it to use proxy media mode... On 12/11/14 9:53 AM, "Steven Ayre" wrote: > Your version is 3 months old. Can you reproduce this on the latest git head? > > On 11 December 2014 at 13:45, Maciej Bylica wrote: >> Hello, >> >> I am running?FreeSWITCH Version >> 1.5.14b+git~20140917T231120Z~8f85b5204c~64bit, proxy media mode and heaving >> problem with some call requests setup. >> Here is one of them: >> >> 2014-12-11 13:38:15.034612 [NOTICE] switch_channel.c:1055 New Channel >> sofia/outside_1/20049112223344 [] >> 2014-12-11 13:38:15.034612 [DEBUG] mod_sofia.c:4579 >> (sofia/outside_1/20049112223344) State Change CS_NEW -> CS_INIT >> 2014-12-11 13:38:15.034612 [DEBUG] switch_core_session.c:1388 Send signal >> sofia/outside_1/20049112223344 [BREAK] >> 2014-12-11 13:38:15.054610 [DEBUG] switch_core_state_machine.c:472 >> (sofia/outside_1/20049112223344) Running State Change CS_INIT >> 2014-12-11 13:38:15.054610 [DEBUG] switch_core_state_machine.c:512 >> (sofia/outside_1/20049112223344) State INIT >> 2014-12-11 13:38:15.054610 [DEBUG] mod_sofia.c:87 >> sofia/outside_1/20049112223344 SOFIA INIT >> 2014-12-11 13:38:15.054610 [DEBUG] switch_core_media.c:7510 >> sofia/outside_1/20049112223344 Patched SDP >> ?--- >> ?v=0 >> ?o=CiscoSystemsSIP-GW-UserAgent 6016 7716 IN IP4 10.10.10.226 >> ?s=SIP Call >> ?c=IN IP4 10.10.10.12 >> ?t=0 0 >> ?m=audio 24782 RTP/AVP 18 19 >> ?c=IN IP4 10.10.10.12 >> ?a=rtpmap:18 G729/8000 >> ?a=fmtp:18 annexb=yes >> ?a=rtpmap:19 CN/8000 >> >> ?+++ >> ?v=0 >> ?o=FreeSWITCH 0255539876 0255539877 IN IP4 10.10.10.166 >> ?s=FreeSWITCH >> ?c=IN IP4 10.10.10.166 >> ?t=0 0 >> ?m=audio 19690 RTP/AVP 18 19 >> ?c=IN IP4 10.10.10.166 >> ?a=rtpmap:18 G729/8000 >> ?a=fmtp:18 annexb=yes >> ?a=rtpmap:19 CN/8000 >> >> ?2014-12-11 13:38:15.054610 [DEBUG] sofia_glue.c:1228 >> sofia/outside_1/20049112223344 sending invite version: 1.5.14b git 8f85b52 >> 2014-09-17 23:11:20Z 64bit >> ?Local SDP: >> ?v=0 >> ?o=FreeSWITCH 0255539876 0255539877 IN IP4 10.10.10.166 >> ?s=FreeSWITCH >> ?c=IN IP4 10.10.10.166 >> ?t=0 0 >> ?m=audio 19690 RTP/AVP 18 19 >> ?c=IN IP4 10.10.10.166 >> ?a=rtpmap:18 G729/8000 >> ?a=fmtp:18 annexb=yes >> ?a=rtpmap:19 CN/8000 >> >> sending out SDP exactly as shown above. >> The other side is responsing with 100, then 180 Ringing wo/SDP then 183 w/SDP >> ? >> ?2014-12-11 14:29:12.214608 [DEBUG] sofia.c:6423 Channel >> sofia/outside_1/20049112223344 entering state [proceeding][183] >> ?2014-12-11 14:29:12.214608 [DEBUG] sofia.c:6433 Remote SDP: >> ?v=0 >> ?o=Dialogic_SDP 11471558 0 IN IP4 10.10.10.218 >> ?s=Dialogic-SIP >> ?c=IN IP4 10.10.10.198 >> ?t=0 0 >> ?m=audio 10024 RTP/AVP 19 18 >> ?a=rtpmap:19 CN/8000 >> ?a=rtpmap:18 G729/8000 >> ?a=fmtp:18 annexb=yes >> ?a=silenceSupp:off - - - - >> ?a=ptime:20 >> >> ?2014-12-11 14:29:12.214608 [DEBUG] switch_core_media.c:7510 >> sofia/outside_1/20049112223344 Patched SDP >> ?--- >> ?v=0 >> ?o=FreeSWITCH 0351957506 0351957507 IN IP4 10.10.10.166 >> ?s=FreeSWITCH >> ?c=IN IP4 10.10.10.166 >> ?t=0 0 >> ?m=audio 29492 RTP/AVP 18 19 >> ?c=IN IP4 10.10.10.166 >> ?a=rtpmap:18 G729/8000 >> ?a=fmtp:18 annexb=yes >> ?a=rtpmap:19 CN/8000 >> >> ?+++ >> ?v=0 >> ?o=FreeSWITCH 0351957506 0351957508 IN IP4 10.10.10.166 >> ?s=FreeSWITCH >> ?c=IN IP4 10.10.10.166 >> ?t=0 0 >> ?m=audio 29492 RTP/AVP 18 19 >> ?c=IN IP4 10.10.10.166 >> ?a=rtpmap:18 G729/8000 >> ?a=fmtp:18 annexb=yes >> ?a=rtpmap:19 CN/8000 >> >> 2014-12-11 14:29:12.214608 [ERR] switch_core_codec.c:651 Invalid codec CN! >> 2014-12-11 14:29:12.214608 [ERR] switch_core_media.c:2294 Can't load codec? >> 2014-12-11 14:29:12.214608 [NOTICE] switch_core_media.c:2295 Hangup >> sofia/outside_1/20049112223344 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] >> 2014-12-11 14:29:12.214608 [DEBUG] switch_channel.c:3222 Send signal >> sofia/outside_1/20049112223344 [KILL] >> >> and Freeswitch is terminating the call by using CANCEL. >> >> My modules.conf.xml config part looks like following: >> ? ? >> ? ? >> ? ? >> ? ? >> ? ? >> ? ? >> ? ? >> ? ? >> ? ? >> ? ? >> ? ? >> ? ? >> ? ? >> >> Is it a problem with CN definition? >> Could you please lead me where the problem is located? >> >> Thanks in advance. >> Mac. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 Twitter: @FreeSWITCH -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141211/20d1f928/attachment-0001.html From mbsip at gazeta.pl Thu Dec 11 19:37:06 2014 From: mbsip at gazeta.pl (Maciej Bylica) Date: Thu, 11 Dec 2014 17:37:06 +0100 Subject: [Freeswitch-users] Invalid codec CN is tearing down the call request In-Reply-To: References: Message-ID: Thanks for prompt reply. Steven, later today i will try to update FS to the latest git and check this out. Ken, i am fine with CN, passing thru is what i really need, but the question is why FS generates such ERRors and how to get out of this. Thanks Mac. 2014-12-11 17:12 GMT+01:00 Ken Rice : > CN is comfort noise... Your cisco is sending it, and freeswitch is > passing it thru because you have told it to use proxy media mode... > > > > On 12/11/14 9:53 AM, "Steven Ayre" wrote: > > Your version is 3 months old. Can you reproduce this on the latest git > head? > > On 11 December 2014 at 13:45, Maciej Bylica wrote: > > Hello, > > I am running FreeSWITCH Version > 1.5.14b+git~20140917T231120Z~8f85b5204c~64bit, proxy media mode and heaving > problem with some call requests setup. > Here is one of them: > > 2014-12-11 13:38:15.034612 [NOTICE] switch_channel.c:1055 New Channel > sofia/outside_1/20049112223344 [] > 2014-12-11 13:38:15.034612 [DEBUG] mod_sofia.c:4579 > (sofia/outside_1/20049112223344) State Change CS_NEW -> CS_INIT > 2014-12-11 13:38:15.034612 [DEBUG] switch_core_session.c:1388 Send signal > sofia/outside_1/20049112223344 [BREAK] > 2014-12-11 13:38:15.054610 [DEBUG] switch_core_state_machine.c:472 > (sofia/outside_1/20049112223344) Running State Change CS_INIT > 2014-12-11 13:38:15.054610 [DEBUG] switch_core_state_machine.c:512 > (sofia/outside_1/20049112223344) State INIT > 2014-12-11 13:38:15.054610 [DEBUG] mod_sofia.c:87 > sofia/outside_1/20049112223344 SOFIA INIT > 2014-12-11 13:38:15.054610 [DEBUG] switch_core_media.c:7510 > sofia/outside_1/20049112223344 Patched SDP > --- > v=0 > o=CiscoSystemsSIP-GW-UserAgent 6016 7716 IN IP4 10.10.10.226 > s=SIP Call > c=IN IP4 10.10.10.12 > t=0 0 > m=audio 24782 RTP/AVP 18 19 > c=IN IP4 10.10.10.12 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=yes > a=rtpmap:19 CN/8000 > > +++ > v=0 > o=FreeSWITCH 0255539876 0255539877 IN IP4 10.10.10.166 > s=FreeSWITCH > c=IN IP4 10.10.10.166 > t=0 0 > m=audio 19690 RTP/AVP 18 19 > c=IN IP4 10.10.10.166 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=yes > a=rtpmap:19 CN/8000 > > 2014-12-11 13:38:15.054610 [DEBUG] sofia_glue.c:1228 > sofia/outside_1/20049112223344 sending invite version: 1.5.14b git 8f85b52 > 2014-09-17 23:11:20Z 64bit > Local SDP: > v=0 > o=FreeSWITCH 0255539876 0255539877 IN IP4 10.10.10.166 > s=FreeSWITCH > c=IN IP4 10.10.10.166 > t=0 0 > m=audio 19690 RTP/AVP 18 19 > c=IN IP4 10.10.10.166 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=yes > a=rtpmap:19 CN/8000 > > sending out SDP exactly as shown above. > The other side is responsing with 100, then 180 Ringing wo/SDP then 183 > w/SDP > > 2014-12-11 14:29:12.214608 [DEBUG] sofia.c:6423 Channel > sofia/outside_1/20049112223344 entering state [proceeding][183] > 2014-12-11 14:29:12.214608 [DEBUG] sofia.c:6433 Remote SDP: > v=0 > o=Dialogic_SDP 11471558 0 IN IP4 10.10.10.218 > s=Dialogic-SIP > c=IN IP4 10.10.10.198 > t=0 0 > m=audio 10024 RTP/AVP 19 18 > a=rtpmap:19 CN/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=yes > a=silenceSupp:off - - - - > a=ptime:20 > > 2014-12-11 14:29:12.214608 [DEBUG] switch_core_media.c:7510 > sofia/outside_1/20049112223344 Patched SDP > --- > v=0 > o=FreeSWITCH 0351957506 0351957507 IN IP4 10.10.10.166 > s=FreeSWITCH > c=IN IP4 10.10.10.166 > t=0 0 > m=audio 29492 RTP/AVP 18 19 > c=IN IP4 10.10.10.166 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=yes > a=rtpmap:19 CN/8000 > > +++ > v=0 > o=FreeSWITCH 0351957506 0351957508 IN IP4 10.10.10.166 > s=FreeSWITCH > c=IN IP4 10.10.10.166 > t=0 0 > m=audio 29492 RTP/AVP 18 19 > c=IN IP4 10.10.10.166 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=yes > a=rtpmap:19 CN/8000 > > 2014-12-11 14:29:12.214608 [ERR] switch_core_codec.c:651 > *Invalid codec CN! *2014-12-11 14:29:12.214608 [ERR] > switch_core_media.c:2294 > *Can't load codec? *2014-12-11 14:29:12.214608 [NOTICE] > switch_core_media.c:2295 Hangup sofia/outside_1/20049112223344 > [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] > 2014-12-11 14:29:12.214608 [DEBUG] switch_channel.c:3222 Send signal > sofia/outside_1/20049112223344 [KILL] > > and Freeswitch is terminating the call by using CANCEL. > > My modules.conf.xml config part looks like following: > > > > > > > > > > > > > > > Is it a problem with CN definition? > Could you please lead me where the problem is located? > > Thanks in advance. > Mac. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 > Twitter: @FreeSWITCH > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141211/dc75c272/attachment.html From anthony.minessale at gmail.com Thu Dec 11 19:41:47 2014 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 11 Dec 2014 10:41:47 -0600 Subject: [Freeswitch-users] Invalid codec CN is tearing down the call request In-Reply-To: References: Message-ID: For years I beg ppl to post issues to jira not the mailing list. if you start typing and keywords like "problem" "not working" "issue" appear in your text, you probably should be filing it to JIRA not here. On Thu, Dec 11, 2014 at 10:37 AM, Maciej Bylica wrote: > Thanks for prompt reply. > > Steven, later today i will try to update FS to the latest git and check > this out. > > Ken, i am fine with CN, passing thru is what i really need, but the > question is why FS generates such ERRors and how to get out of this. > > Thanks > Mac. > > 2014-12-11 17:12 GMT+01:00 Ken Rice : > >> CN is comfort noise... Your cisco is sending it, and freeswitch is >> passing it thru because you have told it to use proxy media mode... >> >> >> >> On 12/11/14 9:53 AM, "Steven Ayre" wrote: >> >> Your version is 3 months old. Can you reproduce this on the latest git >> head? >> >> On 11 December 2014 at 13:45, Maciej Bylica wrote: >> >> Hello, >> >> I am running FreeSWITCH Version >> 1.5.14b+git~20140917T231120Z~8f85b5204c~64bit, proxy media mode and heaving >> problem with some call requests setup. >> Here is one of them: >> >> 2014-12-11 13:38:15.034612 [NOTICE] switch_channel.c:1055 New Channel >> sofia/outside_1/20049112223344 [] >> 2014-12-11 13:38:15.034612 [DEBUG] mod_sofia.c:4579 >> (sofia/outside_1/20049112223344) State Change CS_NEW -> CS_INIT >> 2014-12-11 13:38:15.034612 [DEBUG] switch_core_session.c:1388 Send signal >> sofia/outside_1/20049112223344 [BREAK] >> 2014-12-11 13:38:15.054610 [DEBUG] switch_core_state_machine.c:472 >> (sofia/outside_1/20049112223344) Running State Change CS_INIT >> 2014-12-11 13:38:15.054610 [DEBUG] switch_core_state_machine.c:512 >> (sofia/outside_1/20049112223344) State INIT >> 2014-12-11 13:38:15.054610 [DEBUG] mod_sofia.c:87 >> sofia/outside_1/20049112223344 SOFIA INIT >> 2014-12-11 13:38:15.054610 [DEBUG] switch_core_media.c:7510 >> sofia/outside_1/20049112223344 Patched SDP >> --- >> v=0 >> o=CiscoSystemsSIP-GW-UserAgent 6016 7716 IN IP4 10.10.10.226 >> s=SIP Call >> c=IN IP4 10.10.10.12 >> t=0 0 >> m=audio 24782 RTP/AVP 18 19 >> c=IN IP4 10.10.10.12 >> a=rtpmap:18 G729/8000 >> a=fmtp:18 annexb=yes >> a=rtpmap:19 CN/8000 >> >> +++ >> v=0 >> o=FreeSWITCH 0255539876 0255539877 IN IP4 10.10.10.166 >> s=FreeSWITCH >> c=IN IP4 10.10.10.166 >> t=0 0 >> m=audio 19690 RTP/AVP 18 19 >> c=IN IP4 10.10.10.166 >> a=rtpmap:18 G729/8000 >> a=fmtp:18 annexb=yes >> a=rtpmap:19 CN/8000 >> >> 2014-12-11 13:38:15.054610 [DEBUG] sofia_glue.c:1228 >> sofia/outside_1/20049112223344 sending invite version: 1.5.14b git 8f85b52 >> 2014-09-17 23:11:20Z 64bit >> Local SDP: >> v=0 >> o=FreeSWITCH 0255539876 0255539877 IN IP4 10.10.10.166 >> s=FreeSWITCH >> c=IN IP4 10.10.10.166 >> t=0 0 >> m=audio 19690 RTP/AVP 18 19 >> c=IN IP4 10.10.10.166 >> a=rtpmap:18 G729/8000 >> a=fmtp:18 annexb=yes >> a=rtpmap:19 CN/8000 >> >> sending out SDP exactly as shown above. >> The other side is responsing with 100, then 180 Ringing wo/SDP then 183 >> w/SDP >> >> 2014-12-11 14:29:12.214608 [DEBUG] sofia.c:6423 Channel >> sofia/outside_1/20049112223344 entering state [proceeding][183] >> 2014-12-11 14:29:12.214608 [DEBUG] sofia.c:6433 Remote SDP: >> v=0 >> o=Dialogic_SDP 11471558 0 IN IP4 10.10.10.218 >> s=Dialogic-SIP >> c=IN IP4 10.10.10.198 >> t=0 0 >> m=audio 10024 RTP/AVP 19 18 >> a=rtpmap:19 CN/8000 >> a=rtpmap:18 G729/8000 >> a=fmtp:18 annexb=yes >> a=silenceSupp:off - - - - >> a=ptime:20 >> >> 2014-12-11 14:29:12.214608 [DEBUG] switch_core_media.c:7510 >> sofia/outside_1/20049112223344 Patched SDP >> --- >> v=0 >> o=FreeSWITCH 0351957506 0351957507 IN IP4 10.10.10.166 >> s=FreeSWITCH >> c=IN IP4 10.10.10.166 >> t=0 0 >> m=audio 29492 RTP/AVP 18 19 >> c=IN IP4 10.10.10.166 >> a=rtpmap:18 G729/8000 >> a=fmtp:18 annexb=yes >> a=rtpmap:19 CN/8000 >> >> +++ >> v=0 >> o=FreeSWITCH 0351957506 0351957508 IN IP4 10.10.10.166 >> s=FreeSWITCH >> c=IN IP4 10.10.10.166 >> t=0 0 >> m=audio 29492 RTP/AVP 18 19 >> c=IN IP4 10.10.10.166 >> a=rtpmap:18 G729/8000 >> a=fmtp:18 annexb=yes >> a=rtpmap:19 CN/8000 >> >> 2014-12-11 14:29:12.214608 [ERR] switch_core_codec.c:651 >> *Invalid codec CN! *2014-12-11 14:29:12.214608 [ERR] >> switch_core_media.c:2294 >> *Can't load codec? *2014-12-11 14:29:12.214608 [NOTICE] >> switch_core_media.c:2295 Hangup sofia/outside_1/20049112223344 >> [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] >> 2014-12-11 14:29:12.214608 [DEBUG] switch_channel.c:3222 Send signal >> sofia/outside_1/20049112223344 [KILL] >> >> and Freeswitch is terminating the call by using CANCEL. >> >> My modules.conf.xml config part looks like following: >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> Is it a problem with CN definition? >> Could you please lead me where the problem is located? >> >> Thanks in advance. >> Mac. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> ------------------------------ >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 >> Twitter: @FreeSWITCH >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 ? @anthmfs ? @FreeSWITCH ? ? http://freeswitch.org/ ? http://cluecon.com/ ? http://twitter.com/FreeSWITCH ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ * ClueCon Weekly Development Call ? sip:888 at conference.freeswitch.org ? +19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141211/fe7f528d/attachment-0001.html From sudhansh at gmail.com Thu Dec 11 20:05:55 2014 From: sudhansh at gmail.com (Sudhanshu) Date: Thu, 11 Dec 2014 22:35:55 +0530 Subject: [Freeswitch-users] Immediate disconnect while setting up incoming Message-ID: The setup is a GSM gateway, where I am trying to bridge all incoming calls to the included sims to an external number. The logs are: https://pastebin.freeswitch.org/23714 While trying a call to one of the sims, a ring or two can be heard followed by an immediate busy tone, signifying disconnect, which is corroborated by the logs. Why is this sudden disconnect originating? Can someone kindly help? -- Sudhanshu -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141211/6a13c199/attachment.html From sudhansh at gmail.com Thu Dec 11 20:10:35 2014 From: sudhansh at gmail.com (Sudhanshu) Date: Thu, 11 Dec 2014 22:40:35 +0530 Subject: [Freeswitch-users] fs_cli hangs In-Reply-To: References: Message-ID: Enable debug logging and then check freeswitch.log. What happens when you start freeswitch as anormal process (and not as a daemon)? -- Sudhanshu On Thu, Dec 11, 2014 at 2:30 PM, akhil garg wrote: > running "fs_cli -H 127.0.0.1 -P 8021 -d 7" gives different outputs but no > success. > > > > ------------------------------------------------------------------------------------------------------------------------------------------------ > OUTPUT 1: > > ------------------------------------------------------------------------------------------------------------------------------------------------ > [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is > /root/.fs_cli_conf. > [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is > /etc/fs_cli.conf. > [DEBUG] fs_cli.c:1438 main() profile default does not exist using builtin > profile > [DEBUG] fs_cli.c:1468 main() Using profile internal [127.0.0.1] > > ------------------------------------------------------------------------------------------------------------------------------------------------ > > > > > > ------------------------------------------------------------------------------------------------------------------------------------------------ > OUTPUT 2: > > ------------------------------------------------------------------------------------------------------------------------------------------------ > [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is > /root/.fs_cli_conf. > [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is > /etc/fs_cli.conf. > [DEBUG] fs_cli.c:1438 main() profile default does not exist using builtin > profile > [DEBUG] fs_cli.c:1468 main() Using profile internal [127.0.0.1] > [DEBUG] esl.c:1265 esl_recv_event() RECV HEADER [Content-Type] = > [auth/request] > [DEBUG] esl.c:1437 esl_recv_event() RECV MESSAGE > Event-Name: SOCKET_DATA > Content-Type: auth/request > > > [DEBUG] esl.c:1465 esl_send() SEND > auth ClueCon > > ------------------------------------------------------------------------------------------------------------------------------------------------ > > > > > > ------------------------------------------------------------------------------------------------------------------------------------------------ > OUTPUT 3: > > ------------------------------------------------------------------------------------------------------------------------------------------------ > [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is > /root/.fs_cli_conf. > [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is > /etc/fs_cli.conf. > [DEBUG] fs_cli.c:1438 main() profile default does not exist using builtin > profile > [DEBUG] fs_cli.c:1468 main() Using profile internal [127.0.0.1] > [DEBUG] esl.c:1265 esl_recv_event() RECV HEADER [Content-Type] = > [auth/request] > [DEBUG] esl.c:1437 esl_recv_event() RECV MESSAGE > Event-Name: SOCKET_DATA > Content-Type: auth/request > > > [DEBUG] esl.c:1465 esl_send() SEND > auth ClueCon > > > [DEBUG] esl.c:1265 esl_recv_event() RECV HEADER [Content-Type] = > [command/reply] > [DEBUG] esl.c:1265 esl_recv_event() RECV HEADER [Reply-Text] = [+OK > accepted] > [DEBUG] esl.c:1437 esl_recv_event() RECV MESSAGE > Event-Name: SOCKET_DATA > Content-Type: command/reply > Reply-Text: +OK accepted > > > [DEBUG] esl.c:1465 esl_send() SEND > log > > ------------------------------------------------------------------------------------------------------------------------------------------------ > > > > -- > regards, > akhil > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141211/d2c2f0f7/attachment.html From idokan at gmail.com Thu Dec 11 20:32:30 2014 From: idokan at gmail.com (ik) Date: Thu, 11 Dec 2014 19:32:30 +0200 Subject: [Freeswitch-users] Controling playback recording position Message-ID: Hello, How can I control the position of the recording that is played ? For example, go 10 seconds forward or backward, like with Asterisk's ControlPlayback cmd. Thanks, Ido -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141211/c5b20bb3/attachment.html From mbsip at gazeta.pl Thu Dec 11 20:47:18 2014 From: mbsip at gazeta.pl (Maciej Bylica) Date: Thu, 11 Dec 2014 18:47:18 +0100 Subject: [Freeswitch-users] Invalid codec CN is tearing down the call request In-Reply-To: References: Message-ID: Sure i'll try with jira Mac. 2014-12-11 17:41 GMT+01:00 Anthony Minessale : > For years I beg ppl to post issues to jira not the mailing list. > if you start typing and keywords like "problem" "not working" "issue" > appear in your text, you probably should be filing it to JIRA not here. > > > On Thu, Dec 11, 2014 at 10:37 AM, Maciej Bylica wrote: > >> Thanks for prompt reply. >> >> Steven, later today i will try to update FS to the latest git and check >> this out. >> >> Ken, i am fine with CN, passing thru is what i really need, but the >> question is why FS generates such ERRors and how to get out of this. >> >> Thanks >> Mac. >> >> 2014-12-11 17:12 GMT+01:00 Ken Rice : >> >>> CN is comfort noise... Your cisco is sending it, and freeswitch is >>> passing it thru because you have told it to use proxy media mode... >>> >>> >>> >>> On 12/11/14 9:53 AM, "Steven Ayre" wrote: >>> >>> Your version is 3 months old. Can you reproduce this on the latest git >>> head? >>> >>> On 11 December 2014 at 13:45, Maciej Bylica wrote: >>> >>> Hello, >>> >>> I am running FreeSWITCH Version >>> 1.5.14b+git~20140917T231120Z~8f85b5204c~64bit, proxy media mode and heaving >>> problem with some call requests setup. >>> Here is one of them: >>> >>> 2014-12-11 13:38:15.034612 [NOTICE] switch_channel.c:1055 New Channel >>> sofia/outside_1/20049112223344 [] >>> 2014-12-11 13:38:15.034612 [DEBUG] mod_sofia.c:4579 >>> (sofia/outside_1/20049112223344) State Change CS_NEW -> CS_INIT >>> 2014-12-11 13:38:15.034612 [DEBUG] switch_core_session.c:1388 Send >>> signal sofia/outside_1/20049112223344 [BREAK] >>> 2014-12-11 13:38:15.054610 [DEBUG] switch_core_state_machine.c:472 >>> (sofia/outside_1/20049112223344) Running State Change CS_INIT >>> 2014-12-11 13:38:15.054610 [DEBUG] switch_core_state_machine.c:512 >>> (sofia/outside_1/20049112223344) State INIT >>> 2014-12-11 13:38:15.054610 [DEBUG] mod_sofia.c:87 >>> sofia/outside_1/20049112223344 SOFIA INIT >>> 2014-12-11 13:38:15.054610 [DEBUG] switch_core_media.c:7510 >>> sofia/outside_1/20049112223344 Patched SDP >>> --- >>> v=0 >>> o=CiscoSystemsSIP-GW-UserAgent 6016 7716 IN IP4 10.10.10.226 >>> s=SIP Call >>> c=IN IP4 10.10.10.12 >>> t=0 0 >>> m=audio 24782 RTP/AVP 18 19 >>> c=IN IP4 10.10.10.12 >>> a=rtpmap:18 G729/8000 >>> a=fmtp:18 annexb=yes >>> a=rtpmap:19 CN/8000 >>> >>> +++ >>> v=0 >>> o=FreeSWITCH 0255539876 0255539877 IN IP4 10.10.10.166 >>> s=FreeSWITCH >>> c=IN IP4 10.10.10.166 >>> t=0 0 >>> m=audio 19690 RTP/AVP 18 19 >>> c=IN IP4 10.10.10.166 >>> a=rtpmap:18 G729/8000 >>> a=fmtp:18 annexb=yes >>> a=rtpmap:19 CN/8000 >>> >>> 2014-12-11 13:38:15.054610 [DEBUG] sofia_glue.c:1228 >>> sofia/outside_1/20049112223344 sending invite version: 1.5.14b git 8f85b52 >>> 2014-09-17 23:11:20Z 64bit >>> Local SDP: >>> v=0 >>> o=FreeSWITCH 0255539876 0255539877 IN IP4 10.10.10.166 >>> s=FreeSWITCH >>> c=IN IP4 10.10.10.166 >>> t=0 0 >>> m=audio 19690 RTP/AVP 18 19 >>> c=IN IP4 10.10.10.166 >>> a=rtpmap:18 G729/8000 >>> a=fmtp:18 annexb=yes >>> a=rtpmap:19 CN/8000 >>> >>> sending out SDP exactly as shown above. >>> The other side is responsing with 100, then 180 Ringing wo/SDP then 183 >>> w/SDP >>> >>> 2014-12-11 14:29:12.214608 [DEBUG] sofia.c:6423 Channel >>> sofia/outside_1/20049112223344 entering state [proceeding][183] >>> 2014-12-11 14:29:12.214608 [DEBUG] sofia.c:6433 Remote SDP: >>> v=0 >>> o=Dialogic_SDP 11471558 0 IN IP4 10.10.10.218 >>> s=Dialogic-SIP >>> c=IN IP4 10.10.10.198 >>> t=0 0 >>> m=audio 10024 RTP/AVP 19 18 >>> a=rtpmap:19 CN/8000 >>> a=rtpmap:18 G729/8000 >>> a=fmtp:18 annexb=yes >>> a=silenceSupp:off - - - - >>> a=ptime:20 >>> >>> 2014-12-11 14:29:12.214608 [DEBUG] switch_core_media.c:7510 >>> sofia/outside_1/20049112223344 Patched SDP >>> --- >>> v=0 >>> o=FreeSWITCH 0351957506 0351957507 IN IP4 10.10.10.166 >>> s=FreeSWITCH >>> c=IN IP4 10.10.10.166 >>> t=0 0 >>> m=audio 29492 RTP/AVP 18 19 >>> c=IN IP4 10.10.10.166 >>> a=rtpmap:18 G729/8000 >>> a=fmtp:18 annexb=yes >>> a=rtpmap:19 CN/8000 >>> >>> +++ >>> v=0 >>> o=FreeSWITCH 0351957506 0351957508 IN IP4 10.10.10.166 >>> s=FreeSWITCH >>> c=IN IP4 10.10.10.166 >>> t=0 0 >>> m=audio 29492 RTP/AVP 18 19 >>> c=IN IP4 10.10.10.166 >>> a=rtpmap:18 G729/8000 >>> a=fmtp:18 annexb=yes >>> a=rtpmap:19 CN/8000 >>> >>> 2014-12-11 14:29:12.214608 [ERR] switch_core_codec.c:651 >>> *Invalid codec CN! *2014-12-11 14:29:12.214608 [ERR] >>> switch_core_media.c:2294 >>> *Can't load codec? *2014-12-11 14:29:12.214608 [NOTICE] >>> switch_core_media.c:2295 Hangup sofia/outside_1/20049112223344 >>> [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] >>> 2014-12-11 14:29:12.214608 [DEBUG] switch_channel.c:3222 Send signal >>> sofia/outside_1/20049112223344 [KILL] >>> >>> and Freeswitch is terminating the call by using CANCEL. >>> >>> My modules.conf.xml config part looks like following: >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> Is it a problem with CN definition? >>> Could you please lead me where the problem is located? >>> >>> Thanks in advance. >>> Mac. >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> ------------------------------ >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 >>> Twitter: @FreeSWITCH >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 ? @anthmfs ? @FreeSWITCH ? > > ? http://freeswitch.org/ ? http://cluecon.com/ ? > http://twitter.com/FreeSWITCH > ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ > * > > ClueCon Weekly Development Call > ? sip:888 at conference.freeswitch.org ? +19193869900 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141211/20d4dbe7/attachment-0001.html From brian at freeswitch.org Thu Dec 11 21:07:23 2014 From: brian at freeswitch.org (Brian West) Date: Thu, 11 Dec 2014 12:07:23 -0600 Subject: [Freeswitch-users] Invalid codec CN is tearing down the call request In-Reply-To: References: Message-ID: Funny, payload 13 is CN, which is 0x13 in base16 or 19 in base10, Please file a JIRA, I would need to see a packet capture for this one. On Thu, Dec 11, 2014 at 11:47 AM, Maciej Bylica wrote: > Sure i'll try with jira > > Mac. > > 2014-12-11 17:41 GMT+01:00 Anthony Minessale > : > >> For years I beg ppl to post issues to jira not the mailing list. >> if you start typing and keywords like "problem" "not working" "issue" >> appear in your text, you probably should be filing it to JIRA not here. >> >> >> On Thu, Dec 11, 2014 at 10:37 AM, Maciej Bylica wrote: >> >>> Thanks for prompt reply. >>> >>> Steven, later today i will try to update FS to the latest git and check >>> this out. >>> >>> Ken, i am fine with CN, passing thru is what i really need, but the >>> question is why FS generates such ERRors and how to get out of this. >>> >>> Thanks >>> Mac. >>> >>> 2014-12-11 17:12 GMT+01:00 Ken Rice : >>> >>>> CN is comfort noise... Your cisco is sending it, and freeswitch is >>>> passing it thru because you have told it to use proxy media mode... >>>> >>>> >>>> >>>> On 12/11/14 9:53 AM, "Steven Ayre" wrote: >>>> >>>> Your version is 3 months old. Can you reproduce this on the latest git >>>> head? >>>> >>>> On 11 December 2014 at 13:45, Maciej Bylica wrote: >>>> >>>> Hello, >>>> >>>> I am running FreeSWITCH Version >>>> 1.5.14b+git~20140917T231120Z~8f85b5204c~64bit, proxy media mode and heaving >>>> problem with some call requests setup. >>>> Here is one of them: >>>> >>>> 2014-12-11 13:38:15.034612 [NOTICE] switch_channel.c:1055 New Channel >>>> sofia/outside_1/20049112223344 [] >>>> 2014-12-11 13:38:15.034612 [DEBUG] mod_sofia.c:4579 >>>> (sofia/outside_1/20049112223344) State Change CS_NEW -> CS_INIT >>>> 2014-12-11 13:38:15.034612 [DEBUG] switch_core_session.c:1388 Send >>>> signal sofia/outside_1/20049112223344 [BREAK] >>>> 2014-12-11 13:38:15.054610 [DEBUG] switch_core_state_machine.c:472 >>>> (sofia/outside_1/20049112223344) Running State Change CS_INIT >>>> 2014-12-11 13:38:15.054610 [DEBUG] switch_core_state_machine.c:512 >>>> (sofia/outside_1/20049112223344) State INIT >>>> 2014-12-11 13:38:15.054610 [DEBUG] mod_sofia.c:87 >>>> sofia/outside_1/20049112223344 SOFIA INIT >>>> 2014-12-11 13:38:15.054610 [DEBUG] switch_core_media.c:7510 >>>> sofia/outside_1/20049112223344 Patched SDP >>>> --- >>>> v=0 >>>> o=CiscoSystemsSIP-GW-UserAgent 6016 7716 IN IP4 10.10.10.226 >>>> s=SIP Call >>>> c=IN IP4 10.10.10.12 >>>> t=0 0 >>>> m=audio 24782 RTP/AVP 18 19 >>>> c=IN IP4 10.10.10.12 >>>> a=rtpmap:18 G729/8000 >>>> a=fmtp:18 annexb=yes >>>> a=rtpmap:19 CN/8000 >>>> >>>> +++ >>>> v=0 >>>> o=FreeSWITCH 0255539876 0255539877 IN IP4 10.10.10.166 >>>> s=FreeSWITCH >>>> c=IN IP4 10.10.10.166 >>>> t=0 0 >>>> m=audio 19690 RTP/AVP 18 19 >>>> c=IN IP4 10.10.10.166 >>>> a=rtpmap:18 G729/8000 >>>> a=fmtp:18 annexb=yes >>>> a=rtpmap:19 CN/8000 >>>> >>>> 2014-12-11 13:38:15.054610 [DEBUG] sofia_glue.c:1228 >>>> sofia/outside_1/20049112223344 sending invite version: 1.5.14b git 8f85b52 >>>> 2014-09-17 23:11:20Z 64bit >>>> Local SDP: >>>> v=0 >>>> o=FreeSWITCH 0255539876 0255539877 IN IP4 10.10.10.166 >>>> s=FreeSWITCH >>>> c=IN IP4 10.10.10.166 >>>> t=0 0 >>>> m=audio 19690 RTP/AVP 18 19 >>>> c=IN IP4 10.10.10.166 >>>> a=rtpmap:18 G729/8000 >>>> a=fmtp:18 annexb=yes >>>> a=rtpmap:19 CN/8000 >>>> >>>> sending out SDP exactly as shown above. >>>> The other side is responsing with 100, then 180 Ringing wo/SDP then 183 >>>> w/SDP >>>> >>>> 2014-12-11 14:29:12.214608 [DEBUG] sofia.c:6423 Channel >>>> sofia/outside_1/20049112223344 entering state [proceeding][183] >>>> 2014-12-11 14:29:12.214608 [DEBUG] sofia.c:6433 Remote SDP: >>>> v=0 >>>> o=Dialogic_SDP 11471558 0 IN IP4 10.10.10.218 >>>> s=Dialogic-SIP >>>> c=IN IP4 10.10.10.198 >>>> t=0 0 >>>> m=audio 10024 RTP/AVP 19 18 >>>> a=rtpmap:19 CN/8000 >>>> a=rtpmap:18 G729/8000 >>>> a=fmtp:18 annexb=yes >>>> a=silenceSupp:off - - - - >>>> a=ptime:20 >>>> >>>> 2014-12-11 14:29:12.214608 [DEBUG] switch_core_media.c:7510 >>>> sofia/outside_1/20049112223344 Patched SDP >>>> --- >>>> v=0 >>>> o=FreeSWITCH 0351957506 0351957507 IN IP4 10.10.10.166 >>>> s=FreeSWITCH >>>> c=IN IP4 10.10.10.166 >>>> t=0 0 >>>> m=audio 29492 RTP/AVP 18 19 >>>> c=IN IP4 10.10.10.166 >>>> a=rtpmap:18 G729/8000 >>>> a=fmtp:18 annexb=yes >>>> a=rtpmap:19 CN/8000 >>>> >>>> +++ >>>> v=0 >>>> o=FreeSWITCH 0351957506 0351957508 IN IP4 10.10.10.166 >>>> s=FreeSWITCH >>>> c=IN IP4 10.10.10.166 >>>> t=0 0 >>>> m=audio 29492 RTP/AVP 18 19 >>>> c=IN IP4 10.10.10.166 >>>> a=rtpmap:18 G729/8000 >>>> a=fmtp:18 annexb=yes >>>> a=rtpmap:19 CN/8000 >>>> >>>> 2014-12-11 14:29:12.214608 [ERR] switch_core_codec.c:651 >>>> *Invalid codec CN! *2014-12-11 14:29:12.214608 [ERR] >>>> switch_core_media.c:2294 >>>> *Can't load codec? *2014-12-11 14:29:12.214608 [NOTICE] >>>> switch_core_media.c:2295 Hangup sofia/outside_1/20049112223344 >>>> [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] >>>> 2014-12-11 14:29:12.214608 [DEBUG] switch_channel.c:3222 Send signal >>>> sofia/outside_1/20049112223344 [KILL] >>>> >>>> and Freeswitch is terminating the call by using CANCEL. >>>> >>>> My modules.conf.xml config part looks like following: >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> Is it a problem with CN definition? >>>> Could you please lead me where the problem is located? >>>> >>>> Thanks in advance. >>>> Mac. >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> ------------------------------ >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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 >>>> Twitter: @FreeSWITCH >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 ? @anthmfs ? @FreeSWITCH ? >> >> ? http://freeswitch.org/ ? http://cluecon.com/ ? >> http://twitter.com/FreeSWITCH >> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ >> * >> >> ClueCon Weekly Development Call >> ? sip:888 at conference.freeswitch.org ? +19193869900 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141211/3f8b55f3/attachment-0001.html From brian at freeswitch.org Thu Dec 11 21:08:13 2014 From: brian at freeswitch.org (Brian West) Date: Thu, 11 Dec 2014 12:08:13 -0600 Subject: [Freeswitch-users] Controling playback recording position In-Reply-To: References: Message-ID: See scripts/lua/callback.lua, javascript has a similar method too. On Thu, Dec 11, 2014 at 11:32 AM, ik wrote: > Hello, > > How can I control the position of the recording that is played ? > For example, go 10 seconds forward or backward, like with Asterisk's > ControlPlayback cmd. > > Thanks, > Ido > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141211/2badecd1/attachment.html From arsen.semionov at gmail.com Thu Dec 11 21:44:13 2014 From: arsen.semionov at gmail.com (Arsen) Date: Thu, 11 Dec 2014 20:44:13 +0200 Subject: [Freeswitch-users] Upgrade 1.4.14 TLS issue Message-ID: Hi guys, I've upgraded my FS from 1.2 to 1.4.14 without changes in the config files and TLS stopped working. I can't register any more (503 service unavailable error). I don't see any logs in the console, but with ngrep I can see that the traffic hits the server on port 5061. tls_version = sslv23 if I change to tlsv1 I can register but outgoing calls don't work Any ideas? -- Regards, Arsen. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141211/ff7df494/attachment.html From blasterjr at gmail.com Thu Dec 11 21:51:59 2014 From: blasterjr at gmail.com (Chris Tunbridge) Date: Thu, 11 Dec 2014 11:51:59 -0700 Subject: [Freeswitch-users] Caller ID - Not Defined when executing dialplan, although Lua script is pulling CID from Database In-Reply-To: <9312DD37-1999-4B29-A2C0-BC8429C6B866@jerris.com> References: <9312DD37-1999-4B29-A2C0-BC8429C6B866@jerris.com> Message-ID: Errr yeah, sorry Michael is right they're origination not effective :P On Thu, Dec 11, 2014 at 8:34 AM, Michael Jerris wrote: > you should be looking at the origination_caller_id_* vars, not effective > > On Dec 11, 2014, at 10:26 AM, Chris Tunbridge wrote: > > The variable outbound_caller_id_number and outbound_caller_id_name are not > related to the caller id on outbound calls. > > On your outbound dial plan you need to set something like the following > > data="effective_caller_id_name=${user_data(${username}@${domain_name} var > outbound_caller_id_name)}"/> > data="effective_caller_id_number=${user_data(${username}@${domain_name} > var outbound_caller_id_number)}"/> > > This will cause the system to pull the settings from the users profile and > use it for the outgoing call. > > On Wed, Dec 10, 2014 at 12:27 PM, Joel White wrote: > >> I have gone over the config with a fine tooth comb, it matches another >> config of a server in which the caller id works fine. What I am seeing >> however is that in this system the variable is not exported to the >> dialplan. I may be missing something, and most likely I am. I do have a >> question though. Is there a way to see what variables are defined for a >> particular user in the FreeSWITCH console? >> >> >> Here is some output of the Lua script on the server not pushing caller id >> >> 2014-12-09 16:54:22.819251 [NOTICE] switch_cpp.cpp:1328 Debug from >> gen_dir_user_xml.lua, generated XML: >> >> >>
>> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >>
>> >> >> >> And some output from the Lua script on the server with CID functioning >> >> 2014-12-09 21:50:55.458996 [NOTICE] switch_cpp.cpp:1328 Debug from >> gen_dir_user_xml.lua, generated XML: >> >> >>
>> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >>
>> >> >> >> Of course I removed any identifiable information, but it looks like the >> CID is being set. What am I missing here that is not allowing for the >> variable to be passed to the dialplan? >> >> >> This is what I get when the dialplan executes >> >> EXECUTE sofia/internal/26342 at voip.net >> set(effective_caller_id_number=) >> 2014-12-09 17:00:24.839237 [DEBUG] mod_dptools.c:1435 sofia/internal/ >> 26342 at voip.net SET [effective_caller_id_number]=[UNDEF] >> >> >> Kinda strange and I could not find a discrepancy between the dialplan >> configuration of the working server vs the non-working server >> >> >> >> Here is the version running on the server that works properly >> >> FreeSWITCH Version 1.4.13+git~20141103T195300Z~b942d0faa8~64bit (git >> b942d0f 2014-11-03 19:53:00Z 64bit) >> >> And the version of the server having issue with CID >> >> FreeSWITCH Version 1.4.13+git~20141103T195300Z~ >> b942d0faa8~64bit (git b942d0f 2014-11-03 19:53:00Z 64bit) >> >> >> >> I used diff and compared both servers conf directory recursively. I >> could not find a discrepancy in the files aside from Switch name, etc. >> >> >> What am I missing? >> >> Could there be anything that that would overwrite the CID variable after >> it is set by Lua (Generating the user profile)? >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141211/72b3534d/attachment.html From blasterjr at gmail.com Thu Dec 11 21:54:03 2014 From: blasterjr at gmail.com (Chris Tunbridge) Date: Thu, 11 Dec 2014 11:54:03 -0700 Subject: [Freeswitch-users] fs_cli hangs In-Reply-To: References: Message-ID: Can you do a pastebin of the contents of your conf/autoload_configs/event_socket.conf.xml On Thu, Dec 11, 2014 at 10:10 AM, Sudhanshu wrote: > Enable debug logging and then check freeswitch.log. > What happens when you start freeswitch as anormal process (and not as a > daemon)? > > -- > Sudhanshu > > On Thu, Dec 11, 2014 at 2:30 PM, akhil garg wrote: > >> running "fs_cli -H 127.0.0.1 -P 8021 -d 7" gives different outputs but no >> success. >> >> >> >> ------------------------------------------------------------------------------------------------------------------------------------------------ >> OUTPUT 1: >> >> ------------------------------------------------------------------------------------------------------------------------------------------------ >> [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is >> /root/.fs_cli_conf. >> [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is >> /etc/fs_cli.conf. >> [DEBUG] fs_cli.c:1438 main() profile default does not exist using builtin >> profile >> [DEBUG] fs_cli.c:1468 main() Using profile internal [127.0.0.1] >> >> ------------------------------------------------------------------------------------------------------------------------------------------------ >> >> >> >> >> >> ------------------------------------------------------------------------------------------------------------------------------------------------ >> OUTPUT 2: >> >> ------------------------------------------------------------------------------------------------------------------------------------------------ >> [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is >> /root/.fs_cli_conf. >> [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is >> /etc/fs_cli.conf. >> [DEBUG] fs_cli.c:1438 main() profile default does not exist using builtin >> profile >> [DEBUG] fs_cli.c:1468 main() Using profile internal [127.0.0.1] >> [DEBUG] esl.c:1265 esl_recv_event() RECV HEADER [Content-Type] = >> [auth/request] >> [DEBUG] esl.c:1437 esl_recv_event() RECV MESSAGE >> Event-Name: SOCKET_DATA >> Content-Type: auth/request >> >> >> [DEBUG] esl.c:1465 esl_send() SEND >> auth ClueCon >> >> ------------------------------------------------------------------------------------------------------------------------------------------------ >> >> >> >> >> >> ------------------------------------------------------------------------------------------------------------------------------------------------ >> OUTPUT 3: >> >> ------------------------------------------------------------------------------------------------------------------------------------------------ >> [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is >> /root/.fs_cli_conf. >> [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is >> /etc/fs_cli.conf. >> [DEBUG] fs_cli.c:1438 main() profile default does not exist using builtin >> profile >> [DEBUG] fs_cli.c:1468 main() Using profile internal [127.0.0.1] >> [DEBUG] esl.c:1265 esl_recv_event() RECV HEADER [Content-Type] = >> [auth/request] >> [DEBUG] esl.c:1437 esl_recv_event() RECV MESSAGE >> Event-Name: SOCKET_DATA >> Content-Type: auth/request >> >> >> [DEBUG] esl.c:1465 esl_send() SEND >> auth ClueCon >> >> >> [DEBUG] esl.c:1265 esl_recv_event() RECV HEADER [Content-Type] = >> [command/reply] >> [DEBUG] esl.c:1265 esl_recv_event() RECV HEADER [Reply-Text] = [+OK >> accepted] >> [DEBUG] esl.c:1437 esl_recv_event() RECV MESSAGE >> Event-Name: SOCKET_DATA >> Content-Type: command/reply >> Reply-Text: +OK accepted >> >> >> [DEBUG] esl.c:1465 esl_send() SEND >> log >> >> ------------------------------------------------------------------------------------------------------------------------------------------------ >> >> >> >> -- >> regards, >> akhil >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141211/1418cf0b/attachment-0001.html From my.post at hotmail.com Thu Dec 11 22:25:42 2014 From: my.post at hotmail.com (Post) Date: Thu, 11 Dec 2014 22:25:42 +0300 Subject: [Freeswitch-users] How to play custom ringback using outbound ES. Message-ID: Hi everyone, This is probably very simple, but I am stuck. I am using simple perl script to listen for incoming connection from freeswitch. I am willing to learn how to send a custom ringback to legA user. So far I managed to play some wav file in the following way: $con->execute("pre_answer", "", $uuid); $con->execute("playback", "somefile.wav", $uuid); So this works ok. Now I am trying to send a custom ringback (here are my guesses): 1) $con->execute("set", "ringback=\$\${ru-ring}", $uuid); $con->execute("pre_answer", "", $uuid); doesn't work (hear silence on legA user, see 183 with SDP in fs_cli sofia trace) 2) $con->execute("set", "ringback=\$\${ru-ring}", $uuid); $con->execute("ring_ready", "", $uuid); doesn't work either (hear RBT, pre-programmed in endpoint, see 180 w/o SDP in fs_cli sofia trace). Please give some advice. Regards, Pavel. From krice at freeswitch.org Thu Dec 11 22:37:10 2014 From: krice at freeswitch.org (Ken Rice) Date: Thu, 11 Dec 2014 13:37:10 -0600 Subject: [Freeswitch-users] Upgrade 1.4.14 TLS issue In-Reply-To: Message-ID: This could be because anything below TLSv1 is no longer support due to the various attacks and downgrade attacks (like POODLE etc) If you are using something below TLSv1 you might as well just turn it off On 12/11/14 12:44 PM, "Arsen" wrote: > Hi guys, > > I've upgraded my FS from 1.2 to 1.4.14 without changes in the config files and > TLS stopped working. I can't register any more (503 service unavailable > error). > I don't see any logs in the console, but with ngrep I can see that the traffic > hits the server on port 5061. > tls_version = ?sslv23 > if I change to tlsv1 I can register but outgoing calls don't work > Any ideas? -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141211/4cd74509/attachment.html From arsen.semionov at gmail.com Thu Dec 11 22:47:54 2014 From: arsen.semionov at gmail.com (Arsen) Date: Thu, 11 Dec 2014 21:47:54 +0200 Subject: [Freeswitch-users] Upgrade 1.4.14 TLS issue In-Reply-To: References: Message-ID: Thanks for responce. Sip profile is configured with SSLv23, I've just recreated ssl certificates still getting 503 Service unavailable while registering... On Thursday, December 11, 2014, Ken Rice wrote: > This could be because anything below TLSv1 is no longer support due to > the various attacks and downgrade attacks (like POODLE etc) > > If you are using something below TLSv1 you might as well just turn it off > > > On 12/11/14 12:44 PM, "Arsen" wrote: > > Hi guys, > > I've upgraded my FS from 1.2 to 1.4.14 without changes in the config files > and TLS stopped working. I can't register any more (503 service unavailable > error). > I don't see any logs in the console, but with ngrep I can see that the > traffic hits the server on port 5061. > tls_version = sslv23 > if I change to tlsv1 I can register but outgoing calls don't work > Any ideas? > > > -- > Ken > > > > *http://www.FreeSWITCH.org > http://www.ClueCon.com http://www.OSTAG.org > *irc.freenode.net #freeswitch > Twitter: @FreeSWITCH > > -- Regards, Arsen. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141211/f93f3ba2/attachment.html From pjintheusa at gmail.com Thu Dec 11 23:35:15 2014 From: pjintheusa at gmail.com (Phillip Jones) Date: Thu, 11 Dec 2014 15:35:15 -0500 Subject: [Freeswitch-users] Dialplan - Lua in-line expansion Message-ID: I am going crazy - please tell me what I am missing: I have a dialplan: I have a lua script: freeswitch.consoleLog("NOTICE","util_get10digits.lua \n") freeswitch.consoleLog("NOTICE","Argument passed [" .. argv[1] .."]\n") local result = "none" *result = string.sub(argv[1], -10)* freeswitch.consoleLog("NOTICE", "Result: [" .. result .."]\n") *return result* I run the dialplan: Dialplan: sofia/Carrier/2154791697 at 64.158.162.74 Action set(VUM_SAY_CID=${lua(util_get10digits.lua ${VUM_Outgoing_CallerID_Number})}) I 2014-12-11 15:20:21.795274 [NOTICE] switch_cpp.cpp:1328 util_get10digits.lua 2014-12-11 15:20:21.795274 [NOTICE] switch_cpp.cpp:1328 Argument passed [+12154791697] *2014-12-11 15:20:21.795274 [NOTICE] switch_cpp.cpp:1328 Result: [2154791697]* 2014-12-11 15:20:21.795274 [DEBUG] switch_cpp.cpp:1075 sofia/Carrier/ 2154791697 at 64.158.162.74 destroy/unlink session from object EXECUTE sofia/Carrier/2154791697 at 64.158.162.74 set(VUM_SAY_CID=) *2014-12-11 15:20:21.795274 [DEBUG] mod_dptools.c:1435 sofia/Carrier/2154791697 at 64.158.162.74 <2154791697 at 64.158.162.74> SET [VUM_SAY_CID]=[UNDEF]* The lua script runs correctly - how ever the result is not returned the VUM_SAY_CID variable in the dialplan The work around is to set the session variable in the script. But I wondering what I am missing. Is "return result" not how you return a result from Lua? Thanks for any help. Phil -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141211/2c0817f8/attachment.html From chad at apartmentlines.com Thu Dec 11 23:46:56 2014 From: chad at apartmentlines.com (Chad Phillips) Date: Thu, 11 Dec 2014 13:46:56 -0700 Subject: [Freeswitch-users] Dialplan - Lua in-line expansion In-Reply-To: References: Message-ID: how about setting the channel var directly in the script? session:setVariable("VUM_SAY_CID", result); although for what you?re doing, i really think there?s a cleaner way to do it straight from the dialplan XML... On Thursday, December 11, 2014 at 1:35 PM, Phillip Jones wrote: > I am going crazy - please tell me what I am missing: > > I have a dialplan: > > > > > I have a lua script: > > > freeswitch.consoleLog("NOTICE","util_get10digits.lua \n") > freeswitch.consoleLog("NOTICE","Argument passed [" .. argv[1] .."]\n") > local result = "none" > result = string.sub(argv[1], -10) > freeswitch.consoleLog("NOTICE", "Result: [" .. result .."]\n") > return result > > > I run the dialplan: > > Dialplan: sofia/Carrier/2154791697 at 64.158.162.74 (mailto:2154791697 at 64.158.162.74) Action set(VUM_SAY_CID=${lua(util_get10digits.lua ${VUM_Outgoing_CallerID_Number})}) I > 2014-12-11 15:20:21.795274 [NOTICE] switch_cpp.cpp:1328 util_get10digits.lua > 2014-12-11 15:20:21.795274 [NOTICE] switch_cpp.cpp:1328 Argument passed [+12154791697] > 2014-12-11 15:20:21.795274 [NOTICE] switch_cpp.cpp:1328 Result: [2154791697] > 2014-12-11 15:20:21.795274 [DEBUG] switch_cpp.cpp:1075 sofia/Carrier/2154791697 at 64.158.162.74 (mailto:2154791697 at 64.158.162.74) destroy/unlink session from object > EXECUTE sofia/Carrier/2154791697 at 64.158.162.74 (mailto:2154791697 at 64.158.162.74) set(VUM_SAY_CID=) > 2014-12-11 15:20:21.795274 [DEBUG] mod_dptools.c:1435 sofia/Carrier/2154791697 at 64.158.162.74 (mailto:2154791697 at 64.158.162.74) SET [VUM_SAY_CID]=[UNDEF] > > > > The lua script runs correctly - how ever the result is not returned the VUM_SAY_CID variable in the dialplan > > > The work around is to set the session variable in the script. But I wondering what I am missing. Is "return result" not how you return a result from Lua? > > > Thanks for any help. > > > Phil > _________________________________________________________________________ > 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://confluence.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/20141211/5a40bd02/attachment.html From joseph at onsip.com Thu Dec 11 23:43:08 2014 From: joseph at onsip.com (Joseph Frazier) Date: Thu, 11 Dec 2014 15:43:08 -0500 Subject: [Freeswitch-users] Incomplete UPDATEs after Attended Transfer In-Reply-To: References: Message-ID: I tried the same scenario with proxy-refer, and found that it's still the case that no UPDATE is sent to the Referee, but also that the UPDATE sent to the refer target is incorrect. I reported the latter bug to JIRA as FS-7080: https://freeswitch.org/jira/browse/FS-7080 I'm still unclear on whether I should expect the Referee to receive an UPDATE. I did see in mod_sofia.c that the nua_update calls are guarded by User-Agent checks, none of which appear to match the Grandstream: https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/src/mod/endpoints/mod_sofia/mod_sofia.c#1630 Perhaps that is the problem? Thanks, Joseph On Tue, Dec 9, 2014 at 9:13 PM, Joseph Frazier wrote: > The Referee is a "Grandstream GXV3275 1.0.2.15". For context, here's where > it OKs the INVITE from the first call: > > > https://gist.github.com/joseph-onsip/de90a66afdf37994c334#file-gistfile1-txt-L1813-L1838 > > Thanks, > Joseph > > On Tue, Dec 9, 2014 at 6:54 PM, Brian West wrote: > >> What device is the Referee? >> >> On Tue, Dec 9, 2014 at 4:30 PM, Joseph Frazier wrote: >> >>> Hi everyone, >>> >>> I'm trying to do attended transfers on my FS 1.5.14 box and have the >>> endpoints receive UPDATE requests indicating that the identity of the other >>> leg has changed. I installed the v1.5.14 tag from source and uncommented >>> the following line of sip_profiles/internal.xml: >>> >>> >>> >>> Other than the above line, my configuration is completely vanilla. >>> >>> I have three extensions registered: 1000, 1007, 1006. My scenario is: >>> >>> - 1000 calls 1007 >>> - 1000 calls 1006 >>> - 1000 attended transfers 1007 to 1006 >>> >>> Here is my fs_cli with the following debugging options: >>> >>> sofia global siptrace on >>>> sofia loglevel all 9 >>>> sofia tracelevel alert >>>> console loglevel debug >>>> fsctl debug_level 10 >>> >>> >>> https://gist.github.com/joseph-onsip/de90a66afdf37994c334 >>> >>> After receiving the REFER, FS sends an UPDATE to the refer target, but >>> not to the referee. Is there a configuration option I can set in order to >>> have an UPDATE sent to the referee as well? >>> >>> Thanks, >>> Joseph >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> >> *Brian West* >> brian at freeswitch.org >> >> >> *Twitter: @FreeSWITCH , @briankwest* >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141211/39c4be17/attachment-0001.html From pjintheusa at gmail.com Fri Dec 12 00:32:05 2014 From: pjintheusa at gmail.com (Phillip Jones) Date: Thu, 11 Dec 2014 16:32:05 -0500 Subject: [Freeswitch-users] Dialplan - Lua in-line expansion In-Reply-To: References: Message-ID: Yeah - thanks - I realize I can do that - but it just drives me crazy I can not get in-line expansion to work. I can't see what I am missing. On Thu, Dec 11, 2014 at 3:46 PM, Chad Phillips wrote: > how about setting the channel var directly in the script? > > session:setVariable("VUM_SAY_CID", result); > > although for what you?re doing, i really think there?s a cleaner way to do > it straight from the dialplan XML... > > On Thursday, December 11, 2014 at 1:35 PM, Phillip Jones wrote: > > I am going crazy - please tell me what I am missing: > > I have a dialplan: > > > > > I have a lua script: > > > freeswitch.consoleLog("NOTICE","util_get10digits.lua \n") > freeswitch.consoleLog("NOTICE","Argument passed [" .. argv[1] .."]\n") > local result = "none" > *result = string.sub(argv[1], -10)* > freeswitch.consoleLog("NOTICE", "Result: [" .. result .."]\n") > *return result* > > > I run the dialplan: > > Dialplan: sofia/Carrier/2154791697 at 64.158.162.74 Action > set(VUM_SAY_CID=${lua(util_get10digits.lua > ${VUM_Outgoing_CallerID_Number})}) I > 2014-12-11 15:20:21.795274 [NOTICE] switch_cpp.cpp:1328 > util_get10digits.lua > 2014-12-11 15:20:21.795274 [NOTICE] switch_cpp.cpp:1328 Argument passed > [+12154791697] > *2014-12-11 15:20:21.795274 [NOTICE] switch_cpp.cpp:1328 Result: > [2154791697 <%5B2154791697>]* > 2014-12-11 15:20:21.795274 [DEBUG] switch_cpp.cpp:1075 sofia/Carrier/ > 2154791697 at 64.158.162.74 destroy/unlink session from object > EXECUTE sofia/Carrier/2154791697 at 64.158.162.74 set(VUM_SAY_CID=) > *2014-12-11 15:20:21.795274 [DEBUG] mod_dptools.c:1435 > sofia/Carrier/2154791697 at 64.158.162.74 <2154791697 at 64.158.162.74> SET > [VUM_SAY_CID]=[UNDEF]* > > > The lua script runs correctly - how ever the result is not returned the > VUM_SAY_CID variable in the dialplan > > > The work around is to set the session variable in the script. But I > wondering what I am missing. Is "return result" not how you return a result > from Lua? > > > Thanks for any help. > > > Phil > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141211/1a943cd3/attachment.html From mbsip at gazeta.pl Fri Dec 12 02:03:58 2014 From: mbsip at gazeta.pl (Maciej Bylica) Date: Fri, 12 Dec 2014 00:03:58 +0100 Subject: [Freeswitch-users] Invalid codec CN is tearing down the call request In-Reply-To: References: Message-ID: Steven, i have the same issue on FreeSWITCH Version 1.5.15b+git~20141211T210353Z~be0e09bd7f I am about to rise this ticket on jira. Thanks anyway. BR Mac. 2014-12-11 18:47 GMT+01:00 Maciej Bylica : > Sure i'll try with jira > > Mac. > > 2014-12-11 17:41 GMT+01:00 Anthony Minessale > : > >> For years I beg ppl to post issues to jira not the mailing list. >> if you start typing and keywords like "problem" "not working" "issue" >> appear in your text, you probably should be filing it to JIRA not here. >> >> >> On Thu, Dec 11, 2014 at 10:37 AM, Maciej Bylica wrote: >> >>> Thanks for prompt reply. >>> >>> Steven, later today i will try to update FS to the latest git and check >>> this out. >>> >>> Ken, i am fine with CN, passing thru is what i really need, but the >>> question is why FS generates such ERRors and how to get out of this. >>> >>> Thanks >>> Mac. >>> >>> 2014-12-11 17:12 GMT+01:00 Ken Rice : >>> >>>> CN is comfort noise... Your cisco is sending it, and freeswitch is >>>> passing it thru because you have told it to use proxy media mode... >>>> >>>> >>>> >>>> On 12/11/14 9:53 AM, "Steven Ayre" wrote: >>>> >>>> Your version is 3 months old. Can you reproduce this on the latest git >>>> head? >>>> >>>> On 11 December 2014 at 13:45, Maciej Bylica wrote: >>>> >>>> Hello, >>>> >>>> I am running FreeSWITCH Version >>>> 1.5.14b+git~20140917T231120Z~8f85b5204c~64bit, proxy media mode and heaving >>>> problem with some call requests setup. >>>> Here is one of them: >>>> >>>> 2014-12-11 13:38:15.034612 [NOTICE] switch_channel.c:1055 New Channel >>>> sofia/outside_1/20049112223344 [] >>>> 2014-12-11 13:38:15.034612 [DEBUG] mod_sofia.c:4579 >>>> (sofia/outside_1/20049112223344) State Change CS_NEW -> CS_INIT >>>> 2014-12-11 13:38:15.034612 [DEBUG] switch_core_session.c:1388 Send >>>> signal sofia/outside_1/20049112223344 [BREAK] >>>> 2014-12-11 13:38:15.054610 [DEBUG] switch_core_state_machine.c:472 >>>> (sofia/outside_1/20049112223344) Running State Change CS_INIT >>>> 2014-12-11 13:38:15.054610 [DEBUG] switch_core_state_machine.c:512 >>>> (sofia/outside_1/20049112223344) State INIT >>>> 2014-12-11 13:38:15.054610 [DEBUG] mod_sofia.c:87 >>>> sofia/outside_1/20049112223344 SOFIA INIT >>>> 2014-12-11 13:38:15.054610 [DEBUG] switch_core_media.c:7510 >>>> sofia/outside_1/20049112223344 Patched SDP >>>> --- >>>> v=0 >>>> o=CiscoSystemsSIP-GW-UserAgent 6016 7716 IN IP4 10.10.10.226 >>>> s=SIP Call >>>> c=IN IP4 10.10.10.12 >>>> t=0 0 >>>> m=audio 24782 RTP/AVP 18 19 >>>> c=IN IP4 10.10.10.12 >>>> a=rtpmap:18 G729/8000 >>>> a=fmtp:18 annexb=yes >>>> a=rtpmap:19 CN/8000 >>>> >>>> +++ >>>> v=0 >>>> o=FreeSWITCH 0255539876 0255539877 IN IP4 10.10.10.166 >>>> s=FreeSWITCH >>>> c=IN IP4 10.10.10.166 >>>> t=0 0 >>>> m=audio 19690 RTP/AVP 18 19 >>>> c=IN IP4 10.10.10.166 >>>> a=rtpmap:18 G729/8000 >>>> a=fmtp:18 annexb=yes >>>> a=rtpmap:19 CN/8000 >>>> >>>> 2014-12-11 13:38:15.054610 [DEBUG] sofia_glue.c:1228 >>>> sofia/outside_1/20049112223344 sending invite version: 1.5.14b git 8f85b52 >>>> 2014-09-17 23:11:20Z 64bit >>>> Local SDP: >>>> v=0 >>>> o=FreeSWITCH 0255539876 0255539877 IN IP4 10.10.10.166 >>>> s=FreeSWITCH >>>> c=IN IP4 10.10.10.166 >>>> t=0 0 >>>> m=audio 19690 RTP/AVP 18 19 >>>> c=IN IP4 10.10.10.166 >>>> a=rtpmap:18 G729/8000 >>>> a=fmtp:18 annexb=yes >>>> a=rtpmap:19 CN/8000 >>>> >>>> sending out SDP exactly as shown above. >>>> The other side is responsing with 100, then 180 Ringing wo/SDP then 183 >>>> w/SDP >>>> >>>> 2014-12-11 14:29:12.214608 [DEBUG] sofia.c:6423 Channel >>>> sofia/outside_1/20049112223344 entering state [proceeding][183] >>>> 2014-12-11 14:29:12.214608 [DEBUG] sofia.c:6433 Remote SDP: >>>> v=0 >>>> o=Dialogic_SDP 11471558 0 IN IP4 10.10.10.218 >>>> s=Dialogic-SIP >>>> c=IN IP4 10.10.10.198 >>>> t=0 0 >>>> m=audio 10024 RTP/AVP 19 18 >>>> a=rtpmap:19 CN/8000 >>>> a=rtpmap:18 G729/8000 >>>> a=fmtp:18 annexb=yes >>>> a=silenceSupp:off - - - - >>>> a=ptime:20 >>>> >>>> 2014-12-11 14:29:12.214608 [DEBUG] switch_core_media.c:7510 >>>> sofia/outside_1/20049112223344 Patched SDP >>>> --- >>>> v=0 >>>> o=FreeSWITCH 0351957506 0351957507 IN IP4 10.10.10.166 >>>> s=FreeSWITCH >>>> c=IN IP4 10.10.10.166 >>>> t=0 0 >>>> m=audio 29492 RTP/AVP 18 19 >>>> c=IN IP4 10.10.10.166 >>>> a=rtpmap:18 G729/8000 >>>> a=fmtp:18 annexb=yes >>>> a=rtpmap:19 CN/8000 >>>> >>>> +++ >>>> v=0 >>>> o=FreeSWITCH 0351957506 0351957508 IN IP4 10.10.10.166 >>>> s=FreeSWITCH >>>> c=IN IP4 10.10.10.166 >>>> t=0 0 >>>> m=audio 29492 RTP/AVP 18 19 >>>> c=IN IP4 10.10.10.166 >>>> a=rtpmap:18 G729/8000 >>>> a=fmtp:18 annexb=yes >>>> a=rtpmap:19 CN/8000 >>>> >>>> 2014-12-11 14:29:12.214608 [ERR] switch_core_codec.c:651 >>>> *Invalid codec CN! *2014-12-11 14:29:12.214608 [ERR] >>>> switch_core_media.c:2294 >>>> *Can't load codec? *2014-12-11 14:29:12.214608 [NOTICE] >>>> switch_core_media.c:2295 Hangup sofia/outside_1/20049112223344 >>>> [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] >>>> 2014-12-11 14:29:12.214608 [DEBUG] switch_channel.c:3222 Send signal >>>> sofia/outside_1/20049112223344 [KILL] >>>> >>>> and Freeswitch is terminating the call by using CANCEL. >>>> >>>> My modules.conf.xml config part looks like following: >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> Is it a problem with CN definition? >>>> Could you please lead me where the problem is located? >>>> >>>> Thanks in advance. >>>> Mac. >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> ------------------------------ >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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 >>>> Twitter: @FreeSWITCH >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 ? @anthmfs ? @FreeSWITCH ? >> >> ? http://freeswitch.org/ ? http://cluecon.com/ ? >> http://twitter.com/FreeSWITCH >> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ >> * >> >> ClueCon Weekly Development Call >> ? sip:888 at conference.freeswitch.org ? +19193869900 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141212/2de41e7a/attachment-0001.html From mbsip at gazeta.pl Fri Dec 12 03:00:32 2014 From: mbsip at gazeta.pl (Maciej Bylica) Date: Fri, 12 Dec 2014 01:00:32 +0100 Subject: [Freeswitch-users] Invalid codec CN is tearing down the call request In-Reply-To: References: Message-ID: I will gather packet capture and file JIRA. Thank You Maciej. Funny, payload 13 is CN, which is 0x13 in base16 or 19 in base10, Please > file a JIRA, I would need to see a packet capture for this one. > > > On Thu, Dec 11, 2014 at 11:47 AM, Maciej Bylica > wrote: > >> Sure i'll try with jira >> >> Mac. >> >> 2014-12-11 17:41 GMT+01:00 Anthony Minessale > >: >> >>> For years I beg ppl to post issues to jira not the mailing list. >>> if you start typing and keywords like "problem" "not working" "issue" >>> appear in your text, you probably should be filing it to JIRA not here. >>> >>> >>> On Thu, Dec 11, 2014 at 10:37 AM, Maciej Bylica >> > wrote: >>> >>>> Thanks for prompt reply. >>>> >>>> Steven, later today i will try to update FS to the latest git and check >>>> this out. >>>> >>>> Ken, i am fine with CN, passing thru is what i really need, but the >>>> question is why FS generates such ERRors and how to get out of this. >>>> >>>> Thanks >>>> Mac. >>>> >>>> 2014-12-11 17:12 GMT+01:00 Ken Rice >>> >: >>>> >>>>> CN is comfort noise... Your cisco is sending it, and freeswitch is >>>>> passing it thru because you have told it to use proxy media mode... >>>>> >>>>> >>>>> >>>>> On 12/11/14 9:53 AM, "Steven Ayre" wrote: >>>>> >>>>> Your version is 3 months old. Can you reproduce this on the latest git >>>>> head? >>>>> >>>>> On 11 December 2014 at 13:45, Maciej Bylica wrote: >>>>> >>>>> Hello, >>>>> >>>>> I am running FreeSWITCH Version >>>>> 1.5.14b+git~20140917T231120Z~8f85b5204c~64bit, proxy media mode and heaving >>>>> problem with some call requests setup. >>>>> Here is one of them: >>>>> >>>>> 2014-12-11 13:38:15.034612 [NOTICE] switch_channel.c:1055 New Channel >>>>> sofia/outside_1/20049112223344 [] >>>>> 2014-12-11 13:38:15.034612 [DEBUG] mod_sofia.c:4579 >>>>> (sofia/outside_1/20049112223344) State Change CS_NEW -> CS_INIT >>>>> 2014-12-11 13:38:15.034612 [DEBUG] switch_core_session.c:1388 Send >>>>> signal sofia/outside_1/20049112223344 [BREAK] >>>>> 2014-12-11 13:38:15.054610 [DEBUG] switch_core_state_machine.c:472 >>>>> (sofia/outside_1/20049112223344) Running State Change CS_INIT >>>>> 2014-12-11 13:38:15.054610 [DEBUG] switch_core_state_machine.c:512 >>>>> (sofia/outside_1/20049112223344) State INIT >>>>> 2014-12-11 13:38:15.054610 [DEBUG] mod_sofia.c:87 >>>>> sofia/outside_1/20049112223344 SOFIA INIT >>>>> 2014-12-11 13:38:15.054610 [DEBUG] switch_core_media.c:7510 >>>>> sofia/outside_1/20049112223344 Patched SDP >>>>> --- >>>>> v=0 >>>>> o=CiscoSystemsSIP-GW-UserAgent 6016 7716 IN IP4 10.10.10.226 >>>>> s=SIP Call >>>>> c=IN IP4 10.10.10.12 >>>>> t=0 0 >>>>> m=audio 24782 RTP/AVP 18 19 >>>>> c=IN IP4 10.10.10.12 >>>>> a=rtpmap:18 G729/8000 >>>>> a=fmtp:18 annexb=yes >>>>> a=rtpmap:19 CN/8000 >>>>> >>>>> +++ >>>>> v=0 >>>>> o=FreeSWITCH 0255539876 0255539877 IN IP4 10.10.10.166 >>>>> s=FreeSWITCH >>>>> c=IN IP4 10.10.10.166 >>>>> t=0 0 >>>>> m=audio 19690 RTP/AVP 18 19 >>>>> c=IN IP4 10.10.10.166 >>>>> a=rtpmap:18 G729/8000 >>>>> a=fmtp:18 annexb=yes >>>>> a=rtpmap:19 CN/8000 >>>>> >>>>> 2014-12-11 13:38:15.054610 [DEBUG] sofia_glue.c:1228 >>>>> sofia/outside_1/20049112223344 sending invite version: 1.5.14b git 8f85b52 >>>>> 2014-09-17 23:11:20Z 64bit >>>>> Local SDP: >>>>> v=0 >>>>> o=FreeSWITCH 0255539876 0255539877 IN IP4 10.10.10.166 >>>>> s=FreeSWITCH >>>>> c=IN IP4 10.10.10.166 >>>>> t=0 0 >>>>> m=audio 19690 RTP/AVP 18 19 >>>>> c=IN IP4 10.10.10.166 >>>>> a=rtpmap:18 G729/8000 >>>>> a=fmtp:18 annexb=yes >>>>> a=rtpmap:19 CN/8000 >>>>> >>>>> sending out SDP exactly as shown above. >>>>> The other side is responsing with 100, then 180 Ringing wo/SDP then >>>>> 183 w/SDP >>>>> >>>>> 2014-12-11 14:29:12.214608 [DEBUG] sofia.c:6423 Channel >>>>> sofia/outside_1/20049112223344 entering state [proceeding][183] >>>>> 2014-12-11 14:29:12.214608 [DEBUG] sofia.c:6433 Remote SDP: >>>>> v=0 >>>>> o=Dialogic_SDP 11471558 0 IN IP4 10.10.10.218 >>>>> s=Dialogic-SIP >>>>> c=IN IP4 10.10.10.198 >>>>> t=0 0 >>>>> m=audio 10024 RTP/AVP 19 18 >>>>> a=rtpmap:19 CN/8000 >>>>> a=rtpmap:18 G729/8000 >>>>> a=fmtp:18 annexb=yes >>>>> a=silenceSupp:off - - - - >>>>> a=ptime:20 >>>>> >>>>> 2014-12-11 14:29:12.214608 [DEBUG] switch_core_media.c:7510 >>>>> sofia/outside_1/20049112223344 Patched SDP >>>>> --- >>>>> v=0 >>>>> o=FreeSWITCH 0351957506 0351957507 IN IP4 10.10.10.166 >>>>> s=FreeSWITCH >>>>> c=IN IP4 10.10.10.166 >>>>> t=0 0 >>>>> m=audio 29492 RTP/AVP 18 19 >>>>> c=IN IP4 10.10.10.166 >>>>> a=rtpmap:18 G729/8000 >>>>> a=fmtp:18 annexb=yes >>>>> a=rtpmap:19 CN/8000 >>>>> >>>>> +++ >>>>> v=0 >>>>> o=FreeSWITCH 0351957506 0351957508 IN IP4 10.10.10.166 >>>>> s=FreeSWITCH >>>>> c=IN IP4 10.10.10.166 >>>>> t=0 0 >>>>> m=audio 29492 RTP/AVP 18 19 >>>>> c=IN IP4 10.10.10.166 >>>>> a=rtpmap:18 G729/8000 >>>>> a=fmtp:18 annexb=yes >>>>> a=rtpmap:19 CN/8000 >>>>> >>>>> 2014-12-11 14:29:12.214608 [ERR] switch_core_codec.c:651 >>>>> *Invalid codec CN! *2014-12-11 14:29:12.214608 [ERR] >>>>> switch_core_media.c:2294 >>>>> *Can't load codec? *2014-12-11 14:29:12.214608 [NOTICE] >>>>> switch_core_media.c:2295 Hangup sofia/outside_1/20049112223344 >>>>> [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] >>>>> 2014-12-11 14:29:12.214608 [DEBUG] switch_channel.c:3222 Send signal >>>>> sofia/outside_1/20049112223344 [KILL] >>>>> >>>>> and Freeswitch is terminating the call by using CANCEL. >>>>> >>>>> My modules.conf.xml config part looks like following: >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> Is it a problem with CN definition? >>>>> Could you please lead me where the problem is located? >>>>> >>>>> Thanks in advance. >>>>> Mac. >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> ------------------------------ >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at 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 >>>>> Twitter: @FreeSWITCH >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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 ? @anthmfs ? @FreeSWITCH ? >>> >>> ? http://freeswitch.org/ ? http://cluecon.com/ ? >>> http://twitter.com/FreeSWITCH >>> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ >>> * >>> >>> ClueCon Weekly Development Call >>> ? sip:888 at conference.freeswitch.org >>> ? >>> +19193869900 >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > > *Brian West* > brian at freeswitch.org > > > > *Twitter: @FreeSWITCH , @briankwest* > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141212/b7e17e3c/attachment-0001.html From jungleboogie0 at gmail.com Fri Dec 12 03:33:19 2014 From: jungleboogie0 at gmail.com (jungle Boogie) Date: Thu, 11 Dec 2014 16:33:19 -0800 Subject: [Freeswitch-users] TLS versions and PFS settings In-Reply-To: References: Message-ID: Hi Daniel, On 13 November 2014 at 12:15, Daniel Ivanov wrote: > > Thanks, Brian. Precise as always. And yet how do i enable perfect forward > secrecy? > Do you get PFS to work for you? > 13 ????. 2014 ?. 18:20 ???????????? "Brian West" > ???????: > > in a sofia profile: >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> On Thu, Nov 13, 2014 at 8:55 AM, Daniel Ivanov wrote: >> >>> What are the options to enable PFS on TLS support, i found the ability >>> added in a changelog from feb,2014, but can't find the corresponding config >>> file params. >>> Also can i specify a list of allowed ciphers? >>> >>> >> -- >> >> *Brian West* >> brian at freeswitch.org >> >> -- ------- inum: 883510009027723 sip: jungleboogie at sip2sip.info xmpp: jungle-boogie at jit.si -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141211/c41ec14a/attachment.html From akhilgarg7 at gmail.com Fri Dec 12 09:29:39 2014 From: akhilgarg7 at gmail.com (akhil garg) Date: Fri, 12 Dec 2014 11:59:39 +0530 Subject: [Freeswitch-users] fs_cli hangs In-Reply-To: References: Message-ID: pastebin: 23716 regards, akhil On Fri, Dec 12, 2014 at 12:24 AM, Chris Tunbridge wrote: > > Can you do a pastebin of the contents of your > conf/autoload_configs/event_socket.conf.xml > > > On Thu, Dec 11, 2014 at 10:10 AM, Sudhanshu wrote: > >> Enable debug logging and then check freeswitch.log. >> What happens when you start freeswitch as anormal process (and not as a >> daemon)? >> >> -- >> Sudhanshu >> >> On Thu, Dec 11, 2014 at 2:30 PM, akhil garg wrote: >> >>> running "fs_cli -H 127.0.0.1 -P 8021 -d 7" gives different outputs but >>> no success. >>> >>> >>> >>> ------------------------------------------------------------------------------------------------------------------------------------------------ >>> OUTPUT 1: >>> >>> ------------------------------------------------------------------------------------------------------------------------------------------------ >>> [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is >>> /root/.fs_cli_conf. >>> [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is >>> /etc/fs_cli.conf. >>> [DEBUG] fs_cli.c:1438 main() profile default does not exist using >>> builtin profile >>> [DEBUG] fs_cli.c:1468 main() Using profile internal [127.0.0.1] >>> >>> ------------------------------------------------------------------------------------------------------------------------------------------------ >>> >>> >>> >>> >>> >>> ------------------------------------------------------------------------------------------------------------------------------------------------ >>> OUTPUT 2: >>> >>> ------------------------------------------------------------------------------------------------------------------------------------------------ >>> [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is >>> /root/.fs_cli_conf. >>> [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is >>> /etc/fs_cli.conf. >>> [DEBUG] fs_cli.c:1438 main() profile default does not exist using >>> builtin profile >>> [DEBUG] fs_cli.c:1468 main() Using profile internal [127.0.0.1] >>> [DEBUG] esl.c:1265 esl_recv_event() RECV HEADER [Content-Type] = >>> [auth/request] >>> [DEBUG] esl.c:1437 esl_recv_event() RECV MESSAGE >>> Event-Name: SOCKET_DATA >>> Content-Type: auth/request >>> >>> >>> [DEBUG] esl.c:1465 esl_send() SEND >>> auth ClueCon >>> >>> ------------------------------------------------------------------------------------------------------------------------------------------------ >>> >>> >>> >>> >>> >>> ------------------------------------------------------------------------------------------------------------------------------------------------ >>> OUTPUT 3: >>> >>> ------------------------------------------------------------------------------------------------------------------------------------------------ >>> [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is >>> /root/.fs_cli_conf. >>> [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is >>> /etc/fs_cli.conf. >>> [DEBUG] fs_cli.c:1438 main() profile default does not exist using >>> builtin profile >>> [DEBUG] fs_cli.c:1468 main() Using profile internal [127.0.0.1] >>> [DEBUG] esl.c:1265 esl_recv_event() RECV HEADER [Content-Type] = >>> [auth/request] >>> [DEBUG] esl.c:1437 esl_recv_event() RECV MESSAGE >>> Event-Name: SOCKET_DATA >>> Content-Type: auth/request >>> >>> >>> [DEBUG] esl.c:1465 esl_send() SEND >>> auth ClueCon >>> >>> >>> [DEBUG] esl.c:1265 esl_recv_event() RECV HEADER [Content-Type] = >>> [command/reply] >>> [DEBUG] esl.c:1265 esl_recv_event() RECV HEADER [Reply-Text] = [+OK >>> accepted] >>> [DEBUG] esl.c:1437 esl_recv_event() RECV MESSAGE >>> Event-Name: SOCKET_DATA >>> Content-Type: command/reply >>> Reply-Text: +OK accepted >>> >>> >>> [DEBUG] esl.c:1465 esl_send() SEND >>> log >>> >>> ------------------------------------------------------------------------------------------------------------------------------------------------ >>> >>> >>> >>> -- >>> regards, >>> akhil >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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, akhil -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141212/08f1a01b/attachment.html From akhilgarg7 at gmail.com Fri Dec 12 09:36:30 2014 From: akhilgarg7 at gmail.com (akhil garg) Date: Fri, 12 Dec 2014 12:06:30 +0530 Subject: [Freeswitch-users] fs_cli hangs In-Reply-To: References: Message-ID: Please see the logs: pastebin : 23717 regards, akhil On Thu, Dec 11, 2014 at 10:40 PM, Sudhanshu wrote: > > Enable debug logging and then check freeswitch.log. > What happens when you start freeswitch as anormal process (and not as a > daemon)? > > -- > Sudhanshu > > On Thu, Dec 11, 2014 at 2:30 PM, akhil garg wrote: > >> running "fs_cli -H 127.0.0.1 -P 8021 -d 7" gives different outputs but no >> success. >> >> >> >> ------------------------------------------------------------------------------------------------------------------------------------------------ >> OUTPUT 1: >> >> ------------------------------------------------------------------------------------------------------------------------------------------------ >> [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is >> /root/.fs_cli_conf. >> [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is >> /etc/fs_cli.conf. >> [DEBUG] fs_cli.c:1438 main() profile default does not exist using builtin >> profile >> [DEBUG] fs_cli.c:1468 main() Using profile internal [127.0.0.1] >> >> ------------------------------------------------------------------------------------------------------------------------------------------------ >> >> >> >> >> >> ------------------------------------------------------------------------------------------------------------------------------------------------ >> OUTPUT 2: >> >> ------------------------------------------------------------------------------------------------------------------------------------------------ >> [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is >> /root/.fs_cli_conf. >> [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is >> /etc/fs_cli.conf. >> [DEBUG] fs_cli.c:1438 main() profile default does not exist using builtin >> profile >> [DEBUG] fs_cli.c:1468 main() Using profile internal [127.0.0.1] >> [DEBUG] esl.c:1265 esl_recv_event() RECV HEADER [Content-Type] = >> [auth/request] >> [DEBUG] esl.c:1437 esl_recv_event() RECV MESSAGE >> Event-Name: SOCKET_DATA >> Content-Type: auth/request >> >> >> [DEBUG] esl.c:1465 esl_send() SEND >> auth ClueCon >> >> ------------------------------------------------------------------------------------------------------------------------------------------------ >> >> >> >> >> >> ------------------------------------------------------------------------------------------------------------------------------------------------ >> OUTPUT 3: >> >> ------------------------------------------------------------------------------------------------------------------------------------------------ >> [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is >> /root/.fs_cli_conf. >> [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is >> /etc/fs_cli.conf. >> [DEBUG] fs_cli.c:1438 main() profile default does not exist using builtin >> profile >> [DEBUG] fs_cli.c:1468 main() Using profile internal [127.0.0.1] >> [DEBUG] esl.c:1265 esl_recv_event() RECV HEADER [Content-Type] = >> [auth/request] >> [DEBUG] esl.c:1437 esl_recv_event() RECV MESSAGE >> Event-Name: SOCKET_DATA >> Content-Type: auth/request >> >> >> [DEBUG] esl.c:1465 esl_send() SEND >> auth ClueCon >> >> >> [DEBUG] esl.c:1265 esl_recv_event() RECV HEADER [Content-Type] = >> [command/reply] >> [DEBUG] esl.c:1265 esl_recv_event() RECV HEADER [Reply-Text] = [+OK >> accepted] >> [DEBUG] esl.c:1437 esl_recv_event() RECV MESSAGE >> Event-Name: SOCKET_DATA >> Content-Type: command/reply >> Reply-Text: +OK accepted >> >> >> [DEBUG] esl.c:1465 esl_send() SEND >> log >> >> ------------------------------------------------------------------------------------------------------------------------------------------------ >> >> >> >> -- >> regards, >> akhil >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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, akhil -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141212/f4e69f0d/attachment-0001.html From bordmi at rarus.ru Fri Dec 12 09:59:01 2014 From: bordmi at rarus.ru (=?UTF-8?B?0JHQvtGA0LjRgdC+0LIsINCU0LzQuNGC0YDQuNC5?=) Date: Fri, 12 Dec 2014 10:59:01 +0400 Subject: [Freeswitch-users] Dialplan - Lua in-line expansion In-Reply-To: References: Message-ID: change return result to io.write(result) 2014-12-12 0:32 GMT+03:00 Phillip Jones : > Yeah - thanks - I realize I can do that - but it just drives me crazy I > can not get in-line expansion to work. I can't see what I am missing. > > On Thu, Dec 11, 2014 at 3:46 PM, Chad Phillips > wrote: > >> how about setting the channel var directly in the script? >> >> session:setVariable("VUM_SAY_CID", result); >> >> although for what you?re doing, i really think there?s a cleaner way to >> do it straight from the dialplan XML... >> >> On Thursday, December 11, 2014 at 1:35 PM, Phillip Jones wrote: >> >> I am going crazy - please tell me what I am missing: >> >> I have a dialplan: >> >> >> >> >> I have a lua script: >> >> >> freeswitch.consoleLog("NOTICE","util_get10digits.lua \n") >> freeswitch.consoleLog("NOTICE","Argument passed [" .. argv[1] .."]\n") >> local result = "none" >> *result = string.sub(argv[1], -10)* >> freeswitch.consoleLog("NOTICE", "Result: [" .. result .."]\n") >> *return result* >> >> >> I run the dialplan: >> >> Dialplan: sofia/Carrier/2154791697 at 64.158.162.74 Action >> set(VUM_SAY_CID=${lua(util_get10digits.lua >> ${VUM_Outgoing_CallerID_Number})}) I >> 2014-12-11 15:20:21.795274 [NOTICE] switch_cpp.cpp:1328 >> util_get10digits.lua >> 2014-12-11 15:20:21.795274 [NOTICE] switch_cpp.cpp:1328 Argument passed >> [+12154791697] >> *2014-12-11 15:20:21.795274 [NOTICE] switch_cpp.cpp:1328 Result: >> [2154791697 <%5B2154791697>]* >> 2014-12-11 15:20:21.795274 [DEBUG] switch_cpp.cpp:1075 sofia/Carrier/ >> 2154791697 at 64.158.162.74 destroy/unlink session from object >> EXECUTE sofia/Carrier/2154791697 at 64.158.162.74 set(VUM_SAY_CID=) >> *2014-12-11 15:20:21.795274 [DEBUG] mod_dptools.c:1435 >> sofia/Carrier/2154791697 at 64.158.162.74 <2154791697 at 64.158.162.74> SET >> [VUM_SAY_CID]=[UNDEF]* >> >> >> The lua script runs correctly - how ever the result is not returned the >> VUM_SAY_CID variable in the dialplan >> >> >> The work around is to set the session variable in the script. But I >> wondering what I am missing. Is "return result" not how you return a result >> from Lua? >> >> >> Thanks for any help. >> >> >> Phil >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- with best regards, Dmitriy Borisov -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141212/29535b96/attachment.html From bordmi at rarus.ru Fri Dec 12 11:52:38 2014 From: bordmi at rarus.ru (=?UTF-8?B?0JHQvtGA0LjRgdC+0LIsINCU0LzQuNGC0YDQuNC5?=) Date: Fri, 12 Dec 2014 12:52:38 +0400 Subject: [Freeswitch-users] Dialplan - Lua in-line expansion In-Reply-To: References: Message-ID: Oh, sorry, not *io.write(result)* but *stream:write(result)* 2014-12-12 9:59 GMT+03:00 ???????, ??????? : > change > return result > to > io.write(result) > > 2014-12-12 0:32 GMT+03:00 Phillip Jones : > >> Yeah - thanks - I realize I can do that - but it just drives me crazy I >> can not get in-line expansion to work. I can't see what I am missing. >> >> On Thu, Dec 11, 2014 at 3:46 PM, Chad Phillips >> wrote: >> >>> how about setting the channel var directly in the script? >>> >>> session:setVariable("VUM_SAY_CID", result); >>> >>> although for what you?re doing, i really think there?s a cleaner way to >>> do it straight from the dialplan XML... >>> >>> On Thursday, December 11, 2014 at 1:35 PM, Phillip Jones wrote: >>> >>> I am going crazy - please tell me what I am missing: >>> >>> I have a dialplan: >>> >>> >>> >>> >>> I have a lua script: >>> >>> >>> freeswitch.consoleLog("NOTICE","util_get10digits.lua \n") >>> freeswitch.consoleLog("NOTICE","Argument passed [" .. argv[1] .."]\n") >>> local result = "none" >>> *result = string.sub(argv[1], -10)* >>> freeswitch.consoleLog("NOTICE", "Result: [" .. result .."]\n") >>> *return result* >>> >>> >>> I run the dialplan: >>> >>> Dialplan: sofia/Carrier/2154791697 at 64.158.162.74 Action >>> set(VUM_SAY_CID=${lua(util_get10digits.lua >>> ${VUM_Outgoing_CallerID_Number})}) I >>> 2014-12-11 15:20:21.795274 [NOTICE] switch_cpp.cpp:1328 >>> util_get10digits.lua >>> 2014-12-11 15:20:21.795274 [NOTICE] switch_cpp.cpp:1328 Argument passed >>> [+12154791697] >>> *2014-12-11 15:20:21.795274 [NOTICE] switch_cpp.cpp:1328 Result: >>> [2154791697 <%5B2154791697>]* >>> 2014-12-11 15:20:21.795274 [DEBUG] switch_cpp.cpp:1075 sofia/Carrier/ >>> 2154791697 at 64.158.162.74 destroy/unlink session from object >>> EXECUTE sofia/Carrier/2154791697 at 64.158.162.74 set(VUM_SAY_CID=) >>> *2014-12-11 15:20:21.795274 [DEBUG] mod_dptools.c:1435 >>> sofia/Carrier/2154791697 at 64.158.162.74 <2154791697 at 64.158.162.74> SET >>> [VUM_SAY_CID]=[UNDEF]* >>> >>> >>> The lua script runs correctly - how ever the result is not returned the >>> VUM_SAY_CID variable in the dialplan >>> >>> >>> The work around is to set the session variable in the script. But I >>> wondering what I am missing. Is "return result" not how you return a result >>> from Lua? >>> >>> >>> Thanks for any help. >>> >>> >>> Phil >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > with best regards, > Dmitriy Borisov > -- with best regards, Dmitriy Borisov -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141212/b2eebd9a/attachment-0001.html From idokan at gmail.com Fri Dec 12 13:22:59 2014 From: idokan at gmail.com (ik) Date: Fri, 12 Dec 2014 12:22:59 +0200 Subject: [Freeswitch-users] Controling playback recording position In-Reply-To: References: Message-ID: That's very simple and easy Thank you On Thu, Dec 11, 2014 at 8:08 PM, Brian West wrote: > > See scripts/lua/callback.lua, javascript has a similar method too. > > > > On Thu, Dec 11, 2014 at 11:32 AM, ik wrote: > >> Hello, >> >> How can I control the position of the recording that is played ? >> For example, go 10 seconds forward or backward, like with Asterisk's >> ControlPlayback cmd. >> >> Thanks, >> Ido >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > > *Brian West* > brian at freeswitch.org > > > *Twitter: @FreeSWITCH , @briankwest* > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141212/bd97ffbf/attachment.html From sertys at gmail.com Fri Dec 12 15:33:58 2014 From: sertys at gmail.com (Daniel Ivanov) Date: Fri, 12 Dec 2014 13:33:58 +0100 Subject: [Freeswitch-users] TLS versions and PFS settings In-Reply-To: References: Message-ID: Unfortunately not. Still looking for a way to do it. On Fri, Dec 12, 2014 at 1:33 AM, jungle Boogie wrote: > Hi Daniel, > On 13 November 2014 at 12:15, Daniel Ivanov wrote: >> >> Thanks, Brian. Precise as always. And yet how do i enable perfect forward >> secrecy? >> > > Do you get PFS to work for you? > > >> 13 ????. 2014 ?. 18:20 ???????????? "Brian West" >> ???????: >> >> in a sofia profile: >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> On Thu, Nov 13, 2014 at 8:55 AM, Daniel Ivanov wrote: >>> >>>> What are the options to enable PFS on TLS support, i found the ability >>>> added in a changelog from feb,2014, but can't find the corresponding config >>>> file params. >>>> Also can i specify a list of allowed ciphers? >>>> >>>> >>> -- >>> >>> *Brian West* >>> brian at freeswitch.org >>> >>> -- > ------- > inum: 883510009027723 > sip: jungleboogie at sip2sip.info > xmpp: jungle-boogie at jit.si > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141212/992299f8/attachment.html From aqsyounas at gmail.com Fri Dec 12 15:39:30 2014 From: aqsyounas at gmail.com (Aqs Younas) Date: Fri, 12 Dec 2014 17:39:30 +0500 Subject: [Freeswitch-users] Giving up resync after 1024 bytes Message-ID: Hi, All I am newbie to freeswitch, while playing multiple stream with mod_shout my freeswitch logs flood with this type of errors. [layer3.c:454] error: big_values too large! [layer3.c:454] error: big_values too large! Note: Illegal Audio-MPEG-Header 0xbbb7df0d at offset 19830. Note: Trying to resync... Note: Hit end of (available) data during resync. 2014-12-12 12:35:42.564619 [WARNING] switch_core_file.c:230 File has 2 channels, muxing to 1 channel will occur. Note: Illegal Audio-MPEG-Header 0xbbb7df0d at offset 19830. Note: Trying to resync... Note: Skipped 1024 bytes in input. [parse.c:682] error: Giving up resync after 1024 bytes - your stream is not nice... (maybe increasing resync limit could help). Note: Illegal Audio-MPEG-Header 0xef0000a8 at offset 20857. Note: Trying to resync... Note: Skipped 64 bytes in input. Could someone please help me in understanding or solving this problem. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141212/2e4a560d/attachment.html From thomas.granvej6 at gmail.com Fri Dec 12 16:04:24 2014 From: thomas.granvej6 at gmail.com (=?UTF-8?Q?Thomas_L=C3=B8cke?=) Date: Fri, 12 Dec 2014 14:04:24 +0100 Subject: [Freeswitch-users] FreeBSD or Debian Message-ID: Hey all, I'm involved in a project where we'll soon need to spin up a couple of FreeSWITCH instances, and we're a bit torn between Debian and FreeBSD. Anybody out there with experience with both? Are there any differences? Will Debian be "better" due to FreeSWITCH primarily targeting Linux? We're very slightly leaning towards FreeBSD due to jails, ZFS and other nice FreeBSD tools, but we're not averse to Debian in any way. We're probably at 52% FreeBSD and 48% Debian. The instances will run on hardware with decent specs. No less than 16 cores (probably 32) and no less than 32GB RAM and SSD in every single bay. We're not targeting low spec. Any and all advice is welcomed with open arms. :o) Thomas L?cke -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141212/e411a56c/attachment.html From karl-theo_hofer at inteli-sim.com Fri Dec 12 16:42:00 2014 From: karl-theo_hofer at inteli-sim.com (kthofer) Date: Fri, 12 Dec 2014 14:42:00 +0100 Subject: [Freeswitch-users] mkdir not working Message-ID: <548AF0A8.90909@inteli-sim.com> Hi there we try to create a new directory $con->execute("mkdir", "$vmail_base"); to store some Voice mail files But for some reason FS is not even trying to create the directory we checked the user rights the spelling but still nothing not even in the linux logs we can not see that FS is trying it atall. Any suggestions?? what is going wrong? Please let me know if you have a solution or workaround. -- With best regards Karl Theo Hofer From alipey at gmail.com Fri Dec 12 17:54:07 2014 From: alipey at gmail.com (Ali Pey) Date: Fri, 12 Dec 2014 09:54:07 -0500 Subject: [Freeswitch-users] Record sound quality not good but pcap is quite good Message-ID: Hello, I use record to record part of a call and sometimes I get some static noises and distortion; however when I listen to the call on pcap capture, it's quite clear and no noise. I record to a wav file and then use sox to convert it to ulaw. Any suggestions? Any help would be greatly appreciated. Thanks, Ali Pey -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141212/674a9c96/attachment.html From bordmi at rarus.ru Fri Dec 12 17:55:48 2014 From: bordmi at rarus.ru (=?UTF-8?B?0JHQvtGA0LjRgdC+0LIsINCU0LzQuNGC0YDQuNC5IC8gRG1pdHJpeSBCb3Jpc292?=) Date: Fri, 12 Dec 2014 18:55:48 +0400 Subject: [Freeswitch-users] FreeBSD or Debian In-Reply-To: References: Message-ID: We have test platform on FreeBSD and working platform on Debian. I found only one pricipial difference between platforms: on Debian we have native Corosync/Pacemaker, which are the best tool for clustering until mod_ha_cluster will released. But our platform is not working under huge load, we are not released our prudct still. 2014-12-12 16:04 GMT+03:00 Thomas L?cke : > Hey all, > > I'm involved in a project where we'll soon need to spin up a couple of > FreeSWITCH instances, and we're a bit torn between Debian and FreeBSD. > > Anybody out there with experience with both? Are there any differences? > Will Debian be "better" due to FreeSWITCH primarily targeting Linux? > > We're very slightly leaning towards FreeBSD due to jails, ZFS and other > nice FreeBSD tools, but we're not averse to Debian in any way. We're > probably at 52% FreeBSD and 48% Debian. > > The instances will run on hardware with decent specs. No less than 16 > cores (probably 32) and no less than 32GB RAM and SSD in every single bay. > We're not targeting low spec. > > Any and all advice is welcomed with open arms. > > :o) > Thomas L?cke > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- with best regards, Dmitriy Borisov -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141212/98eef024/attachment.html From krice at freeswitch.org Fri Dec 12 18:02:24 2014 From: krice at freeswitch.org (Ken Rice) Date: Fri, 12 Dec 2014 15:02:24 +0000 Subject: [Freeswitch-users] FreeSWITCH Friday FreeForAll Reminder! Message-ID: <548b0380440a6_363a80b33046726@ip-10-155-242-219.mail> FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888 at conference.freeswitch.org or see http://ift.tt/1prwIZL for access info! -- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141212/d8e67c42/attachment.html From mbsip at gazeta.pl Fri Dec 12 18:07:14 2014 From: mbsip at gazeta.pl (Maciej Bylica) Date: Fri, 12 Dec 2014 16:07:14 +0100 Subject: [Freeswitch-users] Invalid codec CN is tearing down the call request In-Reply-To: References: Message-ID: 1. FS-7086 2014-12-12 1:00 GMT+01:00 Maciej Bylica : > I will gather packet capture and file JIRA. > > Thank You > Maciej. > > > Funny, payload 13 is CN, which is 0x13 in base16 or 19 in base10, Please >> file a JIRA, I would need to see a packet capture for this one. >> >> >> On Thu, Dec 11, 2014 at 11:47 AM, Maciej Bylica wrote: >> >>> Sure i'll try with jira >>> >>> Mac. >>> >>> 2014-12-11 17:41 GMT+01:00 Anthony Minessale < >>> anthony.minessale at gmail.com>: >>> >>>> For years I beg ppl to post issues to jira not the mailing list. >>>> if you start typing and keywords like "problem" "not working" "issue" >>>> appear in your text, you probably should be filing it to JIRA not here. >>>> >>>> >>>> On Thu, Dec 11, 2014 at 10:37 AM, Maciej Bylica >>>> wrote: >>>> >>>>> Thanks for prompt reply. >>>>> >>>>> Steven, later today i will try to update FS to the latest git and >>>>> check this out. >>>>> >>>>> Ken, i am fine with CN, passing thru is what i really need, but the >>>>> question is why FS generates such ERRors and how to get out of this. >>>>> >>>>> Thanks >>>>> Mac. >>>>> >>>>> 2014-12-11 17:12 GMT+01:00 Ken Rice : >>>>> >>>>>> CN is comfort noise... Your cisco is sending it, and freeswitch is >>>>>> passing it thru because you have told it to use proxy media mode... >>>>>> >>>>>> >>>>>> >>>>>> On 12/11/14 9:53 AM, "Steven Ayre" wrote: >>>>>> >>>>>> Your version is 3 months old. Can you reproduce this on the latest >>>>>> git head? >>>>>> >>>>>> On 11 December 2014 at 13:45, Maciej Bylica wrote: >>>>>> >>>>>> Hello, >>>>>> >>>>>> I am running FreeSWITCH Version >>>>>> 1.5.14b+git~20140917T231120Z~8f85b5204c~64bit, proxy media mode and heaving >>>>>> problem with some call requests setup. >>>>>> Here is one of them: >>>>>> >>>>>> 2014-12-11 13:38:15.034612 [NOTICE] switch_channel.c:1055 New Channel >>>>>> sofia/outside_1/20049112223344 [] >>>>>> 2014-12-11 13:38:15.034612 [DEBUG] mod_sofia.c:4579 >>>>>> (sofia/outside_1/20049112223344) State Change CS_NEW -> CS_INIT >>>>>> 2014-12-11 13:38:15.034612 [DEBUG] switch_core_session.c:1388 Send >>>>>> signal sofia/outside_1/20049112223344 [BREAK] >>>>>> 2014-12-11 13:38:15.054610 [DEBUG] switch_core_state_machine.c:472 >>>>>> (sofia/outside_1/20049112223344) Running State Change CS_INIT >>>>>> 2014-12-11 13:38:15.054610 [DEBUG] switch_core_state_machine.c:512 >>>>>> (sofia/outside_1/20049112223344) State INIT >>>>>> 2014-12-11 13:38:15.054610 [DEBUG] mod_sofia.c:87 >>>>>> sofia/outside_1/20049112223344 SOFIA INIT >>>>>> 2014-12-11 13:38:15.054610 [DEBUG] switch_core_media.c:7510 >>>>>> sofia/outside_1/20049112223344 Patched SDP >>>>>> --- >>>>>> v=0 >>>>>> o=CiscoSystemsSIP-GW-UserAgent 6016 7716 IN IP4 10.10.10.226 >>>>>> s=SIP Call >>>>>> c=IN IP4 10.10.10.12 >>>>>> t=0 0 >>>>>> m=audio 24782 RTP/AVP 18 19 >>>>>> c=IN IP4 10.10.10.12 >>>>>> a=rtpmap:18 G729/8000 >>>>>> a=fmtp:18 annexb=yes >>>>>> a=rtpmap:19 CN/8000 >>>>>> >>>>>> +++ >>>>>> v=0 >>>>>> o=FreeSWITCH 0255539876 0255539877 IN IP4 10.10.10.166 >>>>>> s=FreeSWITCH >>>>>> c=IN IP4 10.10.10.166 >>>>>> t=0 0 >>>>>> m=audio 19690 RTP/AVP 18 19 >>>>>> c=IN IP4 10.10.10.166 >>>>>> a=rtpmap:18 G729/8000 >>>>>> a=fmtp:18 annexb=yes >>>>>> a=rtpmap:19 CN/8000 >>>>>> >>>>>> 2014-12-11 13:38:15.054610 [DEBUG] sofia_glue.c:1228 >>>>>> sofia/outside_1/20049112223344 sending invite version: 1.5.14b git 8f85b52 >>>>>> 2014-09-17 23:11:20Z 64bit >>>>>> Local SDP: >>>>>> v=0 >>>>>> o=FreeSWITCH 0255539876 0255539877 IN IP4 10.10.10.166 >>>>>> s=FreeSWITCH >>>>>> c=IN IP4 10.10.10.166 >>>>>> t=0 0 >>>>>> m=audio 19690 RTP/AVP 18 19 >>>>>> c=IN IP4 10.10.10.166 >>>>>> a=rtpmap:18 G729/8000 >>>>>> a=fmtp:18 annexb=yes >>>>>> a=rtpmap:19 CN/8000 >>>>>> >>>>>> sending out SDP exactly as shown above. >>>>>> The other side is responsing with 100, then 180 Ringing wo/SDP then >>>>>> 183 w/SDP >>>>>> >>>>>> 2014-12-11 14:29:12.214608 [DEBUG] sofia.c:6423 Channel >>>>>> sofia/outside_1/20049112223344 entering state [proceeding][183] >>>>>> 2014-12-11 14:29:12.214608 [DEBUG] sofia.c:6433 Remote SDP: >>>>>> v=0 >>>>>> o=Dialogic_SDP 11471558 0 IN IP4 10.10.10.218 >>>>>> s=Dialogic-SIP >>>>>> c=IN IP4 10.10.10.198 >>>>>> t=0 0 >>>>>> m=audio 10024 RTP/AVP 19 18 >>>>>> a=rtpmap:19 CN/8000 >>>>>> a=rtpmap:18 G729/8000 >>>>>> a=fmtp:18 annexb=yes >>>>>> a=silenceSupp:off - - - - >>>>>> a=ptime:20 >>>>>> >>>>>> 2014-12-11 14:29:12.214608 [DEBUG] switch_core_media.c:7510 >>>>>> sofia/outside_1/20049112223344 Patched SDP >>>>>> --- >>>>>> v=0 >>>>>> o=FreeSWITCH 0351957506 0351957507 IN IP4 10.10.10.166 >>>>>> s=FreeSWITCH >>>>>> c=IN IP4 10.10.10.166 >>>>>> t=0 0 >>>>>> m=audio 29492 RTP/AVP 18 19 >>>>>> c=IN IP4 10.10.10.166 >>>>>> a=rtpmap:18 G729/8000 >>>>>> a=fmtp:18 annexb=yes >>>>>> a=rtpmap:19 CN/8000 >>>>>> >>>>>> +++ >>>>>> v=0 >>>>>> o=FreeSWITCH 0351957506 0351957508 IN IP4 10.10.10.166 >>>>>> s=FreeSWITCH >>>>>> c=IN IP4 10.10.10.166 >>>>>> t=0 0 >>>>>> m=audio 29492 RTP/AVP 18 19 >>>>>> c=IN IP4 10.10.10.166 >>>>>> a=rtpmap:18 G729/8000 >>>>>> a=fmtp:18 annexb=yes >>>>>> a=rtpmap:19 CN/8000 >>>>>> >>>>>> 2014-12-11 14:29:12.214608 [ERR] switch_core_codec.c:651 >>>>>> *Invalid codec CN! *2014-12-11 14:29:12.214608 [ERR] >>>>>> switch_core_media.c:2294 >>>>>> *Can't load codec? *2014-12-11 14:29:12.214608 [NOTICE] >>>>>> switch_core_media.c:2295 Hangup sofia/outside_1/20049112223344 >>>>>> [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] >>>>>> 2014-12-11 14:29:12.214608 [DEBUG] switch_channel.c:3222 Send signal >>>>>> sofia/outside_1/20049112223344 [KILL] >>>>>> >>>>>> and Freeswitch is terminating the call by using CANCEL. >>>>>> >>>>>> My modules.conf.xml config part looks like following: >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Is it a problem with CN definition? >>>>>> Could you please lead me where the problem is located? >>>>>> >>>>>> Thanks in advance. >>>>>> Mac. >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://confluence.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>>> ------------------------------ >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://confluence.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at 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 >>>>>> Twitter: @FreeSWITCH >>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://confluence.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at 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 ? @anthmfs ? @FreeSWITCH ? >>>> >>>> ? http://freeswitch.org/ ? http://cluecon.com/ ? >>>> http://twitter.com/FreeSWITCH >>>> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ >>>> * >>>> >>>> ClueCon Weekly Development Call >>>> ? sip:888 at conference.freeswitch.org ? +19193869900 >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> >> *Brian West* >> brian at freeswitch.org >> >> >> *Twitter: @FreeSWITCH , @briankwest* >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141212/af64dead/attachment-0001.html From norm at goes.com Fri Dec 12 17:07:59 2014 From: norm at goes.com (Norm) Date: Fri, 12 Dec 2014 09:07:59 -0500 Subject: [Freeswitch-users] FreeBSD or Debian In-Reply-To: References: Message-ID: <70910985-9657-41DF-93DA-58AD7D4B862D@goes.com> Debian > On Dec 12, 2014, at 8:04 AM, Thomas L?cke wrote: > > Hey all, > > I'm involved in a project where we'll soon need to spin up a couple of FreeSWITCH instances, and we're a bit torn between Debian and FreeBSD. > > Anybody out there with experience with both? Are there any differences? Will Debian be "better" due to FreeSWITCH primarily targeting Linux? > > We're very slightly leaning towards FreeBSD due to jails, ZFS and other nice FreeBSD tools, but we're not averse to Debian in any way. We're probably at 52% FreeBSD and 48% Debian. > > The instances will run on hardware with decent specs. No less than 16 cores (probably 32) and no less than 32GB RAM and SSD in every single bay. We're not targeting low spec. > > Any and all advice is welcomed with open arms. > > :o) > Thomas L?cke > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 michel.brabants at gmail.com Fri Dec 12 18:23:56 2014 From: michel.brabants at gmail.com (Michel Brabants) Date: Fri, 12 Dec 2014 16:23:56 +0100 Subject: [Freeswitch-users] freeswitch 1.4 and encryption/rtcp-mux Message-ID: Hello, I recently started upgrading to FS 1.4, but I encountered 2 difficulties of which I'm still looking into one: 1) DTLS-configuration seems to be required, although we don't use it currently. We use normal sip-profiles (no webrtc). The option to disable it, is "webrtc_enable_dtls=false", which can b set in the dialplan. But why is it trying to enable it by default? Can you disable it also in a profile? 2) Also a change because of webrtc seemingly. When receiving an invite (without sdp - 3pcc-request), freeswitch in the end response in its 200 OK with a rtcp-mux-line in its sdp. We don't want rtcp-mux, just rtp-port+1 for rtcp. When looking at the code, I don't currently know why FS sends back the myx-parameter as it seems only enabled when the other ends proposes it or am I missing something? Thanks, Michel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141212/c026b9c6/attachment.html From grcamauer at gmail.com Fri Dec 12 18:34:05 2014 From: grcamauer at gmail.com (Guillermo Ruiz Camauer) Date: Fri, 12 Dec 2014 12:34:05 -0300 Subject: [Freeswitch-users] FreeBSD or Debian In-Reply-To: <70910985-9657-41DF-93DA-58AD7D4B862D@goes.com> References: <70910985-9657-41DF-93DA-58AD7D4B862D@goes.com> Message-ID: I tend to use machines with ZFS only as dedicated file servers. ZFS is a great file system, but it takes over the machine and uses as much memory and resources as it can get it's hands on. I would not use it for FreeSwitch if there will be a lot of file-io going on. Guillermo On Fri, Dec 12, 2014 at 11:07 AM, Norm wrote: > Debian > > > On Dec 12, 2014, at 8:04 AM, Thomas L?cke > wrote: > > > > Hey all, > > > > I'm involved in a project where we'll soon need to spin up a couple of > FreeSWITCH instances, and we're a bit torn between Debian and FreeBSD. > > > > Anybody out there with experience with both? Are there any differences? > Will Debian be "better" due to FreeSWITCH primarily targeting Linux? > > > > We're very slightly leaning towards FreeBSD due to jails, ZFS and other > nice FreeBSD tools, but we're not averse to Debian in any way. We're > probably at 52% FreeBSD and 48% Debian. > > > > The instances will run on hardware with decent specs. No less than 16 > cores (probably 32) and no less than 32GB RAM and SSD in every single bay. > We're not targeting low spec. > > > > Any and all advice is welcomed with open arms. > > > > :o) > > Thomas L?cke > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://confluence.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Guillermo Ruiz Camauer -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141212/e338f048/attachment.html From pjintheusa at gmail.com Fri Dec 12 18:44:09 2014 From: pjintheusa at gmail.com (Phillip Jones) Date: Fri, 12 Dec 2014 10:44:09 -0500 Subject: [Freeswitch-users] Dialplan - Lua in-line expansion In-Reply-To: References: Message-ID: Perfect - thanks very much. On Fri, Dec 12, 2014 at 3:52 AM, ???????, ??????? wrote: > > Oh, sorry, not *io.write(result)* but *stream:write(result)* > > 2014-12-12 9:59 GMT+03:00 ???????, ??????? : > >> change >> return result >> to >> io.write(result) >> >> 2014-12-12 0:32 GMT+03:00 Phillip Jones : >> >>> Yeah - thanks - I realize I can do that - but it just drives me crazy I >>> can not get in-line expansion to work. I can't see what I am missing. >>> >>> On Thu, Dec 11, 2014 at 3:46 PM, Chad Phillips >>> wrote: >>> >>>> how about setting the channel var directly in the script? >>>> >>>> session:setVariable("VUM_SAY_CID", result); >>>> >>>> although for what you?re doing, i really think there?s a cleaner way to >>>> do it straight from the dialplan XML... >>>> >>>> On Thursday, December 11, 2014 at 1:35 PM, Phillip Jones wrote: >>>> >>>> I am going crazy - please tell me what I am missing: >>>> >>>> I have a dialplan: >>>> >>>> >>>> >>>> >>>> I have a lua script: >>>> >>>> >>>> freeswitch.consoleLog("NOTICE","util_get10digits.lua \n") >>>> freeswitch.consoleLog("NOTICE","Argument passed [" .. argv[1] .."]\n") >>>> local result = "none" >>>> *result = string.sub(argv[1], -10)* >>>> freeswitch.consoleLog("NOTICE", "Result: [" .. result .."]\n") >>>> *return result* >>>> >>>> >>>> I run the dialplan: >>>> >>>> Dialplan: sofia/Carrier/2154791697 at 64.158.162.74 Action >>>> set(VUM_SAY_CID=${lua(util_get10digits.lua >>>> ${VUM_Outgoing_CallerID_Number})}) I >>>> 2014-12-11 15:20:21.795274 [NOTICE] switch_cpp.cpp:1328 >>>> util_get10digits.lua >>>> 2014-12-11 15:20:21.795274 [NOTICE] switch_cpp.cpp:1328 Argument passed >>>> [+12154791697] >>>> *2014-12-11 15:20:21.795274 [NOTICE] switch_cpp.cpp:1328 Result: >>>> [2154791697 <%5B2154791697>]* >>>> 2014-12-11 15:20:21.795274 [DEBUG] switch_cpp.cpp:1075 sofia/Carrier/ >>>> 2154791697 at 64.158.162.74 destroy/unlink session from object >>>> EXECUTE sofia/Carrier/2154791697 at 64.158.162.74 set(VUM_SAY_CID=) >>>> *2014-12-11 15:20:21.795274 [DEBUG] mod_dptools.c:1435 >>>> sofia/Carrier/2154791697 at 64.158.162.74 <2154791697 at 64.158.162.74> SET >>>> [VUM_SAY_CID]=[UNDEF]* >>>> >>>> >>>> The lua script runs correctly - how ever the result is not returned the >>>> VUM_SAY_CID variable in the dialplan >>>> >>>> >>>> The work around is to set the session variable in the script. But I >>>> wondering what I am missing. Is "return result" not how you return a result >>>> from Lua? >>>> >>>> >>>> Thanks for any help. >>>> >>>> >>>> Phil >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> with best regards, >> Dmitriy Borisov >> > > > > -- > with best regards, > Dmitriy Borisov > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141212/ca63a0c6/attachment-0001.html From areski at gmail.com Fri Dec 12 18:56:23 2014 From: areski at gmail.com (Areski) Date: Fri, 12 Dec 2014 16:56:23 +0100 Subject: [Freeswitch-users] [Freeswitch-docs] Doc-Sprint Friday 12 December 2014 In-Reply-To: References: Message-ID: We will be starting in 5 minutes, come and join us. IRC: #freeswitch-docs (freenode) On Wed, Dec 10, 2014 at 12:47 PM, ?talo Rossi wrote: > > Confirmed so far: > > - Italo Rossi (+4) > - Areski Belaid > - Iwada Eja > - Adolphe Cher-Aime > - Bote > > Who else? > > On Fri, Dec 5, 2014 at 12:14 PM, Adolphe Cher-Aime > wrote: > >> I'm in. >> >> >> -- >> Adolphe >> >> On Thu, Dec 4, 2014 at 3:00 PM, Iwada Eja wrote: >> >>> I'm In >>> >>> On Thu, Dec 4, 2014 at 1:20 PM, Areski wrote: >>> >>>> Hi everyone, >>>> >>>> We are planning to organize an other doc sprint on *Friday 12 December >>>> at 10am CT*. >>>> It will be 4 hours long but you can join for less time. >>>> >>>> The Doc-sprint will focus on migrating the remaining pages from >>>> MediaWiki (https://wiki.freeswitch.org) to Confluence Wiki ( >>>> https://freeswitch.org/confluence). >>>> >>>> We will use an FS IRC channel during the sprint: *#freeswitch-docs* >>>> and will track our work on the spreadsheet: >>>> https://docs.google.com/spreadsheets/d/1qsG-kRymvKlNBapnBLw86W130VdbnK6naYapbR_UNds/edit?pli=1#gid=1187898333 >>>> >>>> During the sprint, please change the URL's "Status" you are working on >>>> to "Editing" with your name next to it so we don't duplicate work. >>>> >>>> Some extra information: >>>> - https://freeswitch.org/confluence/display/FREESWITCH/Wiki+Migration >>>> - >>>> https://freeswitch.org/confluence/display/FREESWITCH/Contributing+Documentation >>>> >>>> We hope to get a maximum number of people signed up! >>>> >>>> Peoples confirmed so far: >>>> - Italo Rossi (+4) >>>> - Areski Belaid >>>> >>>> >>>> So, who is in? >>>> >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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 >>> Iwada >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > ?talo Rossi > > _______________________________________________ > Freeswitch-docs mailing list > Freeswitch-docs at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-docs > > -- Kind regards, /Areski ---- Arezqui Belaid, Founder at Star2Billing (www.star2billing.com) Tel: +34650784355 Twitter: http://twitter.com/areskib LinkedIn: http://www.linkedin.com/in/areski -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141212/913cf88e/attachment.html From anuragrana31189 at gmail.com Fri Dec 12 20:25:10 2014 From: anuragrana31189 at gmail.com (Anurag Rana) Date: Fri, 12 Dec 2014 22:55:10 +0530 Subject: [Freeswitch-users] Error while making gsmopen module in FS. Message-ID: Hi All, I installed FreeSwitch and now I am trying to add gsmopen module. I edited the modules.conf file and uncommented the line "endpoints/mod_gsmopen" Now While running the command "make install all" , I am getting this error - ================================================== making install mod_gsmopen make[4]: Entering directory `/usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen' make install-am make[5]: Entering directory `/usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen' CXX mod_gsmopen_la-mod_gsmopen.lo In file included from mod_gsmopen.cpp:34:0: gsmopen.h:103:26: fatal error: ctb-0.16/ctb.h: No such file or directory #include "ctb-0.16/ctb.h" ^ compilation terminated. make[5]: *** [mod_gsmopen_la-mod_gsmopen.lo] Error 1 make[5]: Leaving directory `/usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen' make[4]: *** [install] Error 2 make[4]: Leaving directory `/usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen' make[3]: *** [mod_gsmopen-install] Error 1 make[3]: Leaving directory `/usr/local/src/freeswitch/src/mod' make[2]: *** [install-recursive] Error 1 make[2]: Leaving directory `/usr/local/src/freeswitch/src' make[1]: *** [install-recursive] Error 1 make[1]: Leaving directory `/usr/local/src/freeswitch' make: *** [install] Error 2 ======================================================== ?[image: Inline image 1]? ?Please help me resolving this error.? Anurag Rana M.Tech CSE, IIIT-Delhi, https://sites.google.com/site/homepagerana/ ?? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141212/9d143c5d/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image.png Type: image/png Size: 47388 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141212/9d143c5d/attachment-0001.png From brian at freeswitch.org Fri Dec 12 20:44:37 2014 From: brian at freeswitch.org (Brian West) Date: Fri, 12 Dec 2014 11:44:37 -0600 Subject: [Freeswitch-users] Error while making gsmopen module in FS. In-Reply-To: References: Message-ID: I would review the confluence or wiki pages on setting this up, you're missing the ilbs/files to compile the module properly. On Fri, Dec 12, 2014 at 11:25 AM, Anurag Rana wrote: > > Hi All, > > I installed FreeSwitch and now I am trying to add gsmopen module. > I edited the modules.conf file and uncommented the line > "endpoints/mod_gsmopen" > > Now While running the command "make install all" , I am getting this error > - > ================================================== > making install mod_gsmopen > make[4]: Entering directory > `/usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen' > make install-am > make[5]: Entering directory > `/usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen' > CXX mod_gsmopen_la-mod_gsmopen.lo > In file included from mod_gsmopen.cpp:34:0: > gsmopen.h:103:26: fatal error: ctb-0.16/ctb.h: No such file or directory > #include "ctb-0.16/ctb.h" > ^ > compilation terminated. > make[5]: *** [mod_gsmopen_la-mod_gsmopen.lo] Error 1 > make[5]: Leaving directory > `/usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen' > make[4]: *** [install] Error 2 > make[4]: Leaving directory > `/usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen' > make[3]: *** [mod_gsmopen-install] Error 1 > make[3]: Leaving directory `/usr/local/src/freeswitch/src/mod' > make[2]: *** [install-recursive] Error 1 > make[2]: Leaving directory `/usr/local/src/freeswitch/src' > make[1]: *** [install-recursive] Error 1 > make[1]: Leaving directory `/usr/local/src/freeswitch' > make: *** [install] Error 2 > ======================================================== > > ?[image: Inline image 1]? > > > > ?Please help me resolving this error.? > > > Anurag Rana > M.Tech CSE, IIIT-Delhi, > https://sites.google.com/site/homepagerana/ > ?? > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141212/c4f58df9/attachment.html From gmaruzz at gmail.com Fri Dec 12 20:48:07 2014 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Fri, 12 Dec 2014 18:48:07 +0100 Subject: [Freeswitch-users] Error while making gsmopen module in FS. In-Reply-To: References: Message-ID: The wiki page describes the prerequisites and the way to install. sent from my mobile, Giovanni Maruzzelli cell: +39 347 266 56 18 On Dec 12, 2014 6:45 PM, "Brian West" wrote: > I would review the confluence or wiki pages on setting this up, you're > missing the ilbs/files to compile the module properly. > > On Fri, Dec 12, 2014 at 11:25 AM, Anurag Rana > wrote: >> >> Hi All, >> >> I installed FreeSwitch and now I am trying to add gsmopen module. >> I edited the modules.conf file and uncommented the line >> "endpoints/mod_gsmopen" >> >> Now While running the command "make install all" , I am getting this >> error - >> ================================================== >> making install mod_gsmopen >> make[4]: Entering directory >> `/usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen' >> make install-am >> make[5]: Entering directory >> `/usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen' >> CXX mod_gsmopen_la-mod_gsmopen.lo >> In file included from mod_gsmopen.cpp:34:0: >> gsmopen.h:103:26: fatal error: ctb-0.16/ctb.h: No such file or directory >> #include "ctb-0.16/ctb.h" >> ^ >> compilation terminated. >> make[5]: *** [mod_gsmopen_la-mod_gsmopen.lo] Error 1 >> make[5]: Leaving directory >> `/usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen' >> make[4]: *** [install] Error 2 >> make[4]: Leaving directory >> `/usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen' >> make[3]: *** [mod_gsmopen-install] Error 1 >> make[3]: Leaving directory `/usr/local/src/freeswitch/src/mod' >> make[2]: *** [install-recursive] Error 1 >> make[2]: Leaving directory `/usr/local/src/freeswitch/src' >> make[1]: *** [install-recursive] Error 1 >> make[1]: Leaving directory `/usr/local/src/freeswitch' >> make: *** [install] Error 2 >> ======================================================== >> >> ?[image: Inline image 1]? >> >> >> >> ?Please help me resolving this error.? >> >> >> Anurag Rana >> M.Tech CSE, IIIT-Delhi, >> https://sites.google.com/site/homepagerana/ >> ?? >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > -- > > *Brian West* > brian at freeswitch.org > > > *Twitter: @FreeSWITCH , @briankwest* > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141212/c5925689/attachment.html From mike at jerris.com Fri Dec 12 21:06:34 2014 From: mike at jerris.com (Michael Jerris) Date: Fri, 12 Dec 2014 13:06:34 -0500 Subject: [Freeswitch-users] freeswitch 1.4 and encryption/rtcp-mux In-Reply-To: References: Message-ID: <8B568748-EEE1-4E26-A3BB-302345FB0A01@jerris.com> > On Dec 12, 2014, at 10:23 AM, Michel Brabants wrote: > > Hello, > > I recently started upgrading to FS 1.4, but I encountered 2 difficulties of which I'm still looking into one: > > 1) DTLS-configuration seems to be required, although we don't use it currently. We use normal sip-profiles (no webrtc). The option to disable it, is "webrtc_enable_dtls=false", which can b set in the dialplan. But why is it trying to enable it by default? Can you disable it also in a profile? > In what way do you think that some configuration is required? > 2) Also a change because of webrtc seemingly. When receiving an invite (without sdp - 3pcc-request), freeswitch in the end response in its 200 OK with a rtcp-mux-line in its sdp. We don't want rtcp-mux, just rtp-port+1 for rtcp. When looking at the code, I don't currently know why FS sends back the myx-parameter as it seems only enabled when the other ends proposes it or am I missing something? > Please report a bug on this. From anuragrana31189 at gmail.com Fri Dec 12 21:24:35 2014 From: anuragrana31189 at gmail.com (Anurag Rana) Date: Fri, 12 Dec 2014 23:54:35 +0530 Subject: [Freeswitch-users] Error while making gsmopen module in FS. In-Reply-To: References: Message-ID: According to Wiki page I need to compile and install gsmlib and libctb. These both are available in folder. */usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen* Earlier make command was complaining about not finding "ctb-0.16/ctb.h", So I *copied* the folder "ctb-0.16" from */endpoints/mod_gsmopen/libctb-0.16/include* to */endpoints/mod_gsmopen/ . *now this error is gone but a new error is coming up. /usr/bin/ld: cannot find -lctb-0.16 this is because of below line in MAKEFILE *mod_gsmopen_la_LDFLAGS = -avoid-version -module -no-undefined -lctb-0.16 -lgsmme* any suggestion please? Anurag Rana M.Tech CSE, IIIT-Delhi, https://sites.google.com/site/homepagerana/ On Fri, Dec 12, 2014 at 11:18 PM, Giovanni Maruzzelli wrote: > > The wiki page describes the prerequisites and the way to install. > > sent from my mobile, > Giovanni Maruzzelli > cell: +39 347 266 56 18 > On Dec 12, 2014 6:45 PM, "Brian West" wrote: > >> I would review the confluence or wiki pages on setting this up, you're >> missing the ilbs/files to compile the module properly. >> >> On Fri, Dec 12, 2014 at 11:25 AM, Anurag Rana >> wrote: >>> >>> Hi All, >>> >>> I installed FreeSwitch and now I am trying to add gsmopen module. >>> I edited the modules.conf file and uncommented the line >>> "endpoints/mod_gsmopen" >>> >>> Now While running the command "make install all" , I am getting this >>> error - >>> ================================================== >>> making install mod_gsmopen >>> make[4]: Entering directory >>> `/usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen' >>> make install-am >>> make[5]: Entering directory >>> `/usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen' >>> CXX mod_gsmopen_la-mod_gsmopen.lo >>> In file included from mod_gsmopen.cpp:34:0: >>> gsmopen.h:103:26: fatal error: ctb-0.16/ctb.h: No such file or directory >>> #include "ctb-0.16/ctb.h" >>> ^ >>> compilation terminated. >>> make[5]: *** [mod_gsmopen_la-mod_gsmopen.lo] Error 1 >>> make[5]: Leaving directory >>> `/usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen' >>> make[4]: *** [install] Error 2 >>> make[4]: Leaving directory >>> `/usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen' >>> make[3]: *** [mod_gsmopen-install] Error 1 >>> make[3]: Leaving directory `/usr/local/src/freeswitch/src/mod' >>> make[2]: *** [install-recursive] Error 1 >>> make[2]: Leaving directory `/usr/local/src/freeswitch/src' >>> make[1]: *** [install-recursive] Error 1 >>> make[1]: Leaving directory `/usr/local/src/freeswitch' >>> make: *** [install] Error 2 >>> ======================================================== >>> >>> ?[image: Inline image 1]? >>> >>> >>> >>> ?Please help me resolving this error.? >>> >>> >>> Anurag Rana >>> M.Tech CSE, IIIT-Delhi, >>> https://sites.google.com/site/homepagerana/ >>> ?? >>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> -- >> >> *Brian West* >> brian at freeswitch.org >> >> >> *Twitter: @FreeSWITCH , @briankwest* >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141212/03abac63/attachment.html From gmaruzz at gmail.com Fri Dec 12 21:31:21 2014 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Fri, 12 Dec 2014 19:31:21 +0100 Subject: [Freeswitch-users] Error while making gsmopen module in FS. In-Reply-To: References: Message-ID: Follow strictly, step by step and from the beginning the instructions in the wiki page. Everyone succeeds, you'll eventually succeed too. sent from my mobile, Giovanni Maruzzelli cell: +39 347 266 56 18 On Dec 12, 2014 7:27 PM, "Anurag Rana" wrote: > According to Wiki page I need to compile and install gsmlib and libctb. > > These both are available in folder. > */usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen* > > Earlier make command was complaining about not finding "ctb-0.16/ctb.h", > So I *copied* the folder "ctb-0.16" from > */endpoints/mod_gsmopen/libctb-0.16/include* to */endpoints/mod_gsmopen/ > . *now this error is gone but a new error is coming up. > > /usr/bin/ld: cannot find -lctb-0.16 > > this is because of below line in MAKEFILE > > *mod_gsmopen_la_LDFLAGS = -avoid-version -module -no-undefined -lctb-0.16 > -lgsmme* > > > any suggestion please? > > > > Anurag Rana > M.Tech CSE, IIIT-Delhi, > https://sites.google.com/site/homepagerana/ > > > > > > On Fri, Dec 12, 2014 at 11:18 PM, Giovanni Maruzzelli > wrote: >> >> The wiki page describes the prerequisites and the way to install. >> >> sent from my mobile, >> Giovanni Maruzzelli >> cell: +39 347 266 56 18 >> On Dec 12, 2014 6:45 PM, "Brian West" wrote: >> >>> I would review the confluence or wiki pages on setting this up, you're >>> missing the ilbs/files to compile the module properly. >>> >>> On Fri, Dec 12, 2014 at 11:25 AM, Anurag Rana >> > wrote: >>>> >>>> Hi All, >>>> >>>> I installed FreeSwitch and now I am trying to add gsmopen module. >>>> I edited the modules.conf file and uncommented the line >>>> "endpoints/mod_gsmopen" >>>> >>>> Now While running the command "make install all" , I am getting this >>>> error - >>>> ================================================== >>>> making install mod_gsmopen >>>> make[4]: Entering directory >>>> `/usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen' >>>> make install-am >>>> make[5]: Entering directory >>>> `/usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen' >>>> CXX mod_gsmopen_la-mod_gsmopen.lo >>>> In file included from mod_gsmopen.cpp:34:0: >>>> gsmopen.h:103:26: fatal error: ctb-0.16/ctb.h: No such file or directory >>>> #include "ctb-0.16/ctb.h" >>>> ^ >>>> compilation terminated. >>>> make[5]: *** [mod_gsmopen_la-mod_gsmopen.lo] Error 1 >>>> make[5]: Leaving directory >>>> `/usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen' >>>> make[4]: *** [install] Error 2 >>>> make[4]: Leaving directory >>>> `/usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen' >>>> make[3]: *** [mod_gsmopen-install] Error 1 >>>> make[3]: Leaving directory `/usr/local/src/freeswitch/src/mod' >>>> make[2]: *** [install-recursive] Error 1 >>>> make[2]: Leaving directory `/usr/local/src/freeswitch/src' >>>> make[1]: *** [install-recursive] Error 1 >>>> make[1]: Leaving directory `/usr/local/src/freeswitch' >>>> make: *** [install] Error 2 >>>> ======================================================== >>>> >>>> ?[image: Inline image 1]? >>>> >>>> >>>> >>>> ?Please help me resolving this error.? >>>> >>>> >>>> Anurag Rana >>>> M.Tech CSE, IIIT-Delhi, >>>> https://sites.google.com/site/homepagerana/ >>>> ?? >>>> >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> -- >>> >>> *Brian West* >>> brian at freeswitch.org >>> >>> >>> *Twitter: @FreeSWITCH , @briankwest* >>> http://www.freeswitchbook.com >>> http://www.freeswitchcookbook.com >>> >>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141212/a501b48c/attachment-0001.html From anuragrana31189 at gmail.com Fri Dec 12 21:45:30 2014 From: anuragrana31189 at gmail.com (Anurag Rana) Date: Sat, 13 Dec 2014 00:15:30 +0530 Subject: [Freeswitch-users] Error while making gsmopen module in FS. In-Reply-To: References: Message-ID: Done. Thanks. Anurag Rana M.Tech CSE, IIIT-Delhi, https://sites.google.com/site/homepagerana/ On Sat, Dec 13, 2014 at 12:01 AM, Giovanni Maruzzelli wrote: > > Follow strictly, step by step and from the beginning the instructions in > the wiki page. > > Everyone succeeds, you'll eventually succeed too. > > sent from my mobile, > Giovanni Maruzzelli > cell: +39 347 266 56 18 > On Dec 12, 2014 7:27 PM, "Anurag Rana" wrote: > >> According to Wiki page I need to compile and install gsmlib and libctb. >> >> These both are available in folder. >> */usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen* >> >> Earlier make command was complaining about not finding "ctb-0.16/ctb.h", >> So I *copied* the folder "ctb-0.16" from >> */endpoints/mod_gsmopen/libctb-0.16/include* to */endpoints/mod_gsmopen/ >> . *now this error is gone but a new error is coming up. >> >> /usr/bin/ld: cannot find -lctb-0.16 >> >> this is because of below line in MAKEFILE >> >> *mod_gsmopen_la_LDFLAGS = -avoid-version -module -no-undefined -lctb-0.16 >> -lgsmme* >> >> >> any suggestion please? >> >> >> >> Anurag Rana >> M.Tech CSE, IIIT-Delhi, >> https://sites.google.com/site/homepagerana/ >> >> >> >> >> >> On Fri, Dec 12, 2014 at 11:18 PM, Giovanni Maruzzelli >> wrote: >>> >>> The wiki page describes the prerequisites and the way to install. >>> >>> sent from my mobile, >>> Giovanni Maruzzelli >>> cell: +39 347 266 56 18 >>> On Dec 12, 2014 6:45 PM, "Brian West" wrote: >>> >>>> I would review the confluence or wiki pages on setting this up, you're >>>> missing the ilbs/files to compile the module properly. >>>> >>>> On Fri, Dec 12, 2014 at 11:25 AM, Anurag Rana < >>>> anuragrana31189 at gmail.com> wrote: >>>>> >>>>> Hi All, >>>>> >>>>> I installed FreeSwitch and now I am trying to add gsmopen module. >>>>> I edited the modules.conf file and uncommented the line >>>>> "endpoints/mod_gsmopen" >>>>> >>>>> Now While running the command "make install all" , I am getting this >>>>> error - >>>>> ================================================== >>>>> making install mod_gsmopen >>>>> make[4]: Entering directory >>>>> `/usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen' >>>>> make install-am >>>>> make[5]: Entering directory >>>>> `/usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen' >>>>> CXX mod_gsmopen_la-mod_gsmopen.lo >>>>> In file included from mod_gsmopen.cpp:34:0: >>>>> gsmopen.h:103:26: fatal error: ctb-0.16/ctb.h: No such file or >>>>> directory >>>>> #include "ctb-0.16/ctb.h" >>>>> ^ >>>>> compilation terminated. >>>>> make[5]: *** [mod_gsmopen_la-mod_gsmopen.lo] Error 1 >>>>> make[5]: Leaving directory >>>>> `/usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen' >>>>> make[4]: *** [install] Error 2 >>>>> make[4]: Leaving directory >>>>> `/usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen' >>>>> make[3]: *** [mod_gsmopen-install] Error 1 >>>>> make[3]: Leaving directory `/usr/local/src/freeswitch/src/mod' >>>>> make[2]: *** [install-recursive] Error 1 >>>>> make[2]: Leaving directory `/usr/local/src/freeswitch/src' >>>>> make[1]: *** [install-recursive] Error 1 >>>>> make[1]: Leaving directory `/usr/local/src/freeswitch' >>>>> make: *** [install] Error 2 >>>>> ======================================================== >>>>> >>>>> ?[image: Inline image 1]? >>>>> >>>>> >>>>> >>>>> ?Please help me resolving this error.? >>>>> >>>>> >>>>> Anurag Rana >>>>> M.Tech CSE, IIIT-Delhi, >>>>> https://sites.google.com/site/homepagerana/ >>>>> ?? >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> -- >>>> >>>> *Brian West* >>>> brian at freeswitch.org >>>> >>>> >>>> *Twitter: @FreeSWITCH , @briankwest* >>>> http://www.freeswitchbook.com >>>> http://www.freeswitchcookbook.com >>>> >>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141213/3c838765/attachment.html From sphotos1982 at gmail.com Fri Dec 12 22:20:40 2014 From: sphotos1982 at gmail.com (spros p) Date: Fri, 12 Dec 2014 21:20:40 +0200 Subject: [Freeswitch-users] LUA ESL Help Message-ID: Hello, I am trying to run a simple ESL script using Lua. I have installed Lua 5.1 and the respective dev packages on my linux machine. Make luamod in /libs/esl/ directory works just fine. However when executing: freeswitch/libs/esl/lua/single_command.lua, i receive the following: lua: lua/single_command.lua:2: module 'ESL' not found: no field package.preload['ESL'] no file './ESL.lua' no file '/usr/share/lua/5.1/ESL.lua' no file '/usr/share/lua/5.1/ESL/init.lua' no file '/usr/lib64/lua/5.1/ESL.lua' no file '/usr/lib64/lua/5.1/ESL/init.lua' no file './ESL.so' no file '/usr/lib64/lua/5.1/ESL.so' no file '/usr/lib64/lua/5.1/loadall.so' stack traceback: [C]: in function 'require' lua/single_command.lua:2: in main chunk [C]: ? Any help would be greatly appreciated Best, SP -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141212/e113a113/attachment-0001.html From aqsyounas at gmail.com Fri Dec 12 22:37:33 2014 From: aqsyounas at gmail.com (Aqs Younas) Date: Sat, 13 Dec 2014 00:37:33 +0500 Subject: [Freeswitch-users] How to play a stream other than mp3 with mod_shout. Message-ID: Hi, All How can i play a live stream other than mp3 with mod_shout or any module.? Is there any way to buffer the stream before playing it with mod_shout. Currently i have a list of streams and when i play them with mod_shout some work fine but others give (time out) error. How can i play mostly stream in freeswitch? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141213/7eb65ade/attachment.html From ahabiba at gmail.com Fri Dec 12 22:39:16 2014 From: ahabiba at gmail.com (Ahmed Habiba) Date: Fri, 12 Dec 2014 22:39:16 +0300 Subject: [Freeswitch-users] SIP Header Modification / Manipulation In-Reply-To: <29A45828-2427-492A-B18E-9260B4880745@gmail.com> References: <29A45828-2427-492A-B18E-9260B4880745@gmail.com> Message-ID: <0C78DA91-56C6-4901-B1BA-676257133525@gmail.com> Dears, Kindly help me on how can I modify the SIP header in the re-invite in the below scenario: [A Number] [FreeSWitch][B Number] 1-A Number Call B Number (Sip Invite sent from A number) 2-B number is not registered. 3-Call logic let FreeSwitch answer the call( Freeswitch send re-invite to A number) in the part number 3 I like to do modification on the SIP headers, by for example changing the From or To. Your help will be appreciated. Thanks, Ahmed Habiba. From jason.holden at start.ca Fri Dec 12 22:40:12 2014 From: jason.holden at start.ca (Jason Holden) Date: Fri, 12 Dec 2014 14:40:12 -0500 Subject: [Freeswitch-users] intigration of sms using an ip provider Message-ID: Anyone have any information on this? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141212/2556a40f/attachment.html From andrew at cassidywebservices.co.uk Sat Dec 13 00:16:48 2014 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Fri, 12 Dec 2014 21:16:48 +0000 Subject: [Freeswitch-users] FreeBSD or Debian In-Reply-To: References: <70910985-9657-41DF-93DA-58AD7D4B862D@goes.com> Message-ID: debian-kfreebsd? :P Debian on the freebsd kernel. Sounds rather sadomasochistic to me... On 12 December 2014 at 15:34, Guillermo Ruiz Camauer wrote: > > I tend to use machines with ZFS only as dedicated file servers. ZFS is a > great file system, but it takes over the machine and uses as much memory > and resources as it can get it's hands on. I would not use it for > FreeSwitch if there will be a lot of file-io going on. > > Guillermo > > On Fri, Dec 12, 2014 at 11:07 AM, Norm wrote: > >> Debian >> >> > On Dec 12, 2014, at 8:04 AM, Thomas L?cke >> wrote: >> > >> > Hey all, >> > >> > I'm involved in a project where we'll soon need to spin up a couple of >> FreeSWITCH instances, and we're a bit torn between Debian and FreeBSD. >> > >> > Anybody out there with experience with both? Are there any differences? >> Will Debian be "better" due to FreeSWITCH primarily targeting Linux? >> > >> > We're very slightly leaning towards FreeBSD due to jails, ZFS and other >> nice FreeBSD tools, but we're not averse to Debian in any way. We're >> probably at 52% FreeBSD and 48% Debian. >> > >> > The instances will run on hardware with decent specs. No less than 16 >> cores (probably 32) and no less than 32GB RAM and SSD in every single bay. >> We're not targeting low spec. >> > >> > Any and all advice is welcomed with open arms. >> > >> > :o) >> > Thomas L?cke >> > >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://confluence.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Guillermo Ruiz Camauer > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20141212/c0e51a34/attachment.html From mike at jerris.com Sat Dec 13 00:37:42 2014 From: mike at jerris.com (Michael Jerris) Date: Fri, 12 Dec 2014 16:37:42 -0500 Subject: [Freeswitch-users] SIP Header Modification / Manipulation In-Reply-To: <0C78DA91-56C6-4901-B1BA-676257133525@gmail.com> References: <29A45828-2427-492A-B18E-9260B4880745@gmail.com> <0C78DA91-56C6-4901-B1BA-676257133525@gmail.com> Message-ID: <6F5A84DB-2678-4BA7-80C3-05E62F19B271@jerris.com> if you are looking to update the display of the number that is being called on the caller phone, you want to be looking at display udpates, and callee_id_name and callee_id_number params. > On Dec 12, 2014, at 2:39 PM, Ahmed Habiba wrote: > > Dears, > > Kindly help me on how can I modify the SIP header in the re-invite in the below scenario: > > > [A Number] [FreeSWitch][B Number] > > > 1-A Number Call B Number (Sip Invite sent from A number) > 2-B number is not registered. > 3-Call logic let FreeSwitch answer the call( Freeswitch send re-invite to A number) > > in the part number 3 I like to do modification on the SIP headers, by for example changing the From or To. > > > Your help will be appreciated. > From anthony.minessale at gmail.com Sat Dec 13 00:43:49 2014 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 12 Dec 2014 15:43:49 -0600 Subject: [Freeswitch-users] Record sound quality not good but pcap is quite good In-Reply-To: References: Message-ID: Try latest master. On Fri, Dec 12, 2014 at 8:54 AM, Ali Pey wrote: > > Hello, > > I use record to record part of a call and sometimes I get some static > noises and distortion; however when I listen to the call on pcap capture, > it's quite clear and no noise. > > I record to a wav file and then use sox to convert it to ulaw. > > Any suggestions? Any help would be greatly appreciated. > > > Thanks, > Ali Pey > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 ? @anthmfs ? @FreeSWITCH ? ? http://freeswitch.org/ ? http://cluecon.com/ ? http://twitter.com/FreeSWITCH ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ * ClueCon Weekly Development Call ? sip:888 at conference.freeswitch.org ? +19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141212/ee227644/attachment-0001.html From blasterjr at gmail.com Sat Dec 13 03:48:22 2014 From: blasterjr at gmail.com (Chris Tunbridge) Date: Fri, 12 Dec 2014 17:48:22 -0700 Subject: [Freeswitch-users] mkdir not working In-Reply-To: <548AF0A8.90909@inteli-sim.com> References: <548AF0A8.90909@inteli-sim.com> Message-ID: You could try using something like the following instead. $con->execute("system", "mkdir -p $vmail_base"); but i'm not certain as to where this is being called. On Fri, Dec 12, 2014 at 6:42 AM, kthofer wrote: > > Hi there > we try to create a new directory > $con->execute("mkdir", "$vmail_base"); > to store some Voice mail files > > But for some reason FS is not even trying to create the directory > we checked the user rights > the spelling > but still nothing not even in the linux logs we can not see that FS is > trying it atall. > > Any suggestions?? > what is going wrong? > Please let me know if you have a solution or workaround. > > -- > With best regards > > Karl Theo Hofer > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141212/09988837/attachment.html From chad at apartmentlines.com Sat Dec 13 08:13:41 2014 From: chad at apartmentlines.com (Chad Phillips) Date: Fri, 12 Dec 2014 22:13:41 -0700 Subject: [Freeswitch-users] LUA ESL Help In-Reply-To: References: Message-ID: The paths listed in the error message are the paths that Lua looks in for the module. So you either need to put your ESL.so in one of those locations, or add its location to path list. Check http://www.lua.org/pil/8.1.html for detailed info on how Lua does path handling. On Friday, December 12, 2014 at 12:20 PM, spros p wrote: > Hello, > > I am trying to run a simple ESL script using Lua. I have installed Lua 5.1 and the respective dev packages on my linux machine. Make luamod in /libs/esl/ directory works just fine. > > However when executing: freeswitch/libs/esl/lua/single_command.lua, i receive the following: > > lua: lua/single_command.lua:2: module 'ESL' not found: > no field package.preload['ESL'] > no file './ESL.lua' > no file '/usr/share/lua/5.1/ESL.lua' > no file '/usr/share/lua/5.1/ESL/init.lua' > no file '/usr/lib64/lua/5.1/ESL.lua' > no file '/usr/lib64/lua/5.1/ESL/init.lua' > no file './ESL.so' > no file '/usr/lib64/lua/5.1/ESL.so' > no file '/usr/lib64/lua/5.1/loadall.so' > stack traceback: > [C]: in function 'require' > lua/single_command.lua:2: in main chunk > [C]: ? > > Any help would be greatly appreciated > > Best, > SP > > > _________________________________________________________________________ > 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://confluence.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/20141212/0b4d9f07/attachment.html From omortimer at gmail.com Sat Dec 13 11:42:29 2014 From: omortimer at gmail.com (Oz Mortimer) Date: Sat, 13 Dec 2014 08:42:29 +0000 Subject: [Freeswitch-users] mkdir not working In-Reply-To: References: <548AF0A8.90909@inteli-sim.com> Message-ID: <4D344D60-43F9-47EE-92D6-246697C07ADB@gmail.com> Assuming your using php, you don't need to use freeswitch. system("mkdir -p $vmail_base"); Making sure that the script owner has permissions to create this directory. > On 13 Dec 2014, at 00:48, Chris Tunbridge wrote: > > You could try using something like the following instead. > > $con->execute("system", "mkdir -p $vmail_base"); > > but i'm not certain as to where this is being called. > >> On Fri, Dec 12, 2014 at 6:42 AM, kthofer wrote: >> Hi there >> we try to create a new directory >> $con->execute("mkdir", "$vmail_base"); >> to store some Voice mail files >> >> But for some reason FS is not even trying to create the directory >> we checked the user rights >> the spelling >> but still nothing not even in the linux logs we can not see that FS is >> trying it atall. >> >> Any suggestions?? >> what is going wrong? >> Please let me know if you have a solution or workaround. >> >> -- >> With best regards >> >> Karl Theo Hofer >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141213/c7314ec8/attachment.html From sudhansh at gmail.com Sat Dec 13 15:08:16 2014 From: sudhansh at gmail.com (Sudhanshu) Date: Sat, 13 Dec 2014 17:38:16 +0530 Subject: [Freeswitch-users] fs_cli hangs In-Reply-To: References: Message-ID: Always use syntax highlighting whenever possible. Have you tried disabling the modules which fail to load in your modules.conf.xml? Ideally freeswitch should handle it, but still, it is worth a shot. Also, What happens when you start freeswitch as a normal process (and not as a daemon)? Do you see a freeswitch command prompt? -- Sudhanshu On Fri, Dec 12, 2014 at 12:06 PM, akhil garg wrote: > > Please see the logs: pastebin : 23717 > > > > regards, > akhil > > On Thu, Dec 11, 2014 at 10:40 PM, Sudhanshu wrote: >> >> Enable debug logging and then check freeswitch.log. >> What happens when you start freeswitch as anormal process (and not as a >> daemon)? >> >> -- >> Sudhanshu >> >> On Thu, Dec 11, 2014 at 2:30 PM, akhil garg wrote: >> >>> running "fs_cli -H 127.0.0.1 -P 8021 -d 7" gives different outputs but >>> no success. >>> >>> >>> >>> ------------------------------------------------------------------------------------------------------------------------------------------------ >>> OUTPUT 1: >>> >>> ------------------------------------------------------------------------------------------------------------------------------------------------ >>> [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is >>> /root/.fs_cli_conf. >>> [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is >>> /etc/fs_cli.conf. >>> [DEBUG] fs_cli.c:1438 main() profile default does not exist using >>> builtin profile >>> [DEBUG] fs_cli.c:1468 main() Using profile internal [127.0.0.1] >>> >>> ------------------------------------------------------------------------------------------------------------------------------------------------ >>> >>> >>> >>> >>> >>> ------------------------------------------------------------------------------------------------------------------------------------------------ >>> OUTPUT 2: >>> >>> ------------------------------------------------------------------------------------------------------------------------------------------------ >>> [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is >>> /root/.fs_cli_conf. >>> [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is >>> /etc/fs_cli.conf. >>> [DEBUG] fs_cli.c:1438 main() profile default does not exist using >>> builtin profile >>> [DEBUG] fs_cli.c:1468 main() Using profile internal [127.0.0.1] >>> [DEBUG] esl.c:1265 esl_recv_event() RECV HEADER [Content-Type] = >>> [auth/request] >>> [DEBUG] esl.c:1437 esl_recv_event() RECV MESSAGE >>> Event-Name: SOCKET_DATA >>> Content-Type: auth/request >>> >>> >>> [DEBUG] esl.c:1465 esl_send() SEND >>> auth ClueCon >>> >>> ------------------------------------------------------------------------------------------------------------------------------------------------ >>> >>> >>> >>> >>> >>> ------------------------------------------------------------------------------------------------------------------------------------------------ >>> OUTPUT 3: >>> >>> ------------------------------------------------------------------------------------------------------------------------------------------------ >>> [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is >>> /root/.fs_cli_conf. >>> [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is >>> /etc/fs_cli.conf. >>> [DEBUG] fs_cli.c:1438 main() profile default does not exist using >>> builtin profile >>> [DEBUG] fs_cli.c:1468 main() Using profile internal [127.0.0.1] >>> [DEBUG] esl.c:1265 esl_recv_event() RECV HEADER [Content-Type] = >>> [auth/request] >>> [DEBUG] esl.c:1437 esl_recv_event() RECV MESSAGE >>> Event-Name: SOCKET_DATA >>> Content-Type: auth/request >>> >>> >>> [DEBUG] esl.c:1465 esl_send() SEND >>> auth ClueCon >>> >>> >>> [DEBUG] esl.c:1265 esl_recv_event() RECV HEADER [Content-Type] = >>> [command/reply] >>> [DEBUG] esl.c:1265 esl_recv_event() RECV HEADER [Reply-Text] = [+OK >>> accepted] >>> [DEBUG] esl.c:1437 esl_recv_event() RECV MESSAGE >>> Event-Name: SOCKET_DATA >>> Content-Type: command/reply >>> Reply-Text: +OK accepted >>> >>> >>> [DEBUG] esl.c:1465 esl_send() SEND >>> log >>> >>> ------------------------------------------------------------------------------------------------------------------------------------------------ >>> >>> >>> >>> -- >>> regards, >>> akhil >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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, > akhil > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141213/edd9b8a2/attachment-0001.html From karl-theo_hofer at inteli-sim.com Sat Dec 13 16:57:46 2014 From: karl-theo_hofer at inteli-sim.com (kthofer) Date: Sat, 13 Dec 2014 14:57:46 +0100 Subject: [Freeswitch-users] mkdir not working In-Reply-To: <4D344D60-43F9-47EE-92D6-246697C07ADB@gmail.com> References: <548AF0A8.90909@inteli-sim.com> <4D344D60-43F9-47EE-92D6-246697C07ADB@gmail.com> Message-ID: Hi there i am using perl Will try the suggestions But still strange that FS is not even trying to create a directory With best regards Karl Theo Hofer > Am 13 dec 2014 um 09:42 schrieb Oz Mortimer : > > Assuming your using php, you don't need to use freeswitch. > system("mkdir -p $vmail_base"); > > Making sure that the script owner has permissions to create this directory. > > > >> On 13 Dec 2014, at 00:48, Chris Tunbridge wrote: >> >> You could try using something like the following instead. >> >> $con->execute("system", "mkdir -p $vmail_base"); >> >> but i'm not certain as to where this is being called. >> >>> On Fri, Dec 12, 2014 at 6:42 AM, kthofer wrote: >>> Hi there >>> we try to create a new directory >>> $con->execute("mkdir", "$vmail_base"); >>> to store some Voice mail files >>> >>> But for some reason FS is not even trying to create the directory >>> we checked the user rights >>> the spelling >>> but still nothing not even in the linux logs we can not see that FS is >>> trying it atall. >>> >>> Any suggestions?? >>> what is going wrong? >>> Please let me know if you have a solution or workaround. >>> >>> -- >>> With best regards >>> >>> Karl Theo Hofer >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141213/29337066/attachment.html From max at nysolutions.com Sun Dec 14 03:04:40 2014 From: max at nysolutions.com (Moishe Grunstein) Date: Sun, 14 Dec 2014 00:04:40 +0000 Subject: [Freeswitch-users] How to play a stream other than mp3 with mod_shout. In-Reply-To: References: Message-ID: You can try mod_vlc Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support at nysolutions.com Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [cid:image001.jpg at 01C72F94.9EE45D60] Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Aqs Younas Sent: Friday, December 12, 2014 2:38 PM To: FreeSWITCH Users Help Subject: [Freeswitch-users] How to play a stream other than mp3 with mod_shout. Hi, All How can i play a live stream other than mp3 with mod_shout or any module.? Is there any way to buffer the stream before playing it with mod_shout. Currently i have a list of streams and when i play them with mod_shout some work fine but others give (time out) error. How can i play mostly stream in freeswitch? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141214/89218a5a/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 2424 bytes Desc: image001.jpg Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141214/89218a5a/attachment.jpg From aqsyounas at gmail.com Sun Dec 14 17:46:35 2014 From: aqsyounas at gmail.com (aqs younas) Date: Sun, 14 Dec 2014 19:46:35 +0500 Subject: [Freeswitch-users] How to play a stream other than mp3 withmod_shout. In-Reply-To: References: Message-ID: <548da2cb.85f0c20a.2718.1cb2@mx.google.com> Many thanks for your reply.I will give it a try. -----Original Message----- From: "Moishe Grunstein" Sent: ?12/?14/?2014 5:05 AM To: "FreeSWITCH Users Help" Subject: Re: [Freeswitch-users] How to play a stream other than mp3 withmod_shout. You can try mod_vlc Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support at nysolutions.com Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Aqs Younas Sent: Friday, December 12, 2014 2:38 PM To: FreeSWITCH Users Help Subject: [Freeswitch-users] How to play a stream other than mp3 with mod_shout. Hi, All How can i play a live stream other than mp3 with mod_shout or any module.? Is there any way to buffer the stream before playing it with mod_shout. Currently i have a list of streams and when i play them with mod_shout some work fine but others give (time out) error. How can i play mostly stream in freeswitch? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141214/20b0e8e6/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 2424 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141214/20b0e8e6/attachment-0001.jpg From danny.gershman at gmail.com Sun Dec 14 22:37:56 2014 From: danny.gershman at gmail.com (Danny Gershman) Date: Sun, 14 Dec 2014 14:37:56 -0500 Subject: [Freeswitch-users] How to play a stream other than mp3 with mod_shout. In-Reply-To: References: Message-ID: Also mod_rtmp lets you play from an FMS server. On Friday, December 12, 2014, Aqs Younas wrote: > Hi, All > > How can i play a live stream other than mp3 with mod_shout or any module.? > Is there any way to buffer the stream before playing it with mod_shout. > > > Currently i have a list of streams and when i play them with mod_shout > some work fine but others give (time out) error. > > How can i play mostly stream in freeswitch? > > Thanks > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141214/f31cfec1/attachment.html From steveayre at gmail.com Mon Dec 15 03:32:03 2014 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 15 Dec 2014 00:32:03 +0000 Subject: [Freeswitch-users] Invalid codec CN is tearing down the call request In-Reply-To: References: Message-ID: That shouldn't really matter though should it? IIRC static payload types can be used for codecs other than the one statically assigned by IANA if a=rtpmap is given... so "a=rtpmap:19 CN/8000" should allow 19 to be CN, overriding the default 19 assignment (which is 'Reserved' apparently) On 11 December 2014 at 18:07, Brian West wrote: > > Funny, payload 13 is CN, which is 0x13 in base16 or 19 in base10, Please > file a JIRA, I would need to see a packet capture for this one. > > > On Thu, Dec 11, 2014 at 11:47 AM, Maciej Bylica wrote: > >> Sure i'll try with jira >> >> Mac. >> >> 2014-12-11 17:41 GMT+01:00 Anthony Minessale > >: >> >>> For years I beg ppl to post issues to jira not the mailing list. >>> if you start typing and keywords like "problem" "not working" "issue" >>> appear in your text, you probably should be filing it to JIRA not here. >>> >>> >>> On Thu, Dec 11, 2014 at 10:37 AM, Maciej Bylica wrote: >>> >>>> Thanks for prompt reply. >>>> >>>> Steven, later today i will try to update FS to the latest git and check >>>> this out. >>>> >>>> Ken, i am fine with CN, passing thru is what i really need, but the >>>> question is why FS generates such ERRors and how to get out of this. >>>> >>>> Thanks >>>> Mac. >>>> >>>> 2014-12-11 17:12 GMT+01:00 Ken Rice : >>>> >>>>> CN is comfort noise... Your cisco is sending it, and freeswitch is >>>>> passing it thru because you have told it to use proxy media mode... >>>>> >>>>> >>>>> >>>>> On 12/11/14 9:53 AM, "Steven Ayre" wrote: >>>>> >>>>> Your version is 3 months old. Can you reproduce this on the latest git >>>>> head? >>>>> >>>>> On 11 December 2014 at 13:45, Maciej Bylica wrote: >>>>> >>>>> Hello, >>>>> >>>>> I am running FreeSWITCH Version >>>>> 1.5.14b+git~20140917T231120Z~8f85b5204c~64bit, proxy media mode and heaving >>>>> problem with some call requests setup. >>>>> Here is one of them: >>>>> >>>>> 2014-12-11 13:38:15.034612 [NOTICE] switch_channel.c:1055 New Channel >>>>> sofia/outside_1/20049112223344 [] >>>>> 2014-12-11 13:38:15.034612 [DEBUG] mod_sofia.c:4579 >>>>> (sofia/outside_1/20049112223344) State Change CS_NEW -> CS_INIT >>>>> 2014-12-11 13:38:15.034612 [DEBUG] switch_core_session.c:1388 Send >>>>> signal sofia/outside_1/20049112223344 [BREAK] >>>>> 2014-12-11 13:38:15.054610 [DEBUG] switch_core_state_machine.c:472 >>>>> (sofia/outside_1/20049112223344) Running State Change CS_INIT >>>>> 2014-12-11 13:38:15.054610 [DEBUG] switch_core_state_machine.c:512 >>>>> (sofia/outside_1/20049112223344) State INIT >>>>> 2014-12-11 13:38:15.054610 [DEBUG] mod_sofia.c:87 >>>>> sofia/outside_1/20049112223344 SOFIA INIT >>>>> 2014-12-11 13:38:15.054610 [DEBUG] switch_core_media.c:7510 >>>>> sofia/outside_1/20049112223344 Patched SDP >>>>> --- >>>>> v=0 >>>>> o=CiscoSystemsSIP-GW-UserAgent 6016 7716 IN IP4 10.10.10.226 >>>>> s=SIP Call >>>>> c=IN IP4 10.10.10.12 >>>>> t=0 0 >>>>> m=audio 24782 RTP/AVP 18 19 >>>>> c=IN IP4 10.10.10.12 >>>>> a=rtpmap:18 G729/8000 >>>>> a=fmtp:18 annexb=yes >>>>> a=rtpmap:19 CN/8000 >>>>> >>>>> +++ >>>>> v=0 >>>>> o=FreeSWITCH 0255539876 0255539877 IN IP4 10.10.10.166 >>>>> s=FreeSWITCH >>>>> c=IN IP4 10.10.10.166 >>>>> t=0 0 >>>>> m=audio 19690 RTP/AVP 18 19 >>>>> c=IN IP4 10.10.10.166 >>>>> a=rtpmap:18 G729/8000 >>>>> a=fmtp:18 annexb=yes >>>>> a=rtpmap:19 CN/8000 >>>>> >>>>> 2014-12-11 13:38:15.054610 [DEBUG] sofia_glue.c:1228 >>>>> sofia/outside_1/20049112223344 sending invite version: 1.5.14b git 8f85b52 >>>>> 2014-09-17 23:11:20Z 64bit >>>>> Local SDP: >>>>> v=0 >>>>> o=FreeSWITCH 0255539876 0255539877 IN IP4 10.10.10.166 >>>>> s=FreeSWITCH >>>>> c=IN IP4 10.10.10.166 >>>>> t=0 0 >>>>> m=audio 19690 RTP/AVP 18 19 >>>>> c=IN IP4 10.10.10.166 >>>>> a=rtpmap:18 G729/8000 >>>>> a=fmtp:18 annexb=yes >>>>> a=rtpmap:19 CN/8000 >>>>> >>>>> sending out SDP exactly as shown above. >>>>> The other side is responsing with 100, then 180 Ringing wo/SDP then >>>>> 183 w/SDP >>>>> >>>>> 2014-12-11 14:29:12.214608 [DEBUG] sofia.c:6423 Channel >>>>> sofia/outside_1/20049112223344 entering state [proceeding][183] >>>>> 2014-12-11 14:29:12.214608 [DEBUG] sofia.c:6433 Remote SDP: >>>>> v=0 >>>>> o=Dialogic_SDP 11471558 0 IN IP4 10.10.10.218 >>>>> s=Dialogic-SIP >>>>> c=IN IP4 10.10.10.198 >>>>> t=0 0 >>>>> m=audio 10024 RTP/AVP 19 18 >>>>> a=rtpmap:19 CN/8000 >>>>> a=rtpmap:18 G729/8000 >>>>> a=fmtp:18 annexb=yes >>>>> a=silenceSupp:off - - - - >>>>> a=ptime:20 >>>>> >>>>> 2014-12-11 14:29:12.214608 [DEBUG] switch_core_media.c:7510 >>>>> sofia/outside_1/20049112223344 Patched SDP >>>>> --- >>>>> v=0 >>>>> o=FreeSWITCH 0351957506 0351957507 IN IP4 10.10.10.166 >>>>> s=FreeSWITCH >>>>> c=IN IP4 10.10.10.166 >>>>> t=0 0 >>>>> m=audio 29492 RTP/AVP 18 19 >>>>> c=IN IP4 10.10.10.166 >>>>> a=rtpmap:18 G729/8000 >>>>> a=fmtp:18 annexb=yes >>>>> a=rtpmap:19 CN/8000 >>>>> >>>>> +++ >>>>> v=0 >>>>> o=FreeSWITCH 0351957506 0351957508 IN IP4 10.10.10.166 >>>>> s=FreeSWITCH >>>>> c=IN IP4 10.10.10.166 >>>>> t=0 0 >>>>> m=audio 29492 RTP/AVP 18 19 >>>>> c=IN IP4 10.10.10.166 >>>>> a=rtpmap:18 G729/8000 >>>>> a=fmtp:18 annexb=yes >>>>> a=rtpmap:19 CN/8000 >>>>> >>>>> 2014-12-11 14:29:12.214608 [ERR] switch_core_codec.c:651 >>>>> *Invalid codec CN! *2014-12-11 14:29:12.214608 [ERR] >>>>> switch_core_media.c:2294 >>>>> *Can't load codec? *2014-12-11 14:29:12.214608 [NOTICE] >>>>> switch_core_media.c:2295 Hangup sofia/outside_1/20049112223344 >>>>> [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] >>>>> 2014-12-11 14:29:12.214608 [DEBUG] switch_channel.c:3222 Send signal >>>>> sofia/outside_1/20049112223344 [KILL] >>>>> >>>>> and Freeswitch is terminating the call by using CANCEL. >>>>> >>>>> My modules.conf.xml config part looks like following: >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> Is it a problem with CN definition? >>>>> Could you please lead me where the problem is located? >>>>> >>>>> Thanks in advance. >>>>> Mac. >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> ------------------------------ >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at 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 >>>>> Twitter: @FreeSWITCH >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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 ? @anthmfs ? @FreeSWITCH ? >>> >>> ? http://freeswitch.org/ ? http://cluecon.com/ ? >>> http://twitter.com/FreeSWITCH >>> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ >>> * >>> >>> ClueCon Weekly Development Call >>> ? sip:888 at conference.freeswitch.org ? +19193869900 >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > > *Brian West* > brian at freeswitch.org > > > *Twitter: @FreeSWITCH , @briankwest* > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141215/c3bb629f/attachment-0001.html From anthony.minessale at gmail.com Mon Dec 15 04:18:19 2014 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 14 Dec 2014 19:18:19 -0600 Subject: [Freeswitch-users] Invalid codec CN is tearing down the call request In-Reply-To: References: Message-ID: It doesn't matter. Also bug should be fixed, it was not related to cn. On Sunday, December 14, 2014, Steven Ayre wrote: > That shouldn't really matter though should it? IIRC static payload types > can be used for codecs other than the one statically assigned by IANA if > a=rtpmap is given... so "a=rtpmap:19 CN/8000" should allow 19 to be CN, > overriding the default 19 assignment (which is 'Reserved' apparently) > > > On 11 December 2014 at 18:07, Brian West > wrote: >> >> Funny, payload 13 is CN, which is 0x13 in base16 or 19 in base10, Please >> file a JIRA, I would need to see a packet capture for this one. >> >> >> On Thu, Dec 11, 2014 at 11:47 AM, Maciej Bylica > > wrote: >> >>> Sure i'll try with jira >>> >>> Mac. >>> >>> 2014-12-11 17:41 GMT+01:00 Anthony Minessale < >>> anthony.minessale at gmail.com >>> >: >>> >>>> For years I beg ppl to post issues to jira not the mailing list. >>>> if you start typing and keywords like "problem" "not working" "issue" >>>> appear in your text, you probably should be filing it to JIRA not here. >>>> >>>> >>>> On Thu, Dec 11, 2014 at 10:37 AM, Maciej Bylica >>> > wrote: >>>> >>>>> Thanks for prompt reply. >>>>> >>>>> Steven, later today i will try to update FS to the latest git and >>>>> check this out. >>>>> >>>>> Ken, i am fine with CN, passing thru is what i really need, but the >>>>> question is why FS generates such ERRors and how to get out of this. >>>>> >>>>> Thanks >>>>> Mac. >>>>> >>>>> 2014-12-11 17:12 GMT+01:00 Ken Rice >>>> >: >>>>> >>>>>> CN is comfort noise... Your cisco is sending it, and freeswitch is >>>>>> passing it thru because you have told it to use proxy media mode... >>>>>> >>>>>> >>>>>> >>>>>> On 12/11/14 9:53 AM, "Steven Ayre" wrote: >>>>>> >>>>>> Your version is 3 months old. Can you reproduce this on the latest >>>>>> git head? >>>>>> >>>>>> On 11 December 2014 at 13:45, Maciej Bylica wrote: >>>>>> >>>>>> Hello, >>>>>> >>>>>> I am running FreeSWITCH Version >>>>>> 1.5.14b+git~20140917T231120Z~8f85b5204c~64bit, proxy media mode and heaving >>>>>> problem with some call requests setup. >>>>>> Here is one of them: >>>>>> >>>>>> 2014-12-11 13:38:15.034612 [NOTICE] switch_channel.c:1055 New Channel >>>>>> sofia/outside_1/20049112223344 [] >>>>>> 2014-12-11 13:38:15.034612 [DEBUG] mod_sofia.c:4579 >>>>>> (sofia/outside_1/20049112223344) State Change CS_NEW -> CS_INIT >>>>>> 2014-12-11 13:38:15.034612 [DEBUG] switch_core_session.c:1388 Send >>>>>> signal sofia/outside_1/20049112223344 [BREAK] >>>>>> 2014-12-11 13:38:15.054610 [DEBUG] switch_core_state_machine.c:472 >>>>>> (sofia/outside_1/20049112223344) Running State Change CS_INIT >>>>>> 2014-12-11 13:38:15.054610 [DEBUG] switch_core_state_machine.c:512 >>>>>> (sofia/outside_1/20049112223344) State INIT >>>>>> 2014-12-11 13:38:15.054610 [DEBUG] mod_sofia.c:87 >>>>>> sofia/outside_1/20049112223344 SOFIA INIT >>>>>> 2014-12-11 13:38:15.054610 [DEBUG] switch_core_media.c:7510 >>>>>> sofia/outside_1/20049112223344 Patched SDP >>>>>> --- >>>>>> v=0 >>>>>> o=CiscoSystemsSIP-GW-UserAgent 6016 7716 IN IP4 10.10.10.226 >>>>>> s=SIP Call >>>>>> c=IN IP4 10.10.10.12 >>>>>> t=0 0 >>>>>> m=audio 24782 RTP/AVP 18 19 >>>>>> c=IN IP4 10.10.10.12 >>>>>> a=rtpmap:18 G729/8000 >>>>>> a=fmtp:18 annexb=yes >>>>>> a=rtpmap:19 CN/8000 >>>>>> >>>>>> +++ >>>>>> v=0 >>>>>> o=FreeSWITCH 0255539876 0255539877 IN IP4 10.10.10.166 >>>>>> s=FreeSWITCH >>>>>> c=IN IP4 10.10.10.166 >>>>>> t=0 0 >>>>>> m=audio 19690 RTP/AVP 18 19 >>>>>> c=IN IP4 10.10.10.166 >>>>>> a=rtpmap:18 G729/8000 >>>>>> a=fmtp:18 annexb=yes >>>>>> a=rtpmap:19 CN/8000 >>>>>> >>>>>> 2014-12-11 13:38:15.054610 [DEBUG] sofia_glue.c:1228 >>>>>> sofia/outside_1/20049112223344 sending invite version: 1.5.14b git 8f85b52 >>>>>> 2014-09-17 23:11:20Z 64bit >>>>>> Local SDP: >>>>>> v=0 >>>>>> o=FreeSWITCH 0255539876 0255539877 IN IP4 10.10.10.166 >>>>>> s=FreeSWITCH >>>>>> c=IN IP4 10.10.10.166 >>>>>> t=0 0 >>>>>> m=audio 19690 RTP/AVP 18 19 >>>>>> c=IN IP4 10.10.10.166 >>>>>> a=rtpmap:18 G729/8000 >>>>>> a=fmtp:18 annexb=yes >>>>>> a=rtpmap:19 CN/8000 >>>>>> >>>>>> sending out SDP exactly as shown above. >>>>>> The other side is responsing with 100, then 180 Ringing wo/SDP then >>>>>> 183 w/SDP >>>>>> >>>>>> 2014-12-11 14:29:12.214608 [DEBUG] sofia.c:6423 Channel >>>>>> sofia/outside_1/20049112223344 entering state [proceeding][183] >>>>>> 2014-12-11 14:29:12.214608 [DEBUG] sofia.c:6433 Remote SDP: >>>>>> v=0 >>>>>> o=Dialogic_SDP 11471558 0 IN IP4 10.10.10.218 >>>>>> s=Dialogic-SIP >>>>>> c=IN IP4 10.10.10.198 >>>>>> t=0 0 >>>>>> m=audio 10024 RTP/AVP 19 18 >>>>>> a=rtpmap:19 CN/8000 >>>>>> a=rtpmap:18 G729/8000 >>>>>> a=fmtp:18 annexb=yes >>>>>> a=silenceSupp:off - - - - >>>>>> a=ptime:20 >>>>>> >>>>>> 2014-12-11 14:29:12.214608 [DEBUG] switch_core_media.c:7510 >>>>>> sofia/outside_1/20049112223344 Patched SDP >>>>>> --- >>>>>> v=0 >>>>>> o=FreeSWITCH 0351957506 0351957507 IN IP4 10.10.10.166 >>>>>> s=FreeSWITCH >>>>>> c=IN IP4 10.10.10.166 >>>>>> t=0 0 >>>>>> m=audio 29492 RTP/AVP 18 19 >>>>>> c=IN IP4 10.10.10.166 >>>>>> a=rtpmap:18 G729/8000 >>>>>> a=fmtp:18 annexb=yes >>>>>> a=rtpmap:19 CN/8000 >>>>>> >>>>>> +++ >>>>>> v=0 >>>>>> o=FreeSWITCH 0351957506 0351957508 IN IP4 10.10.10.166 >>>>>> s=FreeSWITCH >>>>>> c=IN IP4 10.10.10.166 >>>>>> t=0 0 >>>>>> m=audio 29492 RTP/AVP 18 19 >>>>>> c=IN IP4 10.10.10.166 >>>>>> a=rtpmap:18 G729/8000 >>>>>> a=fmtp:18 annexb=yes >>>>>> a=rtpmap:19 CN/8000 >>>>>> >>>>>> 2014-12-11 14:29:12.214608 [ERR] switch_core_codec.c:651 >>>>>> *Invalid codec CN! *2014-12-11 14:29:12.214608 [ERR] >>>>>> switch_core_media.c:2294 >>>>>> *Can't load codec? *2014-12-11 14:29:12.214608 [NOTICE] >>>>>> switch_core_media.c:2295 Hangup sofia/outside_1/20049112223344 >>>>>> [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] >>>>>> 2014-12-11 14:29:12.214608 [DEBUG] switch_channel.c:3222 Send signal >>>>>> sofia/outside_1/20049112223344 [KILL] >>>>>> >>>>>> and Freeswitch is terminating the call by using CANCEL. >>>>>> >>>>>> My modules.conf.xml config part looks like following: >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Is it a problem with CN definition? >>>>>> Could you please lead me where the problem is located? >>>>>> >>>>>> Thanks in advance. >>>>>> Mac. >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://confluence.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>>> ------------------------------ >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://confluence.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at 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 >>>>>> Twitter: @FreeSWITCH >>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://confluence.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at 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 ? @anthmfs ? @FreeSWITCH ? >>>> >>>> ? http://freeswitch.org/ ? http://cluecon.com/ ? >>>> http://twitter.com/FreeSWITCH >>>> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ >>>> * >>>> >>>> ClueCon Weekly Development Call >>>> ? sip:888 at conference.freeswitch.org >>>> ? >>>> +19193869900 >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> >> *Brian West* >> brian at freeswitch.org >> >> >> >> *Twitter: @FreeSWITCH , @briankwest* >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 ? @anthmfs ? @FreeSWITCH ? ? http://freeswitch.org/ ? http://cluecon.com/ ? http://twitter.com/FreeSWITCH ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ * ClueCon Weekly Development Call ? sip:888 at conference.freeswitch.org ? +19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141214/4e21b103/attachment-0001.html From sphotos1982 at gmail.com Sat Dec 13 15:46:36 2014 From: sphotos1982 at gmail.com (spros p) Date: Sat, 13 Dec 2014 14:46:36 +0200 Subject: [Freeswitch-users] LUA ESL Help In-Reply-To: References: Message-ID: Thanks Chad - I don't think copying the ESL.so in the paths is the correct approach for resolving. I have in any case made a symbolic link of the ESL.so file to both /usr/share/lua/5.1 and /usr/lib64/lua/5.1 with no avail. My suspicion is that LUA_PATH and LUA_CPATH need to be defined with ESL details. This is fine for the CPATH, as i know where ESL.so resides. However, I cannot define PATH, as there doesn't seem to be a ESL.lua file. Best, Spyros On Sat, Dec 13, 2014 at 7:13 AM, Chad Phillips wrote: > > The paths listed in the error message are the paths that Lua looks in for > the module. So you either need to put your ESL.so in one of those > locations, or add its location to path list. Check > http://www.lua.org/pil/8.1.html for detailed info on how Lua does path > handling. > > On Friday, December 12, 2014 at 12:20 PM, spros p wrote: > > Hello, > > I am trying to run a simple ESL script using Lua. I have installed Lua 5.1 > and the respective dev packages on my linux machine. Make luamod in > /libs/esl/ directory works just fine. > > However when executing: freeswitch/libs/esl/lua/single_command.lua, i > receive the following: > > lua: lua/single_command.lua:2: module 'ESL' not found: > no field package.preload['ESL'] > no file './ESL.lua' > no file '/usr/share/lua/5.1/ESL.lua' > no file '/usr/share/lua/5.1/ESL/init.lua' > no file '/usr/lib64/lua/5.1/ESL.lua' > no file '/usr/lib64/lua/5.1/ESL/init.lua' > no file './ESL.so' > no file '/usr/lib64/lua/5.1/ESL.so' > no file '/usr/lib64/lua/5.1/loadall.so' > stack traceback: > [C]: in function 'require' > lua/single_command.lua:2: in main chunk > [C]: ? > > Any help would be greatly appreciated > > Best, > SP > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141213/e1082e5b/attachment.html From michael.traut at gmail.com Mon Dec 15 04:17:20 2014 From: michael.traut at gmail.com (Michael Traut) Date: Mon, 15 Dec 2014 02:17:20 +0100 Subject: [Freeswitch-users] Freeswitch based door communication Message-ID: Hi, I'm trying to build a Freeswitch / Raspberry based door communication. While there are some examples for such a constellation for Asterisk, i didn't find (and was not able to invent myself so far) a Freeswitch example. The target scenario: * A SIP based door module, * 1-n Pushbuttons, one for each floor * 1-n SIP Phones per floor When a push button is pressed i want to initiate a call to the door (that will be accepted automatically) and a call to all phones on the floor. The first one to answer gets connected to the door. Is this something i can do with "originate" ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141215/ca3f961f/attachment.html From joelewhite at gmail.com Mon Dec 15 09:13:28 2014 From: joelewhite at gmail.com (Joel White) Date: Mon, 15 Dec 2014 01:13:28 -0500 Subject: [Freeswitch-users] Caller ID - Not Defined when executing dialplan, although Lua script is pulling CID from Database In-Reply-To: References: <9312DD37-1999-4B29-A2C0-BC8429C6B866@jerris.com> Message-ID: I will implement this tomorrow. I do have a question though, with all that being said.... the system I have running in the cloud does send out the CID specified in the Database using this methodology. Why would it work on one and not the other when their configs are identical? Just curious On Thu, Dec 11, 2014 at 1:51 PM, Chris Tunbridge wrote: > > Errr yeah, sorry Michael is right they're origination not effective :P > > On Thu, Dec 11, 2014 at 8:34 AM, Michael Jerris wrote: > >> you should be looking at the origination_caller_id_* vars, not effective >> >> On Dec 11, 2014, at 10:26 AM, Chris Tunbridge >> wrote: >> >> The variable outbound_caller_id_number and outbound_caller_id_name are >> not related to the caller id on outbound calls. >> >> On your outbound dial plan you need to set something like the following >> >> > data="effective_caller_id_name=${user_data(${username}@${domain_name} >> var outbound_caller_id_name)}"/> >> > data="effective_caller_id_number=${user_data(${username}@${domain_name} >> var outbound_caller_id_number)}"/> >> >> This will cause the system to pull the settings from the users profile >> and use it for the outgoing call. >> >> On Wed, Dec 10, 2014 at 12:27 PM, Joel White >> wrote: >> >>> I have gone over the config with a fine tooth comb, it matches another >>> config of a server in which the caller id works fine. What I am seeing >>> however is that in this system the variable is not exported to the >>> dialplan. I may be missing something, and most likely I am. I do have a >>> question though. Is there a way to see what variables are defined for a >>> particular user in the FreeSWITCH console? >>> >>> >>> Here is some output of the Lua script on the server not pushing caller id >>> >>> 2014-12-09 16:54:22.819251 [NOTICE] switch_cpp.cpp:1328 Debug from >>> gen_dir_user_xml.lua, generated XML: >>> >>> >>>
>>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >> value="1321XXXXXXX"/> >>> >>> >>> >>> >>> >>>
>>> >>> >>> >>> And some output from the Lua script on the server with CID functioning >>> >>> 2014-12-09 21:50:55.458996 [NOTICE] switch_cpp.cpp:1328 Debug from >>> gen_dir_user_xml.lua, generated XML: >>> >>> >>>
>>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>>
>>> >>> >>> >>> Of course I removed any identifiable information, but it looks like the >>> CID is being set. What am I missing here that is not allowing for the >>> variable to be passed to the dialplan? >>> >>> >>> This is what I get when the dialplan executes >>> >>> EXECUTE sofia/internal/26342 at voip.net >>> set(effective_caller_id_number=) >>> 2014-12-09 17:00:24.839237 [DEBUG] mod_dptools.c:1435 sofia/internal/ >>> 26342 at voip.net SET [effective_caller_id_number]=[UNDEF] >>> >>> >>> Kinda strange and I could not find a discrepancy between the dialplan >>> configuration of the working server vs the non-working server >>> >>> >>> >>> Here is the version running on the server that works properly >>> >>> FreeSWITCH Version 1.4.13+git~20141103T195300Z~b942d0faa8~64bit (git >>> b942d0f 2014-11-03 19:53:00Z 64bit) >>> >>> And the version of the server having issue with CID >>> >>> FreeSWITCH Version 1.4.13+git~20141103T195300Z~ >>> b942d0faa8~64bit (git b942d0f 2014-11-03 19:53:00Z 64bit) >>> >>> >>> >>> I used diff and compared both servers conf directory recursively. I >>> could not find a discrepancy in the files aside from Switch name, etc. >>> >>> >>> What am I missing? >>> >>> Could there be anything that that would overwrite the CID variable after >>> it is set by Lua (Generating the user profile)? >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141215/0114c658/attachment-0001.html From peter at olssononline.se Mon Dec 15 10:20:45 2014 From: peter at olssononline.se (Peter Olsson) Date: Mon, 15 Dec 2014 08:20:45 +0100 Subject: [Freeswitch-users] mkdir not working In-Reply-To: References: <548AF0A8.90909@inteli-sim.com> <4D344D60-43F9-47EE-92D6-246697C07ADB@gmail.com> Message-ID: If you're not using the "system" command as suggested, it's not that strange really. "mkdir" is not a valid FreeSWITCH command to execute. 2014-12-13 14:57 GMT+01:00 kthofer : > > > Hi there i am using perl > Will try the suggestions > But still strange that FS is not even trying to create a directory > > > With best regards > > Karl Theo Hofer > > Am 13 dec 2014 um 09:42 schrieb Oz Mortimer : > > Assuming your using php, you don't need to use freeswitch. > system("mkdir -p $vmail_base"); > > Making sure that the script owner has permissions to create this directory. > > > > On 13 Dec 2014, at 00:48, Chris Tunbridge wrote: > > You could try using something like the following instead. > > $con->execute("system", "mkdir -p $vmail_base"); > > but i'm not certain as to where this is being called. > > On Fri, Dec 12, 2014 at 6:42 AM, kthofer > wrote: >> >> Hi there >> we try to create a new directory >> $con->execute("mkdir", "$vmail_base"); >> to store some Voice mail files >> >> But for some reason FS is not even trying to create the directory >> we checked the user rights >> the spelling >> but still nothing not even in the linux logs we can not see that FS is >> trying it atall. >> >> Any suggestions?? >> what is going wrong? >> Please let me know if you have a solution or workaround. >> >> -- >> With best regards >> >> Karl Theo Hofer >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141215/014a53cf/attachment.html From mkvonarx at gmail.com Mon Dec 15 11:08:05 2014 From: mkvonarx at gmail.com (Markus von Arx) Date: Mon, 15 Dec 2014 09:08:05 +0100 Subject: [Freeswitch-users] Windows build system In-Reply-To: References: Message-ID: Well, we consider it clean and best practice to put all tools that are required by our build scripts into the repo itself. Our goal is that you can checkout any repo (or any label) on any clean PC that does not even need to have internet access and immediately being able build from the working directory by double clicking something like a build.bat script file without any dependencies to the outside (of the repo). We sometimes make exceptions for huge stuff (like e.g. VS itself) or software that cannot run without running an installer (again VS - but we add the installers to the repo in that case). Working like this has many advantages, e.g. you can use different versions of the same tool on the same machine because these versions are inside the various different repo working directories and not on PATH; also this is a really good archiving method that allows us to build even very old repos and old labels of a product with the tools that have been used n years ago. We do basically the same thing with libraries. Even though we use NuGet now for download libraries and their dependencies, we always add the downloaded libraries to the repo too in order to never being dependent on some outside server that may or may not be online at the time when we want to build our software. The price of this is big (sometimes huge) repositories and working directories of course. But we gladly pay that price to have the advantages. Markus 2014-12-08 18:22 GMT+01:00 Michael Jerris : > > I'm a bit out of date on what these tools provide and need someone to get > me up to speed on what is available. Can you help me understand what the > options are for maintaining c libraries in a way that is cleaner than > needing to stuff them all into our build. > > > On Dec 8, 2014, at 5:29 AM, Markus von Arx wrote: > > We are fine with your Visual Studio plans. We only use VS2012 at the > moment and actually would like to migrate to VS2013. > > Not so sure about the use of chocolatey though. Are you planning to use it > to download libraries required by the build or for installing > software/tools required for the build? Both look a bit problematic to me, > because chocolatey (as far as I understand it) always acts globally on the > target machine, meaning that it installs the libs/tools/software not > locally inside the build directory but globally on the machine. I wouldn't > like that at all. I don't want a build tool/process to install anything > outside the build directory. If you manage to use chocolatey to only work > in the build directory that's fine for me. But I'd strongly vote against > any use of chocolatey to install libraries, tools or any software globally > or outside the build directory. I wouldn't like a build tool/process/system > installing anything on my machine for me automatically. Kind of like > calling apt-get on a Linux machine from a build script. I think that is a > no-go. Also, chocolatey is not so good detecting installed software that > was not installed by chocolatey itself and would often try to re-install > software that is already there. I'd vote against using chocolatey in the > FreeSWITCH build if you ask me. Wouldn't nuget be the more natural choice > anyway to install modules/libraries in Visual Studio? Just my opinion. > > Markus > > > 2014-12-03 21:31 GMT+01:00 Michael Jerris : > >> Given the recent announcements by Microsoft about the community edition >> 2013 being available, we are working to migrate the build system towards >> using that as our primary build. As part of this process we will be very >> soon dropping support for any version of Visual Studio prior to 2012. If >> you feel strongly about needing support for these older versions, please >> speak up now with an offer to maintain these legacy build systems. We are >> also investigating moving to using chocolatey as a new system to manage >> dependencies on windows instead of maintaining the build for all our deps >> ourselves. It is also possible we will drop support for the 2012 build >> system in the not so distant future. Could the community chime in here as >> to what their needs are, and what they are willing to do to help support >> the windows builds so we can determine what we plan to support going >> forward. >> >> Thanks >> Mike >> >> https://chocolatey.org/ >> http://www.visualstudio.com/en-us/news/vs2013-community-vs.aspx >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141215/29c6cf7e/attachment.html From aqsyounas at gmail.com Mon Dec 15 15:03:07 2014 From: aqsyounas at gmail.com (Aqs Younas) Date: Mon, 15 Dec 2014 17:03:07 +0500 Subject: [Freeswitch-users] How to play a stream other than mp3 with mod_shout. In-Reply-To: References: Message-ID: Thanks for your reply. I will try that too. Many thanks for your valuable replies. On 15 December 2014 at 00:37, Danny Gershman wrote: > > Also mod_rtmp lets you play from an FMS server. > > > On Friday, December 12, 2014, Aqs Younas wrote: > >> Hi, All >> >> How can i play a live stream other than mp3 with mod_shout or any >> module.? Is there any way to buffer the stream before playing it with >> mod_shout. >> >> >> Currently i have a list of streams and when i play them with mod_shout >> some work fine but others give (time out) error. >> >> How can i play mostly stream in freeswitch? >> >> Thanks >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141215/d1dbd3d2/attachment-0001.html From paul.atreides83 at googlemail.com Mon Dec 15 14:44:35 2014 From: paul.atreides83 at googlemail.com (Paul Atreides) Date: Mon, 15 Dec 2014 12:44:35 +0100 Subject: [Freeswitch-users] Call transfer with bind_digit_action? Message-ID: During a call I want to type a extension, for example 20, and transfer the current call to this extension. How can I realize that with bind_digit_action? Any help would be appreciated. Thank you Paul -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141215/0793a1d5/attachment.html From chad at apartmentlines.com Mon Dec 15 19:37:20 2014 From: chad at apartmentlines.com (Chad Phillips) Date: Mon, 15 Dec 2014 09:37:20 -0700 Subject: [Freeswitch-users] LUA ESL Help In-Reply-To: References: Message-ID: You shouldn?t need any ESL.lua ? pretty sure that?s just part of the search when ?require ?ESL?? is called, and /usr/lib64/lua/5.1/ESL.so should satisfy that search. Not sure why the symlink isn?t working, as an experiment, did you just put the ESL.so there directly? I?d also try adding to CPATH. On Saturday, December 13, 2014 at 5:46 AM, spros p wrote: > Thanks Chad - I don't think copying the ESL.so in the paths is the correct approach for resolving. I have in any case made a symbolic link of the ESL.so file to both /usr/share/lua/5.1 and /usr/lib64/lua/5.1 with no avail. > > My suspicion is that LUA_PATH and LUA_CPATH need to be defined with ESL details. This is fine for the CPATH, as i know where ESL.so resides. > > However, I cannot define PATH, as there doesn't seem to be a ESL.lua file. > > Best, > Spyros > > > On Sat, Dec 13, 2014 at 7:13 AM, Chad Phillips wrote: > > The paths listed in the error message are the paths that Lua looks in for the module. So you either need to put your ESL.so in one of those locations, or add its location to path list. Check http://www.lua.org/pil/8.1.html for detailed info on how Lua does path handling. > > > > > > On Friday, December 12, 2014 at 12:20 PM, spros p wrote: > > > > > > > > > Hello, > > > > > > I am trying to run a simple ESL script using Lua. I have installed Lua 5.1 and the respective dev packages on my linux machine. Make luamod in /libs/esl/ directory works just fine. > > > > > > However when executing: freeswitch/libs/esl/lua/single_command.lua, i receive the following: > > > > > > lua: lua/single_command.lua:2: module 'ESL' not found: > > > no field package.preload['ESL'] > > > no file './ESL.lua' > > > no file '/usr/share/lua/5.1/ESL.lua' > > > no file '/usr/share/lua/5.1/ESL/init.lua' > > > no file '/usr/lib64/lua/5.1/ESL.lua' > > > no file '/usr/lib64/lua/5.1/ESL/init.lua' > > > no file './ESL.so' > > > no file '/usr/lib64/lua/5.1/ESL.so' > > > no file '/usr/lib64/lua/5.1/loadall.so' > > > stack traceback: > > > [C]: in function 'require' > > > lua/single_command.lua:2: in main chunk > > > [C]: ? > > > > > > Any help would be greatly appreciated > > > > > > Best, > > > SP > > > > > > > > > > > > > > > _________________________________________________________________________ > > > 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://confluence.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://confluence.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://confluence.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/20141215/3bbef7fa/attachment.html From mike at jerris.com Mon Dec 15 19:47:58 2014 From: mike at jerris.com (Michael Jerris) Date: Mon, 15 Dec 2014 11:47:58 -0500 Subject: [Freeswitch-users] Freeswitch based door communication In-Reply-To: References: Message-ID: To be clear, you are trying to figure out config for the client running on raspberry (using fs) or for another server? yes, you can do originate to multiple destinations, and the first one wins. This is built in to originate. I don't think we have any code to receive auto_answer indication, we only send it. It would require some small code changes to add this. Mike > On Dec 14, 2014, at 8:17 PM, Michael Traut wrote: > > Hi, > > I'm trying to build a Freeswitch / Raspberry based door communication. While there are some examples for such a constellation for Asterisk, i didn't find (and was not able to invent myself so far) a Freeswitch example. > > The target scenario: > > * A SIP based door module, > * 1-n Pushbuttons, one for each floor > * 1-n SIP Phones per floor > > When a push button is pressed i want to initiate a call to the door (that will be accepted automatically) and a call to all phones on the floor. The first one to answer gets connected to the door. > > Is this something i can do with "originate" ? From mike at jerris.com Mon Dec 15 19:48:52 2014 From: mike at jerris.com (Michael Jerris) Date: Mon, 15 Dec 2014 11:48:52 -0500 Subject: [Freeswitch-users] Caller ID - Not Defined when executing dialplan, although Lua script is pulling CID from Database In-Reply-To: References: <9312DD37-1999-4B29-A2C0-BC8429C6B866@jerris.com> Message-ID: <6EA81D72-BDBA-485B-912F-7300A5949F42@jerris.com> Double check configs and the sip traffic for differences. Also double check firmware on devices. > On Dec 15, 2014, at 1:13 AM, Joel White wrote: > > I will implement this tomorrow. I do have a question though, with all that being said.... the system I have running in the cloud does send out the CID specified in the Database using this methodology. Why would it work on one and not the other when their configs are identical? > > Just curious > > On Thu, Dec 11, 2014 at 1:51 PM, Chris Tunbridge > wrote: > Errr yeah, sorry Michael is right they're origination not effective :P > > On Thu, Dec 11, 2014 at 8:34 AM, Michael Jerris > wrote: > you should be looking at the origination_caller_id_* vars, not effective > >> On Dec 11, 2014, at 10:26 AM, Chris Tunbridge > wrote: >> >> The variable outbound_caller_id_number and outbound_caller_id_name are not related to the caller id on outbound calls. >> >> On your outbound dial plan you need to set something like the following >> >> >> >> >> This will cause the system to pull the settings from the users profile and use it for the outgoing call. >> >> On Wed, Dec 10, 2014 at 12:27 PM, Joel White > wrote: >> I have gone over the config with a fine tooth comb, it matches another config of a server in which the caller id works fine. What I am seeing however is that in this system the variable is not exported to the dialplan. I may be missing something, and most likely I am. I do have a question though. Is there a way to see what variables are defined for a particular user in the FreeSWITCH console? >> >> >> Here is some output of the Lua script on the server not pushing caller id >> >> 2014-12-09 16:54:22.819251 [NOTICE] switch_cpp.cpp:1328 Debug from gen_dir_user_xml.lua, generated XML: >> >> >>
>> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >>
>> >> >> >> And some output from the Lua script on the server with CID functioning >> >> 2014-12-09 21:50:55.458996 [NOTICE] switch_cpp.cpp:1328 Debug from gen_dir_user_xml.lua, generated XML: >> >> >>
>> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >>
>> >> >> >> Of course I removed any identifiable information, but it looks like the CID is being set. What am I missing here that is not allowing for the variable to be passed to the dialplan? >> >> >> This is what I get when the dialplan executes >> >> EXECUTE sofia/internal/26342 at voip.net >> set(effective_caller_id_number=) >> 2014-12-09 17:00:24.839237 [DEBUG] mod_dptools.c:1435 sofia/internal/26342 at voip.net SET [effective_caller_id_number]=[UNDEF] >> >> >> Kinda strange and I could not find a discrepancy between the dialplan configuration of the working server vs the non-working server >> >> >> >> Here is the version running on the server that works properly >> >> FreeSWITCH Version 1.4.13+git~20141103T195300Z~b942d0faa8~64bit (git b942d0f 2014-11-03 19:53:00Z 64bit) >> >> And the version of the server having issue with CID >> >> FreeSWITCH Version 1.4.13+git~20141103T195300Z~ >> b942d0faa8~64bit (git b942d0f 2014-11-03 19:53:00Z 64bit) >> >> >> >> I used diff and compared both servers conf directory recursively. I could not find a discrepancy in the files aside from Switch name, etc. >> >> >> What am I missing? >> >> Could there be anything that that would overwrite the CID variable after it is set by Lua (Generating the user profile)? >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141215/20c2ecc8/attachment-0001.html From joelewhite at gmail.com Mon Dec 15 20:14:54 2014 From: joelewhite at gmail.com (Joel White) Date: Mon, 15 Dec 2014 12:14:54 -0500 Subject: [Freeswitch-users] Issues with rtp audio going in slow motion In-Reply-To: References: Message-ID: Interesting information I found out about this issue. One person stated that they had an hour long call. About mid way through the call the caller dropped down to this "slow motion" slower sample rate. After about 30 seconds of this slow motion, the call went back to a normal state and worked fine for the rest of the call. I am forcing G711u on all inbound and outbound calls heading the to PSTN / VoIP carrier. This is an interesting question and makes me wonder if I can disable the write resampler. Any suggestions? I have not set the phone to only use G711u yet, I am trying to make this work on server side only. On Mon, Nov 3, 2014 at 12:00 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > > This should be handled in JIRA not the mailing list. > If problem or issue seems prudent in the subject, think JIRA > http://jira.freeswitch.org > That way you can post logs etc and not flood the list with help questions. > > > On Mon, Nov 3, 2014 at 10:49 AM, Joel White wrote: > >> FreeSWITCH Version 1.4.9+git~20140929T194948Z~ae069dcca7~64bit (git >> ae069dc 2014-09-29 19:49:48Z 64bit) >> >> Current version on the server >> >> On Mon, Nov 3, 2014 at 11:44 AM, Joel White wrote: >> >>> I will work on getting a sip trace. I am seeing this happen when >>> monitoring the system. The calls sound fine and then it sounds like the >>> audio slows to a crawl. It does not appear to happen regularly and I have >>> seen it happen on individual calls and on conferences. >>> >>> The conferences have external DIDs for outside participants. The >>> internal extensions dial a 5 digit extension. >>> >>> >>> >>> >>> On Mon, Nov 3, 2014 at 10:42 AM, Brian West >>> wrote: >>> >>>> What rev are you on? How are you replicating this? Can you get a >>>> siptrace and logs and file a JIRA so we can get this bug fixed. Sounds >>>> like the codec is probably changing during the call, are you doing a >>>> hold/unhold operation? >>>> >>>> On Tue, Oct 28, 2014 at 6:23 PM, Chris Tunbridge >>>> wrote: >>>> >>>>> Joel is there any way you can test with one polycom set to g711u. >>>>> >>>>> This sounds to me like a 16k vs 8k transcoding issue, maybe setup your >>>>> codecs on the freeswitch side to only allow 8k g722 (if that's even >>>>> possible) >>>>> >>>>> On Tue, Oct 21, 2014 at 8:22 AM, Joel White >>>>> wrote: >>>>> >>>>>> Also, just to mention. The CPU is never above 2% and the current >>>>>> load is about 20 simultaneous calls >>>>>> >>>>>> On Tue, Oct 21, 2014 at 10:16 AM, Joel White >>>>>> wrote: >>>>>> >>>>>>> Is this an issue with transcoding? >>>>>>> >>>>>>> I am wondering how I lock it down so that transcoding only happens >>>>>>> at a minimum >>>>>>> >>>>>>> Also, would it be bad to completely remove G722 from the codec list? >>>>>>> >>>>>>> I think the issue is that our provider is sending G711 and all of >>>>>>> the Polycom Phones are trying to use G722 >>>>>>> >>>>>>> On Mon, Oct 20, 2014 at 11:43 AM, Joel White >>>>>>> wrote: >>>>>>> >>>>>>>> I am having an issue where some calls start out fine and then go >>>>>>>> into slow motion (distorted slow speech) >>>>>>>> >>>>>>>> I noticed something in the logs stating >>>>>>>> >>>>>>>> Activating write resampler >>>>>>>> >>>>>>>> >>>>>>>> What does this mean and how do I avoid this in the future? >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> Thank you in advance >>>>>>>> >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://confluence.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>> http://www.cudatel.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> >>>> -- >>>> >>>> *Brian West* >>>> brian at freeswitch.org >>>> >>>> >>>> *Twitter: @FreeSWITCH , @briankwest* >>>> http://www.freeswitchbook.com >>>> http://www.freeswitchcookbook.com >>>> >>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 ? @anthmfs ? @FreeSWITCH ? > > ? http://freeswitch.org/ ? http://cluecon.com/ ? > http://twitter.com/FreeSWITCH > ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ > * > > ClueCon Weekly Development Call > ? sip:888 at conference.freeswitch.org ? +19193869900 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141215/6ffb3fca/attachment.html From joelewhite at gmail.com Mon Dec 15 20:15:31 2014 From: joelewhite at gmail.com (Joel White) Date: Mon, 15 Dec 2014 12:15:31 -0500 Subject: [Freeswitch-users] Issues with rtp audio going in slow motion In-Reply-To: References: Message-ID: Anthony, I am not signed up to JIRA and have never used it. Do you have any information I need to know to contribute? On Mon, Dec 15, 2014 at 12:14 PM, Joel White wrote: > > Interesting information I found out about this issue. One person stated > that they had an hour long call. About mid way through the call the caller > dropped down to this "slow motion" slower sample rate. After about 30 > seconds of this slow motion, the call went back to a normal state and > worked fine for the rest of the call. I am forcing G711u on all inbound > and outbound calls heading the to PSTN / VoIP carrier. This is an > interesting question and makes me wonder if I can disable the write > resampler. > > Any suggestions? > > I have not set the phone to only use G711u yet, I am trying to make this > work on server side only. > > > > On Mon, Nov 3, 2014 at 12:00 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: >> >> This should be handled in JIRA not the mailing list. >> If problem or issue seems prudent in the subject, think JIRA >> http://jira.freeswitch.org >> That way you can post logs etc and not flood the list with help questions. >> >> >> On Mon, Nov 3, 2014 at 10:49 AM, Joel White wrote: >> >>> FreeSWITCH Version 1.4.9+git~20140929T194948Z~ae069dcca7~64bit (git >>> ae069dc 2014-09-29 19:49:48Z 64bit) >>> >>> Current version on the server >>> >>> On Mon, Nov 3, 2014 at 11:44 AM, Joel White >>> wrote: >>> >>>> I will work on getting a sip trace. I am seeing this happen when >>>> monitoring the system. The calls sound fine and then it sounds like the >>>> audio slows to a crawl. It does not appear to happen regularly and I have >>>> seen it happen on individual calls and on conferences. >>>> >>>> The conferences have external DIDs for outside participants. The >>>> internal extensions dial a 5 digit extension. >>>> >>>> >>>> >>>> >>>> On Mon, Nov 3, 2014 at 10:42 AM, Brian West >>>> wrote: >>>> >>>>> What rev are you on? How are you replicating this? Can you get a >>>>> siptrace and logs and file a JIRA so we can get this bug fixed. Sounds >>>>> like the codec is probably changing during the call, are you doing a >>>>> hold/unhold operation? >>>>> >>>>> On Tue, Oct 28, 2014 at 6:23 PM, Chris Tunbridge >>>>> wrote: >>>>> >>>>>> Joel is there any way you can test with one polycom set to g711u. >>>>>> >>>>>> This sounds to me like a 16k vs 8k transcoding issue, maybe setup >>>>>> your codecs on the freeswitch side to only allow 8k g722 (if that's even >>>>>> possible) >>>>>> >>>>>> On Tue, Oct 21, 2014 at 8:22 AM, Joel White >>>>>> wrote: >>>>>> >>>>>>> Also, just to mention. The CPU is never above 2% and the current >>>>>>> load is about 20 simultaneous calls >>>>>>> >>>>>>> On Tue, Oct 21, 2014 at 10:16 AM, Joel White >>>>>>> wrote: >>>>>>> >>>>>>>> Is this an issue with transcoding? >>>>>>>> >>>>>>>> I am wondering how I lock it down so that transcoding only happens >>>>>>>> at a minimum >>>>>>>> >>>>>>>> Also, would it be bad to completely remove G722 from the codec list? >>>>>>>> >>>>>>>> I think the issue is that our provider is sending G711 and all of >>>>>>>> the Polycom Phones are trying to use G722 >>>>>>>> >>>>>>>> On Mon, Oct 20, 2014 at 11:43 AM, Joel White >>>>>>>> wrote: >>>>>>>> >>>>>>>>> I am having an issue where some calls start out fine and then go >>>>>>>>> into slow motion (distorted slow speech) >>>>>>>>> >>>>>>>>> I noticed something in the logs stating >>>>>>>>> >>>>>>>>> Activating write resampler >>>>>>>>> >>>>>>>>> >>>>>>>>> What does this mean and how do I avoid this in the future? >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> Thank you in advance >>>>>>>>> >>>>>>>> >>>>>>>> >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://confluence.freeswitch.org >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>>> http://www.cudatel.com >>>>>>> >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://confluence.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>> >>>>> >>>>> >>>>> -- >>>>> >>>>> *Brian West* >>>>> brian at freeswitch.org >>>>> >>>>> >>>>> *Twitter: @FreeSWITCH , @briankwest* >>>>> http://www.freeswitchbook.com >>>>> http://www.freeswitchcookbook.com >>>>> >>>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >>>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 ? @anthmfs ? @FreeSWITCH ? >> >> ? http://freeswitch.org/ ? http://cluecon.com/ ? >> http://twitter.com/FreeSWITCH >> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ >> * >> >> ClueCon Weekly Development Call >> ? sip:888 at conference.freeswitch.org ? +19193869900 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141215/f42029ad/attachment-0001.html From anthony.minessale at gmail.com Mon Dec 15 20:20:03 2014 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 15 Dec 2014 11:20:03 -0600 Subject: [Freeswitch-users] Issues with rtp audio going in slow motion In-Reply-To: References: Message-ID: You sign up for an account, log in, create an issue and fill out the fields. Make sure you have reproduced your problem on the very latest master from git before reporting because its a mandatory question. On Mon, Dec 15, 2014 at 11:15 AM, Joel White wrote: > > Anthony, I am not signed up to JIRA and have never used it. Do you have > any information I need to know to contribute? > > On Mon, Dec 15, 2014 at 12:14 PM, Joel White wrote: >> >> Interesting information I found out about this issue. One person stated >> that they had an hour long call. About mid way through the call the caller >> dropped down to this "slow motion" slower sample rate. After about 30 >> seconds of this slow motion, the call went back to a normal state and >> worked fine for the rest of the call. I am forcing G711u on all inbound >> and outbound calls heading the to PSTN / VoIP carrier. This is an >> interesting question and makes me wonder if I can disable the write >> resampler. >> >> Any suggestions? >> >> I have not set the phone to only use G711u yet, I am trying to make this >> work on server side only. >> >> >> >> On Mon, Nov 3, 2014 at 12:00 PM, Anthony Minessale < >> anthony.minessale at gmail.com> wrote: >>> >>> This should be handled in JIRA not the mailing list. >>> If problem or issue seems prudent in the subject, think JIRA >>> http://jira.freeswitch.org >>> That way you can post logs etc and not flood the list with help >>> questions. >>> >>> >>> On Mon, Nov 3, 2014 at 10:49 AM, Joel White >>> wrote: >>> >>>> FreeSWITCH Version 1.4.9+git~20140929T194948Z~ae069dcca7~64bit (git >>>> ae069dc 2014-09-29 19:49:48Z 64bit) >>>> >>>> Current version on the server >>>> >>>> On Mon, Nov 3, 2014 at 11:44 AM, Joel White >>>> wrote: >>>> >>>>> I will work on getting a sip trace. I am seeing this happen when >>>>> monitoring the system. The calls sound fine and then it sounds like the >>>>> audio slows to a crawl. It does not appear to happen regularly and I have >>>>> seen it happen on individual calls and on conferences. >>>>> >>>>> The conferences have external DIDs for outside participants. The >>>>> internal extensions dial a 5 digit extension. >>>>> >>>>> >>>>> >>>>> >>>>> On Mon, Nov 3, 2014 at 10:42 AM, Brian West >>>>> wrote: >>>>> >>>>>> What rev are you on? How are you replicating this? Can you get a >>>>>> siptrace and logs and file a JIRA so we can get this bug fixed. Sounds >>>>>> like the codec is probably changing during the call, are you doing a >>>>>> hold/unhold operation? >>>>>> >>>>>> On Tue, Oct 28, 2014 at 6:23 PM, Chris Tunbridge >>>>> > wrote: >>>>>> >>>>>>> Joel is there any way you can test with one polycom set to g711u. >>>>>>> >>>>>>> This sounds to me like a 16k vs 8k transcoding issue, maybe setup >>>>>>> your codecs on the freeswitch side to only allow 8k g722 (if that's even >>>>>>> possible) >>>>>>> >>>>>>> On Tue, Oct 21, 2014 at 8:22 AM, Joel White >>>>>>> wrote: >>>>>>> >>>>>>>> Also, just to mention. The CPU is never above 2% and the current >>>>>>>> load is about 20 simultaneous calls >>>>>>>> >>>>>>>> On Tue, Oct 21, 2014 at 10:16 AM, Joel White >>>>>>>> wrote: >>>>>>>> >>>>>>>>> Is this an issue with transcoding? >>>>>>>>> >>>>>>>>> I am wondering how I lock it down so that transcoding only happens >>>>>>>>> at a minimum >>>>>>>>> >>>>>>>>> Also, would it be bad to completely remove G722 from the codec >>>>>>>>> list? >>>>>>>>> >>>>>>>>> I think the issue is that our provider is sending G711 and all of >>>>>>>>> the Polycom Phones are trying to use G722 >>>>>>>>> >>>>>>>>> On Mon, Oct 20, 2014 at 11:43 AM, Joel White >>>>>>>> > wrote: >>>>>>>>> >>>>>>>>>> I am having an issue where some calls start out fine and then go >>>>>>>>>> into slow motion (distorted slow speech) >>>>>>>>>> >>>>>>>>>> I noticed something in the logs stating >>>>>>>>>> >>>>>>>>>> Activating write resampler >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> What does this mean and how do I avoid this in the future? >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> Thank you in advance >>>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> _________________________________________________________________________ >>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>> consulting at freeswitch.org >>>>>>>> http://www.freeswitchsolutions.com >>>>>>>> >>>>>>>> Official FreeSWITCH Sites >>>>>>>> http://www.freeswitch.org >>>>>>>> http://confluence.freeswitch.org >>>>>>>> http://www.cluecon.com >>>>>>>> >>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>>>> http://www.cudatel.com >>>>>>>> >>>>>>>> FreeSWITCH-users mailing list >>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>> UNSUBSCRIBE: >>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>> http://www.freeswitch.org >>>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://confluence.freeswitch.org >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> >>>>>> *Brian West* >>>>>> brian at freeswitch.org >>>>>> >>>>>> >>>>>> *Twitter: @FreeSWITCH , @briankwest* >>>>>> http://www.freeswitchbook.com >>>>>> http://www.freeswitchcookbook.com >>>>>> >>>>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >>>>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://confluence.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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 ? @anthmfs ? @FreeSWITCH ? >>> >>> ? http://freeswitch.org/ ? http://cluecon.com/ ? >>> http://twitter.com/FreeSWITCH >>> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ >>> * >>> >>> ClueCon Weekly Development Call >>> ? sip:888 at conference.freeswitch.org ? +19193869900 >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 ? @anthmfs ? @FreeSWITCH ? ? http://freeswitch.org/ ? http://cluecon.com/ ? http://twitter.com/FreeSWITCH ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ * ClueCon Weekly Development Call ? sip:888 at conference.freeswitch.org ? +19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141215/df93b9ed/attachment-0001.html From krice at freeswitch.org Mon Dec 15 20:19:51 2014 From: krice at freeswitch.org (Ken Rice) Date: Mon, 15 Dec 2014 11:19:51 -0600 Subject: [Freeswitch-users] Issues with rtp audio going in slow motion In-Reply-To: Message-ID: To file a jira, just go to freeswitch.org/jira and follow the signup link it takes about a minute, once you have you account, you can then open tickets... See https://freeswitch.org/confluence/display/FREESWITCH/Reporting+Bugs+to+JIRA for more details on filing the jira On 12/15/14 11:15 AM, "Joel White" wrote: > Anthony, I am not signed up to JIRA and have never used it.? Do you have any > information I need to know to contribute? > > On Mon, Dec 15, 2014 at 12:14 PM, Joel White wrote: >> Interesting information I found out about this issue.? One person stated that >> they had an hour long call.? About mid way through the call the caller >> dropped down to this "slow motion" slower sample rate.? After about 30 >> seconds of this slow motion, the call went back to a normal state and worked >> fine for the rest of the call.? I am forcing G711u on all inbound and >> outbound calls heading the to PSTN / VoIP carrier.? This is an interesting >> question and makes me wonder if I can disable the write resampler. >> >> Any suggestions? >> >> I have not set the phone to only use G711u yet, I am trying to make this work >> on server side only. >> >> >> >> On Mon, Nov 3, 2014 at 12:00 PM, Anthony Minessale >> wrote: >>> This should be handled in JIRA not the mailing list. >>> If problem or issue seems prudent in the subject, think JIRA >>> http://jira.freeswitch.org >>> That way you can post logs etc and not flood the list with help questions. >>> ? >>> >>> On Mon, Nov 3, 2014 at 10:49 AM, Joel White wrote: >>>> FreeSWITCH Version 1.4.9+git~20140929T194948Z~ae069dcca7~64bit (git ae069dc >>>> 2014-09-29 19:49:48Z 64bit) >>>> >>>> Current version on the server >>>> >>>> On Mon, Nov 3, 2014 at 11:44 AM, Joel White wrote: >>>>> I will work on getting a sip trace.? I am seeing this happen when >>>>> monitoring the system.? The calls sound fine and then it sounds like the >>>>> audio slows to a crawl.? It does not appear to happen regularly and I have >>>>> seen it happen on individual calls and on conferences.? >>>>> >>>>> The conferences have external DIDs for outside participants.? The internal >>>>> extensions dial a 5 digit extension. >>>>> >>>>> >>>>> >>>>> >>>>> On Mon, Nov 3, 2014 at 10:42 AM, Brian West wrote: >>>>>> What rev are you on? How are you replicating this? Can you get a siptrace >>>>>> and logs and file a JIRA so we can get this bug fixed.? Sounds like the >>>>>> codec is probably changing during the call, are you doing a hold/unhold >>>>>> operation? >>>>>> >>>>>> On Tue, Oct 28, 2014 at 6:23 PM, Chris Tunbridge >>>>>> wrote: >>>>>>> Joel is there any way you can test with one polycom set to g711u. >>>>>>> >>>>>>> This sounds to me like a 16k vs 8k transcoding issue, maybe setup your >>>>>>> codecs on the freeswitch side to only allow 8k g722 (if that's even >>>>>>> possible) >>>>>>> >>>>>>> On Tue, Oct 21, 2014 at 8:22 AM, Joel White >>>>>>> wrote: Also, just to mention.? The CPU is never above 2% and the current load is about 20 simultaneous calls On Tue, Oct 21, 2014 at 10:16 AM, Joel White wrote: Is this an issue with transcoding? I am wondering how I lock it down so that transcoding only happens at a minimum Also, would it be bad to completely remove G722 from the codec list? I think the issue is that our provider is sending G711 and all of the Polycom Phones are trying to use G722 On Mon, Oct 20, 2014 at 11:43 AM, Joel White wrote: I am having an issue where some calls start out fine and then go into slow motion (distorted slow speech) I noticed something in the logs stating ?Activating write resampler What does this mean and how do I avoid this in the future? Thank you in advance _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>>> ________________________________________________________________________>>>>>>> _ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://confluence.freeswitch.org >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users at 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 Twitter: @FreeSWITCH -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141215/401caec6/attachment.html From joelewhite at gmail.com Mon Dec 15 20:21:50 2014 From: joelewhite at gmail.com (Joel White) Date: Mon, 15 Dec 2014 12:21:50 -0500 Subject: [Freeswitch-users] Issues with rtp audio going in slow motion In-Reply-To: References: Message-ID: 2014-12-15 10:13:41.728748 [DEBUG] switch_core_media.c:5083 Audio params are unchanged for sofia/internal/51024 at voip.net. 2014-12-15 10:13:41.748724 [DEBUG] switch_core_media.c:2381 Changing Codec from PCMU at 20ms@8000hz to G722 at 20ms@8000hz 2014-12-15 10:13:41.768769 [DEBUG] switch_core_session.c:1053 Send signal sofia/external/14073134300 at 69.174.248.123 [BREAK] 2014-12-15 10:13:41.788777 [DEBUG] switch_core_media.c:2473 Set Codec sofia/internal/51024 at bcvoip.net G722/8000 20 ms 160 samples 64000 bits 1 channels 2014-12-15 10:13:41.788777 [DEBUG] switch_core_codec.c:123 sofia/internal/ 51024 at voip.net Original read codec replaced with G722:9 2014-12-15 10:13:41.808784 [NOTICE] switch_core_io.c:1287 Activating write resampler 2014-12-15 10:13:41.808784 [DEBUG] switch_core_media.c:2055 alternate payload received (received 0, expecting 9) 2014-12-15 10:13:41.808784 [WARNING] switch_core_media.c:2066 Changing current codec to PCMU (payload type 0). 2014-12-15 10:13:41.808784 [NOTICE] switch_core_io.c:1287 Activating write resampler 2014-12-15 10:13:41.828776 [NOTICE] switch_core_media.c:2374 Deactivating write resampler 2014-12-15 10:13:41.828776 [DEBUG] switch_core_media.c:2381 Changing Codec from G722 at 20ms@8000hz to PCMU at 20ms@8000hz 2014-12-15 10:13:41.848763 [NOTICE] switch_core_io.c:1287 Activating write resampler 2014-12-15 10:13:41.868765 [DEBUG] switch_core_media.c:2473 Set Codec sofia/internal/51024 at voip.net PCMU/8000 20 ms 160 samples 64000 bits 1 channels 2014-12-15 10:13:41.868765 [DEBUG] switch_core_codec.c:123 sofia/internal/ 51024 at voip.net Original read codec replaced with PCMU:0 Some of the messages I see that concern me about the call On Mon, Dec 15, 2014 at 12:15 PM, Joel White wrote: > > Anthony, I am not signed up to JIRA and have never used it. Do you have > any information I need to know to contribute? > > On Mon, Dec 15, 2014 at 12:14 PM, Joel White wrote: >> >> Interesting information I found out about this issue. One person stated >> that they had an hour long call. About mid way through the call the caller >> dropped down to this "slow motion" slower sample rate. After about 30 >> seconds of this slow motion, the call went back to a normal state and >> worked fine for the rest of the call. I am forcing G711u on all inbound >> and outbound calls heading the to PSTN / VoIP carrier. This is an >> interesting question and makes me wonder if I can disable the write >> resampler. >> >> Any suggestions? >> >> I have not set the phone to only use G711u yet, I am trying to make this >> work on server side only. >> >> >> >> On Mon, Nov 3, 2014 at 12:00 PM, Anthony Minessale < >> anthony.minessale at gmail.com> wrote: >>> >>> This should be handled in JIRA not the mailing list. >>> If problem or issue seems prudent in the subject, think JIRA >>> http://jira.freeswitch.org >>> That way you can post logs etc and not flood the list with help >>> questions. >>> >>> >>> On Mon, Nov 3, 2014 at 10:49 AM, Joel White >>> wrote: >>> >>>> FreeSWITCH Version 1.4.9+git~20140929T194948Z~ae069dcca7~64bit (git >>>> ae069dc 2014-09-29 19:49:48Z 64bit) >>>> >>>> Current version on the server >>>> >>>> On Mon, Nov 3, 2014 at 11:44 AM, Joel White >>>> wrote: >>>> >>>>> I will work on getting a sip trace. I am seeing this happen when >>>>> monitoring the system. The calls sound fine and then it sounds like the >>>>> audio slows to a crawl. It does not appear to happen regularly and I have >>>>> seen it happen on individual calls and on conferences. >>>>> >>>>> The conferences have external DIDs for outside participants. The >>>>> internal extensions dial a 5 digit extension. >>>>> >>>>> >>>>> >>>>> >>>>> On Mon, Nov 3, 2014 at 10:42 AM, Brian West >>>>> wrote: >>>>> >>>>>> What rev are you on? How are you replicating this? Can you get a >>>>>> siptrace and logs and file a JIRA so we can get this bug fixed. Sounds >>>>>> like the codec is probably changing during the call, are you doing a >>>>>> hold/unhold operation? >>>>>> >>>>>> On Tue, Oct 28, 2014 at 6:23 PM, Chris Tunbridge >>>>> > wrote: >>>>>> >>>>>>> Joel is there any way you can test with one polycom set to g711u. >>>>>>> >>>>>>> This sounds to me like a 16k vs 8k transcoding issue, maybe setup >>>>>>> your codecs on the freeswitch side to only allow 8k g722 (if that's even >>>>>>> possible) >>>>>>> >>>>>>> On Tue, Oct 21, 2014 at 8:22 AM, Joel White >>>>>>> wrote: >>>>>>> >>>>>>>> Also, just to mention. The CPU is never above 2% and the current >>>>>>>> load is about 20 simultaneous calls >>>>>>>> >>>>>>>> On Tue, Oct 21, 2014 at 10:16 AM, Joel White >>>>>>>> wrote: >>>>>>>> >>>>>>>>> Is this an issue with transcoding? >>>>>>>>> >>>>>>>>> I am wondering how I lock it down so that transcoding only happens >>>>>>>>> at a minimum >>>>>>>>> >>>>>>>>> Also, would it be bad to completely remove G722 from the codec >>>>>>>>> list? >>>>>>>>> >>>>>>>>> I think the issue is that our provider is sending G711 and all of >>>>>>>>> the Polycom Phones are trying to use G722 >>>>>>>>> >>>>>>>>> On Mon, Oct 20, 2014 at 11:43 AM, Joel White >>>>>>>> > wrote: >>>>>>>>> >>>>>>>>>> I am having an issue where some calls start out fine and then go >>>>>>>>>> into slow motion (distorted slow speech) >>>>>>>>>> >>>>>>>>>> I noticed something in the logs stating >>>>>>>>>> >>>>>>>>>> Activating write resampler >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> What does this mean and how do I avoid this in the future? >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> Thank you in advance >>>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> _________________________________________________________________________ >>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>> consulting at freeswitch.org >>>>>>>> http://www.freeswitchsolutions.com >>>>>>>> >>>>>>>> Official FreeSWITCH Sites >>>>>>>> http://www.freeswitch.org >>>>>>>> http://confluence.freeswitch.org >>>>>>>> http://www.cluecon.com >>>>>>>> >>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>>>> http://www.cudatel.com >>>>>>>> >>>>>>>> FreeSWITCH-users mailing list >>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>> UNSUBSCRIBE: >>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>> http://www.freeswitch.org >>>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://confluence.freeswitch.org >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> >>>>>> *Brian West* >>>>>> brian at freeswitch.org >>>>>> >>>>>> >>>>>> *Twitter: @FreeSWITCH , @briankwest* >>>>>> http://www.freeswitchbook.com >>>>>> http://www.freeswitchcookbook.com >>>>>> >>>>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >>>>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://confluence.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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 ? @anthmfs ? @FreeSWITCH ? >>> >>> ? http://freeswitch.org/ ? http://cluecon.com/ ? >>> http://twitter.com/FreeSWITCH >>> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ >>> * >>> >>> ClueCon Weekly Development Call >>> ? sip:888 at conference.freeswitch.org ? +19193869900 >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141215/7edfeeab/attachment-0001.html From joelewhite at gmail.com Mon Dec 15 20:31:41 2014 From: joelewhite at gmail.com (Joel White) Date: Mon, 15 Dec 2014 12:31:41 -0500 Subject: [Freeswitch-users] Caller ID - Not Defined when executing dialplan, although Lua script is pulling CID from Database In-Reply-To: <6EA81D72-BDBA-485B-912F-7300A5949F42@jerris.com> References: <9312DD37-1999-4B29-A2C0-BC8429C6B866@jerris.com> <6EA81D72-BDBA-485B-912F-7300A5949F42@jerris.com> Message-ID: I implemented the suggestion that Chris gave and it works fine. On Mon, Dec 15, 2014 at 11:48 AM, Michael Jerris wrote: > > Double check configs and the sip traffic for differences. Also double > check firmware on devices. > > On Dec 15, 2014, at 1:13 AM, Joel White wrote: > > I will implement this tomorrow. I do have a question though, with all > that being said.... the system I have running in the cloud does send out > the CID specified in the Database using this methodology. Why would it > work on one and not the other when their configs are identical? > > Just curious > > On Thu, Dec 11, 2014 at 1:51 PM, Chris Tunbridge > wrote: >> >> Errr yeah, sorry Michael is right they're origination not effective :P >> >> On Thu, Dec 11, 2014 at 8:34 AM, Michael Jerris wrote: >> >>> you should be looking at the origination_caller_id_* vars, not effective >>> >>> On Dec 11, 2014, at 10:26 AM, Chris Tunbridge >>> wrote: >>> >>> The variable outbound_caller_id_number and outbound_caller_id_name are >>> not related to the caller id on outbound calls. >>> >>> On your outbound dial plan you need to set something like the following >>> >>> >> data="effective_caller_id_name=${user_data(${username}@${domain_name} >>> var outbound_caller_id_name)}"/> >>> >> data="effective_caller_id_number=${user_data(${username}@${domain_name} >>> var outbound_caller_id_number)}"/> >>> >>> This will cause the system to pull the settings from the users profile >>> and use it for the outgoing call. >>> >>> On Wed, Dec 10, 2014 at 12:27 PM, Joel White >>> wrote: >>> >>>> I have gone over the config with a fine tooth comb, it matches another >>>> config of a server in which the caller id works fine. What I am seeing >>>> however is that in this system the variable is not exported to the >>>> dialplan. I may be missing something, and most likely I am. I do have a >>>> question though. Is there a way to see what variables are defined for a >>>> particular user in the FreeSWITCH console? >>>> >>>> >>>> Here is some output of the Lua script on the server not pushing caller >>>> id >>>> >>>> 2014-12-09 16:54:22.819251 [NOTICE] switch_cpp.cpp:1328 Debug from >>>> gen_dir_user_xml.lua, generated XML: >>>> >>>> >>>>
>>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>> value="1321XXXXXXX"/> >>>> >>>> >>>> >>>> >>>> >>>>
>>>> >>>> >>>> >>>> And some output from the Lua script on the server with CID functioning >>>> >>>> 2014-12-09 21:50:55.458996 [NOTICE] switch_cpp.cpp:1328 Debug from >>>> gen_dir_user_xml.lua, generated XML: >>>> >>>> >>>>
>>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>> value="1303XXXXXX"/> >>>> >>>> >>>> >>>> >>>>
>>>> >>>> >>>> >>>> Of course I removed any identifiable information, but it looks like the >>>> CID is being set. What am I missing here that is not allowing for the >>>> variable to be passed to the dialplan? >>>> >>>> >>>> This is what I get when the dialplan executes >>>> >>>> EXECUTE sofia/internal/26342 at voip.net >>>> set(effective_caller_id_number=) >>>> 2014-12-09 17:00:24.839237 [DEBUG] mod_dptools.c:1435 sofia/internal/ >>>> 26342 at voip.net SET [effective_caller_id_number]=[UNDEF] >>>> >>>> >>>> Kinda strange and I could not find a discrepancy between the dialplan >>>> configuration of the working server vs the non-working server >>>> >>>> >>>> >>>> Here is the version running on the server that works properly >>>> >>>> FreeSWITCH Version 1.4.13+git~20141103T195300Z~b942d0faa8~64bit (git >>>> b942d0f 2014-11-03 19:53:00Z 64bit) >>>> >>>> And the version of the server having issue with CID >>>> >>>> FreeSWITCH Version 1.4.13+git~20141103T195300Z~ >>>> b942d0faa8~64bit (git b942d0f 2014-11-03 19:53:00Z 64bit) >>>> >>>> >>>> >>>> I used diff and compared both servers conf directory recursively. I >>>> could not find a discrepancy in the files aside from Switch name, etc. >>>> >>>> >>>> What am I missing? >>>> >>>> Could there be anything that that would overwrite the CID variable >>>> after it is set by Lua (Generating the user profile)? >>>> >>>> >>> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141215/acc520da/attachment.html From joelewhite at gmail.com Mon Dec 15 20:34:37 2014 From: joelewhite at gmail.com (Joel White) Date: Mon, 15 Dec 2014 12:34:37 -0500 Subject: [Freeswitch-users] Caller ID - Not Defined when executing dialplan, although Lua script is pulling CID from Database In-Reply-To: References: <9312DD37-1999-4B29-A2C0-BC8429C6B866@jerris.com> <6EA81D72-BDBA-485B-912F-7300A5949F42@jerris.com> Message-ID: Thank you for your assistance guys :) On Mon, Dec 15, 2014 at 12:31 PM, Joel White wrote: > > I implemented the suggestion that Chris gave and it works fine. > > On Mon, Dec 15, 2014 at 11:48 AM, Michael Jerris wrote: > >> Double check configs and the sip traffic for differences. Also double >> check firmware on devices. >> >> On Dec 15, 2014, at 1:13 AM, Joel White wrote: >> >> I will implement this tomorrow. I do have a question though, with all >> that being said.... the system I have running in the cloud does send out >> the CID specified in the Database using this methodology. Why would it >> work on one and not the other when their configs are identical? >> >> Just curious >> >> On Thu, Dec 11, 2014 at 1:51 PM, Chris Tunbridge >> wrote: >>> >>> Errr yeah, sorry Michael is right they're origination not effective :P >>> >>> On Thu, Dec 11, 2014 at 8:34 AM, Michael Jerris wrote: >>> >>>> you should be looking at the origination_caller_id_* vars, not effective >>>> >>>> On Dec 11, 2014, at 10:26 AM, Chris Tunbridge >>>> wrote: >>>> >>>> The variable outbound_caller_id_number and outbound_caller_id_name are >>>> not related to the caller id on outbound calls. >>>> >>>> On your outbound dial plan you need to set something like the following >>>> >>>> >>> data="effective_caller_id_name=${user_data(${username}@${domain_name} >>>> var outbound_caller_id_name)}"/> >>>> >>> data="effective_caller_id_number=${user_data(${username}@${domain_name} >>>> var outbound_caller_id_number)}"/> >>>> >>>> This will cause the system to pull the settings from the users profile >>>> and use it for the outgoing call. >>>> >>>> On Wed, Dec 10, 2014 at 12:27 PM, Joel White >>>> wrote: >>>> >>>>> I have gone over the config with a fine tooth comb, it matches another >>>>> config of a server in which the caller id works fine. What I am seeing >>>>> however is that in this system the variable is not exported to the >>>>> dialplan. I may be missing something, and most likely I am. I do have a >>>>> question though. Is there a way to see what variables are defined for a >>>>> particular user in the FreeSWITCH console? >>>>> >>>>> >>>>> Here is some output of the Lua script on the server not pushing caller >>>>> id >>>>> >>>>> 2014-12-09 16:54:22.819251 [NOTICE] switch_cpp.cpp:1328 Debug from >>>>> gen_dir_user_xml.lua, generated XML: >>>>> >>>>> >>>>>
>>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>> value="1321XXXXXXX"/> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>>
>>>>> >>>>> >>>>> >>>>> And some output from the Lua script on the server with CID functioning >>>>> >>>>> 2014-12-09 21:50:55.458996 [NOTICE] switch_cpp.cpp:1328 Debug from >>>>> gen_dir_user_xml.lua, generated XML: >>>>> >>>>> >>>>>
>>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>> value="1303XXXXXX"/> >>>>> >>>>> >>>>> >>>>> >>>>>
>>>>> >>>>> >>>>> >>>>> Of course I removed any identifiable information, but it looks like >>>>> the CID is being set. What am I missing here that is not allowing for the >>>>> variable to be passed to the dialplan? >>>>> >>>>> >>>>> This is what I get when the dialplan executes >>>>> >>>>> EXECUTE sofia/internal/26342 at voip.net >>>>> set(effective_caller_id_number=) >>>>> 2014-12-09 17:00:24.839237 [DEBUG] mod_dptools.c:1435 sofia/internal/ >>>>> 26342 at voip.net SET [effective_caller_id_number]=[UNDEF] >>>>> >>>>> >>>>> Kinda strange and I could not find a discrepancy between the dialplan >>>>> configuration of the working server vs the non-working server >>>>> >>>>> >>>>> >>>>> Here is the version running on the server that works properly >>>>> >>>>> FreeSWITCH Version 1.4.13+git~20141103T195300Z~b942d0faa8~64bit (git >>>>> b942d0f 2014-11-03 19:53:00Z 64bit) >>>>> >>>>> And the version of the server having issue with CID >>>>> >>>>> FreeSWITCH Version 1.4.13+git~20141103T195300Z~ >>>>> b942d0faa8~64bit (git b942d0f 2014-11-03 19:53:00Z 64bit) >>>>> >>>>> >>>>> >>>>> I used diff and compared both servers conf directory recursively. I >>>>> could not find a discrepancy in the files aside from Switch name, etc. >>>>> >>>>> >>>>> What am I missing? >>>>> >>>>> Could there be anything that that would overwrite the CID variable >>>>> after it is set by Lua (Generating the user profile)? >>>>> >>>>> >>>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141215/d5fedcdb/attachment-0001.html From joelewhite at gmail.com Mon Dec 15 20:36:36 2014 From: joelewhite at gmail.com (Joel White) Date: Mon, 15 Dec 2014 12:36:36 -0500 Subject: [Freeswitch-users] Issues with rtp audio going in slow motion In-Reply-To: References: Message-ID: Will do, thank you for the information. On Mon, Dec 15, 2014 at 12:19 PM, Ken Rice wrote: > > To file a jira, just go to freeswitch.org/jira and follow the signup > link it takes about a minute, once you have you account, you can then open > tickets... > > See > https://freeswitch.org/confluence/display/FREESWITCH/Reporting+Bugs+to+JIRA > for more details on filing the jira > > > > On 12/15/14 11:15 AM, "Joel White" wrote: > > Anthony, I am not signed up to JIRA and have never used it. Do you have > any information I need to know to contribute? > > On Mon, Dec 15, 2014 at 12:14 PM, Joel White wrote: > > Interesting information I found out about this issue. One person stated > that they had an hour long call. About mid way through the call the caller > dropped down to this "slow motion" slower sample rate. After about 30 > seconds of this slow motion, the call went back to a normal state and > worked fine for the rest of the call. I am forcing G711u on all inbound > and outbound calls heading the to PSTN / VoIP carrier. This is an > interesting question and makes me wonder if I can disable the write > resampler. > > Any suggestions? > > I have not set the phone to only use G711u yet, I am trying to make this > work on server side only. > > > > On Mon, Nov 3, 2014 at 12:00 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > > This should be handled in JIRA not the mailing list. > If problem or issue seems prudent in the subject, think JIRA > http://jira.freeswitch.org > That way you can post logs etc and not flood the list with help questions. > > > On Mon, Nov 3, 2014 at 10:49 AM, Joel White wrote: > > FreeSWITCH Version 1.4.9+git~20140929T194948Z~ae069dcca7~64bit (git > ae069dc 2014-09-29 19:49:48Z 64bit) > > Current version on the server > > On Mon, Nov 3, 2014 at 11:44 AM, Joel White wrote: > > I will work on getting a sip trace. I am seeing this happen when > monitoring the system. The calls sound fine and then it sounds like the > audio slows to a crawl. It does not appear to happen regularly and I have > seen it happen on individual calls and on conferences. > > The conferences have external DIDs for outside participants. The internal > extensions dial a 5 digit extension. > > > > > On Mon, Nov 3, 2014 at 10:42 AM, Brian West wrote: > > What rev are you on? How are you replicating this? Can you get a siptrace > and logs and file a JIRA so we can get this bug fixed. Sounds like the > codec is probably changing during the call, are you doing a hold/unhold > operation? > > On Tue, Oct 28, 2014 at 6:23 PM, Chris Tunbridge > wrote: > > Joel is there any way you can test with one polycom set to g711u. > > This sounds to me like a 16k vs 8k transcoding issue, maybe setup your > codecs on the freeswitch side to only allow 8k g722 (if that's even > possible) > > On Tue, Oct 21, 2014 at 8:22 AM, Joel White wrote: > > Also, just to mention. The CPU is never above 2% and the current load is > about 20 simultaneous calls > > On Tue, Oct 21, 2014 at 10:16 AM, Joel White wrote: > Is this an issue with transcoding? > > I am wondering how I lock it down so that transcoding only happens at a > minimum > > Also, would it be bad to completely remove G722 from the codec list? > > I think the issue is that our provider is sending G711 and all of the > Polycom Phones are trying to use G722 > > On Mon, Oct 20, 2014 at 11:43 AM, Joel White wrote: > I am having an issue where some calls start out fine and then go into slow > motion (distorted slow speech) > > I noticed something in the logs stating > > Activating write resampler > > > What does this mean and how do I avoid this in the future? > > > > Thank you in advance > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 > Twitter: @FreeSWITCH > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141215/7f87ca27/attachment.html From mike at jerris.com Mon Dec 15 20:40:23 2014 From: mike at jerris.com (Michael Jerris) Date: Mon, 15 Dec 2014 12:40:23 -0500 Subject: [Freeswitch-users] Caller ID - Not Defined when executing dialplan, although Lua script is pulling CID from Database In-Reply-To: References: <9312DD37-1999-4B29-A2C0-BC8429C6B866@jerris.com> <6EA81D72-BDBA-485B-912F-7300A5949F42@jerris.com> Message-ID: If you are using effective, you will have issues related to transfers. You should really be using the ones i mentioned. > On Dec 15, 2014, at 12:31 PM, Joel White wrote: > > I implemented the suggestion that Chris gave and it works fine. > > On Mon, Dec 15, 2014 at 11:48 AM, Michael Jerris > wrote: > Double check configs and the sip traffic for differences. Also double check firmware on devices. > >> On Dec 15, 2014, at 1:13 AM, Joel White > wrote: >> >> I will implement this tomorrow. I do have a question though, with all that being said.... the system I have running in the cloud does send out the CID specified in the Database using this methodology. Why would it work on one and not the other when their configs are identical? >> >> Just curious >> >> On Thu, Dec 11, 2014 at 1:51 PM, Chris Tunbridge > wrote: >> Errr yeah, sorry Michael is right they're origination not effective :P >> >> On Thu, Dec 11, 2014 at 8:34 AM, Michael Jerris > wrote: >> you should be looking at the origination_caller_id_* vars, not effective >> >>> On Dec 11, 2014, at 10:26 AM, Chris Tunbridge > wrote: >>> >>> The variable outbound_caller_id_number and outbound_caller_id_name are not related to the caller id on outbound calls. >>> >>> On your outbound dial plan you need to set something like the following >>> >>> >>> >>> >>> This will cause the system to pull the settings from the users profile and use it for the outgoing call. >>> >>> On Wed, Dec 10, 2014 at 12:27 PM, Joel White > wrote: >>> I have gone over the config with a fine tooth comb, it matches another config of a server in which the caller id works fine. What I am seeing however is that in this system the variable is not exported to the dialplan. I may be missing something, and most likely I am. I do have a question though. Is there a way to see what variables are defined for a particular user in the FreeSWITCH console? >>> >>> >>> Here is some output of the Lua script on the server not pushing caller id >>> >>> 2014-12-09 16:54:22.819251 [NOTICE] switch_cpp.cpp:1328 Debug from gen_dir_user_xml.lua, generated XML: >>> >>> >>>
>>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>>
>>> >>> >>> >>> And some output from the Lua script on the server with CID functioning >>> >>> 2014-12-09 21:50:55.458996 [NOTICE] switch_cpp.cpp:1328 Debug from gen_dir_user_xml.lua, generated XML: >>> >>> >>>
>>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>>
>>> >>> >>> >>> Of course I removed any identifiable information, but it looks like the CID is being set. What am I missing here that is not allowing for the variable to be passed to the dialplan? >>> >>> >>> This is what I get when the dialplan executes >>> >>> EXECUTE sofia/internal/26342 at voip.net >>> set(effective_caller_id_number=) >>> 2014-12-09 17:00:24.839237 [DEBUG] mod_dptools.c:1435 sofia/internal/26342 at voip.net SET [effective_caller_id_number]=[UNDEF] >>> >>> >>> Kinda strange and I could not find a discrepancy between the dialplan configuration of the working server vs the non-working server >>> >>> >>> >>> Here is the version running on the server that works properly >>> >>> FreeSWITCH Version 1.4.13+git~20141103T195300Z~b942d0faa8~64bit (git b942d0f 2014-11-03 19:53:00Z 64bit) >>> >>> And the version of the server having issue with CID >>> >>> FreeSWITCH Version 1.4.13+git~20141103T195300Z~ >>> b942d0faa8~64bit (git b942d0f 2014-11-03 19:53:00Z 64bit) >>> >>> >>> >>> I used diff and compared both servers conf directory recursively. I could not find a discrepancy in the files aside from Switch name, etc. >>> >>> >>> What am I missing? >>> >>> Could there be anything that that would overwrite the CID variable after it is set by Lua (Generating the user profile)? >>> >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141215/e31cf0df/attachment-0001.html From joelewhite at gmail.com Mon Dec 15 20:46:55 2014 From: joelewhite at gmail.com (Joel White) Date: Mon, 15 Dec 2014 12:46:55 -0500 Subject: [Freeswitch-users] Caller ID - Not Defined when executing dialplan, although Lua script is pulling CID from Database In-Reply-To: References: <9312DD37-1999-4B29-A2C0-BC8429C6B866@jerris.com> <6EA81D72-BDBA-485B-912F-7300A5949F42@jerris.com> Message-ID: I will definitely use use origination_caller_* Thank you again On Mon, Dec 15, 2014 at 12:40 PM, Michael Jerris wrote: > > If you are using effective, you will have issues related to transfers. > You should really be using the ones i mentioned. > > On Dec 15, 2014, at 12:31 PM, Joel White wrote: > > I implemented the suggestion that Chris gave and it works fine. > > On Mon, Dec 15, 2014 at 11:48 AM, Michael Jerris wrote: >> >> Double check configs and the sip traffic for differences. Also double >> check firmware on devices. >> >> On Dec 15, 2014, at 1:13 AM, Joel White wrote: >> >> I will implement this tomorrow. I do have a question though, with all >> that being said.... the system I have running in the cloud does send out >> the CID specified in the Database using this methodology. Why would it >> work on one and not the other when their configs are identical? >> >> Just curious >> >> On Thu, Dec 11, 2014 at 1:51 PM, Chris Tunbridge >> wrote: >>> >>> Errr yeah, sorry Michael is right they're origination not effective :P >>> >>> On Thu, Dec 11, 2014 at 8:34 AM, Michael Jerris wrote: >>> >>>> you should be looking at the origination_caller_id_* vars, not effective >>>> >>>> On Dec 11, 2014, at 10:26 AM, Chris Tunbridge >>>> wrote: >>>> >>>> The variable outbound_caller_id_number and outbound_caller_id_name are >>>> not related to the caller id on outbound calls. >>>> >>>> On your outbound dial plan you need to set something like the following >>>> >>>> >>> data="effective_caller_id_name=${user_data(${username}@${domain_name} >>>> var outbound_caller_id_name)}"/> >>>> >>> data="effective_caller_id_number=${user_data(${username}@${domain_name} >>>> var outbound_caller_id_number)}"/> >>>> >>>> This will cause the system to pull the settings from the users profile >>>> and use it for the outgoing call. >>>> >>>> On Wed, Dec 10, 2014 at 12:27 PM, Joel White >>>> wrote: >>>> >>>>> I have gone over the config with a fine tooth comb, it matches another >>>>> config of a server in which the caller id works fine. What I am seeing >>>>> however is that in this system the variable is not exported to the >>>>> dialplan. I may be missing something, and most likely I am. I do have a >>>>> question though. Is there a way to see what variables are defined for a >>>>> particular user in the FreeSWITCH console? >>>>> >>>>> >>>>> Here is some output of the Lua script on the server not pushing caller >>>>> id >>>>> >>>>> 2014-12-09 16:54:22.819251 [NOTICE] switch_cpp.cpp:1328 Debug from >>>>> gen_dir_user_xml.lua, generated XML: >>>>> >>>>> >>>>>
>>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>> value="1321XXXXXXX"/> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>>
>>>>> >>>>> >>>>> >>>>> And some output from the Lua script on the server with CID functioning >>>>> >>>>> 2014-12-09 21:50:55.458996 [NOTICE] switch_cpp.cpp:1328 Debug from >>>>> gen_dir_user_xml.lua, generated XML: >>>>> >>>>> >>>>>
>>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>> value="1303XXXXXX"/> >>>>> >>>>> >>>>> >>>>> >>>>>
>>>>> >>>>> >>>>> >>>>> Of course I removed any identifiable information, but it looks like >>>>> the CID is being set. What am I missing here that is not allowing for the >>>>> variable to be passed to the dialplan? >>>>> >>>>> >>>>> This is what I get when the dialplan executes >>>>> >>>>> EXECUTE sofia/internal/26342 at voip.net >>>>> set(effective_caller_id_number=) >>>>> 2014-12-09 17:00:24.839237 [DEBUG] mod_dptools.c:1435 sofia/internal/ >>>>> 26342 at voip.net SET [effective_caller_id_number]=[UNDEF] >>>>> >>>>> >>>>> Kinda strange and I could not find a discrepancy between the dialplan >>>>> configuration of the working server vs the non-working server >>>>> >>>>> >>>>> >>>>> Here is the version running on the server that works properly >>>>> >>>>> FreeSWITCH Version 1.4.13+git~20141103T195300Z~b942d0faa8~64bit (git >>>>> b942d0f 2014-11-03 19:53:00Z 64bit) >>>>> >>>>> And the version of the server having issue with CID >>>>> >>>>> FreeSWITCH Version 1.4.13+git~20141103T195300Z~ >>>>> b942d0faa8~64bit (git b942d0f 2014-11-03 19:53:00Z 64bit) >>>>> >>>>> >>>>> >>>>> I used diff and compared both servers conf directory recursively. I >>>>> could not find a discrepancy in the files aside from Switch name, etc. >>>>> >>>>> >>>>> What am I missing? >>>>> >>>>> Could there be anything that that would overwrite the CID variable >>>>> after it is set by Lua (Generating the user profile)? >>>>> >>>>> >>>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141215/923f6ac6/attachment.html From joelewhite at gmail.com Mon Dec 15 20:47:24 2014 From: joelewhite at gmail.com (Joel White) Date: Mon, 15 Dec 2014 12:47:24 -0500 Subject: [Freeswitch-users] Caller ID - Not Defined when executing dialplan, although Lua script is pulling CID from Database In-Reply-To: References: <9312DD37-1999-4B29-A2C0-BC8429C6B866@jerris.com> <6EA81D72-BDBA-485B-912F-7300A5949F42@jerris.com> Message-ID: And how would I put a conditional if the CID is not specified to set it? On Mon, Dec 15, 2014 at 12:46 PM, Joel White wrote: > > I will definitely use use origination_caller_* > > Thank you again > > On Mon, Dec 15, 2014 at 12:40 PM, Michael Jerris wrote: >> >> If you are using effective, you will have issues related to transfers. >> You should really be using the ones i mentioned. >> >> On Dec 15, 2014, at 12:31 PM, Joel White wrote: >> >> I implemented the suggestion that Chris gave and it works fine. >> >> On Mon, Dec 15, 2014 at 11:48 AM, Michael Jerris wrote: >>> >>> Double check configs and the sip traffic for differences. Also double >>> check firmware on devices. >>> >>> On Dec 15, 2014, at 1:13 AM, Joel White wrote: >>> >>> I will implement this tomorrow. I do have a question though, with all >>> that being said.... the system I have running in the cloud does send out >>> the CID specified in the Database using this methodology. Why would it >>> work on one and not the other when their configs are identical? >>> >>> Just curious >>> >>> On Thu, Dec 11, 2014 at 1:51 PM, Chris Tunbridge >>> wrote: >>>> >>>> Errr yeah, sorry Michael is right they're origination not effective :P >>>> >>>> On Thu, Dec 11, 2014 at 8:34 AM, Michael Jerris >>>> wrote: >>>> >>>>> you should be looking at the origination_caller_id_* vars, not >>>>> effective >>>>> >>>>> On Dec 11, 2014, at 10:26 AM, Chris Tunbridge >>>>> wrote: >>>>> >>>>> The variable outbound_caller_id_number and outbound_caller_id_name are >>>>> not related to the caller id on outbound calls. >>>>> >>>>> On your outbound dial plan you need to set something like the following >>>>> >>>>> >>>> data="effective_caller_id_name=${user_data(${username}@${domain_name} >>>>> var outbound_caller_id_name)}"/> >>>>> >>>> data="effective_caller_id_number=${user_data(${username}@${domain_name} >>>>> var outbound_caller_id_number)}"/> >>>>> >>>>> This will cause the system to pull the settings from the users profile >>>>> and use it for the outgoing call. >>>>> >>>>> On Wed, Dec 10, 2014 at 12:27 PM, Joel White >>>>> wrote: >>>>> >>>>>> I have gone over the config with a fine tooth comb, it matches >>>>>> another config of a server in which the caller id works fine. What I am >>>>>> seeing however is that in this system the variable is not exported to the >>>>>> dialplan. I may be missing something, and most likely I am. I do have a >>>>>> question though. Is there a way to see what variables are defined for a >>>>>> particular user in the FreeSWITCH console? >>>>>> >>>>>> >>>>>> Here is some output of the Lua script on the server not pushing >>>>>> caller id >>>>>> >>>>>> 2014-12-09 16:54:22.819251 [NOTICE] switch_cpp.cpp:1328 Debug from >>>>>> gen_dir_user_xml.lua, generated XML: >>>>>> >>>>>> >>>>>>
>>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>> value="{presence_id=${dialed_user}@ >>>>>> ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>> value="1321XXXXXXX"/> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>
>>>>>> >>>>>> >>>>>> >>>>>> And some output from the Lua script on the server with CID functioning >>>>>> >>>>>> 2014-12-09 21:50:55.458996 [NOTICE] switch_cpp.cpp:1328 Debug from >>>>>> gen_dir_user_xml.lua, generated XML: >>>>>> >>>>>> >>>>>>
>>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>> value="{presence_id=${dialed_user}@ >>>>>> ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>> value="1303XXXXXX"/> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>
>>>>>> >>>>>> >>>>>> >>>>>> Of course I removed any identifiable information, but it looks like >>>>>> the CID is being set. What am I missing here that is not allowing for the >>>>>> variable to be passed to the dialplan? >>>>>> >>>>>> >>>>>> This is what I get when the dialplan executes >>>>>> >>>>>> EXECUTE sofia/internal/26342 at voip.net >>>>>> set(effective_caller_id_number=) >>>>>> 2014-12-09 17:00:24.839237 [DEBUG] mod_dptools.c:1435 sofia/internal/ >>>>>> 26342 at voip.net SET [effective_caller_id_number]=[UNDEF] >>>>>> >>>>>> >>>>>> Kinda strange and I could not find a discrepancy between the dialplan >>>>>> configuration of the working server vs the non-working server >>>>>> >>>>>> >>>>>> >>>>>> Here is the version running on the server that works properly >>>>>> >>>>>> FreeSWITCH Version 1.4.13+git~20141103T195300Z~b942d0faa8~64bit (git >>>>>> b942d0f 2014-11-03 19:53:00Z 64bit) >>>>>> >>>>>> And the version of the server having issue with CID >>>>>> >>>>>> FreeSWITCH Version 1.4.13+git~20141103T195300Z~ >>>>>> b942d0faa8~64bit (git b942d0f 2014-11-03 19:53:00Z 64bit) >>>>>> >>>>>> >>>>>> >>>>>> I used diff and compared both servers conf directory recursively. I >>>>>> could not find a discrepancy in the files aside from Switch name, etc. >>>>>> >>>>>> >>>>>> What am I missing? >>>>>> >>>>>> Could there be anything that that would overwrite the CID variable >>>>>> after it is set by Lua (Generating the user profile)? >>>>>> >>>>>> >>>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141215/22be7584/attachment-0001.html From michael.traut at gmail.com Mon Dec 15 20:44:38 2014 From: michael.traut at gmail.com (Michael Traut) Date: Mon, 15 Dec 2014 18:44:38 +0100 Subject: [Freeswitch-users] Freeswitch based door communication Message-ID: Sorry, i try to restate: Raspberry is running FS, connected are a door SIP and (many) other device in the house. Completely external to the devices and FS an event "bell" is generated (assume by some event on the GPIO). This should trigger FS to connect 1 Leg to the door SIP (which will answer automatically and is always a participant) and the other leg to the first in house phone to answer. The complete session is FS initiated. So i have two successful "originate" calls and need two connect the two ends together... a kind of "two man conference invitation". Am i still obscure? > To be clear, you are trying to figure out config for the client running on raspberry (using fs) or for another server? > > yes, you can do originate to multiple destinations, and the first one wins. This is built in to originate. I don't think we have any code to receive auto_answer indication, we only send it. It would require some small code changes to add this. > Mike >> On Dec 14, 2014, at 8:17 PM, Michael Traut wrote: >> >> Hi, >> >> I'm trying to build a Freeswitch / Raspberry based door communication. While there are some examples for such a constellation for Asterisk, i didn't find (and was not able to invent myself so far) a Freeswitch example. >> >> The target scenario: >> >> * A SIP based door module, >> * 1-n Pushbuttons, one for each floor >> * 1-n SIP Phones per floor >> >> When a push button is pressed i want to initiate a call to the door (that will be accepted automatically) and a call to all phones on the floor. The first one to answer gets connected to the door. >> >> Is this something i can do with "originate" ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141215/a1aebd12/attachment.html From mike at jerris.com Mon Dec 15 20:55:22 2014 From: mike at jerris.com (Michael Jerris) Date: Mon, 15 Dec 2014 12:55:22 -0500 Subject: [Freeswitch-users] Freeswitch based door communication In-Reply-To: References: Message-ID: You originate a call with auto answer to the door phone, and have it call an extension which does the fork dial (just calling bridge with originate syntax for fork dial). There is no conference in play unless you want multiple parties to talk to it at the same time. > On Dec 15, 2014, at 12:44 PM, Michael Traut wrote: > > Sorry, i try to restate: > > Raspberry is running FS, connected are a door SIP and (many) other device in the house. Completely external to the devices and FS an event "bell" is generated (assume by some event on the GPIO). This should trigger FS to connect 1 Leg to the door SIP (which will answer automatically and is always a participant) and the other leg to the first in house phone to answer. The complete session is FS initiated. > > So i have two successful "originate" calls and need two connect the two ends together... a kind of "two man conference invitation". > > Am i still obscure? > > > To be clear, you are trying to figure out config for the client running on raspberry (using fs) or for another server? > > > > yes, you can do originate to multiple destinations, and the first one wins. This is built in to originate. I don't think we have any code to receive auto_answer indication, we only send it. It would require some small code changes to add this. > > Mike > > >> On Dec 14, 2014, at 8:17 PM, Michael Traut > wrote: > >> > >> Hi, > >> > >> I'm trying to build a Freeswitch / Raspberry based door communication. While there are some examples for such a constellation for Asterisk, i didn't find (and was not able to invent myself so far) a Freeswitch example. > >> > >> The target scenario: > >> > >> * A SIP based door module, > >> * 1-n Pushbuttons, one for each floor > >> * 1-n SIP Phones per floor > >> > >> When a push button is pressed i want to initiate a call to the door (that will be accepted automatically) and a call to all phones on the floor. The first one to answer gets connected to the door. > >> > >> Is this something i can do with "originate" ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141215/202c7f3a/attachment.html From krice at freeswitch.org Mon Dec 15 20:59:22 2014 From: krice at freeswitch.org (Ken Rice) Date: Mon, 15 Dec 2014 11:59:22 -0600 Subject: [Freeswitch-users] Freeswitch based door communication In-Reply-To: Message-ID: This should be fairly straight forward... Have a process monitor the GPIO pin on the RPi for the the state change of the button press probably something like have that button pull the GPIO pin low (theres tons of docs on the RPi sites for doing this) On the the process monitoring the GPIO pin can fire an originate command to freeswith Something like # originate endpoint at door default XML ringall DOOR BELL Where endpoint at door is the proper bridge string for the door?s end point default being the dialplan context and XML being the dialplan ringall being an extension destination_number in the dialplan with DOOR being the CIDName BELL being the CIDNumber Then in the dialplan just use fork dialing style of bridging... No conferencing required... Notice the , in the data= string... The comma tells bridge to try each one at the same time and just connect the call to the one that answers first, the others will then be hung up on... On 12/15/14 11:44 AM, "Michael Traut" wrote: > Sorry, i try to restate: > > Raspberry is running FS, connected are a door SIP and (many) other device in > the house. Completely external to the devices and FS an event "bell" is > generated (assume by some event on the GPIO). This should trigger FS to > connect 1 Leg to the door SIP (which will answer automatically and is always a > participant) and the other leg to the first in house phone to answer. The > complete session is FS initiated. > > So i have two successful "originate" calls and need two connect the two ends > together... a kind of "two man conference invitation". > > Am i still obscure? > >> > To be clear, you are trying to figure out config for the client running on >> raspberry (using fs) or for another server? >> > >> >?yes, you can do originate to multiple destinations, and the first one >> wins.? This is built in to originate.? I don't think we have any code to >> receive auto_answer indication, we only send it.? It would require some small >> code changes to add this. >> >?Mike > >>> >> On Dec 14, 2014, at 8:17 PM, Michael Traut >>> wrote: >>> >> >>> >> Hi, >>> >> >>> >> I'm trying to build a Freeswitch / Raspberry based door communication. >>> While there are some examples for such a constellation for Asterisk, i >>> didn't find (and was not able to invent myself so far) a Freeswitch example. >>> >> >>> >> The target scenario: >>> >> >>> >> * A SIP based door module, >>> >> * 1-n Pushbuttons, one for each floor >>> >> * 1-n SIP Phones per floor >>> >> >>> >> When a push button is pressed i want to initiate a call to the door (that >>> will be accepted automatically) and a call to all phones on the floor. The >>> first one to answer gets connected to the door. >>> >> >>> >> Is this something i can do with "originate" ? > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 Twitter: @FreeSWITCH -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141215/f18ed9da/attachment-0001.html From michael.traut at gmail.com Mon Dec 15 21:18:56 2014 From: michael.traut at gmail.com (Michael Traut) Date: Mon, 15 Dec 2014 19:18:56 +0100 Subject: [Freeswitch-users] Freeswitch based door communication In-Reply-To: References: Message-ID: Thank you - the great news is that this is working. I hope i will get the gory details after more documentation study. What i understood is: I initiate an outgoing call , e.g. using the commandline client and do "originate". The leg is immediately bridged to a list of local users, first one winning. The syntax is roughly originate user/door &bridge(user/1001,user/1002,user/1003) ,where i have to do some fine tuning for the real url/adress syntax (sorry, but i'm a very bloody beginner in the PBX business). On Mon, Dec 15, 2014 at 6:55 PM, Michael Jerris wrote: > > You originate a call with auto answer to the door phone, and have it call > an extension which does the fork dial (just calling bridge with originate > syntax for fork dial). There is no conference in play unless you want > multiple parties to talk to it at the same time. > > On Dec 15, 2014, at 12:44 PM, Michael Traut > wrote: > > Sorry, i try to restate: > > Raspberry is running FS, connected are a door SIP and (many) other device > in the house. Completely external to the devices and FS an event "bell" is > generated (assume by some event on the GPIO). This should trigger FS to > connect 1 Leg to the door SIP (which will answer automatically and is > always a participant) and the other leg to the first in house phone to > answer. The complete session is FS initiated. > > So i have two successful "originate" calls and need two connect the two > ends together... a kind of "two man conference invitation". > > Am i still obscure? > > > To be clear, you are trying to figure out config for the client running > on raspberry (using fs) or for another server? > > > > yes, you can do originate to multiple destinations, and the first one > wins. This is built in to originate. I don't think we have any code to > receive auto_answer indication, we only send it. It would require some > small code changes to add this. > > Mike > > >> On Dec 14, 2014, at 8:17 PM, Michael Traut > wrote: > >> > >> Hi, > >> > >> I'm trying to build a Freeswitch / Raspberry based door communication. > While there are some examples for such a constellation for Asterisk, i > didn't find (and was not able to invent myself so far) a Freeswitch example. > >> > >> The target scenario: > >> > >> * A SIP based door module, > >> * 1-n Pushbuttons, one for each floor > >> * 1-n SIP Phones per floor > >> > >> When a push button is pressed i want to initiate a call to the door > (that will be accepted automatically) and a call to all phones on the > floor. The first one to answer gets connected to the door. > >> > >> Is this something i can do with "originate" ? > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141215/a777dad4/attachment.html From michael.traut at gmail.com Mon Dec 15 21:36:25 2014 From: michael.traut at gmail.com (Michael Traut) Date: Mon, 15 Dec 2014 19:36:25 +0100 Subject: [Freeswitch-users] Freeswitch based door communication In-Reply-To: References: Message-ID: thanks a lot for this additional example. i'm still a little confused if i can send this all in one command (where 9999 is door, and 100x are house phones) --> originate user/9999 &bridge(user/1001,user/1002,user/1003) or if i need to split this like in your example. It seems strange to say something like --> originate user/9999 default XML ringall DOOR BEEL and add an extension to the dialplan like Somehow "9999" does not feel like the destination_number, but what else may be the condition? Tomorrow my Grandstream test devices will show up and i am going to experiment... On Mon, Dec 15, 2014 at 6:59 PM, Ken Rice wrote: > > This should be fairly straight forward... > > Have a process monitor the GPIO pin on the RPi for the the state change of > the button press probably something like have that button pull the GPIO pin > low (theres tons of docs on the RPi sites for doing this) > > On the the process monitoring the GPIO pin can fire an originate command > to freeswith > Something like > # originate endpoint at door default XML ringall DOOR BELL > > Where endpoint at door is the proper bridge string for the door?s end point > default being the dialplan context and XML being the dialplan > ringall being an extension destination_number in the dialplan with > DOOR being the CIDName > BELL being the CIDNumber > > Then in the dialplan just use fork dialing style of bridging... No > conferencing required... > > > Notice the , in the data= string... The comma tells bridge to try each one > at the same time and just connect the call to the one that answers first, > the others will then be hung up on... > > > > > > > > > On 12/15/14 11:44 AM, "Michael Traut" wrote: > > Sorry, i try to restate: > > Raspberry is running FS, connected are a door SIP and (many) other device > in the house. Completely external to the devices and FS an event "bell" is > generated (assume by some event on the GPIO). This should trigger FS to > connect 1 Leg to the door SIP (which will answer automatically and is > always a participant) and the other leg to the first in house phone to > answer. The complete session is FS initiated. > > So i have two successful "originate" calls and need two connect the two > ends together... a kind of "two man conference invitation". > > Am i still obscure? > > > To be clear, you are trying to figure out config for the client running > on raspberry (using fs) or for another server? > > > > yes, you can do originate to multiple destinations, and the first one > wins. This is built in to originate. I don't think we have any code to > receive auto_answer indication, we only send it. It would require some > small code changes to add this. > > Mike > > >> On Dec 14, 2014, at 8:17 PM, Michael Traut > wrote: > >> > >> Hi, > >> > >> I'm trying to build a Freeswitch / Raspberry based door communication. > While there are some examples for such a constellation for Asterisk, i > didn't find (and was not able to invent myself so far) a Freeswitch example. > >> > >> The target scenario: > >> > >> * A SIP based door module, > >> * 1-n Pushbuttons, one for each floor > >> * 1-n SIP Phones per floor > >> > >> When a push button is pressed i want to initiate a call to the door > (that will be accepted automatically) and a call to all phones on the > floor. The first one to answer gets connected to the door. > >> > >> Is this something i can do with "originate" ? > > ------------------------------ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 > Twitter: @FreeSWITCH > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141215/1e85eb6d/attachment.html From alipey at gmail.com Mon Dec 15 21:37:50 2014 From: alipey at gmail.com (Ali Pey) Date: Mon, 15 Dec 2014 13:37:50 -0500 Subject: [Freeswitch-users] Freeswitch 1.4.14 Centos RPM (yum) packages missing Message-ID: Hello, There are no 1.4.14 RPM packages in the freeswitch repository. Is there a reason for this? Will they be available? Thanks, Ali Pey -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141215/c80c8961/attachment-0001.html From krice at freeswitch.org Mon Dec 15 21:43:39 2014 From: krice at freeswitch.org (Ken Rice) Date: Mon, 15 Dec 2014 12:43:39 -0600 Subject: [Freeswitch-users] Freeswitch 1.4.14 Centos RPM (yum) packages missing In-Reply-To: Message-ID: They will be available on the next release as they were broken on EL platforms... On 12/15/14 12:37 PM, "Ali Pey" wrote: > Hello, > > There are no 1.4.14 RPM packages in the freeswitch repository. Is there a > reason for this? Will they be available? > > Thanks, > Ali Pey > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 Twitter: @FreeSWITCH -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141215/821d7f95/attachment.html From alipey at gmail.com Mon Dec 15 21:58:02 2014 From: alipey at gmail.com (Ali Pey) Date: Mon, 15 Dec 2014 13:58:02 -0500 Subject: [Freeswitch-users] Freeswitch 1.4.14 Centos RPM (yum) packages missing In-Reply-To: References: Message-ID: Thank you Ken. Is there a time frame for the next release? On Mon, Dec 15, 2014 at 1:43 PM, Ken Rice wrote: > > They will be available on the next release as they were broken on EL > platforms... > > > > On 12/15/14 12:37 PM, "Ali Pey" wrote: > > Hello, > > There are no 1.4.14 RPM packages in the freeswitch repository. Is there a > reason for this? Will they be available? > > Thanks, > Ali Pey > > > ------------------------------ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 > Twitter: @FreeSWITCH > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141215/19312c3c/attachment.html From krice at freeswitch.org Mon Dec 15 22:02:20 2014 From: krice at freeswitch.org (Ken Rice) Date: Mon, 15 Dec 2014 13:02:20 -0600 Subject: [Freeswitch-users] Freeswitch 1.4.14 Centos RPM (yum) packages missing In-Reply-To: Message-ID: I don?t have an exact date but there is one currently in the works On 12/15/14 12:58 PM, "Ali Pey" wrote: > Thank you Ken. > > Is there a time frame for the next release? > > > > On Mon, Dec 15, 2014 at 1:43 PM, Ken Rice wrote: >> They will be available on the next release as they were broken on EL >> platforms... >> >> >> >> On 12/15/14 12:37 PM, "Ali Pey" > >> wrote: >> >>> Hello, >>> >>> There are no 1.4.14 RPM packages in the freeswitch repository. Is there a >>> reason for this? Will they be available? >>> >>> Thanks, >>> Ali Pey >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 Twitter: @FreeSWITCH -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141215/d717313b/attachment.html From alipey at gmail.com Mon Dec 15 22:09:50 2014 From: alipey at gmail.com (Ali Pey) Date: Mon, 15 Dec 2014 14:09:50 -0500 Subject: [Freeswitch-users] Freeswitch 1.4.14 Centos RPM (yum) packages missing In-Reply-To: References: Message-ID: Thanks Ken. On Mon, Dec 15, 2014 at 1:43 PM, Ken Rice wrote: > > They will be available on the next release as they were broken on EL > platforms... > > > > On 12/15/14 12:37 PM, "Ali Pey" wrote: > > Hello, > > There are no 1.4.14 RPM packages in the freeswitch repository. Is there a > reason for this? Will they be available? > > Thanks, > Ali Pey > > > ------------------------------ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 > Twitter: @FreeSWITCH > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141215/951a3fb6/attachment-0001.html From keith at laaks.com Mon Dec 15 22:23:10 2014 From: keith at laaks.com (Keith Laaks) Date: Mon, 15 Dec 2014 21:23:10 +0200 Subject: [Freeswitch-users] How to get the action executed using bind_digit_action ? In-Reply-To: References: Message-ID: <87D46066-A265-40A6-A07B-FE25D8071B1F@laaks.com> Hi Freeswitchers. I am running "FreeSWITCH Version 1.5.15b+git~20141212T030103Z~e1cb0e8632~64bit (git e1cb0e8 2014-12-12 03:01:03Z 64bit)? I have the following actions in an extension being executed: For the ?info? action at the top, I see the normal stuff printed in the console log: EXECUTE sofia/portaprod/27878881000 at 10.10.10.10 info() 2014-12-15 20:52:06.137537 [INFO] mod_dptools.c:1647 CHANNEL_DATA: Channel-State: [CS_EXECUTE] Channel-Call-State: [ACTIVE] Channel-State-Number: [4] Channel-Name: [sofia/portaprod/27878881000 at 10.17.180.204] Unique-ID: [8218a229-562d-4706-85a2-988d2bae0041] Then follows the ?bind_digit_action? and ?digit_action_set_realm? to bind ?##? to executing the ?info? app. The ?echo' application at the very bottom works as per normal and I get my audio echoed back. When I send ?#9? - I get this in the log: 2014-12-15 21:13:32.777537 [DEBUG] switch_rtp.c:6045 RTP RECV DTMF #:800 2014-12-15 21:13:33.637537 [DEBUG] switch_rtp.c:6045 RTP RECV DTMF 9:800 2014-12-15 21:13:33.637537 [DEBUG] mod_dptools.c:132 sofia/portaprod/27878881000 at 10.10.10.10 Digit NOT match binding [#9] So all normal. But here my issue. When I send ?##? - I only see this in the log: 2014-12-15 21:14:58.817537 [DEBUG] switch_rtp.c:6045 RTP RECV DTMF #:800 2014-12-15 21:14:59.317537 [DEBUG] switch_rtp.c:6045 RTP RECV DTMF #:800 2014-12-15 21:14:59.317537 [DEBUG] mod_dptools.c:188 sofia/portaprod/27878881000 at 10.10.10.10 Digit match binding [exec:info][both] 2014-12-15 21:14:59.317537 [DEBUG] switch_core_session.c:1188 Send signal sofia/portaprod/27878881000 at 10.10.10.10 [BREAK] So its finding a match, but the app to be executed never does so. If it had, I would have seen the above ?info? stuff printed to the console. I actually want the call transferred to another extension when the user enters ?##? - but that similarly does not happen. It seems I can?t get any action to actually be executed. So what do I need to set first, such that the specified ?action? gets executed? Hope someone can help me out here. Regards Keith From krice at freeswitch.org Mon Dec 15 22:40:15 2014 From: krice at freeswitch.org (Ken Rice) Date: Mon, 15 Dec 2014 19:40:15 +0000 Subject: [Freeswitch-users] =?utf-8?q?ClueCon_2015_=E2=80=94_Save_the_Date?= =?utf-8?q?!?= Message-ID: <548f391fb41ad_6852ea5330886fe@ip-10-184-129-92.mail> New Post on freeswitch.org from anthm check it out at http://ift.tt/1GpfnGD ClueCon 2015 ? Save the Date! Its not too soon to start planing for ClueCon 2015 August 3-7th 2015 in Chicago IL! Save the date, plan your next talk, become a sponsor!https://ClueCon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141215/97f67087/attachment.html From alipey at gmail.com Tue Dec 16 00:21:53 2014 From: alipey at gmail.com (Ali Pey) Date: Mon, 15 Dec 2014 16:21:53 -0500 Subject: [Freeswitch-users] Record sound quality not good but pcap is quite good In-Reply-To: References: Message-ID: Thanks Anthony. Will it work with 1.4.13? This is a production environment and I need to try a stable release. This is an intermittent issue. On Fri, Dec 12, 2014 at 4:43 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > > Try latest master. > > On Fri, Dec 12, 2014 at 8:54 AM, Ali Pey wrote: > >> Hello, >> >> I use record to record part of a call and sometimes I get some static >> noises and distortion; however when I listen to the call on pcap capture, >> it's quite clear and no noise. >> >> I record to a wav file and then use sox to convert it to ulaw. >> >> Any suggestions? Any help would be greatly appreciated. >> >> >> Thanks, >> Ali Pey >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 ? @anthmfs ? @FreeSWITCH ? > > ? http://freeswitch.org/ ? http://cluecon.com/ ? > http://twitter.com/FreeSWITCH > ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ > * > > ClueCon Weekly Development Call > ? sip:888 at conference.freeswitch.org ? +19193869900 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141215/4500c40e/attachment.html From anthony.minessale at gmail.com Tue Dec 16 00:27:32 2014 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 15 Dec 2014 15:27:32 -0600 Subject: [Freeswitch-users] Record sound quality not good but pcap is quite good In-Reply-To: References: Message-ID: No, the code goes through several stages before its released. So this particular issue is still at the beginning stage so it still only in master. On Mon, Dec 15, 2014 at 3:21 PM, Ali Pey wrote: > > Thanks Anthony. > > Will it work with 1.4.13? This is a production environment and I need to > try a stable release. This is an intermittent issue. > > > > On Fri, Dec 12, 2014 at 4:43 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: >> >> Try latest master. >> >> On Fri, Dec 12, 2014 at 8:54 AM, Ali Pey wrote: >> >>> Hello, >>> >>> I use record to record part of a call and sometimes I get some static >>> noises and distortion; however when I listen to the call on pcap capture, >>> it's quite clear and no noise. >>> >>> I record to a wav file and then use sox to convert it to ulaw. >>> >>> Any suggestions? Any help would be greatly appreciated. >>> >>> >>> Thanks, >>> Ali Pey >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 ? @anthmfs ? @FreeSWITCH ? >> >> ? http://freeswitch.org/ ? http://cluecon.com/ ? >> http://twitter.com/FreeSWITCH >> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ >> * >> >> ClueCon Weekly Development Call >> ? sip:888 at conference.freeswitch.org ? +19193869900 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 ? @anthmfs ? @FreeSWITCH ? ? http://freeswitch.org/ ? http://cluecon.com/ ? http://twitter.com/FreeSWITCH ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ * ClueCon Weekly Development Call ? sip:888 at conference.freeswitch.org ? +19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141215/a79ffda7/attachment-0001.html From mike at jerris.com Tue Dec 16 00:28:39 2014 From: mike at jerris.com (Michael Jerris) Date: Mon, 15 Dec 2014 16:28:39 -0500 Subject: [Freeswitch-users] Record sound quality not good but pcap is quite good In-Reply-To: References: Message-ID: <197FDF8B-7A8E-4C21-8169-52B9576BF82A@jerris.com> It will not work on 1.4.13. This issue was fixed in the last week and we need confirmation if this fix really corrected everyones issues or not. > On Dec 15, 2014, at 4:21 PM, Ali Pey wrote: > > Thanks Anthony. > > Will it work with 1.4.13? This is a production environment and I need to try a stable release. This is an intermittent issue. > > > > On Fri, Dec 12, 2014 at 4:43 PM, Anthony Minessale > wrote: > Try latest master. > > On Fri, Dec 12, 2014 at 8:54 AM, Ali Pey > wrote: > Hello, > > I use record to record part of a call and sometimes I get some static noises and distortion; however when I listen to the call on pcap capture, it's quite clear and no noise. > > I record to a wav file and then use sox to convert it to ulaw. > > Any suggestions? Any help would be greatly appreciated. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141215/00687490/attachment.html From michael.traut at gmail.com Tue Dec 16 01:17:07 2014 From: michael.traut at gmail.com (Michael Traut) Date: Mon, 15 Dec 2014 23:17:07 +0100 Subject: [Freeswitch-users] Freeswitch based door communication, 2nd Message-ID: With your help i managed understand some basics and to do some setup with two android softphones. The softphones register as 1000 and 1001 respectively. Calls between them are working. Now i try to implement the "door comm" scenario: -> originate sofia/internal/1000%192.168.0.170 2001 XML default TEST TEST will as expected call softphone 1000. When i answer, my dialplan extension seems to be found and executed - but the softphone 1001 is not ringing now. After a while the call is discarded. Here's some Log output freeswitch at MIT-T540-W8> originate sofia/internal/1000%192.168.0.170 2001 XML default TEST TEST 2014-12-15 23:12:21.265437 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/1000 [9e94d0e6-80cb-42fd-af72-22140bfaa0e2] 2014-12-15 23:12:21.385007 [NOTICE] sofia.c:6716 Ring-Ready sofia/internal/1000! 2014-12-15 23:12:26.104794 [INFO] switch_core_media.c:5328 Activating RTCP PORT 4031 2014-12-15 23:12:26.104794 [NOTICE] sofia.c:7475 Channel [sofia/internal/1000] has been answered +OK 9e94d0e6-80cb-42fd-af72-22140bfaa0e2 2014-12-15 23:12:26.104794 [NOTICE] switch_ivr.c:1854 Transfer sofia/internal/1000 to XML[2001 at default] 2014-12-15 23:12:26.104794 [INFO] mod_dialplan_xml.c:635 Processing TEST ->2001 in context default 2014-12-15 23:12:26.104794 [INFO] switch_channel.c:3062 sofia/internal/1000 Flipping CID from "TEST" to "Outbound Call" <1000%192.168.0.170> 2014-12-15 23:12:26.125865 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/sip:1001 at 192.168.0.135:42990 [c49443c5-7bc0-4482-8851-8797b5e9102d] freeswitch at MIT-T540-W8> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141215/4a63330a/attachment.html From ahabiba at gmail.com Tue Dec 16 04:01:18 2014 From: ahabiba at gmail.com (Ahmed Habiba) Date: Tue, 16 Dec 2014 04:01:18 +0300 Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 102, Issue 69 In-Reply-To: References: Message-ID: Thanks you really Michael. My point is as below: 1-when FS get Invite from A number the TO is as below: To: 2-and when I run answer application, FS send invite to A number with TO as below: To: 3-I want when I run answer application to edit the TO header and to add some prefix like 505 as below: To: is this possible, by editing sip header, and if so how can I do so? Your kind usual support will be appreciated. > > From: Michael Jerris > > To: FreeSWITCH Users Help > > Date: December 13, 2014 at 12:37:42 AM GMT+3 > Reply-To: FreeSWITCH Users Help > > Subject: Re: [Freeswitch-users] SIP Header Modification / Manipulation > > > if you are looking to update the display of the number that is being called on the caller phone, you want to be looking at display udpates, and callee_id_name and callee_id_number params. > >> On Dec 12, 2014, at 2:39 PM, Ahmed Habiba > wrote: >> >> Dears, >> >> Kindly help me on how can I modify the SIP header in the re-invite in the below scenario: >> >> >> [A Number] [FreeSWitch][B Number] >> >> >> 1-A Number Call B Number (Sip Invite sent from A number) >> 2-B number is not registered. >> 3-Call logic let FreeSwitch answer the call( Freeswitch send re-invite to A number) >> >> in the part number 3 I like to do modification on the SIP headers, by for example changing the From or To. >> >> >> Your help will be appreciated. >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141216/5fe8d7cf/attachment.html From anthony.minessale at gmail.com Tue Dec 16 04:10:51 2014 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 15 Dec 2014 19:10:51 -0600 Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 102, Issue 69 In-Reply-To: References: Message-ID: Please do not reply to the digest-emails. You must subscribe to the full list to participate in conversations. On Mon, Dec 15, 2014 at 7:01 PM, Ahmed Habiba wrote: > > Thanks you really Michael. > > My point is as below: > 1-when FS get Invite from A number the TO is as below: > To: > > 2-and when I run answer application, FS send invite to A number with TO as > below: > To: > > 3-I want when I run answer application to edit the TO header and to add > some prefix like 505 as below: > To: > > is this possible, by editing sip header, and if so how can I do so? > > Your kind usual support will be appreciated. > > > *From: *Michael Jerris > *To: *FreeSWITCH Users Help > *Date: *December 13, 2014 at 12:37:42 AM GMT+3 > *Reply-To: *FreeSWITCH Users Help > *Subject: **Re: [Freeswitch-users] SIP Header Modification / Manipulation* > > > if you are looking to update the display of the number that is being > called on the caller phone, you want to be looking at display udpates, and > callee_id_name and callee_id_number params. > > On Dec 12, 2014, at 2:39 PM, Ahmed Habiba wrote: > > Dears, > > Kindly help me on how can I modify the SIP header in the re-invite in the > below scenario: > > > [A Number] [FreeSWitch][B Number] > > > 1-A Number Call B Number (Sip Invite sent from A number) > 2-B number is not registered. > 3-Call logic let FreeSwitch answer the call( Freeswitch send re-invite to > A number) > > in the part number 3 I like to do modification on the SIP headers, by for > example changing the From or To. > > > Your help will be appreciated. > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 ? @anthmfs ? @FreeSWITCH ? ? http://freeswitch.org/ ? http://cluecon.com/ ? http://twitter.com/FreeSWITCH ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ * ClueCon Weekly Development Call ? sip:888 at conference.freeswitch.org ? +19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141215/b7072a91/attachment-0001.html From ahabiba at gmail.com Tue Dec 16 04:16:48 2014 From: ahabiba at gmail.com (Ahmed Habiba) Date: Tue, 16 Dec 2014 04:16:48 +0300 Subject: [Freeswitch-users] SIP Header Modification / Manipulation In-Reply-To: References: Message-ID: Thanks you really Michael. My point is as below: 1-when FS get Invite from A number the TO is as below: To: > 2-and when I run answer application, FS send invite to A number with TO as below: To: > 3-I want when I run answer application to edit the TO header and to add some prefix like 505 as below: To: > is this possible, by editing sip header, and if so how can I do so? Your kind usual support will be appreciated. > > From: Michael Jerris > > To: FreeSWITCH Users Help > > Date: December 13, 2014 at 12:37:42 AM GMT+3 > Reply-To: FreeSWITCH Users Help > > Subject: Re: [Freeswitch-users] SIP Header Modification / Manipulation > > > if you are looking to update the display of the number that is being called on the caller phone, you want to be looking at display udpates, and callee_id_name and callee_id_number params. > >> On Dec 12, 2014, at 2:39 PM, Ahmed Habiba > wrote: >> >> Dears, >> >> Kindly help me on how can I modify the SIP header in the re-invite in the below scenario: >> >> >> [A Number] [FreeSWitch][B Number] >> >> >> 1-A Number Call B Number (Sip Invite sent from A number) >> 2-B number is not registered. >> 3-Call logic let FreeSwitch answer the call( Freeswitch send re-invite to A number) >> >> in the part number 3 I like to do modification on the SIP headers, by for example changing the From or To. >> >> >> Your help will be appreciated. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141216/fc19ed41/attachment.html From jaimecm at gmail.com Tue Dec 16 06:04:43 2014 From: jaimecm at gmail.com (Jaime) Date: Tue, 16 Dec 2014 00:04:43 -0300 Subject: [Freeswitch-users] Hoot get variables in a LUA Hook Event. Message-ID: Hello, I?m trying to use the Hook event?s with LUA script?s when the channel Hangup a call, but I?m unable to get the variables, for example uuid, ani or billed seconds for the call, I?m using freeswitch 1.4.13 on Centos 6.6 and my configuration is: In /autoload_configs/lua.conf.xml [?] In my catch-event-cdr1.lua script I wrote: ses = freeswitch.Session(); my_uuid = ses:getVariable("uuid"); freeswitch.consoleLog("notice"," uuid=("..my_uuid..")\n") But I?m get the following error: 2014-12-15 23:51:46.100755 [ERR] switch_cpp.cpp:724 session is not initalized 2014-12-15 23:51:46.100755 [NOTICE] switch_cpp.cpp:1328 uuid=() Also, in the wiki doc I see a reference for the ?env? Object, (https://wiki.freeswitch.org/wiki/Mod_lua#Special_Case:_env_object) but if I call it as the example in the same lua script my result is: How I call the env variable: dat = env:serialize() freeswitch.consoleLog("INFO","Here's everything:\n" .. dat .. "\n?) And I got the following error: 2014-12-16 00:00:17.440782 [ERR] mod_lua.cpp:203 /usr/share/freeswitch/scripts/catch-event-reg6.lua:27: attempt to index global 'env' (a nil value) Your help will be appreciated. ? Jamie Jaimecm at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141216/943463b7/attachment.html From michael.traut at gmail.com Tue Dec 16 10:20:02 2014 From: michael.traut at gmail.com (Michael Traut) Date: Tue, 16 Dec 2014 08:20:02 +0100 Subject: [Freeswitch-users] Freeswitch based door communication Message-ID: Further experimenting with another client was successful. So, the example from my previous post is working, thank you all. But: As there's still the possibility that my two hardphones have the same strange behavior - is there a way to get protocol information to detect WHY some machines behave differently? May be there are some options with the session bind to make the other phone happy to answer? The non-working software was CSipSimple version 1.02.03, Android. The working kind was Zoiper 1.19.7 Android. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141216/67fda577/attachment.html From mthakershi at gmail.com Tue Dec 16 14:35:10 2014 From: mthakershi at gmail.com (Malay Thakershi) Date: Tue, 16 Dec 2014 17:05:10 +0530 Subject: [Freeswitch-users] Include intended modules in build Message-ID: Hello, I am trying to upgrade FS to windows 2012 server. Looking at this URL: https://wiki.freeswitch.org/wiki/Installation_for_Windows 1. Can I use VS 2013? 2. How to include necessary modules in the build - e.g. mod_managed or mod_avmd? 3. Once build is done, do I just use C:\FreeSwitch as my deployment directory? Thanks for help and happy holidays. Malay Thakershi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141216/72d8bd6e/attachment.html From akhilgarg7 at gmail.com Tue Dec 16 14:49:26 2014 From: akhilgarg7 at gmail.com (akhil garg) Date: Tue, 16 Dec 2014 17:19:26 +0530 Subject: [Freeswitch-users] fs_cli hangs In-Reply-To: References: Message-ID: Dear Sudhanshu, I have tried disabling the modules giving error on startup of freeswitch but its not helping out. It seems there is some issue related to tcp socket when using loopback adddress. Regards, Akhil On Sat, Dec 13, 2014 at 5:38 PM, Sudhanshu wrote: > > Always use syntax highlighting whenever possible. > > Have you tried disabling the modules which fail to load in your > modules.conf.xml? Ideally freeswitch should handle it, but still, it is > worth a shot. > > Also, What happens when you start freeswitch as a normal process (and not > as a daemon)? Do you see a freeswitch command prompt? > > -- > Sudhanshu > > On Fri, Dec 12, 2014 at 12:06 PM, akhil garg wrote: >> >> Please see the logs: pastebin : 23717 >> >> >> >> regards, >> akhil >> >> On Thu, Dec 11, 2014 at 10:40 PM, Sudhanshu wrote: >>> >>> Enable debug logging and then check freeswitch.log. >>> What happens when you start freeswitch as anormal process (and not as a >>> daemon)? >>> >>> -- >>> Sudhanshu >>> >>> On Thu, Dec 11, 2014 at 2:30 PM, akhil garg >>> wrote: >>> >>>> running "fs_cli -H 127.0.0.1 -P 8021 -d 7" gives different outputs but >>>> no success. >>>> >>>> >>>> >>>> ------------------------------------------------------------------------------------------------------------------------------------------------ >>>> OUTPUT 1: >>>> >>>> ------------------------------------------------------------------------------------------------------------------------------------------------ >>>> [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is >>>> /root/.fs_cli_conf. >>>> [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is >>>> /etc/fs_cli.conf. >>>> [DEBUG] fs_cli.c:1438 main() profile default does not exist using >>>> builtin profile >>>> [DEBUG] fs_cli.c:1468 main() Using profile internal [127.0.0.1] >>>> >>>> ------------------------------------------------------------------------------------------------------------------------------------------------ >>>> >>>> >>>> >>>> >>>> >>>> ------------------------------------------------------------------------------------------------------------------------------------------------ >>>> OUTPUT 2: >>>> >>>> ------------------------------------------------------------------------------------------------------------------------------------------------ >>>> [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is >>>> /root/.fs_cli_conf. >>>> [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is >>>> /etc/fs_cli.conf. >>>> [DEBUG] fs_cli.c:1438 main() profile default does not exist using >>>> builtin profile >>>> [DEBUG] fs_cli.c:1468 main() Using profile internal [127.0.0.1] >>>> [DEBUG] esl.c:1265 esl_recv_event() RECV HEADER [Content-Type] = >>>> [auth/request] >>>> [DEBUG] esl.c:1437 esl_recv_event() RECV MESSAGE >>>> Event-Name: SOCKET_DATA >>>> Content-Type: auth/request >>>> >>>> >>>> [DEBUG] esl.c:1465 esl_send() SEND >>>> auth ClueCon >>>> >>>> ------------------------------------------------------------------------------------------------------------------------------------------------ >>>> >>>> >>>> >>>> >>>> >>>> ------------------------------------------------------------------------------------------------------------------------------------------------ >>>> OUTPUT 3: >>>> >>>> ------------------------------------------------------------------------------------------------------------------------------------------------ >>>> [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is >>>> /root/.fs_cli_conf. >>>> [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is >>>> /etc/fs_cli.conf. >>>> [DEBUG] fs_cli.c:1438 main() profile default does not exist using >>>> builtin profile >>>> [DEBUG] fs_cli.c:1468 main() Using profile internal [127.0.0.1] >>>> [DEBUG] esl.c:1265 esl_recv_event() RECV HEADER [Content-Type] = >>>> [auth/request] >>>> [DEBUG] esl.c:1437 esl_recv_event() RECV MESSAGE >>>> Event-Name: SOCKET_DATA >>>> Content-Type: auth/request >>>> >>>> >>>> [DEBUG] esl.c:1465 esl_send() SEND >>>> auth ClueCon >>>> >>>> >>>> [DEBUG] esl.c:1265 esl_recv_event() RECV HEADER [Content-Type] = >>>> [command/reply] >>>> [DEBUG] esl.c:1265 esl_recv_event() RECV HEADER [Reply-Text] = [+OK >>>> accepted] >>>> [DEBUG] esl.c:1437 esl_recv_event() RECV MESSAGE >>>> Event-Name: SOCKET_DATA >>>> Content-Type: command/reply >>>> Reply-Text: +OK accepted >>>> >>>> >>>> [DEBUG] esl.c:1465 esl_send() SEND >>>> log >>>> >>>> ------------------------------------------------------------------------------------------------------------------------------------------------ >>>> >>>> >>>> >>>> -- >>>> regards, >>>> akhil >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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, >> akhil >> > -- regards, akhil -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141216/01201c62/attachment-0001.html From david.witham at netsip.com.au Tue Dec 16 10:43:48 2014 From: david.witham at netsip.com.au (David Witham) Date: Tue, 16 Dec 2014 07:43:48 +0000 Subject: [Freeswitch-users] Hoot get variables in a LUA Hook Event. In-Reply-To: References: Message-ID: <1d3146c630444d80bb9d05e67cab03c0@OTW-BNE-EXC01.otw.internal> Hi Jamie, What we have done is add these lines in the dialplan: Then in /scripts/myscript.lua: cdr_uuid = session:getVariable("cdr_uuid"); This is on a FreeSWITCH 1.2 box. Hope this helps. regards, David ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org on behalf of Jaime Sent: Tuesday, 16 December 2014 13:04 To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Hoot get variables in a LUA Hook Event. Hello, I'm trying to use the Hook event's with LUA script's when the channel Hangup a call, but I'm unable to get the variables, for example uuid, ani or billed seconds for the call, I'm using freeswitch 1.4.13 on Centos 6.6 and my configuration is: In /autoload_configs/lua.conf.xml [...] In my catch-event-cdr1.lua script I wrote: ses = freeswitch.Session(); my_uuid = ses:getVariable("uuid"); freeswitch.consoleLog("notice"," uuid=("..my_uuid..")\n") But I'm get the following error: 2014-12-15 23:51:46.100755 [ERR] switch_cpp.cpp:724 session is not initalized 2014-12-15 23:51:46.100755 [NOTICE] switch_cpp.cpp:1328 uuid=() Also, in the wiki doc I see a reference for the "env" Object, (https://wiki.freeswitch.org/wiki/Mod_lua#Special_Case:_env_object) but if I call it as the example in the same lua script my result is: How I call the env variable: dat = env:serialize() freeswitch.consoleLog("INFO","Here's everything:\n" .. dat .. "\n") And I got the following error: 2014-12-16 00:00:17.440782 [ERR] mod_lua.cpp:203 /usr/share/freeswitch/scripts/catch-event-reg6.lua:27: attempt to index global 'env' (a nil value) Your help will be appreciated. - Jamie Jaimecm at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141216/408a8662/attachment-0001.html From keith at laaks.com Tue Dec 16 12:17:49 2014 From: keith at laaks.com (Keith Laaks) Date: Tue, 16 Dec 2014 11:17:49 +0200 Subject: [Freeswitch-users] How to get the action executed using bind_digit_action ? Message-ID: Hi Freeswitchers. (Reposting under new thread - previous went to the wrong place.) I am running "FreeSWITCH Version 1.5.15b+git~20141215T224714Z~0b414a8de8~64bit (git 0b414a8 2014-12-15 22:47:14Z 64bit)? - updated this morning. I have the following actions in an extension being executed: For the ?info? action at the top, I see the normal stuff printed in the console log: EXECUTE sofia/portaprod/27878881000 at 10.10.10.10 info() 2014-12-15 20:52:06.137537 [INFO] mod_dptools.c:1647 CHANNEL_DATA: Channel-State: [CS_EXECUTE] Channel-Call-State: [ACTIVE] Channel-State-Number: [4] Channel-Name: [sofia/portaprod/27878881000 at 10.17.180.204 ] Unique-ID: [8218a229-562d-4706-85a2-988d2bae0041] Then follows the ?bind_digit_action? and ?digit_action_set_realm? to bind ?##? to executing the ?info? app. The ?echo' application at the very bottom works as per normal and I get my audio echoed back. When I send ?#9? - I get this in the log: 2014-12-15 21:13:32.777537 [DEBUG] switch_rtp.c:6045 RTP RECV DTMF #:800 2014-12-15 21:13:33.637537 [DEBUG] switch_rtp.c:6045 RTP RECV DTMF 9:800 2014-12-15 21:13:33.637537 [DEBUG] mod_dptools.c:132 sofia/portaprod/27878881000 at 10.10.10.10 Digit NOT match binding [#9] So all normal. But here my issue. When I send ?##? - I only see this in the log: 2014-12-15 21:14:58.817537 [DEBUG] switch_rtp.c:6045 RTP RECV DTMF #:800 2014-12-15 21:14:59.317537 [DEBUG] switch_rtp.c:6045 RTP RECV DTMF #:800 2014-12-15 21:14:59.317537 [DEBUG] mod_dptools.c:188 sofia/portaprod/27878881000 at 10.10.10.10 Digit match binding [exec:info][both] 2014-12-15 21:14:59.317537 [DEBUG] switch_core_session.c:1188 Send signal sofia/portaprod/27878881000 at 10.10.10.10 [BREAK] So its finding a match, but the app to be executed never does so. If it had, I would have seen the above ?info? stuff printed to the console. I actually want the call transferred to another extension when the user enters ?##? - but that similarly does not happen. It seems I can?t get any action to actually be executed. So what do I need to set first, such that the specified ?action? gets executed? Hope someone can help me out here. Looking at the code, I see that in mod_dptools.c there seems to be support for ?blocking? and ?non-blocking? execution modes. I figured out that to ?activate? the ?non-blocking mode the above action needs to change to: [This sets ?exec = 2? such that ?switch_core_session_execute_application? is called - otherwise ?switch_ivr_broadcast_in_thread? is called.] But when I try this, as soon as I send the ##, I get a core dump. Regards Keith -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141216/aee46eeb/attachment-0001.html From bilaln018 at gmail.com Tue Dec 16 14:25:03 2014 From: bilaln018 at gmail.com (Bilal Abbasi) Date: Tue, 16 Dec 2014 16:25:03 +0500 Subject: [Freeswitch-users] [ERR][mod_json_cdr.c][Open of json_cdr.conf failed] Message-ID: Hi All, I have successfully compiled mod_json_cdr but when i go to fs_cli and load mod_json_cdr i get the below error please help. Plus there is no file named json_cdr.conf in autoconfig directory. I think FS is unable to locate that file. freeswitch at internal> load mod_json_cdr +OK Reloading XML -ERR [module load file routine returned an error] freeswitch at internal> 2014-12-16 06:37:25.377189 [INFO] switch_time.c:1369 Timezone reloaded 530 definitions 2014-12-16 06:37:25.377189 [ERR] mod_json_cdr.c:598 Open of json_cdr.conf failed 2014-12-16 06:37:25.377189 [CRIT] switch_loadable_module.c:1447 Error Loading module /usr/local/freeswitch/mod/mod_json_cdr.so **Module load routine returned an error** Regards Bilal Abbasi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141216/09cde1d1/attachment-0001.html From doron at lexifone.com Tue Dec 16 15:29:51 2014 From: doron at lexifone.com (Doron Kruh) Date: Tue, 16 Dec 2014 14:29:51 +0200 Subject: [Freeswitch-users] Originate call SIP call-id Message-ID: Hi All, How can I make sure the SIP Call-Id header sent by Freeswitch when originating a new call is in the form of: uuid at host ? Thank you, Doron -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141216/a2d64f34/attachment.html From bilaln018 at gmail.com Tue Dec 16 15:52:54 2014 From: bilaln018 at gmail.com (Bilal Abbasi) Date: Tue, 16 Dec 2014 17:52:54 +0500 Subject: [Freeswitch-users] [ERR][mod_json_cdr.c][Open of json_cdr.conf failed] In-Reply-To: References: Message-ID: Hi All, I have solved that... I have located the json_cdr.conf.xml file from below path /usr/local/src/freeswitch/src/mod/event_handlers/mod_json_cdr/conf/autoload_configs/json_cdr.conf.xml and copied that to the INSTALLED_LOCATION/conf/autoload_configs/json_cdr.conf.xml after that load mod_json_cdr went successfull. Many thanks Bilal Abbasi On Tue, Dec 16, 2014 at 4:25 PM, Bilal Abbasi wrote: > > Hi All, > I have successfully compiled mod_json_cdr but when i go to fs_cli and load > mod_json_cdr i get the below error please help. > Plus there is no file named json_cdr.conf in autoconfig directory. > I think FS is unable to locate that file. > > freeswitch at internal> load mod_json_cdr > +OK Reloading XML > -ERR [module load file routine returned an error] > > freeswitch at internal> 2014-12-16 06:37:25.377189 [INFO] switch_time.c:1369 > Timezone reloaded 530 definitions > 2014-12-16 06:37:25.377189 [ERR] mod_json_cdr.c:598 Open of json_cdr.conf > failed > 2014-12-16 06:37:25.377189 [CRIT] switch_loadable_module.c:1447 Error > Loading module /usr/local/freeswitch/mod/mod_json_cdr.so > **Module load routine returned an error** > > Regards > Bilal Abbasi > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141216/41865217/attachment.html From snabel at lexifone.com Tue Dec 16 15:57:14 2014 From: snabel at lexifone.com (Snabel Kabiya) Date: Tue, 16 Dec 2014 14:57:14 +0200 Subject: [Freeswitch-users] Get caller IP address Message-ID: Hi, How can i get the caller IP from session? In sip trace I've: Call-ID: 7XHCXNSJABE2NGOBAHSXRZIGPA at 1.2.3.4 i want to get the IP 1.2.3.4 i my Lua script, anyone knows how? Thanks, Snabel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141216/fd0be917/attachment.html From sos at sokhapkin.dyndns.org Tue Dec 16 16:06:31 2014 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Tue, 16 Dec 2014 08:06:31 -0500 Subject: [Freeswitch-users] Get caller IP address In-Reply-To: References: Message-ID: <4098462.2GfFJKnuE9@sos> Don't relay on Call-ID header, it may but may not contain IP address. In many cases it could contain internal IP address of NATed client. Many SIP clients do not include IP address to call-id. On Tuesday 16 December 2014 14:57:14 Snabel Kabiya wrote: > Hi, > > How can i get the caller IP from session? > > In sip trace I've: > > Call-ID: 7XHCXNSJABE2NGOBAHSXRZIGPA at 1.2.3.4 > > i want to get the IP 1.2.3.4 i my Lua script, anyone knows how? > > Thanks, > Snabel From aqsyounas at gmail.com Tue Dec 16 16:08:26 2014 From: aqsyounas at gmail.com (Aqs Younas) Date: Tue, 16 Dec 2014 18:08:26 +0500 Subject: [Freeswitch-users] [ERR][mod_json_cdr.c][Open of json_cdr.conf failed] In-Reply-To: References: Message-ID: Ok Abbasi sahb On 16 December 2014 at 17:52, Bilal Abbasi wrote: > > Hi All, > I have solved that... > I have located the json_cdr.conf.xml file from below path > > /usr/local/src/freeswitch/src/mod/event_handlers/mod_json_cdr/conf/autoload_configs/json_cdr.conf.xml > and copied that to the > INSTALLED_LOCATION/conf/autoload_configs/json_cdr.conf.xml > > after that load mod_json_cdr went successfull. > > Many thanks > Bilal Abbasi > > On Tue, Dec 16, 2014 at 4:25 PM, Bilal Abbasi wrote: >> >> Hi All, >> I have successfully compiled mod_json_cdr but when i go to fs_cli and >> load mod_json_cdr i get the below error please help. >> Plus there is no file named json_cdr.conf in autoconfig directory. >> I think FS is unable to locate that file. >> >> freeswitch at internal> load mod_json_cdr >> +OK Reloading XML >> -ERR [module load file routine returned an error] >> >> freeswitch at internal> 2014-12-16 06:37:25.377189 [INFO] >> switch_time.c:1369 Timezone reloaded 530 definitions >> 2014-12-16 06:37:25.377189 [ERR] mod_json_cdr.c:598 Open of >> json_cdr.conf failed >> 2014-12-16 06:37:25.377189 [CRIT] switch_loadable_module.c:1447 Error >> Loading module /usr/local/freeswitch/mod/mod_json_cdr.so >> **Module load routine returned an error** >> >> Regards >> Bilal Abbasi >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141216/0204db99/attachment.html From steveayre at gmail.com Tue Dec 16 17:05:58 2014 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 16 Dec 2014 14:05:58 +0000 Subject: [Freeswitch-users] Originate call SIP call-id In-Reply-To: References: Message-ID: The normal FS behaviour is to just use a uuid, but you could implement this manually with: Of course doing this for every originate/bridge would be tedious. There's no requirement in SIP for it to be of the uuid at host format, it's simply defined as a byte string. Is there a specific reason you need it to be so? If you wish you could submit a patch to add this option via a profile flag. On 16 December 2014 at 12:29, Doron Kruh wrote: > > Hi All, > > How can I make sure the SIP Call-Id header sent by Freeswitch when > originating a new call is in the form of: uuid at host ? > > Thank you, > > Doron > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141216/da025cfb/attachment-0001.html From brian at freeswitch.org Tue Dec 16 17:16:58 2014 From: brian at freeswitch.org (Brian West) Date: Tue, 16 Dec 2014 08:16:58 -0600 Subject: [Freeswitch-users] Originate call SIP call-id In-Reply-To: References: Message-ID: Bigger question is why would the call id format matter? On Tue, Dec 16, 2014 at 8:05 AM, Steven Ayre wrote: > > The normal FS behaviour is to just use a uuid, but you could implement > this manually with: > > > > Of course doing this for every originate/bridge would be tedious. > > There's no requirement in SIP for it to be of the uuid at host format, it's > simply defined as a byte string. Is there a specific reason you need it to > be so? If you wish you could submit a patch to add this option via a > profile flag. > > On 16 December 2014 at 12:29, Doron Kruh wrote: > >> Hi All, >> >> How can I make sure the SIP Call-Id header sent by Freeswitch when >> originating a new call is in the form of: uuid at host ? >> >> Thank you, >> >> Doron >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141216/a15e311e/attachment.html From brian at freeswitch.org Tue Dec 16 17:17:30 2014 From: brian at freeswitch.org (Brian West) Date: Tue, 16 Dec 2014 08:17:30 -0600 Subject: [Freeswitch-users] fs_cli hangs In-Reply-To: References: Message-ID: Debian? On Tue, Dec 16, 2014 at 5:49 AM, akhil garg wrote: > > Dear Sudhanshu, > I have tried disabling the modules giving error on startup of freeswitch > but its not helping out. > It seems there is some issue related to tcp socket when using loopback > adddress. > > > Regards, > Akhil > > On Sat, Dec 13, 2014 at 5:38 PM, Sudhanshu wrote: >> >> Always use syntax highlighting whenever possible. >> >> Have you tried disabling the modules which fail to load in your >> modules.conf.xml? Ideally freeswitch should handle it, but still, it is >> worth a shot. >> >> Also, What happens when you start freeswitch as a normal process (and >> not as a daemon)? Do you see a freeswitch command prompt? >> >> -- >> Sudhanshu >> >> On Fri, Dec 12, 2014 at 12:06 PM, akhil garg >> wrote: >>> >>> Please see the logs: pastebin : 23717 >>> >>> >>> >>> regards, >>> akhil >>> >>> On Thu, Dec 11, 2014 at 10:40 PM, Sudhanshu wrote: >>>> >>>> Enable debug logging and then check freeswitch.log. >>>> What happens when you start freeswitch as anormal process (and not as a >>>> daemon)? >>>> >>>> -- >>>> Sudhanshu >>>> >>>> On Thu, Dec 11, 2014 at 2:30 PM, akhil garg >>>> wrote: >>>> >>>>> running "fs_cli -H 127.0.0.1 -P 8021 -d 7" gives different outputs but >>>>> no success. >>>>> >>>>> >>>>> >>>>> ------------------------------------------------------------------------------------------------------------------------------------------------ >>>>> OUTPUT 1: >>>>> >>>>> ------------------------------------------------------------------------------------------------------------------------------------------------ >>>>> [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is >>>>> /root/.fs_cli_conf. >>>>> [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is >>>>> /etc/fs_cli.conf. >>>>> [DEBUG] fs_cli.c:1438 main() profile default does not exist using >>>>> builtin profile >>>>> [DEBUG] fs_cli.c:1468 main() Using profile internal [127.0.0.1] >>>>> >>>>> ------------------------------------------------------------------------------------------------------------------------------------------------ >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> ------------------------------------------------------------------------------------------------------------------------------------------------ >>>>> OUTPUT 2: >>>>> >>>>> ------------------------------------------------------------------------------------------------------------------------------------------------ >>>>> [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is >>>>> /root/.fs_cli_conf. >>>>> [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is >>>>> /etc/fs_cli.conf. >>>>> [DEBUG] fs_cli.c:1438 main() profile default does not exist using >>>>> builtin profile >>>>> [DEBUG] fs_cli.c:1468 main() Using profile internal [127.0.0.1] >>>>> [DEBUG] esl.c:1265 esl_recv_event() RECV HEADER [Content-Type] = >>>>> [auth/request] >>>>> [DEBUG] esl.c:1437 esl_recv_event() RECV MESSAGE >>>>> Event-Name: SOCKET_DATA >>>>> Content-Type: auth/request >>>>> >>>>> >>>>> [DEBUG] esl.c:1465 esl_send() SEND >>>>> auth ClueCon >>>>> >>>>> ------------------------------------------------------------------------------------------------------------------------------------------------ >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> ------------------------------------------------------------------------------------------------------------------------------------------------ >>>>> OUTPUT 3: >>>>> >>>>> ------------------------------------------------------------------------------------------------------------------------------------------------ >>>>> [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is >>>>> /root/.fs_cli_conf. >>>>> [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is >>>>> /etc/fs_cli.conf. >>>>> [DEBUG] fs_cli.c:1438 main() profile default does not exist using >>>>> builtin profile >>>>> [DEBUG] fs_cli.c:1468 main() Using profile internal [127.0.0.1] >>>>> [DEBUG] esl.c:1265 esl_recv_event() RECV HEADER [Content-Type] = >>>>> [auth/request] >>>>> [DEBUG] esl.c:1437 esl_recv_event() RECV MESSAGE >>>>> Event-Name: SOCKET_DATA >>>>> Content-Type: auth/request >>>>> >>>>> >>>>> [DEBUG] esl.c:1465 esl_send() SEND >>>>> auth ClueCon >>>>> >>>>> >>>>> [DEBUG] esl.c:1265 esl_recv_event() RECV HEADER [Content-Type] = >>>>> [command/reply] >>>>> [DEBUG] esl.c:1265 esl_recv_event() RECV HEADER [Reply-Text] = [+OK >>>>> accepted] >>>>> [DEBUG] esl.c:1437 esl_recv_event() RECV MESSAGE >>>>> Event-Name: SOCKET_DATA >>>>> Content-Type: command/reply >>>>> Reply-Text: +OK accepted >>>>> >>>>> >>>>> [DEBUG] esl.c:1465 esl_send() SEND >>>>> log >>>>> >>>>> ------------------------------------------------------------------------------------------------------------------------------------------------ >>>>> >>>>> >>>>> >>>>> -- >>>>> regards, >>>>> akhil >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at 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, >>> akhil >>> >> > > -- > regards, > akhil > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest 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URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141216/9c995dd2/attachment-0001.html From brian at freeswitch.org Tue Dec 16 17:17:56 2014 From: brian at freeswitch.org (Brian West) Date: Tue, 16 Dec 2014 08:17:56 -0600 Subject: [Freeswitch-users] Get caller IP address In-Reply-To: <4098462.2GfFJKnuE9@sos> References: <4098462.2GfFJKnuE9@sos> Message-ID: network_addr variable will have the real IP in it. On Tue, Dec 16, 2014 at 7:06 AM, Sergey Okhapkin wrote: > > Don't relay on Call-ID header, it may but may not contain IP address. In > many > cases it could contain internal IP address of NATed client. Many SIP > clients > do not include IP address to call-id. > > On Tuesday 16 December 2014 14:57:14 Snabel Kabiya wrote: > > Hi, > > > > How can i get the caller IP from session? > > > > In sip trace I've: > > > > Call-ID: 7XHCXNSJABE2NGOBAHSXRZIGPA at 1.2.3.4 > > > > i want to get the IP 1.2.3.4 i my Lua script, anyone knows how? > > > > Thanks, > > Snabel > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141216/218e18b7/attachment.html From steveayre at gmail.com Tue Dec 16 17:56:25 2014 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 16 Dec 2014 14:56:25 +0000 Subject: [Freeswitch-users] How to get the action executed using bind_digit_action ? In-Reply-To: References: Message-ID: If you get a core dump file a Jira. Core dumps are always a bug. On 16 December 2014 at 09:17, Keith Laaks wrote: > > Hi Freeswitchers. > > (Reposting under new thread - previous went to the wrong place.) > > I am running "FreeSWITCH Version > 1.5.15b+git~20141215T224714Z~0b414a8de8~64bit (git 0b414a8 2014-12-15 > 22:47:14Z 64bit)? - updated this morning. > > > I have the following actions in an extension being executed: > > > data="start_echo,##,exec:info,both,self"/> > > > > For the ?info? action at the top, I see the normal stuff printed in the > console log: > EXECUTE sofia/portaprod/27878881000 at 10.10.10.10 info() > 2014-12-15 20:52:06.137537 [INFO] mod_dptools.c:1647 CHANNEL_DATA: > Channel-State: [CS_EXECUTE] > Channel-Call-State: [ACTIVE] > Channel-State-Number: [4] > Channel-Name: [sofia/portaprod/27878881000 at 10.17.180.204] > Unique-ID: [8218a229-562d-4706-85a2-988d2bae0041] > > > Then follows the ?bind_digit_action? and ?digit_action_set_realm? to bind > ?##? to executing the ?info? app. > > The ?echo' application at the very bottom works as per normal and I get my > audio echoed back. > > When I send ?#9? - I get this in the log: > 2014-12-15 21:13:32.777537 [DEBUG] switch_rtp.c:6045 RTP RECV DTMF #:800 > 2014-12-15 21:13:33.637537 [DEBUG] switch_rtp.c:6045 RTP RECV DTMF 9:800 > 2014-12-15 21:13:33.637537 [DEBUG] mod_dptools.c:132 > sofia/portaprod/27878881000 at 10.10.10.10 Digit NOT match binding [#9] > > So all normal. > > But here my issue. > > When I send ?##? - I only see this in the log: > 2014-12-15 21:14:58.817537 [DEBUG] switch_rtp.c:6045 RTP RECV DTMF #:800 > 2014-12-15 21:14:59.317537 [DEBUG] switch_rtp.c:6045 RTP RECV DTMF #:800 > 2014-12-15 21:14:59.317537 [DEBUG] mod_dptools.c:188 > sofia/portaprod/27878881000 at 10.10.10.10 Digit match binding > [exec:info][both] > 2014-12-15 21:14:59.317537 [DEBUG] switch_core_session.c:1188 Send signal > sofia/portaprod/27878881000 at 10.10.10.10 [BREAK] > > So its finding a match, but the app to be executed never does so. If it > had, I would have seen the above ?info? stuff printed to the console. > I actually want the call transferred to another extension when the user > enters ?##? - but that similarly does not happen. It seems I can?t get any > action to actually be executed. > > So what do I need to set first, such that the specified ?action? gets > executed? > > Hope someone can help me out here. > > Looking at the code, I see that in mod_dptools.c there seems to be support > for ?blocking? and ?non-blocking? execution modes. > I figured out that to ?activate? the ?non-blocking mode the above action > needs to change to: > data="start_echo,##,exec[i]:info,both,self"/> > [This sets ?exec = 2? such that ?switch_core_session_execute_application? > is called - otherwise ?switch_ivr_broadcast_in_thread? is called.] > > But when I try this, as soon as I send the ##, I get a core dump. > > > > Regards > > Keith > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141216/be56e685/attachment.html From nbhatti at gmail.com Tue Dec 16 17:57:27 2014 From: nbhatti at gmail.com (Muhammad Naseer Bhatti) Date: Tue, 16 Dec 2014 17:57:27 +0300 Subject: [Freeswitch-users] Strange xml_curl behavior. Can't see more than 7 variables on POST Message-ID: I am using xml_curl to generate all config dynamically and this is getting very strange, my directory looks like this,? ?
? ?? ? ? ?? ? ?? ? ?? ? ?? ? ? ? ? ? ?? ? ?? ? ?? ? ?? ? ?? ? ?? ? ?? ? ?? ? ?? ? ?? ? ?? ? ? ? ?? ? ?
 The directory is working fine and user is able to login. I have a bunch of variables defined here. When I look at the POST on the web server, I am only able to see first 7 variable no matter what their names are. If I change order for example account_type which is last variable, and make it on top, it shows there in the XML_REQUEST but then hangup_after_bridge goes away. I see only first 7. Are there any limitation on what channel variables can be POSTed to the web server or am I missing something here.? --? Muhammad Naseer Bhatti -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141216/98fdfaa0/attachment-0001.html From steveayre at gmail.com Tue Dec 16 17:58:18 2014 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 16 Dec 2014 14:58:18 +0000 Subject: [Freeswitch-users] How to get the action executed using bind_digit_action ? In-Reply-To: References: Message-ID: Try this (info takes no arguments) Note the double comma On 16 December 2014 at 09:17, Keith Laaks wrote: > > Hi Freeswitchers. > > (Reposting under new thread - previous went to the wrong place.) > > I am running "FreeSWITCH Version > 1.5.15b+git~20141215T224714Z~0b414a8de8~64bit (git 0b414a8 2014-12-15 > 22:47:14Z 64bit)? - updated this morning. > > > I have the following actions in an extension being executed: > > > data="start_echo,##,exec:info,both,self"/> > > > > For the ?info? action at the top, I see the normal stuff printed in the > console log: > EXECUTE sofia/portaprod/27878881000 at 10.10.10.10 info() > 2014-12-15 20:52:06.137537 [INFO] mod_dptools.c:1647 CHANNEL_DATA: > Channel-State: [CS_EXECUTE] > Channel-Call-State: [ACTIVE] > Channel-State-Number: [4] > Channel-Name: [sofia/portaprod/27878881000 at 10.17.180.204] > Unique-ID: [8218a229-562d-4706-85a2-988d2bae0041] > > > Then follows the ?bind_digit_action? and ?digit_action_set_realm? to bind > ?##? to executing the ?info? app. > > The ?echo' application at the very bottom works as per normal and I get my > audio echoed back. > > When I send ?#9? - I get this in the log: > 2014-12-15 21:13:32.777537 [DEBUG] switch_rtp.c:6045 RTP RECV DTMF #:800 > 2014-12-15 21:13:33.637537 [DEBUG] switch_rtp.c:6045 RTP RECV DTMF 9:800 > 2014-12-15 21:13:33.637537 [DEBUG] mod_dptools.c:132 > sofia/portaprod/27878881000 at 10.10.10.10 Digit NOT match binding [#9] > > So all normal. > > But here my issue. > > When I send ?##? - I only see this in the log: > 2014-12-15 21:14:58.817537 [DEBUG] switch_rtp.c:6045 RTP RECV DTMF #:800 > 2014-12-15 21:14:59.317537 [DEBUG] switch_rtp.c:6045 RTP RECV DTMF #:800 > 2014-12-15 21:14:59.317537 [DEBUG] mod_dptools.c:188 > sofia/portaprod/27878881000 at 10.10.10.10 Digit match binding > [exec:info][both] > 2014-12-15 21:14:59.317537 [DEBUG] switch_core_session.c:1188 Send signal > sofia/portaprod/27878881000 at 10.10.10.10 [BREAK] > > So its finding a match, but the app to be executed never does so. If it > had, I would have seen the above ?info? stuff printed to the console. > I actually want the call transferred to another extension when the user > enters ?##? - but that similarly does not happen. It seems I can?t get any > action to actually be executed. > > So what do I need to set first, such that the specified ?action? gets > executed? > > Hope someone can help me out here. > > Looking at the code, I see that in mod_dptools.c there seems to be support > for ?blocking? and ?non-blocking? execution modes. > I figured out that to ?activate? the ?non-blocking mode the above action > needs to change to: > data="start_echo,##,exec[i]:info,both,self"/> > [This sets ?exec = 2? such that ?switch_core_session_execute_application? > is called - otherwise ?switch_ivr_broadcast_in_thread? is called.] > > But when I try this, as soon as I send the ##, I get a core dump. > > > > Regards > > Keith > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141216/3ef41052/attachment.html From nbhatti at gmail.com Tue Dec 16 18:01:19 2014 From: nbhatti at gmail.com (Muhammad Naseer Bhatti) Date: Tue, 16 Dec 2014 07:01:19 -0800 (PST) Subject: [Freeswitch-users] Strange xml_curl behavior. Can't see more than 7 variables on POST In-Reply-To: References: Message-ID: <687890D7-E063-42AA-A049-19F630C2C29B@gmail.com> Forgot to mention that I am also pulling up dialplan configuration from xml_curl. I am seeing all the variables when request for a dialplan. Sent with Unibox > On Dec 16, 2014, at 5:57 PM, Muhammad Naseer Bhatti wrote: > > > I am using xml_curl to generate all config dynamically and this is getting very strange, my directory looks like this,? > > > > > > > ?
> > ? > > ? > > ? > > ? > > ? > > ? > > ? > > ? > > ? > > ? > > ? > > ? > > ? > > ? > > ? > > ? > > ? > > ? > > ? > > ? > > ? > > ? > > ? > > ?
> > > > > > > The directory is working fine and user is able to login. I have a bunch of variables defined here. When I look at the POST on the web server, I am only able to see first 7 variable no matter what their names are. If I change order for example?account_type > which is last variable, and make it on top, it shows there in the XML_REQUEST but then?hangup_after_bridge > goes away. I see only first 7. Are there any limitation on what channel variables can be POSTed to the web server or am I missing something here.? > > --? > Muhammad Naseer Bhatti > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141216/c926e228/attachment.html From brian at freeswitch.org Tue Dec 16 18:19:57 2014 From: brian at freeswitch.org (Brian West) Date: Tue, 16 Dec 2014 09:19:57 -0600 Subject: [Freeswitch-users] Strange xml_curl behavior. Can't see more than 7 variables on POST In-Reply-To: <687890D7-E063-42AA-A049-19F630C2C29B@gmail.com> References: <687890D7-E063-42AA-A049-19F630C2C29B@gmail.com> Message-ID: Does it behave differently without the cacheable tag? On Tue, Dec 16, 2014 at 9:01 AM, Muhammad Naseer Bhatti wrote: > > > Forgot to mention that I am also pulling up dialplan configuration from > xml_curl. I am seeing all the variables when request for a dialplan. > > > Sent with Unibox > > On Dec 16, 2014, at 5:57 PM, Muhammad Naseer Bhatti > wrote: > > I am using xml_curl to generate all config dynamically and this is getting > very strange, my directory looks like this, > > > > > >
> > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > >
> > > > The directory is working fine and user is able to login. I have a bunch of > variables defined here. When I look at the POST on the web server, I am > only able to see first 7 variable no matter what their names are. If I > change order for example *account_type*which is last variable, and make > it on top, it shows there in the XML_REQUEST but then > *hangup_after_bridge*goes away. I see only first 7. Are there any > limitation on what channel variables can be POSTed to the web server or am > I missing something here. > > -- > Muhammad Naseer Bhatti > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141216/3261b728/attachment-0001.html From nbhatti at gmail.com Tue Dec 16 18:26:12 2014 From: nbhatti at gmail.com (Muhammad Naseer Bhatti) Date: Tue, 16 Dec 2014 18:26:12 +0300 Subject: [Freeswitch-users] Strange xml_curl behavior. Can't see more than 7 variables on POST In-Reply-To: References: <687890D7-E063-42AA-A049-19F630C2C29B@gmail.com> Message-ID: No, it?s the same with or without cacheable set to 0 or anything else. Here is the list of all variables I can see,?https://pastebin.freeswitch.org/23740 ? --? Muhammad Naseer Bhatti Sent with Airmail From:?Brian West Reply:?FreeSWITCH Users Help > Date:?December 16, 2014 at 6:20:48 PM To:?FreeSWITCH Users Help > Subject:? Re: [Freeswitch-users] Strange xml_curl behavior. Can't see more than 7 variables on POST Does it behave differently without the cacheable tag? On Tue, Dec 16, 2014 at 9:01 AM, Muhammad Naseer Bhatti wrote: Forgot to mention that I am also pulling up dialplan configuration from xml_curl. I am seeing all the variables when request for a dialplan. Sent with Unibox On Dec 16, 2014, at 5:57 PM, Muhammad Naseer Bhatti wrote: I am using xml_curl to generate all config dynamically and this is getting very strange, my directory looks like this,? ?
? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?
 The directory is working fine and user is able to login. I have a bunch of variables defined here. When I look at the POST on the web server, I am only able to see first 7 variable no matter what their names are. If I change order for example?account_typewhich is last variable, and make it on top, it shows there in the XML_REQUEST but then?hangup_after_bridgegoes away. I see only first 7. Are there any limitation on what channel variables can be POSTed to the web server or am I missing something here.? --? Muhammad Naseer Bhatti _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Brian West brian at freeswitch.org Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378) iNUM:+883 5100 1420 9001 |?ISN:410*543 |?Skype:briankwest _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141216/45b417e5/attachment.html From nbhatti at gmail.com Tue Dec 16 19:31:03 2014 From: nbhatti at gmail.com (Muhammad Naseer Bhatti) Date: Tue, 16 Dec 2014 19:31:03 +0300 Subject: [Freeswitch-users] Strange xml_curl behavior. Can't see more than 7 variables on POST In-Reply-To: References: <687890D7-E063-42AA-A049-19F630C2C29B@gmail.com> Message-ID: Since I am using nginx to serve the requests, it turns out to be?ngx.req.get_post_args() takes a limited number of arguments in the body by default. 100 to be exact by default. If I change the value to a larger number, I am able to see all the variables in POST. ngx.req.read_body() local XML_REQUEST = ngx.req.get_post_args(1000) for k, v in pairs (XML_REQUEST) do ? if (k:match('variable')) then ? ? log.debug ('['..k .. '] : [' .. v..']') ? end end Solved..? --? Muhammad Naseer Bhatti From:?Muhammad Naseer Bhatti Reply:?Muhammad Naseer Bhatti > Date:?December 16, 2014 at 6:26:21 PM To:?FreeSWITCH Users Help >, Brian West > Subject:? Re: [Freeswitch-users] Strange xml_curl behavior. Can't see more than 7 variables on POST No, it?s the same with or without cacheable set to 0 or anything else. Here is the list of all variables I can see,?https://pastebin.freeswitch.org/23740 ? --? Muhammad Naseer Bhatti Sent with Airmail From:?Brian West Reply:?FreeSWITCH Users Help > Date:?December 16, 2014 at 6:20:48 PM To:?FreeSWITCH Users Help > Subject:? Re: [Freeswitch-users] Strange xml_curl behavior. Can't see more than 7 variables on POST Does it behave differently without the cacheable tag? On Tue, Dec 16, 2014 at 9:01 AM, Muhammad Naseer Bhatti wrote: Forgot to mention that I am also pulling up dialplan configuration from xml_curl. I am seeing all the variables when request for a dialplan. Sent with Unibox On Dec 16, 2014, at 5:57 PM, Muhammad Naseer Bhatti wrote: I am using xml_curl to generate all config dynamically and this is getting very strange, my directory looks like this,? ?
? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?
 The directory is working fine and user is able to login. I have a bunch of variables defined here. When I look at the POST on the web server, I am only able to see first 7 variable no matter what their names are. If I change order for example?account_typewhich is last variable, and make it on top, it shows there in the XML_REQUEST but then?hangup_after_bridgegoes away. I see only first 7. Are there any limitation on what channel variables can be POSTed to the web server or am I missing something here.? --? Muhammad Naseer Bhatti _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Brian West brian at freeswitch.org Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378) iNUM:+883 5100 1420 9001 |?ISN:410*543 |?Skype:briankwest _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141216/0864f7c0/attachment-0001.html From frederick at targointernet.com Tue Dec 16 19:41:02 2014 From: frederick at targointernet.com (Frederick Pruneau) Date: Tue, 16 Dec 2014 11:41:02 -0500 Subject: [Freeswitch-users] Issue with Freeswitch Behind nat Message-ID: Hi guys, We have an issue with one freeswitch server behind nat. We have a setup like this: -One master Freeswitch server -One freeswitch server connected to the master (Public IP) - Server A -One freeswitch server connected to the master (behind nat) - Server B If server A call server B, nothing happens. There is no sound. After 30 sec, it times out. We have done a tcpdump. From server A to master packets are ok. From Master to server B, we have seen that there is no source and no destination ports for sip invite. If we use our cellphone and we call server B, there is no problem. I have attached the failed call pcap file and freeswitch's log file so you can take a look at them. Master = Freeswitch v1.4.13 Server A = Freeswitch v.1.4.13 Server B = Freeswitch v.1.4.14 (Updated to latest release since we have issues with this server) Thanks in advance. PS: The failed call is from 514-448-0773. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141216/86e8d1dc/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: fail_remote_dump_SIP_Invite_No_Port.pcap Type: application/octet-stream Size: 35441 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141216/86e8d1dc/attachment-0002.obj -------------- next part -------------- A non-text attachment was scrubbed... Name: freeswitch.log Type: application/octet-stream Size: 4644109 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141216/86e8d1dc/attachment-0003.obj From brian at freeswitch.org Tue Dec 16 20:25:40 2014 From: brian at freeswitch.org (Brian West) Date: Tue, 16 Dec 2014 11:25:40 -0600 Subject: [Freeswitch-users] Issue with Freeswitch Behind nat In-Reply-To: References: Message-ID: On the system behind nat what do you have ext-rtp-ip, ext-sip-ip and local-network-acl set to? On Tue, Dec 16, 2014 at 10:41 AM, Frederick Pruneau < frederick at targointernet.com> wrote: > > Hi guys, > > We have an issue with one freeswitch server behind nat. We have a setup > like this: > > -One master Freeswitch server > > -One freeswitch server connected to the master (Public IP) - Server A > > -One freeswitch server connected to the master (behind nat) - Server B > > If server A call server B, nothing happens. There is no sound. After 30 > sec, it times out. We have done a tcpdump. From server A to master packets > are ok. From Master to server B, we have seen that there is no source and > no destination ports for sip invite. > > If we use our cellphone and we call server B, there is no problem. > > I have attached the failed call pcap file and freeswitch's log file so you > can take a look at them. > > Master = Freeswitch v1.4.13 > Server A = Freeswitch v.1.4.13 > Server B = Freeswitch v.1.4.14 (Updated to latest release since we have > issues with this server) > > Thanks in advance. > > PS: The failed call is from 514-448-0773. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141216/5eec265a/attachment.html From mike at jerris.com Tue Dec 16 20:33:27 2014 From: mike at jerris.com (Michael Jerris) Date: Tue, 16 Dec 2014 12:33:27 -0500 Subject: [Freeswitch-users] SIP Header Modification / Manipulation In-Reply-To: References: Message-ID: <8D690492-722F-4EB3-9A3F-62391C15C191@jerris.com> Your scenario doesn?t make any sense. We do not send an invite on answer. You haven?t actually answered what you are trying to actually accomplish, just what you think you want the sip traffic to be in a scenario that doesn?t exist. > On Dec 15, 2014, at 8:16 PM, Ahmed Habiba wrote: > > Thanks you really Michael. > > My point is as below: > 1-when FS get Invite from A number the TO is as below: > To: > > > 2-and when I run answer application, FS send invite to A number with TO as below: > To: > > > 3-I want when I run answer application to edit the TO header and to add some prefix like 505 as below: > To: > > > is this possible, by editing sip header, and if so how can I do so? > > Your kind usual support will be appreciated. > >> >> From: Michael Jerris > >> To: FreeSWITCH Users Help > >> Date: December 13, 2014 at 12:37:42 AM GMT+3 >> Reply-To: FreeSWITCH Users Help > >> Subject: Re: [Freeswitch-users] SIP Header Modification / Manipulation >> >> >> if you are looking to update the display of the number that is being called on the caller phone, you want to be looking at display udpates, and callee_id_name and callee_id_number params. >> >>> On Dec 12, 2014, at 2:39 PM, Ahmed Habiba > wrote: >>> >>> Dears, >>> >>> Kindly help me on how can I modify the SIP header in the re-invite in the below scenario: >>> >>> >>> [A Number] [FreeSWitch][B Number] >>> >>> >>> 1-A Number Call B Number (Sip Invite sent from A number) >>> 2-B number is not registered. >>> 3-Call logic let FreeSwitch answer the call( Freeswitch send re-invite to A number) >>> >>> in the part number 3 I like to do modification on the SIP headers, by for example changing the From or To. >>> >>> >>> Your help will be appreciated. > _________________________________________________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141216/71337511/attachment.html From mike at jerris.com Tue Dec 16 20:34:48 2014 From: mike at jerris.com (Michael Jerris) Date: Tue, 16 Dec 2014 12:34:48 -0500 Subject: [Freeswitch-users] Include intended modules in build In-Reply-To: References: Message-ID: There are a few outstanding issues for vs 2013. We are working to correct those still. Keep an eye out for fixes. > On Dec 16, 2014, at 6:35 AM, Malay Thakershi wrote: > > Hello, > > I am trying to upgrade FS to windows 2012 server. > > Looking at this URL: https://wiki.freeswitch.org/wiki/Installation_for_Windows > > 1. Can I use VS 2013? > 2. How to include necessary modules in the build - e.g. mod_managed or mod_avmd? > 3. Once build is done, do I just use C:\FreeSwitch as my deployment directory? > > Thanks for help and happy holidays. > > Malay Thakershi > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141216/f9034e8e/attachment.html From mike at jerris.com Tue Dec 16 20:37:48 2014 From: mike at jerris.com (Michael Jerris) Date: Tue, 16 Dec 2014 12:37:48 -0500 Subject: [Freeswitch-users] How to get the action executed using bind_digit_action ? In-Reply-To: References: Message-ID: <78BB442F-25DF-486D-A0C6-635AD3331B2A@jerris.com> Please file a lira for the crash you are getting. (test on master first to confirm we have not already fixed it please). > On Dec 16, 2014, at 4:17 AM, Keith Laaks wrote: > > Hi Freeswitchers. > > (Reposting under new thread - previous went to the wrong place.) > > I am running "FreeSWITCH Version 1.5.15b+git~20141215T224714Z~0b414a8de8~64bit (git 0b414a8 2014-12-15 22:47:14Z 64bit)? - updated this morning. > > > I have the following actions in an extension being executed: > > > > > > > For the ?info? action at the top, I see the normal stuff printed in the console log: > EXECUTE sofia/portaprod/27878881000 at 10.10.10.10 info() > 2014-12-15 20:52:06.137537 [INFO] mod_dptools.c:1647 CHANNEL_DATA: > Channel-State: [CS_EXECUTE] > Channel-Call-State: [ACTIVE] > Channel-State-Number: [4] > Channel-Name: [sofia/portaprod/27878881000 at 10.17.180.204 ] > Unique-ID: [8218a229-562d-4706-85a2-988d2bae0041] > > > Then follows the ?bind_digit_action? and ?digit_action_set_realm? to bind ?##? to executing the ?info? app. > > The ?echo' application at the very bottom works as per normal and I get my audio echoed back. > > When I send ?#9? - I get this in the log: > 2014-12-15 21:13:32.777537 [DEBUG] switch_rtp.c:6045 RTP RECV DTMF #:800 > 2014-12-15 21:13:33.637537 [DEBUG] switch_rtp.c:6045 RTP RECV DTMF 9:800 > 2014-12-15 21:13:33.637537 [DEBUG] mod_dptools.c:132 sofia/portaprod/27878881000 at 10.10.10.10 Digit NOT match binding [#9] > > So all normal. > > But here my issue. > > When I send ?##? - I only see this in the log: > 2014-12-15 21:14:58.817537 [DEBUG] switch_rtp.c:6045 RTP RECV DTMF #:800 > 2014-12-15 21:14:59.317537 [DEBUG] switch_rtp.c:6045 RTP RECV DTMF #:800 > 2014-12-15 21:14:59.317537 [DEBUG] mod_dptools.c:188 sofia/portaprod/27878881000 at 10.10.10.10 Digit match binding [exec:info][both] > 2014-12-15 21:14:59.317537 [DEBUG] switch_core_session.c:1188 Send signal sofia/portaprod/27878881000 at 10.10.10.10 [BREAK] > > So its finding a match, but the app to be executed never does so. If it had, I would have seen the above ?info? stuff printed to the console. > I actually want the call transferred to another extension when the user enters ?##? - but that similarly does not happen. It seems I can?t get any action to actually be executed. > > So what do I need to set first, such that the specified ?action? gets executed? > > Hope someone can help me out here. > > Looking at the code, I see that in mod_dptools.c there seems to be support for ?blocking? and ?non-blocking? execution modes. > I figured out that to ?activate? the ?non-blocking mode the above action needs to change to: > > [This sets ?exec = 2? such that ?switch_core_session_execute_application? is called - otherwise ?switch_ivr_broadcast_in_thread? is called.] > > But when I try this, as soon as I send the ##, I get a core dump. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141216/3ec50f81/attachment-0001.html From support at targointernet.com Tue Dec 16 21:15:45 2014 From: support at targointernet.com (Support Technique) Date: Tue, 16 Dec 2014 13:15:45 -0500 Subject: [Freeswitch-users] Issue with Freeswitch Behind nat In-Reply-To: References: Message-ID: 2014-12-16 12:25 GMT-05:00 Brian West : > > On the system behind nat what do you have ext-rtp-ip, ext-sip-ip and > local-network-acl set to? > > On Tue, Dec 16, 2014 at 10:41 AM, Frederick Pruneau < > frederick at targointernet.com> wrote: > >> Hi guys, >> >> We have an issue with one freeswitch server behind nat. We have a setup >> like this: >> >> -One master Freeswitch server >> >> -One freeswitch server connected to the master (Public IP) - Server A >> >> -One freeswitch server connected to the master (behind nat) - Server B >> >> If server A call server B, nothing happens. There is no sound. After 30 >> sec, it times out. We have done a tcpdump. From server A to master packets >> are ok. From Master to server B, we have seen that there is no source and >> no destination ports for sip invite. >> >> If we use our cellphone and we call server B, there is no problem. >> >> I have attached the failed call pcap file and freeswitch's log file so >> you can take a look at them. >> >> Master = Freeswitch v1.4.13 >> Server A = Freeswitch v.1.4.13 >> Server B = Freeswitch v.1.4.14 (Updated to latest release since we have >> issues with this server) >> >> Thanks in advance. >> >> PS: The failed call is from 514-448-0773. >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > -- > > *Brian West* > brian at freeswitch.org > > > *Twitter: @FreeSWITCH , @briankwest* > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141216/811ae796/attachment.html From brian at freeswitch.org Tue Dec 16 21:55:24 2014 From: brian at freeswitch.org (Brian West) Date: Tue, 16 Dec 2014 12:55:24 -0600 Subject: [Freeswitch-users] Issue with Freeswitch Behind nat In-Reply-To: References: Message-ID: Guessing you don't have UPNP or NAT-PMP on your network, there for that won't work, ext-sip-ip=autonat:x.x.x.x ext-rtp-ip=autonat:x.x.x.x Set local-network-ac to rfc1918.auto On Tue, Dec 16, 2014 at 12:15 PM, Support Technique < support at targointernet.com> wrote: > > > > > > 2014-12-16 12:25 GMT-05:00 Brian West : > >> On the system behind nat what do you have ext-rtp-ip, ext-sip-ip and >> local-network-acl set to? >> >> On Tue, Dec 16, 2014 at 10:41 AM, Frederick Pruneau < >> frederick at targointernet.com> wrote: >> >>> Hi guys, >>> >>> We have an issue with one freeswitch server behind nat. We have a setup >>> like this: >>> >>> -One master Freeswitch server >>> >>> -One freeswitch server connected to the master (Public IP) - Server A >>> >>> -One freeswitch server connected to the master (behind nat) - Server B >>> >>> If server A call server B, nothing happens. There is no sound. After 30 >>> sec, it times out. We have done a tcpdump. From server A to master packets >>> are ok. From Master to server B, we have seen that there is no source and >>> no destination ports for sip invite. >>> >>> If we use our cellphone and we call server B, there is no problem. >>> >>> I have attached the failed call pcap file and freeswitch's log file so >>> you can take a look at them. >>> >>> Master = Freeswitch v1.4.13 >>> Server A = Freeswitch v.1.4.13 >>> Server B = Freeswitch v.1.4.14 (Updated to latest release since we have >>> issues with this server) >>> >>> Thanks in advance. >>> >>> PS: The failed call is from 514-448-0773. >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> -- >> >> *Brian West* >> brian at freeswitch.org >> >> >> *Twitter: @FreeSWITCH , @briankwest* >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141216/e9ed42ce/attachment.html From brian at freeswitch.org Tue Dec 16 22:04:07 2014 From: brian at freeswitch.org (Brian West) Date: Tue, 16 Dec 2014 13:04:07 -0600 Subject: [Freeswitch-users] How to get the action executed using bind_digit_action ? In-Reply-To: <78BB442F-25DF-486D-A0C6-635AD3331B2A@jerris.com> References: <78BB442F-25DF-486D-A0C6-635AD3331B2A@jerris.com> Message-ID: I'm not getting the crash, but I am getting the exec:info no-op, please file the JIRA. On Tue, Dec 16, 2014 at 11:37 AM, Michael Jerris wrote: > > Please file a lira for the crash you are getting. (test on master first to > confirm we have not already fixed it please). > > On Dec 16, 2014, at 4:17 AM, Keith Laaks wrote: > > Hi Freeswitchers. > > (Reposting under new thread - previous went to the wrong place.) > > I am running "FreeSWITCH Version > 1.5.15b+git~20141215T224714Z~0b414a8de8~64bit (git 0b414a8 2014-12-15 > 22:47:14Z 64bit)? - updated this morning. > > > I have the following actions in an extension being executed: > > > data="start_echo,##,exec:info,both,self"/> > > > > For the ?info? action at the top, I see the normal stuff printed in the > console log: > EXECUTE sofia/portaprod/27878881000 at 10.10.10.10 info() > 2014-12-15 20:52:06.137537 [INFO] mod_dptools.c:1647 CHANNEL_DATA: > Channel-State: [CS_EXECUTE] > Channel-Call-State: [ACTIVE] > Channel-State-Number: [4] > Channel-Name: [sofia/portaprod/27878881000 at 10.17.180.204] > Unique-ID: [8218a229-562d-4706-85a2-988d2bae0041] > > > Then follows the ?bind_digit_action? and ?digit_action_set_realm? to bind > ?##? to executing the ?info? app. > > The ?echo' application at the very bottom works as per normal and I get my > audio echoed back. > > When I send ?#9? - I get this in the log: > 2014-12-15 21:13:32.777537 [DEBUG] switch_rtp.c:6045 RTP RECV DTMF #:800 > 2014-12-15 21:13:33.637537 [DEBUG] switch_rtp.c:6045 RTP RECV DTMF 9:800 > 2014-12-15 21:13:33.637537 [DEBUG] mod_dptools.c:132 > sofia/portaprod/27878881000 at 10.10.10.10 Digit NOT match binding [#9] > > So all normal. > > But here my issue. > > When I send ?##? - I only see this in the log: > 2014-12-15 21:14:58.817537 [DEBUG] switch_rtp.c:6045 RTP RECV DTMF #:800 > 2014-12-15 21:14:59.317537 [DEBUG] switch_rtp.c:6045 RTP RECV DTMF #:800 > 2014-12-15 21:14:59.317537 [DEBUG] mod_dptools.c:188 > sofia/portaprod/27878881000 at 10.10.10.10 Digit match binding > [exec:info][both] > 2014-12-15 21:14:59.317537 [DEBUG] switch_core_session.c:1188 Send signal > sofia/portaprod/27878881000 at 10.10.10.10 [BREAK] > > So its finding a match, but the app to be executed never does so. If it > had, I would have seen the above ?info? stuff printed to the console. > I actually want the call transferred to another extension when the user > enters ?##? - but that similarly does not happen. It seems I can?t get any > action to actually be executed. > > So what do I need to set first, such that the specified ?action? gets > executed? > > Hope someone can help me out here. > > Looking at the code, I see that in mod_dptools.c there seems to be support > for ?blocking? and ?non-blocking? execution modes. > I figured out that to ?activate? the ?non-blocking mode the above action > needs to change to: > data="start_echo,##,exec[i]:info,both,self"/> > [This sets ?exec = 2? such that ?switch_core_session_execute_application? > is called - otherwise ?switch_ivr_broadcast_in_thread? is called.] > > But when I try this, as soon as I send the ##, I get a core dump. > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141216/aec26aba/attachment-0001.html From joelewhite at gmail.com Tue Dec 16 22:25:00 2014 From: joelewhite at gmail.com (Joel White) Date: Tue, 16 Dec 2014 14:25:00 -0500 Subject: [Freeswitch-users] CallerID for E911 Message-ID: Hi everyone, I was wondering what the maximum length a caller id number can be. I need to implement appending a location ID to each extension not assigned a DID. This is for purposes of the E911 system. Does anyone have experience with this? Any help would be greatly appreciated -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141216/de77e740/attachment.html From frederick at targointernet.com Tue Dec 16 23:37:10 2014 From: frederick at targointernet.com (Frederick Pruneau) Date: Tue, 16 Dec 2014 15:37:10 -0500 Subject: [Freeswitch-users] Issue with Freeswitch Behind nat In-Reply-To: References: Message-ID: Same problem... 2014-12-16 13:55 GMT-05:00 Brian West : > > Guessing you don't have UPNP or NAT-PMP on your network, there for that > won't work, > > ext-sip-ip=autonat:x.x.x.x > ext-rtp-ip=autonat:x.x.x.x > > Set local-network-ac to rfc1918.auto > > On Tue, Dec 16, 2014 at 12:15 PM, Support Technique < > support at targointernet.com> wrote: >> >> >> >> >> >> 2014-12-16 12:25 GMT-05:00 Brian West : >> >>> On the system behind nat what do you have ext-rtp-ip, ext-sip-ip and >>> local-network-acl set to? >>> >>> On Tue, Dec 16, 2014 at 10:41 AM, Frederick Pruneau < >>> frederick at targointernet.com> wrote: >>> >>>> Hi guys, >>>> >>>> We have an issue with one freeswitch server behind nat. We have a setup >>>> like this: >>>> >>>> -One master Freeswitch server >>>> >>>> -One freeswitch server connected to the master (Public IP) - Server A >>>> >>>> -One freeswitch server connected to the master (behind nat) - Server B >>>> >>>> If server A call server B, nothing happens. There is no sound. After 30 >>>> sec, it times out. We have done a tcpdump. From server A to master packets >>>> are ok. From Master to server B, we have seen that there is no source and >>>> no destination ports for sip invite. >>>> >>>> If we use our cellphone and we call server B, there is no problem. >>>> >>>> I have attached the failed call pcap file and freeswitch's log file so >>>> you can take a look at them. >>>> >>>> Master = Freeswitch v1.4.13 >>>> Server A = Freeswitch v.1.4.13 >>>> Server B = Freeswitch v.1.4.14 (Updated to latest release since we have >>>> issues with this server) >>>> >>>> Thanks in advance. >>>> >>>> PS: The failed call is from 514-448-0773. >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> -- >>> >>> *Brian West* >>> brian at freeswitch.org >>> >>> >>> *Twitter: @FreeSWITCH , @briankwest* >>> http://www.freeswitchbook.com >>> http://www.freeswitchcookbook.com >>> >>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > -- > > *Brian West* > brian at freeswitch.org > > > *Twitter: @FreeSWITCH , @briankwest* > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141216/73df6ea4/attachment.html From brian at freeswitch.org Tue Dec 16 23:39:17 2014 From: brian at freeswitch.org (Brian West) Date: Tue, 16 Dec 2014 14:39:17 -0600 Subject: [Freeswitch-users] CallerID for E911 In-Reply-To: References: Message-ID: How do you need to present this information when you originate a call? On Tue, Dec 16, 2014 at 1:25 PM, Joel White wrote: > > Hi everyone, > > I was wondering what the maximum length a caller id number can be. I need > to implement appending a location ID to each extension not assigned a DID. > This is for purposes of the E911 system. Does anyone have experience with > this? > > Any help would be greatly appreciated > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141216/3910e4b2/attachment.html From brian at freeswitch.org Tue Dec 16 23:44:55 2014 From: brian at freeswitch.org (Brian West) Date: Tue, 16 Dec 2014 14:44:55 -0600 Subject: [Freeswitch-users] Issue with Freeswitch Behind nat In-Reply-To: References: Message-ID: have you looked at the signalling? What does the sip traffic show? Please pastebin that. On Tue, Dec 16, 2014 at 2:37 PM, Frederick Pruneau < frederick at targointernet.com> wrote: > > Same problem... > > 2014-12-16 13:55 GMT-05:00 Brian West : > >> Guessing you don't have UPNP or NAT-PMP on your network, there for that >> won't work, >> >> ext-sip-ip=autonat:x.x.x.x >> ext-rtp-ip=autonat:x.x.x.x >> >> Set local-network-ac to rfc1918.auto >> >> On Tue, Dec 16, 2014 at 12:15 PM, Support Technique < >> support at targointernet.com> wrote: >>> >>> >>> >>> >>> >>> 2014-12-16 12:25 GMT-05:00 Brian West : >>> >>>> On the system behind nat what do you have ext-rtp-ip, ext-sip-ip and >>>> local-network-acl set to? >>>> >>>> On Tue, Dec 16, 2014 at 10:41 AM, Frederick Pruneau < >>>> frederick at targointernet.com> wrote: >>>> >>>>> Hi guys, >>>>> >>>>> We have an issue with one freeswitch server behind nat. We have a >>>>> setup like this: >>>>> >>>>> -One master Freeswitch server >>>>> >>>>> -One freeswitch server connected to the master (Public IP) - Server A >>>>> >>>>> -One freeswitch server connected to the master (behind nat) - Server B >>>>> >>>>> If server A call server B, nothing happens. There is no sound. After >>>>> 30 sec, it times out. We have done a tcpdump. From server A to master >>>>> packets are ok. From Master to server B, we have seen that there is no >>>>> source and no destination ports for sip invite. >>>>> >>>>> If we use our cellphone and we call server B, there is no problem. >>>>> >>>>> I have attached the failed call pcap file and freeswitch's log file so >>>>> you can take a look at them. >>>>> >>>>> Master = Freeswitch v1.4.13 >>>>> Server A = Freeswitch v.1.4.13 >>>>> Server B = Freeswitch v.1.4.14 (Updated to latest release since we >>>>> have issues with this server) >>>>> >>>>> Thanks in advance. >>>>> >>>>> PS: The failed call is from 514-448-0773. >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> -- >>>> >>>> *Brian West* >>>> brian at freeswitch.org >>>> >>>> >>>> *Twitter: @FreeSWITCH , @briankwest* >>>> http://www.freeswitchbook.com >>>> http://www.freeswitchcookbook.com >>>> >>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> -- >> >> *Brian West* >> brian at freeswitch.org >> >> >> *Twitter: @FreeSWITCH , @briankwest* >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141216/4b38080e/attachment-0001.html From fcastelco at gmail.com Tue Dec 16 23:57:01 2014 From: fcastelco at gmail.com (Federico Castro) Date: Tue, 16 Dec 2014 17:57:01 -0300 Subject: [Freeswitch-users] Freeswitch HA advices In-Reply-To: References: Message-ID: Hi Eliot and all, I keep on working in my HA solution. I?m almost done but there are some things I would like to improve. When I finish it I will share my solution with the community. I have two FS Boxes with Postgres running locally. Psql is running in read-write mode in Master node and in read-only mode in Slave. Psql is streaming data asynchronously through nodes. First thing I would like to improve: FS try to DROP some tables and delete records from sip_registrations, channels, etc. when connecting to psql and if it fails mod_sofia stops loading. This behaviour produces (in my scenary) that FS in Slave server does not start properly If it connects to local database because it is in read-only mode. To solve this I configured FS to connect to psql through virtual IP (it is always in Master node) then mod_sofia starts properly in Slave node. The problem here is that FS in slave node delete records from table in Master node when get connected. Is there any way to start FS in a ?standby mode? in slave node to avoid it trying to write database? Thanks! 2014-06-16 12:05 GMT-03:00 Federico Castro : > > Hi Eliot, thanks for your verbose response, it is really useful for me. > > I'm working on a duplicated FS + postgreSQL schema. The two boxes will > have same HW. Both of them will run FS and postgreSQL. One will act as > master and the other one as stand-by waiting for the first to fail. > > FS will not have more than a hundred simultaneous calls. > > I will read about Pacemaker and Corosync and I will update to the list > about the implementation. > > Thanks again. > > > > 2014-06-13 10:45 GMT-03:00 Eliot Gable : > >> On Tue, Jun 10, 2014 at 10:56 AM, Federico Castro >> wrote: >> >>> Hi all, I'm working on a Freeswitch HA solution. Now I'm deciding what >>> method and DB I'll use to track calls. >>> >>> I have installed PostgreSQL on both servers and I configured them to >>> replicate DB asynchronously. >>> >>> I would like to know if someone has experience with this kind of >>> solution and what things do I have to contemplate to deploy a solid >>> solution. >>> >>> >> Lots of people have experience with such a solution; it all depends on >> what you are trying to achieve. >> >> Personally, I recommend you setup Corosync and Pacemaker both on your >> PostgreSQL boxes and on your FreeSWITCH systems. I also recommend you run >> PostgreSQL on a separate set of boxes from FS. Both can use a lot of memory >> if you are running a lot of calls and/or have a lot of clients. If you need >> performance, I recommend using the fastest disks you can get in the >> PostgreSQL systems. Also install as much RAM as you can afford for the >> project in the PGSQL boxes. You will want redundant power supplies in each >> system with each supply plugged into a different circuit. You will also >> want redundant Ethernet connectivity to redundant switches which also have >> redundant power supplies. You will also want redundant cross-over >> connections between the pairs of boxes. >> >> Once you have Corosync and Pacemaker configured to start PGSQL and FS on >> their own boxes and you have tested manual fail-over, then you need to >> start thinking about every possible way you can make either of those two >> systems stop working. Think about hard drives failing, power loss, kernel >> panics, firewall rules blocking communication, someone accidentally >> removing the IP address from one of the systems (it happens), killing >> processes, Sofia profiles failing to load because something else is using >> the port, etc. Make sure you have things set up to detect and recover from >> any such failure. One of the best ways to do this is to actually build an >> external testing system which places real calls through the system and has >> them route back to itself to verify they made it. If it places a call and >> the call does not make it back to itself, then you know something failed >> and you can run more tests to determine what failed and reset it. >> >> Like I said, it all depends on what you are trying to accomplish. If you >> want really good automatic HA, you have to go to some pretty great lengths >> to get it. If you are OK with occasional manual intervention, then you can >> make some assumptions (like nobody accidentally removing your IP from the >> interface or telling it to stop responding to ARP or throwing up a firewall >> rule which blocks something). That makes the setup considerably easier, but >> it also means manual intervention when something like that happens. In >> other words, if something like that happens, you experience an outage which >> the HA system doesn't detect and recover from. When you get calls that >> service stopped working, you then have someone log in and take a look and >> manually fix the issue. This could take anywhere from 5 minutes to an hour >> or more to do, depending on how good your support is and how good your team >> is. >> >> So, probably the first task you should do is list all the things you want >> it to automatically recover from and all the things you are willing to >> accept causing an outage and then work on your implementation based on that >> plan. >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141216/0829b2e5/attachment.html From akhilgarg7 at gmail.com Wed Dec 17 11:56:12 2014 From: akhilgarg7 at gmail.com (akhil garg) Date: Wed, 17 Dec 2014 14:26:12 +0530 Subject: [Freeswitch-users] fs_cli hangs Message-ID: Its not Debian but mips -- regards, akhil -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141217/5bb44e9f/attachment.html From steveayre at gmail.com Wed Dec 17 12:17:02 2014 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 17 Dec 2014 09:17:02 +0000 Subject: [Freeswitch-users] SIP Header Modification / Manipulation In-Reply-To: References: Message-ID: If the INVITE comes from A then on answering the call the response to A will be 200 OK, not another invite. The To header in the 200 OK has to match the value in the INVITE, as they're part of the same dialog. The rules of the SIP protocol mean needs you to reply to A within the call using the values you were originally sent in the original INVITE. If you need to provide information back to A I suggest you look at the sip_rh_ family of variables. They'll allow you to send extra headers (eg X- headers) in the 200 OK. On 16 December 2014 at 01:16, Ahmed Habiba wrote: > > Thanks you really Michael. > > My point is as below: > 1-when FS get Invite from A number the TO is as below: > To: > > 2-and when I run answer application, FS send invite to A number with TO as > below: > To: > > 3-I want when I run answer application to edit the TO header and to add > some prefix like 505 as below: > To: > > is this possible, by editing sip header, and if so how can I do so? > > Your kind usual support will be appreciated. > > > *From: *Michael Jerris > *To: *FreeSWITCH Users Help > *Date: *December 13, 2014 at 12:37:42 AM GMT+3 > *Reply-To: *FreeSWITCH Users Help > *Subject: **Re: [Freeswitch-users] SIP Header Modification / Manipulation* > > > if you are looking to update the display of the number that is being > called on the caller phone, you want to be looking at display udpates, and > callee_id_name and callee_id_number params. > > On Dec 12, 2014, at 2:39 PM, Ahmed Habiba wrote: > > Dears, > > Kindly help me on how can I modify the SIP header in the re-invite in the > below scenario: > > > [A Number] [FreeSWitch][B Number] > > > 1-A Number Call B Number (Sip Invite sent from A number) > 2-B number is not registered. > 3-Call logic let FreeSwitch answer the call( Freeswitch send re-invite to > A number) > > in the part number 3 I like to do modification on the SIP headers, by for > example changing the From or To. > > > Your help will be appreciated. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141217/64d56236/attachment.html From sukithaj at gmail.com Wed Dec 17 14:30:23 2014 From: sukithaj at gmail.com (sukitha jayasinghe) Date: Wed, 17 Dec 2014 17:00:23 +0530 Subject: [Freeswitch-users] Conference no sound Message-ID: Hi All, I have configured freeswitch server with conference feature enabled, When I call the conference number, i can hear hold music. when second call dial the conference number hold music stops, but no voice between legs. what would be the reason for this. Best Regards, Sukitha -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141217/72e60dda/attachment-0001.html From italorossib at gmail.com Wed Dec 17 15:41:08 2014 From: italorossib at gmail.com (=?UTF-8?B?w410YWxvIFJvc3Np?=) Date: Wed, 17 Dec 2014 09:41:08 -0300 Subject: [Freeswitch-users] Conference no sound In-Reply-To: References: Message-ID: Is there audio when you call the other user directly, without the conference? On Wed, Dec 17, 2014 at 8:30 AM, sukitha jayasinghe wrote: > > Hi All, > > I have configured freeswitch server with conference feature enabled, When > I call the conference number, i can hear hold music. when second call dial > the conference number hold music stops, but no voice between legs. what > would be the reason for this. > > Best Regards, > Sukitha > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- ?talo Rossi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141217/b5ddd020/attachment.html From aqsyounas at gmail.com Wed Dec 17 16:34:24 2014 From: aqsyounas at gmail.com (Aqs Younas) Date: Wed, 17 Dec 2014 18:34:24 +0500 Subject: [Freeswitch-users] How to play a stream other than mp3 with mod_shout. In-Reply-To: References: Message-ID: Hi, Moishe Grunstein I am able to successfully configure mod_vlc, but when i play i stream with vlc i hear nothing. Like dead call. Here is my default.xml When call is answered and i hangup the call from my softphone, it does not actually ends on freeswitch side. Also when i type freeswitch at internal> hupall +OK hangup all channels with cause MANAGER_REQUEST But freeswitch at internal> show calls count 2 total. Call doesn't terminate. When i play the stream with mod_shout everything works fine. My OS is Debian 7.6. Can you please provide my with example how to play stream with vlc. Or anything that i am doing wrong. Best Regards. On 15 December 2014 at 17:03, Aqs Younas wrote: > > Thanks for your reply. I will try that too. > Many thanks for your valuable replies. > > On 15 December 2014 at 00:37, Danny Gershman > wrote: > >> Also mod_rtmp lets you play from an FMS server. >> >> >> On Friday, December 12, 2014, Aqs Younas wrote: >> >>> Hi, All >>> >>> How can i play a live stream other than mp3 with mod_shout or any >>> module.? Is there any way to buffer the stream before playing it with >>> mod_shout. >>> >>> >>> Currently i have a list of streams and when i play them with mod_shout >>> some work fine but others give (time out) error. >>> >>> How can i play mostly stream in freeswitch? >>> >>> Thanks >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141217/80018c41/attachment.html From frederick at targointernet.com Wed Dec 17 16:44:10 2014 From: frederick at targointernet.com (Frederick Pruneau) Date: Wed, 17 Dec 2014 08:44:10 -0500 Subject: [Freeswitch-users] Issue with Freeswitch Behind nat In-Reply-To: References: Message-ID: Sorry for this noob question but how can I see sip traffic? Is there a specific command to show this? Is it what we find in freeswitch.log? If so, I attached my log file in my first post. Thanks for you help 2014-12-16 15:44 GMT-05:00 Brian West : > > have you looked at the signalling? What does the sip traffic show? > Please pastebin that. > > On Tue, Dec 16, 2014 at 2:37 PM, Frederick Pruneau < > frederick at targointernet.com> wrote: >> >> Same problem... >> >> 2014-12-16 13:55 GMT-05:00 Brian West : >> >>> Guessing you don't have UPNP or NAT-PMP on your network, there for that >>> won't work, >>> >>> ext-sip-ip=autonat:x.x.x.x >>> ext-rtp-ip=autonat:x.x.x.x >>> >>> Set local-network-ac to rfc1918.auto >>> >>> On Tue, Dec 16, 2014 at 12:15 PM, Support Technique < >>> support at targointernet.com> wrote: >>>> >>>> >>>> >>>> >>>> >>>> 2014-12-16 12:25 GMT-05:00 Brian West : >>>> >>>>> On the system behind nat what do you have ext-rtp-ip, ext-sip-ip and >>>>> local-network-acl set to? >>>>> >>>>> On Tue, Dec 16, 2014 at 10:41 AM, Frederick Pruneau < >>>>> frederick at targointernet.com> wrote: >>>>> >>>>>> Hi guys, >>>>>> >>>>>> We have an issue with one freeswitch server behind nat. We have a >>>>>> setup like this: >>>>>> >>>>>> -One master Freeswitch server >>>>>> >>>>>> -One freeswitch server connected to the master (Public IP) - Server A >>>>>> >>>>>> -One freeswitch server connected to the master (behind nat) - Server B >>>>>> >>>>>> If server A call server B, nothing happens. There is no sound. After >>>>>> 30 sec, it times out. We have done a tcpdump. From server A to master >>>>>> packets are ok. From Master to server B, we have seen that there is no >>>>>> source and no destination ports for sip invite. >>>>>> >>>>>> If we use our cellphone and we call server B, there is no problem. >>>>>> >>>>>> I have attached the failed call pcap file and freeswitch's log file >>>>>> so you can take a look at them. >>>>>> >>>>>> Master = Freeswitch v1.4.13 >>>>>> Server A = Freeswitch v.1.4.13 >>>>>> Server B = Freeswitch v.1.4.14 (Updated to latest release since we >>>>>> have issues with this server) >>>>>> >>>>>> Thanks in advance. >>>>>> >>>>>> PS: The failed call is from 514-448-0773. >>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://confluence.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>> >>>>> >>>>> -- >>>>> >>>>> *Brian West* >>>>> brian at freeswitch.org >>>>> >>>>> >>>>> *Twitter: @FreeSWITCH , @briankwest* >>>>> http://www.freeswitchbook.com >>>>> http://www.freeswitchcookbook.com >>>>> >>>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >>>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> -- >>> >>> *Brian West* >>> brian at freeswitch.org >>> >>> >>> *Twitter: @FreeSWITCH , @briankwest* >>> http://www.freeswitchbook.com >>> http://www.freeswitchcookbook.com >>> >>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > -- > > *Brian West* > brian at freeswitch.org > > > *Twitter: @FreeSWITCH , @briankwest* > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141217/af529881/attachment-0001.html From aqsyounas at gmail.com Wed Dec 17 16:50:00 2014 From: aqsyounas at gmail.com (Aqs Younas) Date: Wed, 17 Dec 2014 18:50:00 +0500 Subject: [Freeswitch-users] How to play a stream other than mp3 with mod_shout. In-Reply-To: References: Message-ID: Also these are logs when playing stream with mod_vlc [0x7f69b00025d8] main input debug: Creating an input for ' http://s9.voscast.com:7584' [0x7f69b00025d8] main input debug: using timeshift granularity of 50 MiB, in path '/tmp' [0x7f69b00025d8] main input debug: `http://s9.voscast.com:7584' gives access `http' demux `' path `s9.voscast.com:7584' [0x7f69b00025d8] main input debug: creating demux: access='http' demux='' location='s9.voscast.com:7584' file='(null)' [0x26cbd48] main demux debug: looking for access_demux module matching "http": 11 candidates [0x26cbd48] main demux debug: no access_demux modules matched [0x7f69b00025d8] main input debug: creating access 'http' location=' s9.voscast.com:7584', path='(null)' [0x26cbc68] main access debug: looking for access module matching "http": 19 candidates [0x26cbc68] access_http access debug: querying proxy for http://s9.voscast.com:7584 Session doesn't terminate even after i apply hupall on cli. Thanks. On 17 December 2014 at 18:34, Aqs Younas wrote: > > Hi, Moishe Grunstein > > I am able to successfully configure mod_vlc, but when i play i stream with > vlc i hear nothing. Like dead call. > > Here is my default.xml > > > > > > > > > > > > > > When call is answered and i hangup the call from my softphone, it does not > actually ends on freeswitch side. > Also when i type > > freeswitch at internal> hupall > +OK hangup all channels with cause MANAGER_REQUEST > > But > freeswitch at internal> show calls count > 2 total. > > Call doesn't terminate. When i play the stream with mod_shout everything > works fine. > > My OS is Debian 7.6. > > Can you please provide my with example how to play stream with vlc. Or > anything that i am doing wrong. > > Best Regards. > > > On 15 December 2014 at 17:03, Aqs Younas wrote: >> >> Thanks for your reply. I will try that too. >> Many thanks for your valuable replies. >> >> On 15 December 2014 at 00:37, Danny Gershman >> wrote: >> >>> Also mod_rtmp lets you play from an FMS server. >>> >>> >>> On Friday, December 12, 2014, Aqs Younas wrote: >>> >>>> Hi, All >>>> >>>> How can i play a live stream other than mp3 with mod_shout or any >>>> module.? Is there any way to buffer the stream before playing it with >>>> mod_shout. >>>> >>>> >>>> Currently i have a list of streams and when i play them with mod_shout >>>> some work fine but others give (time out) error. >>>> >>>> How can i play mostly stream in freeswitch? >>>> >>>> Thanks >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141217/82c184bc/attachment.html From sukithaj at gmail.com Wed Dec 17 16:56:45 2014 From: sukithaj at gmail.com (sukitha jayasinghe) Date: Wed, 17 Dec 2014 19:26:45 +0530 Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 102, Issue 100 In-Reply-To: References: Message-ID: Other calls working fine, in conference also both sides can hear conference instruction messages. On 17 Dec 2014 19:15, 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: Conference no sound (?talo Rossi) > 2. Re: How to play a stream other than mp3 with mod_shout. > (Aqs Younas) > 3. Re: Issue with Freeswitch Behind nat (Frederick Pruneau) > > > ---------- Forwarded message ---------- > From: "?talo Rossi" > To: FreeSWITCH Users Help > Cc: > Date: Wed, 17 Dec 2014 09:41:08 -0300 > Subject: Re: [Freeswitch-users] Conference no sound > Is there audio when you call the other user directly, without the > conference? > > On Wed, Dec 17, 2014 at 8:30 AM, sukitha jayasinghe > wrote: >> >> Hi All, >> >> I have configured freeswitch server with conference feature enabled, When >> I call the conference number, i can hear hold music. when second call dial >> the conference number hold music stops, but no voice between legs. what >> would be the reason for this. >> >> Best Regards, >> Sukitha >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > -- > ?talo Rossi > > > ---------- Forwarded message ---------- > From: Aqs Younas > To: FreeSWITCH Users Help > Cc: > Date: Wed, 17 Dec 2014 18:34:24 +0500 > Subject: Re: [Freeswitch-users] How to play a stream other than mp3 with > mod_shout. > Hi, Moishe Grunstein > > I am able to successfully configure mod_vlc, but when i play i stream with > vlc i hear nothing. Like dead call. > > Here is my default.xml > > > > > > > > > > > > > > When call is answered and i hangup the call from my softphone, it does not > actually ends on freeswitch side. > Also when i type > > freeswitch at internal> hupall > +OK hangup all channels with cause MANAGER_REQUEST > > But > freeswitch at internal> show calls count > 2 total. > > Call doesn't terminate. When i play the stream with mod_shout everything > works fine. > > My OS is Debian 7.6. > > Can you please provide my with example how to play stream with vlc. Or > anything that i am doing wrong. > > Best Regards. > > > On 15 December 2014 at 17:03, Aqs Younas wrote: >> >> Thanks for your reply. I will try that too. >> Many thanks for your valuable replies. >> >> On 15 December 2014 at 00:37, Danny Gershman >> wrote: >> >>> Also mod_rtmp lets you play from an FMS server. >>> >>> >>> On Friday, December 12, 2014, Aqs Younas wrote: >>> >>>> Hi, All >>>> >>>> How can i play a live stream other than mp3 with mod_shout or any >>>> module.? Is there any way to buffer the stream before playing it with >>>> mod_shout. >>>> >>>> >>>> Currently i have a list of streams and when i play them with mod_shout >>>> some work fine but others give (time out) error. >>>> >>>> How can i play mostly stream in freeswitch? >>>> >>>> Thanks >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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: Frederick Pruneau > To: FreeSWITCH Users Help > Cc: > Date: Wed, 17 Dec 2014 08:44:10 -0500 > Subject: Re: [Freeswitch-users] Issue with Freeswitch Behind nat > Sorry for this noob question but how can I see sip traffic? Is there a > specific command to show this? Is it what we find in freeswitch.log? If so, > I attached my log file in my first post. > > Thanks for you help > > 2014-12-16 15:44 GMT-05:00 Brian West : >> >> have you looked at the signalling? What does the sip traffic show? >> Please pastebin that. >> >> On Tue, Dec 16, 2014 at 2:37 PM, Frederick Pruneau < >> frederick at targointernet.com> wrote: >>> >>> Same problem... >>> >>> 2014-12-16 13:55 GMT-05:00 Brian West : >>> >>>> Guessing you don't have UPNP or NAT-PMP on your network, there for that >>>> won't work, >>>> >>>> ext-sip-ip=autonat:x.x.x.x >>>> ext-rtp-ip=autonat:x.x.x.x >>>> >>>> Set local-network-ac to rfc1918.auto >>>> >>>> On Tue, Dec 16, 2014 at 12:15 PM, Support Technique < >>>> support at targointernet.com> wrote: >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> 2014-12-16 12:25 GMT-05:00 Brian West : >>>>> >>>>>> On the system behind nat what do you have ext-rtp-ip, ext-sip-ip and >>>>>> local-network-acl set to? >>>>>> >>>>>> On Tue, Dec 16, 2014 at 10:41 AM, Frederick Pruneau < >>>>>> frederick at targointernet.com> wrote: >>>>>> >>>>>>> Hi guys, >>>>>>> >>>>>>> We have an issue with one freeswitch server behind nat. We have a >>>>>>> setup like this: >>>>>>> >>>>>>> -One master Freeswitch server >>>>>>> >>>>>>> -One freeswitch server connected to the master (Public IP) - Server A >>>>>>> >>>>>>> -One freeswitch server connected to the master (behind nat) - Server >>>>>>> B >>>>>>> >>>>>>> If server A call server B, nothing happens. There is no sound. After >>>>>>> 30 sec, it times out. We have done a tcpdump. From server A to master >>>>>>> packets are ok. From Master to server B, we have seen that there is no >>>>>>> source and no destination ports for sip invite. >>>>>>> >>>>>>> If we use our cellphone and we call server B, there is no problem. >>>>>>> >>>>>>> I have attached the failed call pcap file and freeswitch's log file >>>>>>> so you can take a look at them. >>>>>>> >>>>>>> Master = Freeswitch v1.4.13 >>>>>>> Server A = Freeswitch v.1.4.13 >>>>>>> Server B = Freeswitch v.1.4.14 (Updated to latest release since we >>>>>>> have issues with this server) >>>>>>> >>>>>>> Thanks in advance. >>>>>>> >>>>>>> PS: The failed call is from 514-448-0773. >>>>>>> >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://confluence.freeswitch.org >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> >>>>>> *Brian West* >>>>>> brian at freeswitch.org >>>>>> >>>>>> >>>>>> *Twitter: @FreeSWITCH , @briankwest* >>>>>> http://www.freeswitchbook.com >>>>>> http://www.freeswitchcookbook.com >>>>>> >>>>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >>>>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://confluence.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> -- >>>> >>>> *Brian West* >>>> brian at freeswitch.org >>>> >>>> >>>> *Twitter: @FreeSWITCH , @briankwest* >>>> http://www.freeswitchbook.com >>>> http://www.freeswitchcookbook.com >>>> >>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> -- >> >> *Brian West* >> brian at freeswitch.org >> >> >> *Twitter: @FreeSWITCH , @briankwest* >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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/20141217/7c0fd558/attachment-0001.html From brian at freeswitch.org Wed Dec 17 17:24:04 2014 From: brian at freeswitch.org (Brian West) Date: Wed, 17 Dec 2014 08:24:04 -0600 Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 102, Issue 100 In-Reply-To: References: Message-ID: If you are going to respond and interact with the list we ask that you do not use digest. On Wed, Dec 17, 2014 at 7:56 AM, sukitha jayasinghe wrote: > > Other calls working fine, in conference also both sides can hear > conference instruction messages. > On 17 Dec 2014 19:15, > 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: Conference no sound (?talo Rossi) >> 2. Re: How to play a stream other than mp3 with mod_shout. >> (Aqs Younas) >> 3. Re: Issue with Freeswitch Behind nat (Frederick Pruneau) >> >> >> ---------- Forwarded message ---------- >> From: "?talo Rossi" >> To: FreeSWITCH Users Help >> Cc: >> Date: Wed, 17 Dec 2014 09:41:08 -0300 >> Subject: Re: [Freeswitch-users] Conference no sound >> Is there audio when you call the other user directly, without the >> conference? >> >> On Wed, Dec 17, 2014 at 8:30 AM, sukitha jayasinghe >> wrote: >>> >>> Hi All, >>> >>> I have configured freeswitch server with conference feature enabled, >>> When I call the conference number, i can hear hold music. when second call >>> dial the conference number hold music stops, but no voice between legs. >>> what would be the reason for this. >>> >>> Best Regards, >>> Sukitha >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> -- >> ?talo Rossi >> >> >> ---------- Forwarded message ---------- >> From: Aqs Younas >> To: FreeSWITCH Users Help >> Cc: >> Date: Wed, 17 Dec 2014 18:34:24 +0500 >> Subject: Re: [Freeswitch-users] How to play a stream other than mp3 with >> mod_shout. >> Hi, Moishe Grunstein >> >> I am able to successfully configure mod_vlc, but when i play i stream >> with vlc i hear nothing. Like dead call. >> >> Here is my default.xml >> >> >> >> >> >> >> >> >> >> >> >> >> >> When call is answered and i hangup the call from my softphone, it does >> not actually ends on freeswitch side. >> Also when i type >> >> freeswitch at internal> hupall >> +OK hangup all channels with cause MANAGER_REQUEST >> >> But >> freeswitch at internal> show calls count >> 2 total. >> >> Call doesn't terminate. When i play the stream with mod_shout everything >> works fine. >> >> My OS is Debian 7.6. >> >> Can you please provide my with example how to play stream with vlc. Or >> anything that i am doing wrong. >> >> Best Regards. >> >> >> On 15 December 2014 at 17:03, Aqs Younas wrote: >>> >>> Thanks for your reply. I will try that too. >>> Many thanks for your valuable replies. >>> >>> On 15 December 2014 at 00:37, Danny Gershman >>> wrote: >>> >>>> Also mod_rtmp lets you play from an FMS server. >>>> >>>> >>>> On Friday, December 12, 2014, Aqs Younas wrote: >>>> >>>>> Hi, All >>>>> >>>>> How can i play a live stream other than mp3 with mod_shout or any >>>>> module.? Is there any way to buffer the stream before playing it with >>>>> mod_shout. >>>>> >>>>> >>>>> Currently i have a list of streams and when i play them with mod_shout >>>>> some work fine but others give (time out) error. >>>>> >>>>> How can i play mostly stream in freeswitch? >>>>> >>>>> Thanks >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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: Frederick Pruneau >> To: FreeSWITCH Users Help >> Cc: >> Date: Wed, 17 Dec 2014 08:44:10 -0500 >> Subject: Re: [Freeswitch-users] Issue with Freeswitch Behind nat >> Sorry for this noob question but how can I see sip traffic? Is there a >> specific command to show this? Is it what we find in freeswitch.log? If so, >> I attached my log file in my first post. >> >> Thanks for you help >> >> 2014-12-16 15:44 GMT-05:00 Brian West : >>> >>> have you looked at the signalling? What does the sip traffic show? >>> Please pastebin that. >>> >>> On Tue, Dec 16, 2014 at 2:37 PM, Frederick Pruneau < >>> frederick at targointernet.com> wrote: >>>> >>>> Same problem... >>>> >>>> 2014-12-16 13:55 GMT-05:00 Brian West : >>>> >>>>> Guessing you don't have UPNP or NAT-PMP on your network, there for >>>>> that won't work, >>>>> >>>>> ext-sip-ip=autonat:x.x.x.x >>>>> ext-rtp-ip=autonat:x.x.x.x >>>>> >>>>> Set local-network-ac to rfc1918.auto >>>>> >>>>> On Tue, Dec 16, 2014 at 12:15 PM, Support Technique < >>>>> support at targointernet.com> wrote: >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> 2014-12-16 12:25 GMT-05:00 Brian West : >>>>>> >>>>>>> On the system behind nat what do you have ext-rtp-ip, ext-sip-ip and >>>>>>> local-network-acl set to? >>>>>>> >>>>>>> On Tue, Dec 16, 2014 at 10:41 AM, Frederick Pruneau < >>>>>>> frederick at targointernet.com> wrote: >>>>>>> >>>>>>>> Hi guys, >>>>>>>> >>>>>>>> We have an issue with one freeswitch server behind nat. We have a >>>>>>>> setup like this: >>>>>>>> >>>>>>>> -One master Freeswitch server >>>>>>>> >>>>>>>> -One freeswitch server connected to the master (Public IP) - Server >>>>>>>> A >>>>>>>> >>>>>>>> -One freeswitch server connected to the master (behind nat) - >>>>>>>> Server B >>>>>>>> >>>>>>>> If server A call server B, nothing happens. There is no sound. >>>>>>>> After 30 sec, it times out. We have done a tcpdump. From server A to master >>>>>>>> packets are ok. From Master to server B, we have seen that there is no >>>>>>>> source and no destination ports for sip invite. >>>>>>>> >>>>>>>> If we use our cellphone and we call server B, there is no problem. >>>>>>>> >>>>>>>> I have attached the failed call pcap file and freeswitch's log file >>>>>>>> so you can take a look at them. >>>>>>>> >>>>>>>> Master = Freeswitch v1.4.13 >>>>>>>> Server A = Freeswitch v.1.4.13 >>>>>>>> Server B = Freeswitch v.1.4.14 (Updated to latest release since we >>>>>>>> have issues with this server) >>>>>>>> >>>>>>>> Thanks in advance. >>>>>>>> >>>>>>>> PS: The failed call is from 514-448-0773. >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> _________________________________________________________________________ >>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>> consulting at freeswitch.org >>>>>>>> http://www.freeswitchsolutions.com >>>>>>>> >>>>>>>> Official FreeSWITCH Sites >>>>>>>> http://www.freeswitch.org >>>>>>>> http://confluence.freeswitch.org >>>>>>>> http://www.cluecon.com >>>>>>>> >>>>>>>> FreeSWITCH-users mailing list >>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>> UNSUBSCRIBE: >>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>> http://www.freeswitch.org >>>>>>>> >>>>>>> >>>>>>> >>>>>>> -- >>>>>>> >>>>>>> *Brian West* >>>>>>> brian at freeswitch.org >>>>>>> >>>>>>> >>>>>>> *Twitter: @FreeSWITCH , @briankwest* >>>>>>> http://www.freeswitchbook.com >>>>>>> http://www.freeswitchcookbook.com >>>>>>> >>>>>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >>>>>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://confluence.freeswitch.org >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://confluence.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>> >>>>> >>>>> -- >>>>> >>>>> *Brian West* >>>>> brian at freeswitch.org >>>>> >>>>> >>>>> *Twitter: @FreeSWITCH , @briankwest* >>>>> http://www.freeswitchbook.com >>>>> http://www.freeswitchcookbook.com >>>>> >>>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >>>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> -- >>> >>> *Brian West* >>> brian at freeswitch.org >>> >>> >>> *Twitter: @FreeSWITCH , @briankwest* >>> http://www.freeswitchbook.com >>> http://www.freeswitchcookbook.com >>> >>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141217/456333cc/attachment-0001.html From brian at freeswitch.org Wed Dec 17 17:26:14 2014 From: brian at freeswitch.org (Brian West) Date: Wed, 17 Dec 2014 08:26:14 -0600 Subject: [Freeswitch-users] How to play a stream other than mp3 with mod_shout. In-Reply-To: References: Message-ID: Don't think mod_rtmp can actually do this, from loading it: 2014-12-17 08:25:39.983446 [NOTICE] switch_loadable_module.c:149 Adding Endpoint 'rtmp' 2014-12-17 08:25:39.983446 [NOTICE] switch_loadable_module.c:315 Adding API Function 'rtmp' 2014-12-17 08:25:39.983446 [NOTICE] switch_loadable_module.c:315 Adding API Function 'rtmp_contact' This is all that gets registered. On Sun, Dec 14, 2014 at 1:37 PM, Danny Gershman wrote: > > Also mod_rtmp lets you play from an FMS server. > > > On Friday, December 12, 2014, Aqs Younas wrote: > >> Hi, All >> >> How can i play a live stream other than mp3 with mod_shout or any >> module.? Is there any way to buffer the stream before playing it with >> mod_shout. >> >> >> Currently i have a list of streams and when i play them with mod_shout >> some work fine but others give (time out) error. >> >> How can i play mostly stream in freeswitch? >> >> Thanks >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141217/f4a01788/attachment.html From brian at freeswitch.org Wed Dec 17 17:26:51 2014 From: brian at freeswitch.org (Brian West) Date: Wed, 17 Dec 2014 08:26:51 -0600 Subject: [Freeswitch-users] Conference no sound In-Reply-To: References: Message-ID: Please press 0 to verify that you're not hitting a bug that was recently fix, you also didn't outline the rev you're running. Have you tried master? On Wed, Dec 17, 2014 at 5:30 AM, sukitha jayasinghe wrote: > > Hi All, > > I have configured freeswitch server with conference feature enabled, When > I call the conference number, i can hear hold music. when second call dial > the conference number hold music stops, but no voice between legs. what > would be the reason for this. > > Best Regards, > Sukitha > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141217/3e4b645d/attachment.html From brian at freeswitch.org Wed Dec 17 17:27:06 2014 From: brian at freeswitch.org (Brian West) Date: Wed, 17 Dec 2014 08:27:06 -0600 Subject: [Freeswitch-users] Issue with Freeswitch Behind nat In-Reply-To: References: Message-ID: sofia global siptrace on from fs_cli On Wed, Dec 17, 2014 at 7:44 AM, Frederick Pruneau < frederick at targointernet.com> wrote: > > Sorry for this noob question but how can I see sip traffic? Is there a > specific command to show this? Is it what we find in freeswitch.log? If so, > I attached my log file in my first post. > > Thanks for you help > > 2014-12-16 15:44 GMT-05:00 Brian West : > >> have you looked at the signalling? What does the sip traffic show? >> Please pastebin that. >> >> On Tue, Dec 16, 2014 at 2:37 PM, Frederick Pruneau < >> frederick at targointernet.com> wrote: >>> >>> Same problem... >>> >>> 2014-12-16 13:55 GMT-05:00 Brian West : >>> >>>> Guessing you don't have UPNP or NAT-PMP on your network, there for that >>>> won't work, >>>> >>>> ext-sip-ip=autonat:x.x.x.x >>>> ext-rtp-ip=autonat:x.x.x.x >>>> >>>> Set local-network-ac to rfc1918.auto >>>> >>>> On Tue, Dec 16, 2014 at 12:15 PM, Support Technique < >>>> support at targointernet.com> wrote: >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> 2014-12-16 12:25 GMT-05:00 Brian West : >>>>> >>>>>> On the system behind nat what do you have ext-rtp-ip, ext-sip-ip and >>>>>> local-network-acl set to? >>>>>> >>>>>> On Tue, Dec 16, 2014 at 10:41 AM, Frederick Pruneau < >>>>>> frederick at targointernet.com> wrote: >>>>>> >>>>>>> Hi guys, >>>>>>> >>>>>>> We have an issue with one freeswitch server behind nat. We have a >>>>>>> setup like this: >>>>>>> >>>>>>> -One master Freeswitch server >>>>>>> >>>>>>> -One freeswitch server connected to the master (Public IP) - Server A >>>>>>> >>>>>>> -One freeswitch server connected to the master (behind nat) - Server >>>>>>> B >>>>>>> >>>>>>> If server A call server B, nothing happens. There is no sound. After >>>>>>> 30 sec, it times out. We have done a tcpdump. From server A to master >>>>>>> packets are ok. From Master to server B, we have seen that there is no >>>>>>> source and no destination ports for sip invite. >>>>>>> >>>>>>> If we use our cellphone and we call server B, there is no problem. >>>>>>> >>>>>>> I have attached the failed call pcap file and freeswitch's log file >>>>>>> so you can take a look at them. >>>>>>> >>>>>>> Master = Freeswitch v1.4.13 >>>>>>> Server A = Freeswitch v.1.4.13 >>>>>>> Server B = Freeswitch v.1.4.14 (Updated to latest release since we >>>>>>> have issues with this server) >>>>>>> >>>>>>> Thanks in advance. >>>>>>> >>>>>>> PS: The failed call is from 514-448-0773. >>>>>>> >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://confluence.freeswitch.org >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> >>>>>> *Brian West* >>>>>> brian at freeswitch.org >>>>>> >>>>>> >>>>>> *Twitter: @FreeSWITCH , @briankwest* >>>>>> http://www.freeswitchbook.com >>>>>> http://www.freeswitchcookbook.com >>>>>> >>>>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >>>>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://confluence.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> -- >>>> >>>> *Brian West* >>>> brian at freeswitch.org >>>> >>>> >>>> *Twitter: @FreeSWITCH , @briankwest* >>>> http://www.freeswitchbook.com >>>> http://www.freeswitchcookbook.com >>>> >>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> -- >> >> *Brian West* >> brian at freeswitch.org >> >> >> *Twitter: @FreeSWITCH , @briankwest* >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141217/a318bdbe/attachment-0001.html From brian at freeswitch.org Wed Dec 17 17:28:25 2014 From: brian at freeswitch.org (Brian West) Date: Wed, 17 Dec 2014 08:28:25 -0600 Subject: [Freeswitch-users] fs_cli hangs In-Reply-To: References: Message-ID: One of these things is not like the others... Debian is a linux distro and mips is a CPU, so what linux distro are you running on this mips platform? On Wed, Dec 17, 2014 at 2:56 AM, akhil garg wrote: > > Its not Debian but mips > > -- > regards, > akhil > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141217/a4f6a752/attachment.html From mthakershi at gmail.com Wed Dec 17 17:49:41 2014 From: mthakershi at gmail.com (Malay Thakershi) Date: Wed, 17 Dec 2014 20:19:41 +0530 Subject: [Freeswitch-users] Include intended modules in build In-Reply-To: References: Message-ID: Hello, I built with VS 2013. I only got errors in libv8 and mod_v8 modules. Apart from that all projects built fine. Are the V8 projects mandatory with setup? Have another doubt. Should I keep platform target as "Any CPU" or "x64"? I am planning to put FS on 64 bit windows server in order to improve performance. Thanks. On Tue, Dec 16, 2014 at 11:04 PM, Michael Jerris wrote: > > There are a few outstanding issues for vs 2013. We are working to correct > those still. Keep an eye out for fixes. > > > On Dec 16, 2014, at 6:35 AM, Malay Thakershi wrote: > > Hello, > > I am trying to upgrade FS to windows 2012 server. > > Looking at this URL: > https://wiki.freeswitch.org/wiki/Installation_for_Windows > > 1. Can I use VS 2013? > 2. How to include necessary modules in the build - e.g. mod_managed or > mod_avmd? > 3. Once build is done, do I just use C:\FreeSwitch as my deployment > directory? > > Thanks for help and happy holidays. > > Malay Thakershi > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141217/b0f3f88f/attachment.html From michel.brabants at gmail.com Wed Dec 17 17:55:23 2014 From: michel.brabants at gmail.com (Michel Brabants) Date: Wed, 17 Dec 2014 15:55:23 +0100 Subject: [Freeswitch-users] freeswitch 1.4 and encryption/rtcp-mux In-Reply-To: <8B568748-EEE1-4E26-A3BB-302345FB0A01@jerris.com> References: <8B568748-EEE1-4E26-A3BB-302345FB0A01@jerris.com> Message-ID: Hello, I found the issue: 2 lines of code. I'll see if I can submit a patch, but I'm looking into it why it was added. The problem exists primarily because everywhere where a local sdp is generated, the function set_ice is called, while I don't want it (because I don't need it - nonat - and it generates, ice-not-ready-errors causing rtp to be dropped). The set_ice-function also sets the webrtc-flag (not sure why), causing dtls to become a requirement, which is not true in the current context. Anyway, this is nothing for this list, but I just want to add for any user currently encountering this problem. I'll do my best to generate a usefull patch. Michel On Fri, Dec 12, 2014 at 7:06 PM, Michael Jerris wrote: > > > > On Dec 12, 2014, at 10:23 AM, Michel Brabants > wrote: > > > > Hello, > > > > I recently started upgrading to FS 1.4, but I encountered 2 difficulties > of which I'm still looking into one: > > > > 1) DTLS-configuration seems to be required, although we don't use it > currently. We use normal sip-profiles (no webrtc). The option to disable > it, is "webrtc_enable_dtls=false", which can b set in the dialplan. But why > is it trying to enable it by default? Can you disable it also in a profile? > > > > In what way do you think that some configuration is required? > > > 2) Also a change because of webrtc seemingly. When receiving an invite > (without sdp - 3pcc-request), freeswitch in the end response in its 200 OK > with a rtcp-mux-line in its sdp. We don't want rtcp-mux, just rtp-port+1 > for rtcp. When looking at the code, I don't currently know why FS sends > back the myx-parameter as it seems only enabled when the other ends > proposes it or am I missing something? > > > > Please report a bug on this. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141217/65e4a9d8/attachment.html From aademattia at comcast.net Wed Dec 17 17:55:53 2014 From: aademattia at comcast.net (Andrew) Date: Wed, 17 Dec 2014 09:55:53 -0500 Subject: [Freeswitch-users] Include intended modules in build In-Reply-To: References: Message-ID: <068501d01a09$91f49ee0$b5dddca0$@comcast.net> Hi, I have the same issues and I just remove those projects. I also keep on 64bit. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Malay Thakershi Sent: Wednesday, December 17, 2014 9:50 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Include intended modules in build Hello, I built with VS 2013. I only got errors in libv8 and mod_v8 modules. Apart from that all projects built fine. Are the V8 projects mandatory with setup? Have another doubt. Should I keep platform target as "Any CPU" or "x64"? I am planning to put FS on 64 bit windows server in order to improve performance. Thanks. On Tue, Dec 16, 2014 at 11:04 PM, Michael Jerris > wrote: There are a few outstanding issues for vs 2013. We are working to correct those still. Keep an eye out for fixes. On Dec 16, 2014, at 6:35 AM, Malay Thakershi > wrote: Hello, I am trying to upgrade FS to windows 2012 server. Looking at this URL: https://wiki.freeswitch.org/wiki/Installation_for_Windows 1. Can I use VS 2013? 2. How to include necessary modules in the build - e.g. mod_managed or mod_avmd? 3. Once build is done, do I just use C:\FreeSwitch as my deployment directory? Thanks for help and happy holidays. Malay Thakershi _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141217/21f77537/attachment-0001.html From frederick at targointernet.com Wed Dec 17 18:36:57 2014 From: frederick at targointernet.com (Frederick Pruneau) Date: Wed, 17 Dec 2014 10:36:57 -0500 Subject: [Freeswitch-users] Issue with Freeswitch Behind nat In-Reply-To: References: Message-ID: Here it is: https://pastebin.freeswitch.org/23746 It is my freeswitch log. I have followed this guide before: https://wiki.freeswitch.org/wiki/Sofia#Debugging_Sofia-SIP I have enabled this and make a call: sofia global siptrace on sofia loglevel all 9 sofia tracelevel alert console loglevel debug fsctl debug_level 10 This is what you will get in my pastebin 2014-12-17 9:27 GMT-05:00 Brian West : > > sofia global siptrace on > > from fs_cli > > On Wed, Dec 17, 2014 at 7:44 AM, Frederick Pruneau < > frederick at targointernet.com> wrote: >> >> Sorry for this noob question but how can I see sip traffic? Is there a >> specific command to show this? Is it what we find in freeswitch.log? If so, >> I attached my log file in my first post. >> >> Thanks for you help >> >> 2014-12-16 15:44 GMT-05:00 Brian West : >> >>> have you looked at the signalling? What does the sip traffic show? >>> Please pastebin that. >>> >>> On Tue, Dec 16, 2014 at 2:37 PM, Frederick Pruneau < >>> frederick at targointernet.com> wrote: >>>> >>>> Same problem... >>>> >>>> 2014-12-16 13:55 GMT-05:00 Brian West : >>>> >>>>> Guessing you don't have UPNP or NAT-PMP on your network, there for >>>>> that won't work, >>>>> >>>>> ext-sip-ip=autonat:x.x.x.x >>>>> ext-rtp-ip=autonat:x.x.x.x >>>>> >>>>> Set local-network-ac to rfc1918.auto >>>>> >>>>> On Tue, Dec 16, 2014 at 12:15 PM, Support Technique < >>>>> support at targointernet.com> wrote: >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> 2014-12-16 12:25 GMT-05:00 Brian West : >>>>>> >>>>>>> On the system behind nat what do you have ext-rtp-ip, ext-sip-ip and >>>>>>> local-network-acl set to? >>>>>>> >>>>>>> On Tue, Dec 16, 2014 at 10:41 AM, Frederick Pruneau < >>>>>>> frederick at targointernet.com> wrote: >>>>>>> >>>>>>>> Hi guys, >>>>>>>> >>>>>>>> We have an issue with one freeswitch server behind nat. We have a >>>>>>>> setup like this: >>>>>>>> >>>>>>>> -One master Freeswitch server >>>>>>>> >>>>>>>> -One freeswitch server connected to the master (Public IP) - Server >>>>>>>> A >>>>>>>> >>>>>>>> -One freeswitch server connected to the master (behind nat) - >>>>>>>> Server B >>>>>>>> >>>>>>>> If server A call server B, nothing happens. There is no sound. >>>>>>>> After 30 sec, it times out. We have done a tcpdump. From server A to master >>>>>>>> packets are ok. From Master to server B, we have seen that there is no >>>>>>>> source and no destination ports for sip invite. >>>>>>>> >>>>>>>> If we use our cellphone and we call server B, there is no problem. >>>>>>>> >>>>>>>> I have attached the failed call pcap file and freeswitch's log file >>>>>>>> so you can take a look at them. >>>>>>>> >>>>>>>> Master = Freeswitch v1.4.13 >>>>>>>> Server A = Freeswitch v.1.4.13 >>>>>>>> Server B = Freeswitch v.1.4.14 (Updated to latest release since we >>>>>>>> have issues with this server) >>>>>>>> >>>>>>>> Thanks in advance. >>>>>>>> >>>>>>>> PS: The failed call is from 514-448-0773. >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> _________________________________________________________________________ >>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>> consulting at freeswitch.org >>>>>>>> http://www.freeswitchsolutions.com >>>>>>>> >>>>>>>> Official FreeSWITCH Sites >>>>>>>> http://www.freeswitch.org >>>>>>>> http://confluence.freeswitch.org >>>>>>>> http://www.cluecon.com >>>>>>>> >>>>>>>> FreeSWITCH-users mailing list >>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>> UNSUBSCRIBE: >>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>> http://www.freeswitch.org >>>>>>>> >>>>>>> >>>>>>> >>>>>>> -- >>>>>>> >>>>>>> *Brian West* >>>>>>> brian at freeswitch.org >>>>>>> >>>>>>> >>>>>>> *Twitter: @FreeSWITCH , @briankwest* >>>>>>> http://www.freeswitchbook.com >>>>>>> http://www.freeswitchcookbook.com >>>>>>> >>>>>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >>>>>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://confluence.freeswitch.org >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://confluence.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>> >>>>> >>>>> -- >>>>> >>>>> *Brian West* >>>>> brian at freeswitch.org >>>>> >>>>> >>>>> *Twitter: @FreeSWITCH , @briankwest* >>>>> http://www.freeswitchbook.com >>>>> http://www.freeswitchcookbook.com >>>>> >>>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >>>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> -- >>> >>> *Brian West* >>> brian at freeswitch.org >>> >>> >>> *Twitter: @FreeSWITCH , @briankwest* >>> http://www.freeswitchbook.com >>> http://www.freeswitchcookbook.com >>> >>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > -- > > *Brian West* > brian at freeswitch.org > > > *Twitter: @FreeSWITCH , @briankwest* > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141217/9e4e0635/attachment-0001.html From sukithaj at gmail.com Wed Dec 17 18:38:29 2014 From: sukithaj at gmail.com (sukitha jayasinghe) Date: Wed, 17 Dec 2014 21:08:29 +0530 Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 102, Issue 103 In-Reply-To: References: Message-ID: I am using FreeSWITCH Version 1.5.15b+git~20141031T184939Z~7ca4ac566c~64bit (git 7ca4ac5 2014-10-31 18:49:39Z 64bit) . Do i need to update this in to newer version On 17 Dec 2014 19:57, 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: How to play a stream other than mp3 with mod_shout. > (Brian West) > 2. Re: Conference no sound (Brian West) > 3. Re: Issue with Freeswitch Behind nat (Brian West) > > > ---------- Forwarded message ---------- > From: Brian West > To: FreeSWITCH Users Help > Cc: > Date: Wed, 17 Dec 2014 08:26:14 -0600 > Subject: Re: [Freeswitch-users] How to play a stream other than mp3 with > mod_shout. > Don't think mod_rtmp can actually do this, from loading it: > > 2014-12-17 08:25:39.983446 [NOTICE] switch_loadable_module.c:149 Adding > Endpoint 'rtmp' > > 2014-12-17 08:25:39.983446 [NOTICE] switch_loadable_module.c:315 Adding > API Function 'rtmp' > > 2014-12-17 08:25:39.983446 [NOTICE] switch_loadable_module.c:315 Adding > API Function 'rtmp_contact' > > > This is all that gets registered. > > On Sun, Dec 14, 2014 at 1:37 PM, Danny Gershman > wrote: >> >> Also mod_rtmp lets you play from an FMS server. >> >> >> On Friday, December 12, 2014, Aqs Younas wrote: >> >>> Hi, All >>> >>> How can i play a live stream other than mp3 with mod_shout or any >>> module.? Is there any way to buffer the stream before playing it with >>> mod_shout. >>> >>> >>> Currently i have a list of streams and when i play them with mod_shout >>> some work fine but others give (time out) error. >>> >>> How can i play mostly stream in freeswitch? >>> >>> Thanks >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > -- > > *Brian West* > brian at freeswitch.org > > > *Twitter: @FreeSWITCH , @briankwest* > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest > > > ---------- Forwarded message ---------- > From: Brian West > To: FreeSWITCH Users Help > Cc: > Date: Wed, 17 Dec 2014 08:26:51 -0600 > Subject: Re: [Freeswitch-users] Conference no sound > Please press 0 to verify that you're not hitting a bug that was recently > fix, you also didn't outline the rev you're running. Have you tried master? > > On Wed, Dec 17, 2014 at 5:30 AM, sukitha jayasinghe > wrote: >> >> Hi All, >> >> I have configured freeswitch server with conference feature enabled, When >> I call the conference number, i can hear hold music. when second call dial >> the conference number hold music stops, but no voice between legs. what >> would be the reason for this. >> >> Best Regards, >> Sukitha >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > -- > > *Brian West* > brian at freeswitch.org > > > *Twitter: @FreeSWITCH , @briankwest* > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest > > > ---------- Forwarded message ---------- > From: Brian West > To: FreeSWITCH Users Help > Cc: > Date: Wed, 17 Dec 2014 08:27:06 -0600 > Subject: Re: [Freeswitch-users] Issue with Freeswitch Behind nat > sofia global siptrace on > > from fs_cli > > On Wed, Dec 17, 2014 at 7:44 AM, Frederick Pruneau < > frederick at targointernet.com> wrote: >> >> Sorry for this noob question but how can I see sip traffic? Is there a >> specific command to show this? Is it what we find in freeswitch.log? If so, >> I attached my log file in my first post. >> >> Thanks for you help >> >> 2014-12-16 15:44 GMT-05:00 Brian West : >> >>> have you looked at the signalling? What does the sip traffic show? >>> Please pastebin that. >>> >>> On Tue, Dec 16, 2014 at 2:37 PM, Frederick Pruneau < >>> frederick at targointernet.com> wrote: >>>> >>>> Same problem... >>>> >>>> 2014-12-16 13:55 GMT-05:00 Brian West : >>>> >>>>> Guessing you don't have UPNP or NAT-PMP on your network, there for >>>>> that won't work, >>>>> >>>>> ext-sip-ip=autonat:x.x.x.x >>>>> ext-rtp-ip=autonat:x.x.x.x >>>>> >>>>> Set local-network-ac to rfc1918.auto >>>>> >>>>> On Tue, Dec 16, 2014 at 12:15 PM, Support Technique < >>>>> support at targointernet.com> wrote: >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> 2014-12-16 12:25 GMT-05:00 Brian West : >>>>>> >>>>>>> On the system behind nat what do you have ext-rtp-ip, ext-sip-ip and >>>>>>> local-network-acl set to? >>>>>>> >>>>>>> On Tue, Dec 16, 2014 at 10:41 AM, Frederick Pruneau < >>>>>>> frederick at targointernet.com> wrote: >>>>>>> >>>>>>>> Hi guys, >>>>>>>> >>>>>>>> We have an issue with one freeswitch server behind nat. We have a >>>>>>>> setup like this: >>>>>>>> >>>>>>>> -One master Freeswitch server >>>>>>>> >>>>>>>> -One freeswitch server connected to the master (Public IP) - Server >>>>>>>> A >>>>>>>> >>>>>>>> -One freeswitch server connected to the master (behind nat) - >>>>>>>> Server B >>>>>>>> >>>>>>>> If server A call server B, nothing happens. There is no sound. >>>>>>>> After 30 sec, it times out. We have done a tcpdump. From server A to master >>>>>>>> packets are ok. From Master to server B, we have seen that there is no >>>>>>>> source and no destination ports for sip invite. >>>>>>>> >>>>>>>> If we use our cellphone and we call server B, there is no problem. >>>>>>>> >>>>>>>> I have attached the failed call pcap file and freeswitch's log file >>>>>>>> so you can take a look at them. >>>>>>>> >>>>>>>> Master = Freeswitch v1.4.13 >>>>>>>> Server A = Freeswitch v.1.4.13 >>>>>>>> Server B = Freeswitch v.1.4.14 (Updated to latest release since we >>>>>>>> have issues with this server) >>>>>>>> >>>>>>>> Thanks in advance. >>>>>>>> >>>>>>>> PS: The failed call is from 514-448-0773. >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> _________________________________________________________________________ >>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>> consulting at freeswitch.org >>>>>>>> http://www.freeswitchsolutions.com >>>>>>>> >>>>>>>> Official FreeSWITCH Sites >>>>>>>> http://www.freeswitch.org >>>>>>>> http://confluence.freeswitch.org >>>>>>>> http://www.cluecon.com >>>>>>>> >>>>>>>> FreeSWITCH-users mailing list >>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>> UNSUBSCRIBE: >>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>> http://www.freeswitch.org >>>>>>>> >>>>>>> >>>>>>> >>>>>>> -- >>>>>>> >>>>>>> *Brian West* >>>>>>> brian at freeswitch.org >>>>>>> >>>>>>> >>>>>>> *Twitter: @FreeSWITCH , @briankwest* >>>>>>> http://www.freeswitchbook.com >>>>>>> http://www.freeswitchcookbook.com >>>>>>> >>>>>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >>>>>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://confluence.freeswitch.org >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://confluence.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>> >>>>> >>>>> -- >>>>> >>>>> *Brian West* >>>>> brian at freeswitch.org >>>>> >>>>> >>>>> *Twitter: @FreeSWITCH , @briankwest* >>>>> http://www.freeswitchbook.com >>>>> http://www.freeswitchcookbook.com >>>>> >>>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >>>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> -- >>> >>> *Brian West* >>> brian at freeswitch.org >>> >>> >>> *Twitter: @FreeSWITCH , @briankwest* >>> http://www.freeswitchbook.com >>> http://www.freeswitchcookbook.com >>> >>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > -- > > *Brian West* > brian at freeswitch.org > > > *Twitter: @FreeSWITCH , @briankwest* > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141217/ac9b583e/attachment-0001.html From alhakeem at gmail.com Wed Dec 17 17:47:32 2014 From: alhakeem at gmail.com (Abdul Hakeem) Date: Wed, 17 Dec 2014 14:47:32 -0000 Subject: [Freeswitch-users] Sip_Authentiation_Table Message-ID: Hello, Is there any way to map the sip_authentication_table to an external database or to a memcache table ? Thanks in advance. Regards, Abdul Hakeem From jmesquita at freeswitch.org Wed Dec 17 21:13:12 2014 From: jmesquita at freeswitch.org (=?UTF-8?Q?Jo=C3=A3o_Mesquita?=) Date: Wed, 17 Dec 2014 15:13:12 -0300 Subject: [Freeswitch-users] fs_cli hangs In-Reply-To: References: Message-ID: bkw, why ask if it's a Debian? I'm just curious... JM Jo?o Mesquita FreeSWITCH? Solutions On Wed, Dec 17, 2014 at 11:28 AM, Brian West wrote: > > One of these things is not like the others... Debian is a linux distro and > mips is a CPU, so what linux distro are you running on this mips platform? > > On Wed, Dec 17, 2014 at 2:56 AM, akhil garg wrote: > >> Its not Debian but mips >> >> -- >> regards, >> akhil >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > -- > > *Brian West* > brian at freeswitch.org > > > *Twitter: @FreeSWITCH , @briankwest* > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141217/736124ad/attachment.html From mike at jerris.com Thu Dec 18 01:08:01 2014 From: mike at jerris.com (Michael Jerris) Date: Wed, 17 Dec 2014 17:08:01 -0500 Subject: [Freeswitch-users] Freeswitch HA advices In-Reply-To: References: Message-ID: <37F06F1B-28C5-459C-950C-6BA229400A8C@jerris.com> from switch.conf.xml > On Dec 16, 2014, at 3:57 PM, Federico Castro wrote: > > Hi Eliot and all, I keep on working in my HA solution. I?m almost done but there are some things I would like to improve. When I finish it I will share my solution with the community. > > I have two FS Boxes with Postgres running locally. Psql is running in read-write mode in Master node and in read-only mode in Slave. Psql is streaming data asynchronously through nodes. > > First thing I would like to improve: > > FS try to DROP some tables and delete records from sip_registrations, channels, etc. when connecting to psql and if it fails mod_sofia stops loading. This behaviour produces (in my scenary) that FS in Slave server does not start properly If it connects to local database because it is in read-only mode. > To solve this I configured FS to connect to psql through virtual IP (it is always in Master node) then mod_sofia starts properly in Slave node. The problem here is that FS in slave node delete records from table in Master node when get connected. > > Is there any way to start FS in a ?standby mode? in slave node to avoid it trying to write database? > > Thanks! > > > > 2014-06-16 12:05 GMT-03:00 Federico Castro >: > Hi Eliot, thanks for your verbose response, it is really useful for me. > > I'm working on a duplicated FS + postgreSQL schema. The two boxes will have same HW. Both of them will run FS and postgreSQL. One will act as master and the other one as stand-by waiting for the first to fail. > > FS will not have more than a hundred simultaneous calls. > > I will read about Pacemaker and Corosync and I will update to the list about the implementation. > > Thanks again. > > > > 2014-06-13 10:45 GMT-03:00 Eliot Gable >: > On Tue, Jun 10, 2014 at 10:56 AM, Federico Castro > wrote: > Hi all, I'm working on a Freeswitch HA solution. Now I'm deciding what method and DB I'll use to track calls. > > I have installed PostgreSQL on both servers and I configured them to replicate DB asynchronously. > > I would like to know if someone has experience with this kind of solution and what things do I have to contemplate to deploy a solid solution. > > > Lots of people have experience with such a solution; it all depends on what you are trying to achieve. > > Personally, I recommend you setup Corosync and Pacemaker both on your PostgreSQL boxes and on your FreeSWITCH systems. I also recommend you run PostgreSQL on a separate set of boxes from FS. Both can use a lot of memory if you are running a lot of calls and/or have a lot of clients. If you need performance, I recommend using the fastest disks you can get in the PostgreSQL systems. Also install as much RAM as you can afford for the project in the PGSQL boxes. You will want redundant power supplies in each system with each supply plugged into a different circuit. You will also want redundant Ethernet connectivity to redundant switches which also have redundant power supplies. You will also want redundant cross-over connections between the pairs of boxes. > > Once you have Corosync and Pacemaker configured to start PGSQL and FS on their own boxes and you have tested manual fail-over, then you need to start thinking about every possible way you can make either of those two systems stop working. Think about hard drives failing, power loss, kernel panics, firewall rules blocking communication, someone accidentally removing the IP address from one of the systems (it happens), killing processes, Sofia profiles failing to load because something else is using the port, etc. Make sure you have things set up to detect and recover from any such failure. One of the best ways to do this is to actually build an external testing system which places real calls through the system and has them route back to itself to verify they made it. If it places a call and the call does not make it back to itself, then you know something failed and you can run more tests to determine what failed and reset it. > > Like I said, it all depends on what you are trying to accomplish. If you want really good automatic HA, you have to go to some pretty great lengths to get it. If you are OK with occasional manual intervention, then you can make some assumptions (like nobody accidentally removing your IP from the interface or telling it to stop responding to ARP or throwing up a firewall rule which blocks something). That makes the setup considerably easier, but it also means manual intervention when something like that happens. In other words, if something like that happens, you experience an outage which the HA system doesn't detect and recover from. When you get calls that service stopped working, you then have someone log in and take a look and manually fix the issue. This could take anywhere from 5 minutes to an hour or more to do, depending on how good your support is and how good your team is. > > So, probably the first task you should do is list all the things you want it to automatically recover from and all the things you are willing to accept causing an outage and then work on your implementation based on that plan. > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141217/5baf5f40/attachment-0001.html From brian at freeswitch.org Thu Dec 18 01:55:03 2014 From: brian at freeswitch.org (Brian West) Date: Wed, 17 Dec 2014 16:55:03 -0600 Subject: [Freeswitch-users] fs_cli hangs In-Reply-To: References: Message-ID: Because I have the same thing happen on debian at random that requires fs_cli to be killed, I suspect a bug but can't replicate it enough to really nail it down. On Wed, Dec 17, 2014 at 12:13 PM, Jo?o Mesquita wrote: > > bkw, why ask if it's a Debian? I'm just curious... > > JM > > Jo?o Mesquita > FreeSWITCH? Solutions > > On Wed, Dec 17, 2014 at 11:28 AM, Brian West wrote: >> >> One of these things is not like the others... Debian is a linux distro >> and mips is a CPU, so what linux distro are you running on this mips >> platform? >> >> On Wed, Dec 17, 2014 at 2:56 AM, akhil garg wrote: >> >>> Its not Debian but mips >>> >>> -- >>> regards, >>> akhil >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> -- >> >> *Brian West* >> brian at freeswitch.org >> >> >> *Twitter: @FreeSWITCH , @briankwest* >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141217/f8107dde/attachment.html From jmesquita at freeswitch.org Thu Dec 18 01:59:56 2014 From: jmesquita at freeswitch.org (=?UTF-8?Q?Jo=C3=A3o_Mesquita?=) Date: Wed, 17 Dec 2014 19:59:56 -0300 Subject: [Freeswitch-users] fs_cli hangs In-Reply-To: References: Message-ID: That's funny... I am getting the same thing but I always thought it was network congestion on SSH that created the bottleneck. Next time I get it, I will gcore it.... Jo?o Mesquita FreeSWITCH? Solutions On Wed, Dec 17, 2014 at 7:55 PM, Brian West wrote: > > Because I have the same thing happen on debian at random that requires > fs_cli to be killed, I suspect a bug but can't replicate it enough to > really nail it down. > > On Wed, Dec 17, 2014 at 12:13 PM, Jo?o Mesquita > wrote: >> >> bkw, why ask if it's a Debian? I'm just curious... >> >> JM >> >> Jo?o Mesquita >> FreeSWITCH? Solutions >> >> On Wed, Dec 17, 2014 at 11:28 AM, Brian West >> wrote: >>> >>> One of these things is not like the others... Debian is a linux distro >>> and mips is a CPU, so what linux distro are you running on this mips >>> platform? >>> >>> On Wed, Dec 17, 2014 at 2:56 AM, akhil garg >>> wrote: >>> >>>> Its not Debian but mips >>>> >>>> -- >>>> regards, >>>> akhil >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> -- >>> >>> *Brian West* >>> brian at freeswitch.org >>> >>> >>> *Twitter: @FreeSWITCH , @briankwest* >>> http://www.freeswitchbook.com >>> http://www.freeswitchcookbook.com >>> >>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > -- > > *Brian West* > brian at freeswitch.org > > > *Twitter: @FreeSWITCH , @briankwest* > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141217/a07c0b95/attachment-0001.html From t.mahe at b-and-c.net Thu Dec 18 02:15:33 2014 From: t.mahe at b-and-c.net (=?windows-1252?Q?Tristan_Mah=E9?=) Date: Wed, 17 Dec 2014 15:15:33 -0800 Subject: [Freeswitch-users] fs_cli hangs In-Reply-To: References: Message-ID: <54920E95.3070708@b-and-c.net> Also seeing it sometimes on some servers ( debian wheezy ), fs_cli commands start correctly, but every command sent gets no answer and cli is stuck, I have to kill ssh connection. Doing the same command with fs_cli -x 'cmd' does not show this behaviour. Usually launching fs_cli with -t and -T parameters fix the issue. Le 17/12/2014 14:55, Brian West a ?crit : > Because I have the same thing happen on debian at random that requires > fs_cli to be killed, I suspect a bug but can't replicate it enough to > really nail it down. > > On Wed, Dec 17, 2014 at 12:13 PM, Jo?o Mesquita > > wrote: > > bkw, why ask if it's a Debian? I'm just curious... > > JM > > Jo?o Mesquita > FreeSWITCH? Solutions > > On Wed, Dec 17, 2014 at 11:28 AM, Brian West > wrote: > > One of these things is not like the others... Debian is a linux > distro and mips is a CPU, so what linux distro are you running > on this mips platform? > > On Wed, Dec 17, 2014 at 2:56 AM, akhil garg > > wrote: > > Its not Debian but mips > > -- > regards, > akhil > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -- > > */Brian West/* > brian at freeswitch.org > > > */Twitter: @FreeSWITCH , @briankwest/* > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > *T:*+19184209001 | *F:*+19184209002 > | *M:*+1918424WEST (9378) > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -- > > */Brian West/* > brian at freeswitch.org > > > */Twitter: @FreeSWITCH , @briankwest/* > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 473 bytes Desc: OpenPGP digital signature Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141217/ecbd1405/attachment.bin From gabe at gundy.org Thu Dec 18 02:30:28 2014 From: gabe at gundy.org (Gabriel Gunderson) Date: Wed, 17 Dec 2014 16:30:28 -0700 Subject: [Freeswitch-users] fs_cli hangs In-Reply-To: References: Message-ID: On Wed, Dec 17, 2014 at 3:55 PM, Brian West wrote: > > Because I have the same thing happen on debian at random that requires > fs_cli to be killed, I suspect a bug but can't replicate it enough to > really nail it down. > Brian, Is it fs_cli that hangs, or is it event socket that hangs (causing fs_cli to misbehave)? I only ask as I've been helping someone that seems to have that issue. It's been a bit tricky to pin down. Best, Gabe -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141217/e0392ba6/attachment.html From aademattia at comcast.net Thu Dec 18 02:38:27 2014 From: aademattia at comcast.net (Andrew) Date: Wed, 17 Dec 2014 18:38:27 -0500 Subject: [Freeswitch-users] mod_sndfile.c:207 Error Opening File Message-ID: <07d401d01a52$93151fe0$b93f5fa0$@comcast.net> Hi, I have an odd problem and I have no clue how to fix this. I call about 100+ calls and then files that were working no longer work. I have been working on this for days. I know the files are real and work but for some odd reason it will not open or play any audio. If I restart everything works for the next few hundreds of calls. Working with FreeSWITCH master embedded on windows server x64. Does anyone have any idea? 2014-12-17 18:21:42.154751 [ERR] mod_sndfile.c:207 Error Opening File [c:\dialer\projects\x\audio\introvz.wav] [No Error.] 2014-12-17 18:21:59.434751 [ERR] mod_sndfile.c:207 Error Opening File [c:\dialer\projects\x\audio\introvz2.wav] [No Error.] 2014-12-17 18:23:09.574751 [ERR] mod_sndfile.c:207 Error Opening File [c:\dialer\projects\x\audio\press8.wav] [No Error.] 2014-12-17 18:23:18.634751 [ERR] mod_sndfile.c:207 Error Opening File [c:\dialer\projects\x\audio\closing.wav] [No Error.] -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141217/1ef90221/attachment.html From luis.daniel.lucio at gmail.com Thu Dec 18 03:19:37 2014 From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz) Date: Wed, 17 Dec 2014 19:19:37 -0500 Subject: [Freeswitch-users] Freeswitch HA advices In-Reply-To: <37F06F1B-28C5-459C-950C-6BA229400A8C@jerris.com> References: <37F06F1B-28C5-459C-950C-6BA229400A8C@jerris.com> Message-ID: are your servers in same LAN or they are in the " cloud " ? 2014-12-17 17:08 GMT-05:00 Michael Jerris : > from switch.conf.xml > > > > > > > > > On Dec 16, 2014, at 3:57 PM, Federico Castro wrote: > > Hi Eliot and all, I keep on working in my HA solution. I'm almost done but > there are some things I would like to improve. When I finish it I will share > my solution with the community. > > I have two FS Boxes with Postgres running locally. Psql is running in > read-write mode in Master node and in read-only mode in Slave. Psql is > streaming data asynchronously through nodes. > > First thing I would like to improve: > > FS try to DROP some tables and delete records from sip_registrations, > channels, etc. when connecting to psql and if it fails mod_sofia stops > loading. This behaviour produces (in my scenary) that FS in Slave server > does not start properly If it connects to local database because it is in > read-only mode. > To solve this I configured FS to connect to psql through virtual IP (it is > always in Master node) then mod_sofia starts properly in Slave node. The > problem here is that FS in slave node delete records from table in Master > node when get connected. > > Is there any way to start FS in a "standby mode" in slave node to avoid it > trying to write database? > > Thanks! > > > > 2014-06-16 12:05 GMT-03:00 Federico Castro : >> >> Hi Eliot, thanks for your verbose response, it is really useful for me. >> >> I'm working on a duplicated FS + postgreSQL schema. The two boxes will >> have same HW. Both of them will run FS and postgreSQL. One will act as >> master and the other one as stand-by waiting for the first to fail. >> >> FS will not have more than a hundred simultaneous calls. >> >> I will read about Pacemaker and Corosync and I will update to the list >> about the implementation. >> >> Thanks again. >> >> >> >> 2014-06-13 10:45 GMT-03:00 Eliot Gable : >>> >>> On Tue, Jun 10, 2014 at 10:56 AM, Federico Castro >>> wrote: >>>> >>>> Hi all, I'm working on a Freeswitch HA solution. Now I'm deciding what >>>> method and DB I'll use to track calls. >>>> >>>> I have installed PostgreSQL on both servers and I configured them to >>>> replicate DB asynchronously. >>>> >>>> I would like to know if someone has experience with this kind of >>>> solution and what things do I have to contemplate to deploy a solid >>>> solution. >>>> >>> >>> Lots of people have experience with such a solution; it all depends on >>> what you are trying to achieve. >>> >>> Personally, I recommend you setup Corosync and Pacemaker both on your >>> PostgreSQL boxes and on your FreeSWITCH systems. I also recommend you run >>> PostgreSQL on a separate set of boxes from FS. Both can use a lot of memory >>> if you are running a lot of calls and/or have a lot of clients. If you need >>> performance, I recommend using the fastest disks you can get in the >>> PostgreSQL systems. Also install as much RAM as you can afford for the >>> project in the PGSQL boxes. You will want redundant power supplies in each >>> system with each supply plugged into a different circuit. You will also want >>> redundant Ethernet connectivity to redundant switches which also have >>> redundant power supplies. You will also want redundant cross-over >>> connections between the pairs of boxes. >>> >>> Once you have Corosync and Pacemaker configured to start PGSQL and FS on >>> their own boxes and you have tested manual fail-over, then you need to start >>> thinking about every possible way you can make either of those two systems >>> stop working. Think about hard drives failing, power loss, kernel panics, >>> firewall rules blocking communication, someone accidentally removing the IP >>> address from one of the systems (it happens), killing processes, Sofia >>> profiles failing to load because something else is using the port, etc. Make >>> sure you have things set up to detect and recover from any such failure. One >>> of the best ways to do this is to actually build an external testing system >>> which places real calls through the system and has them route back to itself >>> to verify they made it. If it places a call and the call does not make it >>> back to itself, then you know something failed and you can run more tests to >>> determine what failed and reset it. >>> >>> Like I said, it all depends on what you are trying to accomplish. If you >>> want really good automatic HA, you have to go to some pretty great lengths >>> to get it. If you are OK with occasional manual intervention, then you can >>> make some assumptions (like nobody accidentally removing your IP from the >>> interface or telling it to stop responding to ARP or throwing up a firewall >>> rule which blocks something). That makes the setup considerably easier, but >>> it also means manual intervention when something like that happens. In other >>> words, if something like that happens, you experience an outage which the HA >>> system doesn't detect and recover from. When you get calls that service >>> stopped working, you then have someone log in and take a look and manually >>> fix the issue. This could take anywhere from 5 minutes to an hour or more to >>> do, depending on how good your support is and how good your team is. >>> >>> So, probably the first task you should do is list all the things you want >>> it to automatically recover from and all the things you are willing to >>> accept causing an outage and then work on your implementation based on that >>> plan. >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at 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 at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 grcamauer at gmail.com Thu Dec 18 04:46:30 2014 From: grcamauer at gmail.com (Guillermo Ruiz Camauer) Date: Wed, 17 Dec 2014 22:46:30 -0300 Subject: [Freeswitch-users] mod_sndfile.c:207 Error Opening File In-Reply-To: <07d401d01a52$93151fe0$b93f5fa0$@comcast.net> References: <07d401d01a52$93151fe0$b93f5fa0$@comcast.net> Message-ID: You are probably opening the files, not closing them and therefore running out of file handles. Try increasing file handles. If the problem appears again but after a larger number of calls, then that's your answer. Guillermo Sent from my iPhone > On 17/12/2014, at 20:38, Andrew wrote: > > Hi, > > I have an odd problem and I have no clue how to fix this. > > I call about 100+ calls and then files that were working no longer work. I have been working on this for days. > > I know the files are real and work but for some odd reason it will not open or play any audio. If I restart everything works for the next few hundreds of calls. > > Working with FreeSWITCH master embedded on windows server x64. > > Does anyone have any idea? > > > > 2014-12-17 18:21:42.154751 [ERR] mod_sndfile.c:207 Error Opening File [c:\dialer\projects\x\audio\introvz.wav] [No Error.] > 2014-12-17 18:21:59.434751 [ERR] mod_sndfile.c:207 Error Opening File [c:\dialer\projects\x\audio\introvz2.wav] [No Error.] > 2014-12-17 18:23:09.574751 [ERR] mod_sndfile.c:207 Error Opening File [c:\dialer\projects\x\audio\press8.wav] [No Error.] > 2014-12-17 18:23:18.634751 [ERR] mod_sndfile.c:207 Error Opening File [c:\dialer\projects\x\audio\closing.wav] [No Error.] > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141217/48b60dfb/attachment-0001.html From aademattia at comcast.net Thu Dec 18 05:16:38 2014 From: aademattia at comcast.net (=?utf-8?B?YWFkZW1hdHRpYUBjb21jYXN0Lm5ldA==?=) Date: Wed, 17 Dec 2014 21:16:38 -0500 Subject: [Freeswitch-users] =?utf-8?q?mod=5Fsndfile=2Ec=3A207_Error_Openin?= =?utf-8?q?g_File?= Message-ID: I am doing a session .read() How do I close the audio files? Sent from my HTC ----- Reply message ----- From: "Guillermo Ruiz Camauer" To: "FreeSWITCH Users Help" Subject: [Freeswitch-users] mod_sndfile.c:207 Error Opening File Date: Wed, Dec 17, 2014 8:46 PM You are probably opening the files, not closing them and therefore running out of file handles. Try increasing file handles. If the problem appears again but after a larger number of calls, then that's your answer. Guillermo Sent from my iPhone On 17/12/2014, at 20:38, Andrew wrote: Hi,I have an odd problem and I have no clue how to fix this.I call about 100+ calls and then files that were working no longer work. I have been working on this for days.I know the files are real and work but for some odd reason it will not open or play any audio. If I restart everything works for the next few hundreds of calls.Working with FreeSWITCH master embedded on windows server x64.Does anyone have any idea? 2014-12-17 18:21:42.154751 [ERR] mod_sndfile.c:207 Error Opening File [c:\dialer\projects\x\audio\introvz.wav] [No Error.]2014-12-17 18:21:59.434751 [ERR] mod_sndfile.c:207 Error Opening File [c:\dialer\projects\x\audio\introvz2.wav] [No Error.]2014-12-17 18:23:09.574751 [ERR] mod_sndfile.c:207 Error Opening File [c:\dialer\projects\x\audio\press8.wav] [No Error.]2014-12-17 18:23:18.634751 [ERR] mod_sndfile.c:207 Error Opening File [c:\dialer\projects\x\audio\closing.wav] [No Error.] _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141217/ae7a10c9/attachment.html From aademattia at comcast.net Thu Dec 18 06:42:41 2014 From: aademattia at comcast.net (Andrew) Date: Wed, 17 Dec 2014 22:42:41 -0500 Subject: [Freeswitch-users] mod_sndfile.c:207 Error Opening File Message-ID: <080201d01a74$b17b9820$1472c860$@comcast.net> I see windows can do about 2k file handles if the code is changed. Not knowing C I took a guess and put _setmaxstdio(2048); under sndfile_file_open. I am thinking this is not the correct place to put this code. I can see 512 files running out fast but even when all the channels are closed I still can?t open more files. Makes me think the handlers are not being released. Any ideas?? Andrew From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of aademattia at comcast.net Sent: Wednesday, December 17, 2014 9:17 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] mod_sndfile.c:207 Error Opening File I am doing a session .read() How do I close the audio files? Sent from my HTC ----- Reply message ----- From: "Guillermo Ruiz Camauer" > To: "FreeSWITCH Users Help" > Subject: [Freeswitch-users] mod_sndfile.c:207 Error Opening File Date: Wed, Dec 17, 2014 8:46 PM You are probably opening the files, not closing them and therefore running out of file handles. Try increasing file handles. If the problem appears again but after a larger number of calls, then that's your answer. Guillermo Sent from my iPhone On 17/12/2014, at 20:38, Andrew > wrote: Hi, I have an odd problem and I have no clue how to fix this. I call about 100+ calls and then files that were working no longer work. I have been working on this for days. I know the files are real and work but for some odd reason it will not open or play any audio. If I restart everything works for the next few hundreds of calls. Working with FreeSWITCH master embedded on windows server x64. Does anyone have any idea? 2014-12-17 18:21:42.154751 [ERR] mod_sndfile.c:207 Error Opening File [c:\dialer\projects\x\audio\introvz.wav] [No Error.] 2014-12-17 18:21:59.434751 [ERR] mod_sndfile.c:207 Error Opening File [c:\dialer\projects\x\audio\introvz2.wav] [No Error.] 2014-12-17 18:23:09.574751 [ERR] mod_sndfile.c:207 Error Opening File [c:\dialer\projects\x\audio\press8.wav] [No Error.] 2014-12-17 18:23:18.634751 [ERR] mod_sndfile.c:207 Error Opening File [c:\dialer\projects\x\audio\closing.wav] [No Error.] _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141217/814586c4/attachment.html From bordmi at rarus.ru Thu Dec 18 10:59:21 2014 From: bordmi at rarus.ru (=?UTF-8?B?0JHQvtGA0LjRgdC+0LIsINCU0LzQuNGC0YDQuNC5IC8gRG1pdHJpeSBCb3Jpc292?=) Date: Thu, 18 Dec 2014 11:59:21 +0400 Subject: [Freeswitch-users] Sip_Authentiation_Table In-Reply-To: References: Message-ID: Yes, look at https://wiki.freeswitch.org/wiki/Sofia.conf.xml#odbc-dsn 2014-12-17 17:47 GMT+03:00 Abdul Hakeem : > > Hello, > > Is there any way to map the sip_authentication_table to an external > database or > to a memcache table ? > Thanks in advance. > > Regards, > Abdul Hakeem > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- with best regards, Dmitriy Borisov -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141218/4af79c9d/attachment-0001.html From peter at olssononline.se Thu Dec 18 11:00:12 2014 From: peter at olssononline.se (Peter Olsson) Date: Thu, 18 Dec 2014 09:00:12 +0100 Subject: [Freeswitch-users] Include intended modules in build In-Reply-To: <068501d01a09$91f49ee0$b5dddca0$@comcast.net> References: <068501d01a09$91f49ee0$b5dddca0$@comcast.net> Message-ID: There is a patch on Jira that will fix mod_v8 build: https://freeswitch.org/jira/browse/FS-6520. /Peter 2014-12-17 15:55 GMT+01:00 Andrew : > > Hi, > > I have the same issues and I just remove those projects. > > I also keep on 64bit. > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Malay > Thakershi > *Sent:* Wednesday, December 17, 2014 9:50 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Include intended modules in build > > > > Hello, I built with VS 2013. I only got errors in libv8 and mod_v8 > modules. Apart from that all projects built fine. Are the V8 projects > mandatory with setup? > > > > Have another doubt. Should I keep platform target as "Any CPU" or "x64"? I > am planning to put FS on 64 bit windows server in order to improve > performance. > > > > Thanks. > > > > On Tue, Dec 16, 2014 at 11:04 PM, Michael Jerris wrote: > > There are a few outstanding issues for vs 2013. We are working to correct > those still. Keep an eye out for fixes. > > > > > > On Dec 16, 2014, at 6:35 AM, Malay Thakershi wrote: > > > > Hello, > > > > I am trying to upgrade FS to windows 2012 server. > > > > Looking at this URL: > https://wiki.freeswitch.org/wiki/Installation_for_Windows > > > > 1. Can I use VS 2013? > > 2. How to include necessary modules in the build - e.g. mod_managed or > mod_avmd? > > 3. Once build is done, do I just use C:\FreeSwitch as my deployment > directory? > > > > Thanks for help and happy holidays. > > > > Malay Thakershi > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141218/c7dd85c9/attachment.html From akhilgarg7 at gmail.com Thu Dec 18 12:05:15 2014 From: akhilgarg7 at gmail.com (akhil garg) Date: Thu, 18 Dec 2014 14:35:15 +0530 Subject: [Freeswitch-users] Fwd: fs_cli hangs In-Reply-To: References: Message-ID: running "fs_cli -d 7" gives different outputs randomly but no success. success rate is 1-5 % only I can say. ------------------------------------------------------------------------------------------------------------------------------------------------ OUTPUT CASE 1: ------------------------------------------------------------------------------------------------------------------------------------------------ # fs_cli -d 7 [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is /root/.fs_cli_conf. [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is /etc/fs_cli.conf. [DEBUG] fs_cli.c:1438 main() profile default does not exist using builtin profile [DEBUG] fs_cli.c:1468 main() Using profile internal [127.0.0.1] ------------------------------------------------------------------------------------------------------------------------------------------------ # netstat -anlp | grep 8021 tcp 0 0 127.0.0.1:8021 0.0.0.0:* LISTEN 13050/freeswitch tcp 0 28 127.0.0.1:8021 127.0.0.1:42948 ESTABLISHED 13050/freeswitch tcp 0 0 127.0.0.1:42948 127.0.0.1:8021 ESTABLISHED 13090/ds_cli ------------------------------------------------------------------------------------------------------------------------------------------------ OUTPUT CASE 2: ------------------------------------------------------------------------------------------------------------------------------------------------ # fs_cli -d 7 [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is /root/.fs_cli_conf. [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is /etc/fs_cli.conf. [DEBUG] fs_cli.c:1438 main() profile default does not exist using builtin profile [DEBUG] fs_cli.c:1468 main() Using profile internal [127.0.0.1] [DEBUG] esl.c:1265 esl_recv_event() RECV HEADER [Content-Type] = [auth/request] [DEBUG] esl.c:1437 esl_recv_event() RECV MESSAGE Event-Name: SOCKET_DATA Content-Type: auth/request [DEBUG] esl.c:1465 esl_send() SEND auth ClueCon ------------------------------------------------------------------------------------------------------------------------------------------------ # netstat -anlp | grep 8021 tcp 0 0 127.0.0.1:8021 0.0.0.0:* LISTEN 12872/freeswitch tcp 0 54 127.0.0.1:8021 127.0.0.1:42945 ESTABLISHED 12872/freeswitch tcp 0 0 127.0.0.1:42945 127.0.0.1:8021 ESTABLISHED 12914/ds_cli ------------------------------------------------------------------------------------------------------------------------------------------------ OUTPUT CASE 3: ------------------------------------------------------------------------------------------------------------------------------------------------ # fs_cli -d 7 [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is /root/.fs_cli_conf. [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is /etc/fs_cli.conf. [DEBUG] fs_cli.c:1438 main() profile default does not exist using builtin profile [DEBUG] fs_cli.c:1468 main() Using profile internal [127.0.0.1] [DEBUG] esl.c:1265 esl_recv_event() RECV HEADER [Content-Type] = [auth/request] [DEBUG] esl.c:1437 esl_recv_event() RECV MESSAGE Event-Name: SOCKET_DATA Content-Type: auth/request [DEBUG] esl.c:1465 esl_send() SEND auth ClueCon [DEBUG] esl.c:1265 esl_recv_event() RECV HEADER [Content-Type] = [command/reply] [DEBUG] esl.c:1265 esl_recv_event() RECV HEADER [Reply-Text] = [+OK accepted] [DEBUG] esl.c:1437 esl_recv_event() RECV MESSAGE Event-Name: SOCKET_DATA Content-Type: command/reply Reply-Text: +OK accepted [DEBUG] esl.c:1465 esl_send() SEND log ------------------------------------------------------------------------------------------------------------------------------------------------ Freeswitch release 1.5.6b (git version: 40c105322193b7d0160814ed8ffcd5bf7f566944) -- regards, akhil -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141218/407bf7ce/attachment.html From akhilgarg7 at gmail.com Thu Dec 18 12:11:20 2014 From: akhilgarg7 at gmail.com (akhil garg) Date: Thu, 18 Dec 2014 14:41:20 +0530 Subject: [Freeswitch-users] fs_cli hangs Message-ID: previous message can be seen clearly in pastebin < https://pastebin.freeswitch.org/23749> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141218/b2cfe0e2/attachment.html From akhilgarg7 at gmail.com Thu Dec 18 12:25:32 2014 From: akhilgarg7 at gmail.com (akhil garg) Date: Thu, 18 Dec 2014 14:55:32 +0530 Subject: [Freeswitch-users] fs_cli hangs Message-ID: Please have a look on the pastepin link < https://pastebin.freeswitch.org/23750> Issue is happening when mod_event_socket server is listening on 127.0.0.1 and fs_cli is executed from the same machine where freeswitch is running. Issue is not coming when freeswitch and mod_event_socket server is listening on 192.168.1.1 and fs_cli is running 192.168.1.2 (some different machine on LAN) -- regards, akhil -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141218/4208bed4/attachment-0001.html From mbrancaleoni at voismart.it Thu Dec 18 13:08:34 2014 From: mbrancaleoni at voismart.it (Matteo) Date: Thu, 18 Dec 2014 11:08:34 +0100 (CET) Subject: [Freeswitch-users] fs_cli hangs In-Reply-To: <1415995559.157788.1418897298049.JavaMail.zimbra@voismart.it> References: Message-ID: <43417333.157789.1418897314605.JavaMail.zimbra@voismart.it> Well, ----- Il 17-dic-14, alle 23:55, Brian West brian at freeswitch.org ha scritto: > Because I have the same thing happen on debian at random that requires fs_cli to > be killed, I suspect a bug but can't replicate it enough to really nail it > down. the same happens randomly also on CentOS, but never filled a bug since cannot replicate it enough, too. mat From mbrancaleoni at voismart.it Thu Dec 18 13:10:28 2014 From: mbrancaleoni at voismart.it (Matteo) Date: Thu, 18 Dec 2014 11:10:28 +0100 (CET) Subject: [Freeswitch-users] fs_cli hangs In-Reply-To: <695037380.157801.1418897382934.JavaMail.zimbra@voismart.it> References: Message-ID: <2073162606.157817.1418897428754.JavaMail.zimbra@voismart.it> Hi, ----- Il 18-dic-14, alle 0:30, Gabriel Gunderson gabe at gundy.org ha scritto: > On Wed, Dec 17, 2014 at 3:55 PM, Brian West < brian at freeswitch.org > wrote: > > > Because I have the same thing happen on debian at random that requires fs_cli to > be killed, I suspect a bug but can't replicate it enough to really nail it > down. > > Brian, > > Is it fs_cli that hangs, or is it event socket that hangs (causing fs_cli to > misbehave)? I don't think is event socket, we use the ev socket quite heavily and never misses a bit. Mat From denis at ringme.ru Thu Dec 18 13:26:05 2014 From: denis at ringme.ru (=?UTF-8?B?0JTQtdC90LjRgQ==?=) Date: Thu, 18 Dec 2014 13:26:05 +0300 Subject: [Freeswitch-users] bug with callcenter tier, agent Message-ID: <5492ABBD.5000007@ringme.ru> When i delete not exist tiers, i always got +OK code > version FreeSWITCH Version 1.4.13~64bit ( 64bit) ========= tier freeswitch at 192.168.10.201@internal> callcenter_config tier del none none +OK 2014-12-18 13:08:05.977624 [DEBUG] mod_callcenter.c:1202 Deleted tier Agent none in Queue none freeswitch at 192.168.10.201@internal> callcenter_config tier del none none +OK 2014-12-18 13:08:53.677653 [DEBUG] mod_callcenter.c:1202 Deleted tier Agent none in Queue none ========agent freeswitch at 192.168.10.201@internal> callcenter_config agent del nothing +OK 2014-12-18 13:11:08.177763 [DEBUG] mod_callcenter.c:852 Deleted Agent nothing freeswitch at 192.168.10.201@internal> freeswitch at 192.168.10.201@internal> callcenter_config agent del nothing +OK 2014-12-18 13:11:08.417641 [DEBUG] mod_callcenter.c:852 Deleted Agent nothing From paul.atreides83 at googlemail.com Thu Dec 18 13:53:02 2014 From: paul.atreides83 at googlemail.com (Paul Atreides) Date: Thu, 18 Dec 2014 11:53:02 +0100 Subject: [Freeswitch-users] DTMF detection Message-ID: Hi, can someone please help me with DTMF detection? What am I doing wrong? I searched the net everywhere but couldn?t find a solutions for my problem. I call in with xlite. DTMF is set to "RFC 2833 and SIP INFO" But the console detects nothing. Where is the problem? Thanks for help conf/sip_profiles/internal.xml conf/dialplan/default.xml -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141218/149d0fd6/attachment.html From michael.traut at gmail.com Thu Dec 18 14:18:33 2014 From: michael.traut at gmail.com (Michael Traut) Date: Thu, 18 Dec 2014 12:18:33 +0100 Subject: [Freeswitch-users] DTMF detection In-Reply-To: References: Message-ID: I'm still a noob, but as far as i understood from here https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf you need to enable dtmf for the binding upfront in the dialplan. ** On Thu, Dec 18, 2014 at 11:53 AM, Paul Atreides < paul.atreides83 at googlemail.com> wrote: > > Hi, > > can someone please help me with DTMF detection? What am I doing wrong? > I searched the net everywhere but couldn?t find a solutions for my problem. > I call in with xlite. DTMF is set to "RFC 2833 and SIP INFO" > But the console detects nothing. Where is the problem? > > Thanks for help > > conf/sip_profiles/internal.xml > > > > > > conf/dialplan/default.xml > > > > > > > > > > data="test1,##,exec:playback,ivr/ivr welcome_to_freeswitch.wav"/> > > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141218/50c804f4/attachment.html From bordmi at rarus.ru Thu Dec 18 14:21:57 2014 From: bordmi at rarus.ru (=?UTF-8?B?0JHQvtGA0LjRgdC+0LIsINCU0LzQuNGC0YDQuNC5IC8gRG1pdHJpeSBCb3Jpc292?=) Date: Thu, 18 Dec 2014 15:21:57 +0400 Subject: [Freeswitch-users] DTMF detection In-Reply-To: References: Message-ID: try next: 2014-12-18 13:53 GMT+03:00 Paul Atreides : > > Hi, > > can someone please help me with DTMF detection? What am I doing wrong? > I searched the net everywhere but couldn?t find a solutions for my problem. > I call in with xlite. DTMF is set to "RFC 2833 and SIP INFO" > But the console detects nothing. Where is the problem? > > Thanks for help > > conf/sip_profiles/internal.xml > > > > > > conf/dialplan/default.xml > > > > > > > > > > data="test1,##,exec:playback,ivr/ivr welcome_to_freeswitch.wav"/> > > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- with best regards, Dmitriy Borisov -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141218/765a1f71/attachment-0001.html From fdelawarde at wirelessmundi.com Thu Dec 18 14:21:52 2014 From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde) Date: Thu, 18 Dec 2014 12:21:52 +0100 Subject: [Freeswitch-users] Huge BLF/NOTIFY delay under load Message-ID: <1418901712.10858.70.camel@luna.madrid.commsmundi.com> Hi all, In a decent server (6x core E5-2420 with 16GB RAM) and under a bit of load, I see that FS sends NOTIFY up to a 5 minutes AFTER the actual event, so BLFs are useless when it happens. Is it a normal/expected behavior? Code suggests that FS can queue up to 50000 events, and presence does lots and lots of DB stuff. This particular site handles around 4000-5000 subscriptions and up to 2-3 calls per second under load. I use sqlite3 presence.db in tmpfs. Is there anything that can be done to reduce that huge NOTIFY delay, or do I just have to live with it? Thanks, Fran?ois. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141218/bd6128b1/attachment.html From bordmi at rarus.ru Thu Dec 18 14:24:37 2014 From: bordmi at rarus.ru (=?UTF-8?B?0JHQvtGA0LjRgdC+0LIsINCU0LzQuNGC0YDQuNC5IC8gRG1pdHJpeSBCb3Jpc292?=) Date: Thu, 18 Dec 2014 15:24:37 +0400 Subject: [Freeswitch-users] DTMF detection In-Reply-To: References: Message-ID: not do start_dtmf 2014-12-18 14:18 GMT+03:00 Michael Traut : > > I'm still a noob, but as far as i understood from here > > https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf > > you need to enable dtmf for the binding upfront in the dialplan. > > > > > ** > > > > > > On Thu, Dec 18, 2014 at 11:53 AM, Paul Atreides < > paul.atreides83 at googlemail.com> wrote: > >> Hi, >> >> can someone please help me with DTMF detection? What am I doing wrong? >> I searched the net everywhere but couldn?t find a solutions for my >> problem. >> I call in with xlite. DTMF is set to "RFC 2833 and SIP INFO" >> But the console detects nothing. Where is the problem? >> >> Thanks for help >> >> conf/sip_profiles/internal.xml >> >> >> >> >> >> conf/dialplan/default.xml >> >> >> >> >> >> >> >> >> >> > data="test1,##,exec:playback,ivr/ivr welcome_to_freeswitch.wav"/> >> >> >> >> >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- with best regards, Dmitriy Borisov -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141218/3f2f6339/attachment.html From michael.traut at gmail.com Thu Dec 18 14:27:21 2014 From: michael.traut at gmail.com (Michael Traut) Date: Thu, 18 Dec 2014 12:27:21 +0100 Subject: [Freeswitch-users] DTMF detection In-Reply-To: References: Message-ID: Could you explain what "start_dtmf" is for? Whats the difference in this scenario? Thanks! On Thu, Dec 18, 2014 at 12:24 PM, ???????, ??????? / Dmitriy Borisov < bordmi at rarus.ru> wrote: > > not do start_dtmf > > 2014-12-18 14:18 GMT+03:00 Michael Traut : >> >> I'm still a noob, but as far as i understood from here >> >> https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf >> >> you need to enable dtmf for the binding upfront in the dialplan. >> >> >> >> >> ** >> >> >> >> >> >> On Thu, Dec 18, 2014 at 11:53 AM, Paul Atreides < >> paul.atreides83 at googlemail.com> wrote: >> >>> Hi, >>> >>> can someone please help me with DTMF detection? What am I doing wrong? >>> I searched the net everywhere but couldn?t find a solutions for my >>> problem. >>> I call in with xlite. DTMF is set to "RFC 2833 and SIP INFO" >>> But the console detects nothing. Where is the problem? >>> >>> Thanks for help >>> >>> conf/sip_profiles/internal.xml >>> >>> >>> >>> >>> >>> conf/dialplan/default.xml >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >> data="test1,##,exec:playback,ivr/ivr welcome_to_freeswitch.wav"/> >>> >>> >>> >>> >>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > -- > with best regards, > Dmitriy Borisov > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141218/d37c6aac/attachment-0001.html From bordmi at rarus.ru Thu Dec 18 14:34:14 2014 From: bordmi at rarus.ru (=?UTF-8?B?0JHQvtGA0LjRgdC+0LIsINCU0LzQuNGC0YDQuNC5IC8gRG1pdHJpeSBCb3Jpc292?=) Date: Thu, 18 Dec 2014 15:34:14 +0400 Subject: [Freeswitch-users] DTMF detection In-Reply-To: References: Message-ID: ivr application enables reading of DTMF. If you add start_dtmf there it will be two concurent functions for DTMF reading in channel. I don`t know, how they will work, but may be not properly 2014-12-18 14:27 GMT+03:00 Michael Traut : > > Could you explain what "start_dtmf" is for? Whats the difference in this > scenario? > > Thanks! > > On Thu, Dec 18, 2014 at 12:24 PM, ???????, ??????? / Dmitriy Borisov < > bordmi at rarus.ru> wrote: >> >> not do start_dtmf >> >> 2014-12-18 14:18 GMT+03:00 Michael Traut : >>> >>> I'm still a noob, but as far as i understood from here >>> >>> https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf >>> >>> you need to enable dtmf for the binding upfront in the dialplan. >>> >>> >>> >>> >>> ** >>> >>> >>> >>> >>> >>> On Thu, Dec 18, 2014 at 11:53 AM, Paul Atreides < >>> paul.atreides83 at googlemail.com> wrote: >>> >>>> Hi, >>>> >>>> can someone please help me with DTMF detection? What am I doing wrong? >>>> I searched the net everywhere but couldn?t find a solutions for my >>>> problem. >>>> I call in with xlite. DTMF is set to "RFC 2833 and SIP INFO" >>>> But the console detects nothing. Where is the problem? >>>> >>>> Thanks for help >>>> >>>> conf/sip_profiles/internal.xml >>>> >>>> >>>> >>>> >>>> >>>> conf/dialplan/default.xml >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>> data="test1,##,exec:playback,ivr/ivr welcome_to_freeswitch.wav"/> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> -- >> with best regards, >> Dmitriy Borisov >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- with best regards, Dmitriy Borisov -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141218/d8d1d1ec/attachment.html From paul.atreides83 at googlemail.com Thu Dec 18 14:59:55 2014 From: paul.atreides83 at googlemail.com (Paul Atreides) Date: Thu, 18 Dec 2014 12:59:55 +0100 Subject: [Freeswitch-users] DTMF detection In-Reply-To: References: Message-ID: Thank you Dmitriy, it worked ! On Thu, Dec 18, 2014 at 12:21 PM, ???????, ??????? / Dmitriy Borisov < bordmi at rarus.ru> wrote: > > try next: > > > > > > > > > > > > > > 2014-12-18 13:53 GMT+03:00 Paul Atreides : > >> Hi, >> >> can someone please help me with DTMF detection? What am I doing wrong? >> I searched the net everywhere but couldn?t find a solutions for my >> problem. >> I call in with xlite. DTMF is set to "RFC 2833 and SIP INFO" >> But the console detects nothing. Where is the problem? >> >> Thanks for help >> >> conf/sip_profiles/internal.xml >> >> >> >> >> >> conf/dialplan/default.xml >> >> >> >> >> >> >> >> >> >> > data="test1,##,exec:playback,ivr/ivr welcome_to_freeswitch.wav"/> >> >> >> >> >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > -- > with best regards, > Dmitriy Borisov > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141218/95df433c/attachment-0001.html From kawano at itsherpa.com Thu Dec 18 15:22:14 2014 From: kawano at itsherpa.com (=?UTF-8?B?5bed6YeO?=) Date: Thu, 18 Dec 2014 21:22:14 +0900 Subject: [Freeswitch-users] play sound file issue in conference Message-ID: hello sometimes , conference play command dose not return immediately. Example 1.make default conference 2.enter command in fs_cli many times conference aaaaaa play /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav 3. sometimes not return 4.if input dtmf return prompt why conference play command does not return immediately? freeswitch at internal> version FreeSWITCH Version 1.4.13+git~20141024T064225Z~e8a01c97f9~64bit (git e8a01c9 2014-10-24 06:42:25Z 64bit) uname -a Linux ITS-SIP-07 2.6.32-504.el6.x86_64 #1 SMP Wed Oct 15 04:27:16 UTC 2014 x86_64 x86_64 x86_64 GNU/Linux CentOS release 6.6 (Final) Kernel \r on an \m console log freeswitch at internal> conference aaaaaa play /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav (play) Playing file /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav freeswitch at internal> conference aaaaaa play /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav (play) Playing file /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav freeswitch at internal> conference aaaaaa play /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav (play) Playing file /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav freeswitch at internal> conference aaaaaa play /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav (play) Playing file /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav freeswitch at internal> conference aaaaaa play /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav _yes_no.wav ::::::::: not response conference aaaaaa play /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav conference aaaaaa play /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav conference aaaaaa play /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav conference aaaaaa play /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav 2014-12-19 05:57:45.588937 [DEBUG] switch_rtp.c:6041 RTP RECV DTMF 1:2080 2014-12-19 05:57:45.588937 [DEBUG] switch_channel.c:488 RECV DTMF 1:2080 (play) Playing file /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav freeswitch at internal> conference aaaaaa play /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav (play) Playing file /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav freeswitch at internal> conference aaaaaa play /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav (play) Playing file /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav freeswitch at internal> conference aaaaaa play /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav (play) Playing file /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav freeswitch at internal> conference aaaaaa play /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav (play) Playing file /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav 2014-12-19 05:57:45.608938 [DEBUG] mod_conference.c:5659 Queueing file '/usr/local/freeswitch/sounds/en/us/callie/currency/negative.wav' for play 2014-12-19 05:57:45.608938 [DEBUG] mod_conference.c:5659 Queueing file '/usr/local/freeswitch/sounds/en/us/callie/digits/1.wav' for play freeswitch at internal> conference aaaaaa play /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav (play) Playing file /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav freeswitch at internal> conference aaaaaa play /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav (play) Playing file /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav freeswitch at internal> conference aaaaaa play /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav (play) Playing file /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav From akhilgarg7 at gmail.com Thu Dec 18 16:05:49 2014 From: akhilgarg7 at gmail.com (akhil garg) Date: Thu, 18 Dec 2014 18:35:49 +0530 Subject: [Freeswitch-users] fs_cli hangs Message-ID: I am using linux OpenWRT (BARRIER BREAKER) -- regards, akhil -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141218/c2696494/attachment.html From italorossib at gmail.com Thu Dec 18 16:07:50 2014 From: italorossib at gmail.com (=?UTF-8?B?w410YWxvIFJvc3Np?=) Date: Thu, 18 Dec 2014 10:07:50 -0300 Subject: [Freeswitch-users] play sound file issue in conference In-Reply-To: References: Message-ID: Do your hear the audio without problem? Maybe fs_cli problem? On Thu, Dec 18, 2014 at 9:22 AM, ?? wrote: > > hello > > sometimes , conference play command dose not return immediately. > > Example > 1.make default conference > 2.enter command in fs_cli many times > conference aaaaaa play > > /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav > 3. sometimes not return > 4.if input dtmf return prompt > > why conference play command does not return immediately? > > freeswitch at internal> version > FreeSWITCH Version 1.4.13+git~20141024T064225Z~e8a01c97f9~64bit (git > e8a01c9 2014-10-24 06:42:25Z 64bit) > > uname -a > Linux ITS-SIP-07 2.6.32-504.el6.x86_64 #1 SMP Wed Oct 15 04:27:16 UTC > 2014 x86_64 x86_64 x86_64 GNU/Linux > CentOS release 6.6 (Final) > Kernel \r on an \m > > console log > freeswitch at internal> conference aaaaaa play > > /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav > (play) Playing file > > /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav > > freeswitch at internal> conference aaaaaa play > > /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav > (play) Playing file > > /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav > > freeswitch at internal> conference aaaaaa play > > /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav > (play) Playing file > > /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav > > freeswitch at internal> conference aaaaaa play > > /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav > (play) Playing file > > /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav > > freeswitch at internal> conference aaaaaa play > > /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav > _yes_no.wav ::::::::: not response > conference aaaaaa play > > /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav > conference aaaaaa play > > /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav > conference aaaaaa play > > /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav > conference aaaaaa play > > /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav > 2014-12-19 05:57:45.588937 [DEBUG] switch_rtp.c:6041 RTP RECV DTMF 1:2080 > 2014-12-19 05:57:45.588937 [DEBUG] switch_channel.c:488 RECV DTMF 1:2080 > (play) Playing file > > /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav > > freeswitch at internal> conference aaaaaa play > > /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav > (play) Playing file > > /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav > > freeswitch at internal> conference aaaaaa play > > /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav > (play) Playing file > > /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav > > freeswitch at internal> conference aaaaaa play > > /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav > (play) Playing file > > /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav > > freeswitch at internal> conference aaaaaa play > > /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav > (play) Playing file > > /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav > > 2014-12-19 05:57:45.608938 [DEBUG] mod_conference.c:5659 Queueing file > '/usr/local/freeswitch/sounds/en/us/callie/currency/negative.wav' for > play > 2014-12-19 05:57:45.608938 [DEBUG] mod_conference.c:5659 Queueing file > '/usr/local/freeswitch/sounds/en/us/callie/digits/1.wav' for play > freeswitch at internal> conference aaaaaa play > > /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav > (play) Playing file > > /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav > > freeswitch at internal> conference aaaaaa play > > /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav > (play) Playing file > > /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav > > freeswitch at internal> conference aaaaaa play > > /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav > (play) Playing file > > /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- ?talo Rossi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141218/a2114ef2/attachment.html From bordmi at rarus.ru Thu Dec 18 16:25:57 2014 From: bordmi at rarus.ru (=?UTF-8?B?0JHQvtGA0LjRgdC+0LIsINCU0LzQuNGC0YDQuNC5IC8gRG1pdHJpeSBCb3Jpc292?=) Date: Thu, 18 Dec 2014 17:25:57 +0400 Subject: [Freeswitch-users] mod_event_multicast problem Message-ID: I have some hosts with FS running. One of them working under FreeBSD, other - Debian. FreeBSD host produce next: 15:26:47.173490 IP (tos 0x0, ttl 1, id 54112, offset 0, flags [none], proto UDP (17), length 1276) 225.1.1.1.4242 > 225.1.1.1.4242: [udp sum ok] UDP, length 1248 E....`......................Event-Name: HEARTBEAT Core-UUID: 688a4808-a986-e411-820a-005056a83c1f FreeSWITCH-Hostname: switch.xxx.com FreeSWITCH-Switchname: switch.xxx.com FreeSWITCH-IPv4: xxx.xxx.xxx.36 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2014-12-18%2015%3A26%3A47 Event-Date-GMT: Thu,%2018%20Dec%202014%2012%3A26%3A47%20GMT Event-Date-Timestamp: 1418905607153759 Event-Calling-File: switch_core.c Event-Calling-Function: send_heartbeat Event-Calling-Line-Number: 70 Event-Sequence: 1808 Event-Info: System%20Ready Up-Time: 0%20years,%200%20days,%200%20hours,%2057%20minutes,%2059%20seconds,%20720%20milliseconds,%20943%20microseconds FreeSWITCH-Version: 1.5.15b%2Bgit~20141218T070405Z~d786490584~64bit Uptime-msec: 3479720 Session-Count: 0 Max-Sessions: 1000 Session-Per-Sec: 30 Session-Per-Sec-Max: 0 Session-Per-Sec-FiveMin: 0 Session-Since-Startup: 0 Session-Peak-Max: 0 Session-Peak-FiveMin: 0 Idle-CPU: 100.000000 Multicast-Sender: switch.xxx.com and Debian produce: 15:26:34.703154 IP (tos 0x0, ttl 1, id 51929, offset 0, flags [DF], proto UDP (17), length 1308) xxx.xxx.xxx.40.4242 > 225.1.1.1.4242: [udp sum ok] UDP, length 1280 E..... at ...P....(............Event-Name: HEARTBEAT Core-UUID: 9716f838-8079-11e4-8285-41b93eb8ef1c FreeSWITCH-Hostname: voip-switch-01.xxx.com FreeSWITCH-Switchname: voip-switch-01.xxx.com FreeSWITCH-IPv4: xxx.xxx.xxx.40 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2014-12-18%2015%3A26%3A44 Event-Date-GMT: Thu,%2018%20Dec%202014%2012%3A26%3A44%20GMT Event-Date-Timestamp: 1418905604180868 Event-Calling-File: switch_core.c Event-Calling-Function: send_heartbeat Event-Calling-Line-Number: 69 Event-Sequence: 29748434 Event-Info: System%20Ready Up-Time: 0%20years,%207%20days,%2021%20hours,%2052%20minutes,%2039%20seconds,%20971%20milliseconds,%20109%20microseconds FreeSWITCH-Version: 1.4.13%2Bgit~20141024T064225Z~e8a01c97f9~64bit Uptime-msec: 683559971 Session-Count: 0 Max-Sessions: 1000 Session-Per-Sec: 30 Session-Per-Sec-Max: 6 Session-Per-Sec-FiveMin: 0 Session-Since-Startup: 3827 Session-Peak-Max: 6 Session-Peak-FiveMin: 0 Idle-CPU: 99.500000 Multicast-Sender: voip-switch-01.xxx.com The difference is in source address of packets, and packets from FreeSWITCH, started on FreeBSD are not accepted by FS on linux Where must I find for solution of this problem? -- with best regards, Dmitriy Borisov -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141218/898c24a8/attachment-0001.html From kawano at itsherpa.com Thu Dec 18 16:24:43 2014 From: kawano at itsherpa.com (=?UTF-8?B?5bed6YeO?=) Date: Thu, 18 Dec 2014 22:24:43 +0900 Subject: [Freeswitch-users] play sound file issue in conference In-Reply-To: References: Message-ID: I thank event socket problem. Because, another fs_cli open and execute same command. it's retrun immediately. and it happen perl event socket program. 2014-12-18 22:07 GMT+09:00 ?talo Rossi : > Do your hear the audio without problem? > > Maybe fs_cli problem? > > On Thu, Dec 18, 2014 at 9:22 AM, ?? wrote: >> >> hello >> >> sometimes , conference play command dose not return immediately. >> >> Example >> 1.make default conference >> 2.enter command in fs_cli many times >> conference aaaaaa play >> >> /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav >> 3. sometimes not return >> 4.if input dtmf return prompt >> >> why conference play command does not return immediately? >> >> freeswitch at internal> version >> FreeSWITCH Version 1.4.13+git~20141024T064225Z~e8a01c97f9~64bit (git >> e8a01c9 2014-10-24 06:42:25Z 64bit) >> >> uname -a >> Linux ITS-SIP-07 2.6.32-504.el6.x86_64 #1 SMP Wed Oct 15 04:27:16 UTC >> 2014 x86_64 x86_64 x86_64 GNU/Linux >> CentOS release 6.6 (Final) >> Kernel \r on an \m >> >> console log >> freeswitch at internal> conference aaaaaa play >> >> /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav >> (play) Playing file >> >> /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav >> >> freeswitch at internal> conference aaaaaa play >> >> /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav >> (play) Playing file >> >> /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav >> >> freeswitch at internal> conference aaaaaa play >> >> /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav >> (play) Playing file >> >> /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav >> >> freeswitch at internal> conference aaaaaa play >> >> /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav >> (play) Playing file >> >> /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav >> >> freeswitch at internal> conference aaaaaa play >> >> /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav >> _yes_no.wav ::::::::: not response >> conference aaaaaa play >> >> /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav >> conference aaaaaa play >> >> /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav >> conference aaaaaa play >> >> /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav >> conference aaaaaa play >> >> /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav >> 2014-12-19 05:57:45.588937 [DEBUG] switch_rtp.c:6041 RTP RECV DTMF 1:2080 >> 2014-12-19 05:57:45.588937 [DEBUG] switch_channel.c:488 RECV DTMF 1:2080 >> (play) Playing file >> >> /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav >> >> freeswitch at internal> conference aaaaaa play >> >> /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav >> (play) Playing file >> >> /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav >> >> freeswitch at internal> conference aaaaaa play >> >> /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav >> (play) Playing file >> >> /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav >> >> freeswitch at internal> conference aaaaaa play >> >> /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav >> (play) Playing file >> >> /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav >> >> freeswitch at internal> conference aaaaaa play >> >> /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav >> (play) Playing file >> >> /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav >> >> 2014-12-19 05:57:45.608938 [DEBUG] mod_conference.c:5659 Queueing file >> '/usr/local/freeswitch/sounds/en/us/callie/currency/negative.wav' for >> play >> 2014-12-19 05:57:45.608938 [DEBUG] mod_conference.c:5659 Queueing file >> '/usr/local/freeswitch/sounds/en/us/callie/digits/1.wav' for play >> freeswitch at internal> conference aaaaaa play >> >> /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav >> (play) Playing file >> >> /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav >> >> freeswitch at internal> conference aaaaaa play >> >> /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav >> (play) Playing file >> >> /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav >> >> freeswitch at internal> conference aaaaaa play >> >> /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav >> (play) Playing file >> >> /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > ?talo Rossi > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 kawano at itsherpa.com Thu Dec 18 16:36:03 2014 From: kawano at itsherpa.com (=?UTF-8?B?5bed6YeO?=) Date: Thu, 18 Dec 2014 22:36:03 +0900 Subject: [Freeswitch-users] play sound file issue in conference In-Reply-To: References: Message-ID: >Do your hear the audio without problem? Yes, i can hear the audio. And when return prompt ,I can hear the audio. 2014-12-18 22:07 GMT+09:00 ?talo Rossi : > Do your hear the audio without problem? > > Maybe fs_cli problem? > > On Thu, Dec 18, 2014 at 9:22 AM, ?? wrote: >> >> hello >> >> sometimes , conference play command dose not return immediately. >> >> Example >> 1.make default conference >> 2.enter command in fs_cli many times >> conference aaaaaa play >> >> /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav >> 3. sometimes not return >> 4.if input dtmf return prompt >> >> why conference play command does not return immediately? >> >> freeswitch at internal> version >> FreeSWITCH Version 1.4.13+git~20141024T064225Z~e8a01c97f9~64bit (git >> e8a01c9 2014-10-24 06:42:25Z 64bit) >> >> uname -a >> Linux ITS-SIP-07 2.6.32-504.el6.x86_64 #1 SMP Wed Oct 15 04:27:16 UTC >> 2014 x86_64 x86_64 x86_64 GNU/Linux >> CentOS release 6.6 (Final) >> Kernel \r on an \m >> >> console log >> freeswitch at internal> conference aaaaaa play >> >> /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav >> (play) Playing file >> >> /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav >> >> freeswitch at internal> conference aaaaaa play >> >> /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav >> (play) Playing file >> >> /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav >> >> freeswitch at internal> conference aaaaaa play >> >> /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav >> (play) Playing file >> >> /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav >> >> freeswitch at internal> conference aaaaaa play >> >> /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav >> (play) Playing file >> >> /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav >> >> freeswitch at internal> conference aaaaaa play >> >> /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav >> _yes_no.wav ::::::::: not response >> conference aaaaaa play >> >> /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav >> conference aaaaaa play >> >> /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav >> conference aaaaaa play >> >> /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav >> conference aaaaaa play >> >> /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav >> 2014-12-19 05:57:45.588937 [DEBUG] switch_rtp.c:6041 RTP RECV DTMF 1:2080 >> 2014-12-19 05:57:45.588937 [DEBUG] switch_channel.c:488 RECV DTMF 1:2080 >> (play) Playing file >> >> /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav >> >> freeswitch at internal> conference aaaaaa play >> >> /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav >> (play) Playing file >> >> /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav >> >> freeswitch at internal> conference aaaaaa play >> >> /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav >> (play) Playing file >> >> /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav >> >> freeswitch at internal> conference aaaaaa play >> >> /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav >> (play) Playing file >> >> /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav >> >> freeswitch at internal> conference aaaaaa play >> >> /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav >> (play) Playing file >> >> /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav >> >> 2014-12-19 05:57:45.608938 [DEBUG] mod_conference.c:5659 Queueing file >> '/usr/local/freeswitch/sounds/en/us/callie/currency/negative.wav' for >> play >> 2014-12-19 05:57:45.608938 [DEBUG] mod_conference.c:5659 Queueing file >> '/usr/local/freeswitch/sounds/en/us/callie/digits/1.wav' for play >> freeswitch at internal> conference aaaaaa play >> >> /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav >> (play) Playing file >> >> /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav >> >> freeswitch at internal> conference aaaaaa play >> >> /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav >> (play) Playing file >> >> /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav >> >> freeswitch at internal> conference aaaaaa play >> >> /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav >> (play) Playing file >> >> /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-tutorial_yes_no.wav >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > ?talo Rossi > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 fcastelco at gmail.com Thu Dec 18 17:13:42 2014 From: fcastelco at gmail.com (Federico Castro) Date: Thu, 18 Dec 2014 11:13:42 -0300 Subject: [Freeswitch-users] Freeswitch HA advices In-Reply-To: References: <37F06F1B-28C5-459C-950C-6BA229400A8C@jerris.com> Message-ID: Thanks Mike, I will put these parameters to false. Luis, they are in the same LAN. 2014-12-17 21:19 GMT-03:00 Luis Daniel Lucio Quiroz < luis.daniel.lucio at gmail.com>: > > are your servers in same LAN or they are in the " cloud " ? > > 2014-12-17 17:08 GMT-05:00 Michael Jerris : > > from switch.conf.xml > > > > > > > > > > > > > > > > > > On Dec 16, 2014, at 3:57 PM, Federico Castro > wrote: > > > > Hi Eliot and all, I keep on working in my HA solution. I'm almost done > but > > there are some things I would like to improve. When I finish it I will > share > > my solution with the community. > > > > I have two FS Boxes with Postgres running locally. Psql is running in > > read-write mode in Master node and in read-only mode in Slave. Psql is > > streaming data asynchronously through nodes. > > > > First thing I would like to improve: > > > > FS try to DROP some tables and delete records from sip_registrations, > > channels, etc. when connecting to psql and if it fails mod_sofia stops > > loading. This behaviour produces (in my scenary) that FS in Slave server > > does not start properly If it connects to local database because it is in > > read-only mode. > > To solve this I configured FS to connect to psql through virtual IP (it > is > > always in Master node) then mod_sofia starts properly in Slave node. The > > problem here is that FS in slave node delete records from table in Master > > node when get connected. > > > > Is there any way to start FS in a "standby mode" in slave node to avoid > it > > trying to write database? > > > > Thanks! > > > > > > > > 2014-06-16 12:05 GMT-03:00 Federico Castro : > >> > >> Hi Eliot, thanks for your verbose response, it is really useful for me. > >> > >> I'm working on a duplicated FS + postgreSQL schema. The two boxes will > >> have same HW. Both of them will run FS and postgreSQL. One will act as > >> master and the other one as stand-by waiting for the first to fail. > >> > >> FS will not have more than a hundred simultaneous calls. > >> > >> I will read about Pacemaker and Corosync and I will update to the list > >> about the implementation. > >> > >> Thanks again. > >> > >> > >> > >> 2014-06-13 10:45 GMT-03:00 Eliot Gable : > >>> > >>> On Tue, Jun 10, 2014 at 10:56 AM, Federico Castro > > >>> wrote: > >>>> > >>>> Hi all, I'm working on a Freeswitch HA solution. Now I'm deciding what > >>>> method and DB I'll use to track calls. > >>>> > >>>> I have installed PostgreSQL on both servers and I configured them to > >>>> replicate DB asynchronously. > >>>> > >>>> I would like to know if someone has experience with this kind of > >>>> solution and what things do I have to contemplate to deploy a solid > >>>> solution. > >>>> > >>> > >>> Lots of people have experience with such a solution; it all depends on > >>> what you are trying to achieve. > >>> > >>> Personally, I recommend you setup Corosync and Pacemaker both on your > >>> PostgreSQL boxes and on your FreeSWITCH systems. I also recommend you > run > >>> PostgreSQL on a separate set of boxes from FS. Both can use a lot of > memory > >>> if you are running a lot of calls and/or have a lot of clients. If you > need > >>> performance, I recommend using the fastest disks you can get in the > >>> PostgreSQL systems. Also install as much RAM as you can afford for the > >>> project in the PGSQL boxes. You will want redundant power supplies in > each > >>> system with each supply plugged into a different circuit. You will > also want > >>> redundant Ethernet connectivity to redundant switches which also have > >>> redundant power supplies. You will also want redundant cross-over > >>> connections between the pairs of boxes. > >>> > >>> Once you have Corosync and Pacemaker configured to start PGSQL and FS > on > >>> their own boxes and you have tested manual fail-over, then you need to > start > >>> thinking about every possible way you can make either of those two > systems > >>> stop working. Think about hard drives failing, power loss, kernel > panics, > >>> firewall rules blocking communication, someone accidentally removing > the IP > >>> address from one of the systems (it happens), killing processes, Sofia > >>> profiles failing to load because something else is using the port, > etc. Make > >>> sure you have things set up to detect and recover from any such > failure. One > >>> of the best ways to do this is to actually build an external testing > system > >>> which places real calls through the system and has them route back to > itself > >>> to verify they made it. If it places a call and the call does not make > it > >>> back to itself, then you know something failed and you can run more > tests to > >>> determine what failed and reset it. > >>> > >>> Like I said, it all depends on what you are trying to accomplish. If > you > >>> want really good automatic HA, you have to go to some pretty great > lengths > >>> to get it. If you are OK with occasional manual intervention, then you > can > >>> make some assumptions (like nobody accidentally removing your IP from > the > >>> interface or telling it to stop responding to ARP or throwing up a > firewall > >>> rule which blocks something). That makes the setup considerably > easier, but > >>> it also means manual intervention when something like that happens. In > other > >>> words, if something like that happens, you experience an outage which > the HA > >>> system doesn't detect and recover from. When you get calls that service > >>> stopped working, you then have someone log in and take a look and > manually > >>> fix the issue. This could take anywhere from 5 minutes to an hour or > more to > >>> do, depending on how good your support is and how good your team is. > >>> > >>> So, probably the first task you should do is list all the things you > want > >>> it to automatically recover from and all the things you are willing to > >>> accept causing an outage and then work on your implementation based on > that > >>> plan. > >>> > >>> > >>> > >>> > _________________________________________________________________________ > >>> Professional FreeSWITCH Consulting Services: > >>> consulting at 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 at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >> > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://confluence.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://confluence.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141218/2101bb1e/attachment-0001.html From nreis at wavecom.pt Thu Dec 18 18:25:53 2014 From: nreis at wavecom.pt (Nuno Reis) Date: Thu, 18 Dec 2014 15:25:53 +0000 Subject: [Freeswitch-users] fs_cli hangs In-Reply-To: References: Message-ID: Hello all. I'm experiencing the same from time to time under CentOS 7 (x86_64) with latest v1.4. -- *Nuno Miguel Reis* | *Unified Communication** Systems* M. +351 913907481 | nreis at wavecom.pt WAVECOM-Solu??es R?dio, S.A. Cacia Park | Rua do Progresso, Lote 15 3800-639 AVEIRO | Portugal T. +351 309 700 225 | F. +351 234 919 191 *GPS | www.wavecom.pt ** * [image: Description: Description: WavecomSignature] [image: Publicity] On Wed, Dec 17, 2014 at 10:55 PM, Brian West wrote: > > Because I have the same thing happen on debian at random that requires > fs_cli to be killed, I suspect a bug but can't replicate it enough to > really nail it down. > > On Wed, Dec 17, 2014 at 12:13 PM, Jo?o Mesquita > wrote: >> >> bkw, why ask if it's a Debian? I'm just curious... >> >> JM >> >> Jo?o Mesquita >> FreeSWITCH? Solutions >> >> On Wed, Dec 17, 2014 at 11:28 AM, Brian West >> wrote: >>> >>> One of these things is not like the others... Debian is a linux distro >>> and mips is a CPU, so what linux distro are you running on this mips >>> platform? >>> >>> On Wed, Dec 17, 2014 at 2:56 AM, akhil garg >>> wrote: >>> >>>> Its not Debian but mips >>>> >>>> -- >>>> regards, >>>> akhil >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> -- >>> >>> *Brian West* >>> brian at freeswitch.org >>> >>> >>> *Twitter: @FreeSWITCH , @briankwest* >>> http://www.freeswitchbook.com >>> http://www.freeswitchcookbook.com >>> >>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > -- > > *Brian West* > brian at freeswitch.org > > > *Twitter: @FreeSWITCH , @briankwest* > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141218/216106a0/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.png Type: image/png Size: 16423 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141218/216106a0/attachment-0001.png From fdelawarde at wirelessmundi.com Thu Dec 18 18:50:04 2014 From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde) Date: Thu, 18 Dec 2014 16:50:04 +0100 Subject: [Freeswitch-users] Huge BLF/NOTIFY delay under load In-Reply-To: <1418901712.10858.70.camel@luna.madrid.commsmundi.com> References: <1418901712.10858.70.camel@luna.madrid.commsmundi.com> Message-ID: <1418917804.4455.9.camel@luna.madrid.commsmundi.com> Still not sure if it's a bug and should be added as a JIRA or if it is an expected behavior for that kind of load. PS: I tried tweaking "max-db-handles" but it doesn't seem to make any difference with sqlite3 core db in tmpfs. Regards, Fran?ois. On Thu, 2014-12-18 at 12:21 +0100, Fran?ois Delawarde wrote: > Hi all, > > In a decent server (6x core E5-2420 with 16GB RAM) and under a bit of > load, I see that FS sends NOTIFY up to a 5 minutes AFTER the actual > event, so BLFs are useless when it happens. > > Is it a normal/expected behavior? Code suggests that FS can queue up > to 50000 events, and presence does lots and lots of DB stuff. > > This particular site handles around 4000-5000 subscriptions and up to > 2-3 calls per second under load. I use sqlite3 presence.db in tmpfs. > > Is there anything that can be done to reduce that huge NOTIFY delay, > or do I just have to live with it? > > Thanks, > Fran?ois. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141218/65a7c68d/attachment.html From callum.guy at x-on.co.uk Thu Dec 18 19:39:30 2014 From: callum.guy at x-on.co.uk (Callum Guy) Date: Thu, 18 Dec 2014 16:39:30 +0000 Subject: [Freeswitch-users] RTP NAT No Audio on EC2 Message-ID: Hi All, Its been a while since i've written to this board so i hope you're all well. Today i've been trying out FreeSWITCH on AWS for the first time and it certainly appears to be working as expected however my configuration must be wrong for the environment so i wondered if someone here could offer some advice? A TCP dump shows correct SIP negotiation between my handset and the server over the respective public IP's however when viewing the flow in Wireshark the RTP heads off to my handsets internal IP. I see this in the CLI: 2014-12-18 16:32:28.531022 [DEBUG] switch_core_media.c:5141 AUDIO RTP [sofia/internal/1000 at 54.77.115.40] 172.31.6.36 port 30074 -> 192.168.5.195 port 16468 codec: 0 ms: 20 Does anyone have any keen suggestions? Thanks, Callum ______________________________ Callum Guy X-on Framlingham Technology Centre Station Road, Framlingham, Suffolk, IP13 9EZ T 0333 332 0116 E callum.guy at x-on.co.uk X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD Company Registration No. 2578478 This email has been sent from X-on.The contents and attachments are confidential to the sender and the intended addressees.If the message is received by anyone other than the addressee please return the message to the sender by replying to it and then delete the message from your computer without copying or disclosing the contents to anyone.Opinions, conclusions and statements of intent in this email are those of the sender and do not bind X-on unless confirmed by authorised representatives independently of this message.While best endeavours have been taken to avoid transmission of viruses, it is the responsibility of the recipient to scan for these.Please note emails sent to and from X-on are routinely monitored for record keeping and quality control, to ensure regulatory compliance and prevent unauthorised use of our systems. Please consider the environment before printing this email. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141218/81bb625d/attachment.html From italorossib at gmail.com Thu Dec 18 19:51:10 2014 From: italorossib at gmail.com (=?UTF-8?B?w410YWxvIFJvc3Np?=) Date: Thu, 18 Dec 2014 13:51:10 -0300 Subject: [Freeswitch-users] RTP NAT No Audio on EC2 In-Reply-To: References: Message-ID: Did you changed the rtp-ip and sip-ip vars in your profile? Follow https://freeswitch.org/confluence/display/FREESWITCH/NAT On Thu, Dec 18, 2014 at 1:39 PM, Callum Guy wrote: > > Hi All, > > Its been a while since i've written to this board so i hope you're all > well. Today i've been trying out FreeSWITCH on AWS for the first time and > it certainly appears to be working as expected however my configuration > must be wrong for the environment so i wondered if someone here could offer > some advice? > > A TCP dump shows correct SIP negotiation between my handset and the server > over the respective public IP's however when viewing the flow in Wireshark > the RTP heads off to my handsets internal IP. > > I see this in the CLI: > > 2014-12-18 16:32:28.531022 [DEBUG] switch_core_media.c:5141 AUDIO RTP > [sofia/internal/1000 at 54.77.115.40] 172.31.6.36 port 30074 -> > 192.168.5.195 port 16468 codec: 0 ms: 20 > > Does anyone have any keen suggestions? > > Thanks, > > Callum > > ______________________________ > > Callum Guy > > X-on > Framlingham Technology Centre > Station Road, Framlingham, > Suffolk, IP13 9EZ > > T 0333 332 0116 > E callum.guy at x-on.co.uk > > > X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales > Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD > Company Registration No. 2578478 > > This email has been sent from X-on.The contents and attachments are confidential to the sender and the intended addressees.If the message > is received by anyone other than the addressee please return the message to the sender by replying to it and then delete the message from > your computer without copying or disclosing the contents to anyone.Opinions, conclusions and statements of intent in this email are those of > the sender and do not bind X-on unless confirmed by authorised representatives independently of this message.While best endeavours have > been taken to avoid transmission of viruses, it is the responsibility of the recipient to scan for these.Please note emails sent to and from X-on > are routinely monitored for record keeping and quality control, to ensure regulatory compliance and prevent unauthorised use of our systems. > Please consider the environment before printing this email. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- ?talo Rossi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141218/dafefd37/attachment.html From callum.guy at x-on.co.uk Thu Dec 18 19:58:03 2014 From: callum.guy at x-on.co.uk (Callum Guy) Date: Thu, 18 Dec 2014 16:58:03 +0000 Subject: [Freeswitch-users] RTP NAT No Audio on EC2 In-Reply-To: References: Message-ID: I have not changed those values yet so will look into that immediately and get back to you. Most of my changes are those cited here: https://wiki.freeswitch.org/wiki/Amazon_EC2 which include ext- prefixed versions of those parameters but i haven't changed those without the prefix. Thanks, back soon ______________________________ Callum Guy X-on Framlingham Technology Centre Station Road, Framlingham, Suffolk, IP13 9EZ T 0333 332 0116 E callum.guy at x-on.co.uk X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD Company Registration No. 2578478 This email has been sent from X-on.The contents and attachments are confidential to the sender and the intended addressees.If the message is received by anyone other than the addressee please return the message to the sender by replying to it and then delete the message from your computer without copying or disclosing the contents to anyone.Opinions, conclusions and statements of intent in this email are those of the sender and do not bind X-on unless confirmed by authorised representatives independently of this message.While best endeavours have been taken to avoid transmission of viruses, it is the responsibility of the recipient to scan for these.Please note emails sent to and from X-on are routinely monitored for record keeping and quality control, to ensure regulatory compliance and prevent unauthorised use of our systems. Please consider the environment before printing this email. On 18 December 2014 at 16:51, ?talo Rossi wrote: > > Did you changed the rtp-ip and sip-ip vars in your profile? > > Follow https://freeswitch.org/confluence/display/FREESWITCH/NAT > > On Thu, Dec 18, 2014 at 1:39 PM, Callum Guy wrote: > >> Hi All, >> >> Its been a while since i've written to this board so i hope you're all >> well. Today i've been trying out FreeSWITCH on AWS for the first time and >> it certainly appears to be working as expected however my configuration >> must be wrong for the environment so i wondered if someone here could offer >> some advice? >> >> A TCP dump shows correct SIP negotiation between my handset and the >> server over the respective public IP's however when viewing the flow in >> Wireshark the RTP heads off to my handsets internal IP. >> >> I see this in the CLI: >> >> 2014-12-18 16:32:28.531022 [DEBUG] switch_core_media.c:5141 AUDIO RTP >> [sofia/internal/1000 at 54.77.115.40] 172.31.6.36 port 30074 -> >> 192.168.5.195 port 16468 codec: 0 ms: 20 >> >> Does anyone have any keen suggestions? >> >> Thanks, >> >> Callum >> >> ______________________________ >> >> Callum Guy >> >> X-on >> Framlingham Technology Centre >> Station Road, Framlingham, >> Suffolk, IP13 9EZ >> >> T 0333 332 0116 >> E callum.guy at x-on.co.uk >> >> >> X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales >> Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD >> Company Registration No. 2578478 >> >> This email has been sent from X-on.The contents and attachments are confidential to the sender and the intended addressees.If the message >> is received by anyone other than the addressee please return the message to the sender by replying to it and then delete the message from >> your computer without copying or disclosing the contents to anyone.Opinions, conclusions and statements of intent in this email are those of >> the sender and do not bind X-on unless confirmed by authorised representatives independently of this message.While best endeavours have >> been taken to avoid transmission of viruses, it is the responsibility of the recipient to scan for these.Please note emails sent to and from X-on >> are routinely monitored for record keeping and quality control, to ensure regulatory compliance and prevent unauthorised use of our systems. >> Please consider the environment before printing this email. >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > -- > ?talo Rossi > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141218/89ec8b17/attachment-0001.html From callum.guy at x-on.co.uk Thu Dec 18 20:28:52 2014 From: callum.guy at x-on.co.uk (Callum Guy) Date: Thu, 18 Dec 2014 17:28:52 +0000 Subject: [Freeswitch-users] RTP NAT No Audio on EC2 In-Reply-To: References: Message-ID: ?OK so i have been changing these options as recommended: ? The SIP option I set to my real external (elastic) IP and it broke the SIP comms - i removed that because SIP is working correctly. The RTP option was set as auto but this doesn't appear to have made any noticeable difference. I haven't found any detailed information about what those fields actually do so apologies for the shot-in-the-dark approach here. Any further ideas? ______________________________ Callum Guy X-on Framlingham Technology Centre Station Road, Framlingham, Suffolk, IP13 9EZ T 0333 332 0116 E callum.guy at x-on.co.uk X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD Company Registration No. 2578478 This email has been sent from X-on.The contents and attachments are confidential to the sender and the intended addressees.If the message is received by anyone other than the addressee please return the message to the sender by replying to it and then delete the message from your computer without copying or disclosing the contents to anyone.Opinions, conclusions and statements of intent in this email are those of the sender and do not bind X-on unless confirmed by authorised representatives independently of this message.While best endeavours have been taken to avoid transmission of viruses, it is the responsibility of the recipient to scan for these.Please note emails sent to and from X-on are routinely monitored for record keeping and quality control, to ensure regulatory compliance and prevent unauthorised use of our systems. Please consider the environment before printing this email. On 18 December 2014 at 16:58, Callum Guy wrote: > > I have not changed those values yet so will look into that immediately and > get back to you. > > Most of my changes are those cited here: > https://wiki.freeswitch.org/wiki/Amazon_EC2 which include ext- prefixed > versions of those parameters but i haven't changed those without the prefix. > > Thanks, back soon > > > > ______________________________ > > Callum Guy > > X-on > Framlingham Technology Centre > Station Road, Framlingham, > Suffolk, IP13 9EZ > > T 0333 332 0116 > E callum.guy at x-on.co.uk > > > X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales > Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD > Company Registration No. 2578478 > > This email has been sent from X-on.The contents and attachments are confidential to the sender and the intended addressees.If the message > is received by anyone other than the addressee please return the message to the sender by replying to it and then delete the message from > your computer without copying or disclosing the contents to anyone.Opinions, conclusions and statements of intent in this email are those of > the sender and do not bind X-on unless confirmed by authorised representatives independently of this message.While best endeavours have > been taken to avoid transmission of viruses, it is the responsibility of the recipient to scan for these.Please note emails sent to and from X-on > are routinely monitored for record keeping and quality control, to ensure regulatory compliance and prevent unauthorised use of our systems. > Please consider the environment before printing this email. > > > On 18 December 2014 at 16:51, ?talo Rossi wrote: >> >> Did you changed the rtp-ip and sip-ip vars in your profile? >> >> Follow https://freeswitch.org/confluence/display/FREESWITCH/NAT >> >> On Thu, Dec 18, 2014 at 1:39 PM, Callum Guy >> wrote: >> >>> Hi All, >>> >>> Its been a while since i've written to this board so i hope you're all >>> well. Today i've been trying out FreeSWITCH on AWS for the first time and >>> it certainly appears to be working as expected however my configuration >>> must be wrong for the environment so i wondered if someone here could offer >>> some advice? >>> >>> A TCP dump shows correct SIP negotiation between my handset and the >>> server over the respective public IP's however when viewing the flow in >>> Wireshark the RTP heads off to my handsets internal IP. >>> >>> I see this in the CLI: >>> >>> 2014-12-18 16:32:28.531022 [DEBUG] switch_core_media.c:5141 AUDIO RTP >>> [sofia/internal/1000 at 54.77.115.40] 172.31.6.36 port 30074 -> >>> 192.168.5.195 port 16468 codec: 0 ms: 20 >>> >>> Does anyone have any keen suggestions? >>> >>> Thanks, >>> >>> Callum >>> >>> ______________________________ >>> >>> Callum Guy >>> >>> X-on >>> Framlingham Technology Centre >>> Station Road, Framlingham, >>> Suffolk, IP13 9EZ >>> >>> T 0333 332 0116 >>> E callum.guy at x-on.co.uk >>> >>> >>> X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales >>> Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD >>> Company Registration No. 2578478 >>> >>> This email has been sent from X-on.The contents and attachments are confidential to the sender and the intended addressees.If the message >>> is received by anyone other than the addressee please return the message to the sender by replying to it and then delete the message from >>> your computer without copying or disclosing the contents to anyone.Opinions, conclusions and statements of intent in this email are those of >>> the sender and do not bind X-on unless confirmed by authorised representatives independently of this message.While best endeavours have >>> been taken to avoid transmission of viruses, it is the responsibility of the recipient to scan for these.Please note emails sent to and from X-on >>> are routinely monitored for record keeping and quality control, to ensure regulatory compliance and prevent unauthorised use of our systems. >>> Please consider the environment before printing this email. >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> -- >> ?talo Rossi >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141218/47c026d6/attachment-0001.html From mike at jerris.com Thu Dec 18 21:16:25 2014 From: mike at jerris.com (Michael Jerris) Date: Thu, 18 Dec 2014 13:16:25 -0500 Subject: [Freeswitch-users] freeswitch 1.4 and encryption/rtcp-mux In-Reply-To: References: <8B568748-EEE1-4E26-A3BB-302345FB0A01@jerris.com> Message-ID: <68FF9142-0519-4D45-A006-4713607ED7F3@jerris.com> It does not enable ice by default. If it did, everyone would be having issues. What condition is it triggering to enable ice? Its possible the lines from ice to webrtc triggering each other are a bit crossed, but I would have to see specifics. Again. Please move this to jira so we can discuss and have history of what is found recorded. > On Dec 17, 2014, at 9:55 AM, Michel Brabants wrote: > > Hello, > > I found the issue: 2 lines of code. I'll see if I can submit a patch, but I'm looking into it why it was added. The problem exists primarily because everywhere where a local sdp is generated, the function set_ice is called, while I don't want it (because I don't need it - nonat - and it generates, ice-not-ready-errors causing rtp to be dropped). The set_ice-function also sets the webrtc-flag (not sure why), causing dtls to become a requirement, which is not true in the current context. Anyway, this is nothing for this list, but I just want to add for any user currently encountering this problem. > I'll do my best to generate a usefull patch. > > Michel > > On Fri, Dec 12, 2014 at 7:06 PM, Michael Jerris > wrote: > > > On Dec 12, 2014, at 10:23 AM, Michel Brabants > wrote: > > > > Hello, > > > > I recently started upgrading to FS 1.4, but I encountered 2 difficulties of which I'm still looking into one: > > > > 1) DTLS-configuration seems to be required, although we don't use it currently. We use normal sip-profiles (no webrtc). The option to disable it, is "webrtc_enable_dtls=false", which can b set in the dialplan. But why is it trying to enable it by default? Can you disable it also in a profile? > > > > In what way do you think that some configuration is required? > > > 2) Also a change because of webrtc seemingly. When receiving an invite (without sdp - 3pcc-request), freeswitch in the end response in its 200 OK with a rtcp-mux-line in its sdp. We don't want rtcp-mux, just rtp-port+1 for rtcp. When looking at the code, I don't currently know why FS sends back the myx-parameter as it seems only enabled when the other ends proposes it or am I missing something? > > > > Please report a bug on this. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141218/2287eac1/attachment.html From brian at freeswitch.org Thu Dec 18 23:22:43 2014 From: brian at freeswitch.org (Brian West) Date: Thu, 18 Dec 2014 14:22:43 -0600 Subject: [Freeswitch-users] fs_cli hangs In-Reply-To: References: Message-ID: If its an issue in libedit, or our code we need to nail it down and fix it or figure out the cause. Could one of you file a JIRA. On Thu, Dec 18, 2014 at 9:25 AM, Nuno Reis wrote: > > Hello all. > > I'm experiencing the same from time to time under CentOS 7 (x86_64) with > latest v1.4. > > -- > > *Nuno Miguel Reis* | *Unified Communication** Systems* > M. +351 913907481 | nreis at wavecom.pt > WAVECOM-Solu??es R?dio, S.A. > Cacia Park | Rua do Progresso, Lote 15 > 3800-639 AVEIRO | Portugal > T. +351 309 700 225 | F. +351 234 919 191 > *GPS > > | www.wavecom.pt ** * > > [image: Description: Description: WavecomSignature] > > > [image: Publicity] > > > > On Wed, Dec 17, 2014 at 10:55 PM, Brian West wrote: >> >> Because I have the same thing happen on debian at random that requires >> fs_cli to be killed, I suspect a bug but can't replicate it enough to >> really nail it down. >> >> On Wed, Dec 17, 2014 at 12:13 PM, Jo?o Mesquita > > wrote: >>> >>> bkw, why ask if it's a Debian? I'm just curious... >>> >>> JM >>> >>> Jo?o Mesquita >>> FreeSWITCH? Solutions >>> >>> On Wed, Dec 17, 2014 at 11:28 AM, Brian West >>> wrote: >>>> >>>> One of these things is not like the others... Debian is a linux distro >>>> and mips is a CPU, so what linux distro are you running on this mips >>>> platform? >>>> >>>> On Wed, Dec 17, 2014 at 2:56 AM, akhil garg >>>> wrote: >>>> >>>>> Its not Debian but mips >>>>> >>>>> -- >>>>> regards, >>>>> akhil >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> -- >>>> >>>> *Brian West* >>>> brian at freeswitch.org >>>> >>>> >>>> *Twitter: @FreeSWITCH , @briankwest* >>>> http://www.freeswitchbook.com >>>> http://www.freeswitchcookbook.com >>>> >>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> -- >> >> *Brian West* >> brian at freeswitch.org >> >> >> *Twitter: @FreeSWITCH , @briankwest* >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141218/35c59e60/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.png Type: image/png Size: 16423 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141218/35c59e60/attachment-0001.png From brian at freeswitch.org Thu Dec 18 23:42:08 2014 From: brian at freeswitch.org (Brian West) Date: Thu, 18 Dec 2014 14:42:08 -0600 Subject: [Freeswitch-users] RTP NAT No Audio on EC2 In-Reply-To: References: Message-ID: you need to set ext-sip-ip and ext-rtp-ip, and it looks like your client isn't providing you a valid ip address either. On Thu, Dec 18, 2014 at 11:28 AM, Callum Guy wrote: > > ?OK so i have been changing these options as recommended: > > ? > The SIP option I set to my real external (elastic) IP and it broke the SIP > comms - i removed that because SIP is working correctly. > > The RTP option was set as auto but this doesn't appear to have made any > noticeable difference. > > I haven't found any detailed information about what those fields actually > do so apologies for the shot-in-the-dark approach here. > > Any further ideas? > > ______________________________ > > Callum Guy > > X-on > Framlingham Technology Centre > Station Road, Framlingham, > Suffolk, IP13 9EZ > > T 0333 332 0116 > E callum.guy at x-on.co.uk > > > X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales > Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD > Company Registration No. 2578478 > > This email has been sent from X-on.The contents and attachments are confidential to the sender and the intended addressees.If the message > is received by anyone other than the addressee please return the message to the sender by replying to it and then delete the message from > your computer without copying or disclosing the contents to anyone.Opinions, conclusions and statements of intent in this email are those of > the sender and do not bind X-on unless confirmed by authorised representatives independently of this message.While best endeavours have > been taken to avoid transmission of viruses, it is the responsibility of the recipient to scan for these.Please note emails sent to and from X-on > are routinely monitored for record keeping and quality control, to ensure regulatory compliance and prevent unauthorised use of our systems. > Please consider the environment before printing this email. > > > On 18 December 2014 at 16:58, Callum Guy wrote: >> >> I have not changed those values yet so will look into that immediately >> and get back to you. >> >> Most of my changes are those cited here: >> https://wiki.freeswitch.org/wiki/Amazon_EC2 which include ext- prefixed >> versions of those parameters but i haven't changed those without the prefix. >> >> Thanks, back soon >> >> >> >> ______________________________ >> >> Callum Guy >> >> X-on >> Framlingham Technology Centre >> Station Road, Framlingham, >> Suffolk, IP13 9EZ >> >> T 0333 332 0116 >> E callum.guy at x-on.co.uk >> >> >> X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales >> Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD >> Company Registration No. 2578478 >> >> This email has been sent from X-on.The contents and attachments are confidential to the sender and the intended addressees.If the message >> is received by anyone other than the addressee please return the message to the sender by replying to it and then delete the message from >> your computer without copying or disclosing the contents to anyone.Opinions, conclusions and statements of intent in this email are those of >> the sender and do not bind X-on unless confirmed by authorised representatives independently of this message.While best endeavours have >> been taken to avoid transmission of viruses, it is the responsibility of the recipient to scan for these.Please note emails sent to and from X-on >> are routinely monitored for record keeping and quality control, to ensure regulatory compliance and prevent unauthorised use of our systems. >> Please consider the environment before printing this email. >> >> >> On 18 December 2014 at 16:51, ?talo Rossi wrote: >>> >>> Did you changed the rtp-ip and sip-ip vars in your profile? >>> >>> Follow https://freeswitch.org/confluence/display/FREESWITCH/NAT >>> >>> On Thu, Dec 18, 2014 at 1:39 PM, Callum Guy >>> wrote: >>> >>>> Hi All, >>>> >>>> Its been a while since i've written to this board so i hope you're all >>>> well. Today i've been trying out FreeSWITCH on AWS for the first time and >>>> it certainly appears to be working as expected however my configuration >>>> must be wrong for the environment so i wondered if someone here could offer >>>> some advice? >>>> >>>> A TCP dump shows correct SIP negotiation between my handset and the >>>> server over the respective public IP's however when viewing the flow in >>>> Wireshark the RTP heads off to my handsets internal IP. >>>> >>>> I see this in the CLI: >>>> >>>> 2014-12-18 16:32:28.531022 [DEBUG] switch_core_media.c:5141 AUDIO RTP >>>> [sofia/internal/1000 at 54.77.115.40] 172.31.6.36 port 30074 -> >>>> 192.168.5.195 port 16468 codec: 0 ms: 20 >>>> >>>> Does anyone have any keen suggestions? >>>> >>>> Thanks, >>>> >>>> Callum >>>> >>>> ______________________________ >>>> >>>> Callum Guy >>>> >>>> X-on >>>> Framlingham Technology Centre >>>> Station Road, Framlingham, >>>> Suffolk, IP13 9EZ >>>> >>>> T 0333 332 0116 >>>> E callum.guy at x-on.co.uk >>>> >>>> >>>> X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales >>>> Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD >>>> Company Registration No. 2578478 >>>> >>>> This email has been sent from X-on.The contents and attachments are confidential to the sender and the intended addressees.If the message >>>> is received by anyone other than the addressee please return the message to the sender by replying to it and then delete the message from >>>> your computer without copying or disclosing the contents to anyone.Opinions, conclusions and statements of intent in this email are those of >>>> the sender and do not bind X-on unless confirmed by authorised representatives independently of this message.While best endeavours have >>>> been taken to avoid transmission of viruses, it is the responsibility of the recipient to scan for these.Please note emails sent to and from X-on >>>> are routinely monitored for record keeping and quality control, to ensure regulatory compliance and prevent unauthorised use of our systems. >>>> Please consider the environment before printing this email. >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> -- >>> ?talo Rossi >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141218/3b9cc2cd/attachment-0001.html From grcamauer at gmail.com Fri Dec 19 00:42:54 2014 From: grcamauer at gmail.com (Guillermo Ruiz Camauer) Date: Thu, 18 Dec 2014 18:42:54 -0300 Subject: [Freeswitch-users] Hot to determine DTMF negotiation outcome Message-ID: How can I determine if a call has successfully negotiated a DTMF transfort mechanism? Case in point: if I dial out from FS, when the call is established, I want to know if DTMF can be successfully interchanged between the dialed party and FS. My FS is setup for RCF-2833, but sometimes I get: 2014-12-18 16:59:10.309023 [WARNING] sofia.c:8604 IGNORE INFO DTMF(5) (This channel was not configured to use INFO DTMF!) which to me means that the other party is sending DTMF using SIP INFO. If I only offer RFC-2833 and the other party does not support this, how can I find out about it? Thanks, -- Guillermo Ruiz Camauer -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141218/5e20245a/attachment.html From krice at freeswitch.org Fri Dec 19 01:12:54 2014 From: krice at freeswitch.org (Ken Rice) Date: Thu, 18 Dec 2014 22:12:54 +0000 Subject: [Freeswitch-users] Git Critical Vulnerability Announce! Message-ID: <54935166e90b5_ba157733288452@ip-10-164-132-112.mail> New Post on freeswitch.org from krice387 check it out at http://ift.tt/1wILkXi Git Critical Vulnerability Announce! The Git Team has released a new version of Git to address a critical security vulnerability. >From the Github description of the problem: A critical Git security vulnerability has been announced today, affecting all versions of the official Git client and all related software that interacts with Git repositories, including GitHub for Windows and GitHub for Mac. Because this is a client-side only vulnerability, github.com and GitHub Enterprise are not directly affected. The vulnerability concerns Git and Git-compatible clients that access Git repositories in a case-insensitive or case-normalizing filesystem. An attacker can craft a malicious Git tree that will cause Git to overwrite its own .git/config file when cloning or checking out a repository, leading to arbitrary command execution in the client machine. Git clients running on OS X (HFS+) or any version of Microsoft Windows (NTFS, FAT) are exploitable through this vulnerability. Linux clients are not affected if they run in a case-sensitive filesystem. For more information, see:http://ift.tt/1z9v0RBhttp://ift.tt/1x3mUK2 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141218/e50a9219/attachment.html From bote_radio at botecomm.com Fri Dec 19 01:16:11 2014 From: bote_radio at botecomm.com (Bote Man) Date: Thu, 18 Dec 2014 17:16:11 -0500 Subject: [Freeswitch-users] Hot to determine DTMF negotiation outcome In-Reply-To: References: Message-ID: <013801d01b10$3bc38380$b34a8a80$@com> I believe this is negotiated when the call is set up and is out of your control once it is completed. Check in the XML file that handles these calls under conf/sip_profiles/ to see if the following flag is set: I don't know what the default value is, but if you uncomment that line it should accept the method negotiated with the far end. Hope this helps. John Boteler Bote Communications Fort Lauderdale, FL From: Guillermo Ruiz Camauer Sent: Thursday, 18 December, 2014 16:43 Subject: [Freeswitch-users] Hot to determine DTMF negotiation outcome How can I determine if a call has successfully negotiated a DTMF transfort mechanism? Case in point: if I dial out from FS, when the call is established, I want to know if DTMF can be successfully interchanged between the dialed party and FS. My FS is setup for RCF-2833, but sometimes I get: 2014-12-18 16:59:10.309023 [WARNING] sofia.c:8604 IGNORE INFO DTMF(5) (This channel was not configured to use INFO DTMF!) which to me means that the other party is sending DTMF using SIP INFO. If I only offer RFC-2833 and the other party does not support this, how can I find out about it? Thanks, -- Guillermo Ruiz Camauer -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141218/1e6459c1/attachment.html From callum.guy at x-on.co.uk Fri Dec 19 01:17:59 2014 From: callum.guy at x-on.co.uk (Callum Guy) Date: Thu, 18 Dec 2014 22:17:59 +0000 Subject: [Freeswitch-users] RTP NAT No Audio on EC2 In-Reply-To: References: Message-ID: Thanks for responding. I've retested from another location and the calls are now working as expected without any change so I presume it was an issue with the device/network I was calling from. Sorry to waste your time but I still struggle with a workflow for determining RTP issues. Does anyone have any advice for a good article/book which would help? Cheers On 18 Dec 2014 20:44, "Brian West" wrote: > you need to set ext-sip-ip and ext-rtp-ip, and it looks like your client > isn't providing you a valid ip address either. > > On Thu, Dec 18, 2014 at 11:28 AM, Callum Guy > wrote: >> >> ?OK so i have been changing these options as recommended: >> >> ? >> The SIP option I set to my real external (elastic) IP and it broke the >> SIP comms - i removed that because SIP is working correctly. >> >> The RTP option was set as auto but this doesn't appear to have made any >> noticeable difference. >> >> I haven't found any detailed information about what those fields actually >> do so apologies for the shot-in-the-dark approach here. >> >> Any further ideas? >> >> ______________________________ >> >> Callum Guy >> >> X-on >> Framlingham Technology Centre >> Station Road, Framlingham, >> Suffolk, IP13 9EZ >> >> T 0333 332 0116 >> E callum.guy at x-on.co.uk >> >> >> X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales >> Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD >> Company Registration No. 2578478 >> >> This email has been sent from X-on.The contents and attachments are confidential to the sender and the intended addressees.If the message >> is received by anyone other than the addressee please return the message to the sender by replying to it and then delete the message from >> your computer without copying or disclosing the contents to anyone.Opinions, conclusions and statements of intent in this email are those of >> the sender and do not bind X-on unless confirmed by authorised representatives independently of this message.While best endeavours have >> been taken to avoid transmission of viruses, it is the responsibility of the recipient to scan for these.Please note emails sent to and from X-on >> are routinely monitored for record keeping and quality control, to ensure regulatory compliance and prevent unauthorised use of our systems. >> Please consider the environment before printing this email. >> >> >> On 18 December 2014 at 16:58, Callum Guy wrote: >>> >>> I have not changed those values yet so will look into that immediately >>> and get back to you. >>> >>> Most of my changes are those cited here: >>> https://wiki.freeswitch.org/wiki/Amazon_EC2 which include ext- prefixed >>> versions of those parameters but i haven't changed those without the prefix. >>> >>> Thanks, back soon >>> >>> >>> >>> ______________________________ >>> >>> Callum Guy >>> >>> X-on >>> Framlingham Technology Centre >>> Station Road, Framlingham, >>> Suffolk, IP13 9EZ >>> >>> T 0333 332 0116 >>> E callum.guy at x-on.co.uk >>> >>> >>> X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales >>> Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD >>> Company Registration No. 2578478 >>> >>> This email has been sent from X-on.The contents and attachments are confidential to the sender and the intended addressees.If the message >>> is received by anyone other than the addressee please return the message to the sender by replying to it and then delete the message from >>> your computer without copying or disclosing the contents to anyone.Opinions, conclusions and statements of intent in this email are those of >>> the sender and do not bind X-on unless confirmed by authorised representatives independently of this message.While best endeavours have >>> been taken to avoid transmission of viruses, it is the responsibility of the recipient to scan for these.Please note emails sent to and from X-on >>> are routinely monitored for record keeping and quality control, to ensure regulatory compliance and prevent unauthorised use of our systems. >>> Please consider the environment before printing this email. >>> >>> >>> On 18 December 2014 at 16:51, ?talo Rossi wrote: >>>> >>>> Did you changed the rtp-ip and sip-ip vars in your profile? >>>> >>>> Follow https://freeswitch.org/confluence/display/FREESWITCH/NAT >>>> >>>> On Thu, Dec 18, 2014 at 1:39 PM, Callum Guy >>>> wrote: >>>> >>>>> Hi All, >>>>> >>>>> Its been a while since i've written to this board so i hope you're all >>>>> well. Today i've been trying out FreeSWITCH on AWS for the first time and >>>>> it certainly appears to be working as expected however my configuration >>>>> must be wrong for the environment so i wondered if someone here could offer >>>>> some advice? >>>>> >>>>> A TCP dump shows correct SIP negotiation between my handset and the >>>>> server over the respective public IP's however when viewing the flow in >>>>> Wireshark the RTP heads off to my handsets internal IP. >>>>> >>>>> I see this in the CLI: >>>>> >>>>> 2014-12-18 16:32:28.531022 [DEBUG] switch_core_media.c:5141 AUDIO RTP >>>>> [sofia/internal/1000 at 54.77.115.40] 172.31.6.36 port 30074 -> >>>>> 192.168.5.195 port 16468 codec: 0 ms: 20 >>>>> >>>>> Does anyone have any keen suggestions? >>>>> >>>>> Thanks, >>>>> >>>>> Callum >>>>> >>>>> ______________________________ >>>>> >>>>> Callum Guy >>>>> >>>>> X-on >>>>> Framlingham Technology Centre >>>>> Station Road, Framlingham, >>>>> Suffolk, IP13 9EZ >>>>> >>>>> T 0333 332 0116 >>>>> E callum.guy at x-on.co.uk >>>>> >>>>> >>>>> X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales >>>>> Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD >>>>> Company Registration No. 2578478 >>>>> >>>>> This email has been sent from X-on.The contents and attachments are confidential to the sender and the intended addressees.If the message >>>>> is received by anyone other than the addressee please return the message to the sender by replying to it and then delete the message from >>>>> your computer without copying or disclosing the contents to anyone.Opinions, conclusions and statements of intent in this email are those of >>>>> the sender and do not bind X-on unless confirmed by authorised representatives independently of this message.While best endeavours have >>>>> been taken to avoid transmission of viruses, it is the responsibility of the recipient to scan for these.Please note emails sent to and from X-on >>>>> are routinely monitored for record keeping and quality control, to ensure regulatory compliance and prevent unauthorised use of our systems. >>>>> Please consider the environment before printing this email. >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> -- >>>> ?talo Rossi >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > -- > > *Brian West* > brian at freeswitch.org > > > *Twitter: @FreeSWITCH , @briankwest* > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An 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URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141218/18f6f956/attachment-0001.html From GeorgePhelps at gfphelps.com Fri Dec 19 03:52:01 2014 From: GeorgePhelps at gfphelps.com (George F. Phelps) Date: Thu, 18 Dec 2014 19:52:01 -0500 Subject: [Freeswitch-users] Provider Configuration: switch2voip.us Message-ID: <013901d01b26$005d5c10$01181430$@gfphelps.com> Could someone please provide an example, Freeswitch Provider Configuration for using a SIP trunk from switch2voip.us? I tried to use the tutorial (at the URL below), for the "Digest Authentication (Account & PIN)", as an example - but I have been unsuccessful. http://switch2voip.us/index.php/customer-support/byod/sip-trunking-configura tion Thanks! From brian at freeswitch.org Fri Dec 19 06:12:19 2014 From: brian at freeswitch.org (Brian West) Date: Thu, 18 Dec 2014 21:12:19 -0600 Subject: [Freeswitch-users] RTP NAT No Audio on EC2 In-Reply-To: References: Message-ID: A good understanding of how NAT works is probably the best place to start, I'm not too sure which materials you should read to start off with... anyone have a link to a good primer on what NAT is and how the various versions of NAT work? On Thu, Dec 18, 2014 at 4:17 PM, Callum Guy wrote: > > Thanks for responding. I've retested from another location and the calls > are now working as expected without any change so I presume it was an issue > with the device/network I was calling from. Sorry to waste your time but I > still struggle with a workflow for determining RTP issues. Does anyone have > any advice for a good article/book which would help? Cheers > On 18 Dec 2014 20:44, "Brian West" wrote: > >> you need to set ext-sip-ip and ext-rtp-ip, and it looks like your client >> isn't providing you a valid ip address either. >> >> On Thu, Dec 18, 2014 at 11:28 AM, Callum Guy >> wrote: >>> >>> ?OK so i have been changing these options as recommended: >>> >>> ? >>> The SIP option I set to my real external (elastic) IP and it broke the >>> SIP comms - i removed that because SIP is working correctly. >>> >>> The RTP option was set as auto but this doesn't appear to have made any >>> noticeable difference. >>> >>> I haven't found any detailed information about what those fields >>> actually do so apologies for the shot-in-the-dark approach here. >>> >>> Any further ideas? >>> >>> ______________________________ >>> >>> Callum Guy >>> >>> X-on >>> Framlingham Technology Centre >>> Station Road, Framlingham, >>> Suffolk, IP13 9EZ >>> >>> T 0333 332 0116 >>> E callum.guy at x-on.co.uk >>> >>> >>> X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales >>> Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD >>> Company Registration No. 2578478 >>> >>> This email has been sent from X-on.The contents and attachments are confidential to the sender and the intended addressees.If the message >>> is received by anyone other than the addressee please return the message to the sender by replying to it and then delete the message from >>> your computer without copying or disclosing the contents to anyone.Opinions, conclusions and statements of intent in this email are those of >>> the sender and do not bind X-on unless confirmed by authorised representatives independently of this message.While best endeavours have >>> been taken to avoid transmission of viruses, it is the responsibility of the recipient to scan for these.Please note emails sent to and from X-on >>> are routinely monitored for record keeping and quality control, to ensure regulatory compliance and prevent unauthorised use of our systems. >>> Please consider the environment before printing this email. >>> >>> >>> On 18 December 2014 at 16:58, Callum Guy wrote: >>>> >>>> I have not changed those values yet so will look into that immediately >>>> and get back to you. >>>> >>>> Most of my changes are those cited here: >>>> https://wiki.freeswitch.org/wiki/Amazon_EC2 which include ext- >>>> prefixed versions of those parameters but i haven't changed those without >>>> the prefix. >>>> >>>> Thanks, back soon >>>> >>>> >>>> >>>> ______________________________ >>>> >>>> Callum Guy >>>> >>>> X-on >>>> Framlingham Technology Centre >>>> Station Road, Framlingham, >>>> Suffolk, IP13 9EZ >>>> >>>> T 0333 332 0116 >>>> E callum.guy at x-on.co.uk >>>> >>>> >>>> X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales >>>> Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD >>>> Company Registration No. 2578478 >>>> >>>> This email has been sent from X-on.The contents and attachments are confidential to the sender and the intended addressees.If the message >>>> is received by anyone other than the addressee please return the message to the sender by replying to it and then delete the message from >>>> your computer without copying or disclosing the contents to anyone.Opinions, conclusions and statements of intent in this email are those of >>>> the sender and do not bind X-on unless confirmed by authorised representatives independently of this message.While best endeavours have >>>> been taken to avoid transmission of viruses, it is the responsibility of the recipient to scan for these.Please note emails sent to and from X-on >>>> are routinely monitored for record keeping and quality control, to ensure regulatory compliance and prevent unauthorised use of our systems. >>>> Please consider the environment before printing this email. >>>> >>>> >>>> On 18 December 2014 at 16:51, ?talo Rossi >>>> wrote: >>>>> >>>>> Did you changed the rtp-ip and sip-ip vars in your profile? >>>>> >>>>> Follow https://freeswitch.org/confluence/display/FREESWITCH/NAT >>>>> >>>>> On Thu, Dec 18, 2014 at 1:39 PM, Callum Guy >>>>> wrote: >>>>> >>>>>> Hi All, >>>>>> >>>>>> Its been a while since i've written to this board so i hope you're >>>>>> all well. Today i've been trying out FreeSWITCH on AWS for the first time >>>>>> and it certainly appears to be working as expected however my configuration >>>>>> must be wrong for the environment so i wondered if someone here could offer >>>>>> some advice? >>>>>> >>>>>> A TCP dump shows correct SIP negotiation between my handset and the >>>>>> server over the respective public IP's however when viewing the flow in >>>>>> Wireshark the RTP heads off to my handsets internal IP. >>>>>> >>>>>> I see this in the CLI: >>>>>> >>>>>> 2014-12-18 16:32:28.531022 [DEBUG] switch_core_media.c:5141 AUDIO RTP >>>>>> [sofia/internal/1000 at 54.77.115.40] 172.31.6.36 port 30074 -> >>>>>> 192.168.5.195 port 16468 codec: 0 ms: 20 >>>>>> >>>>>> Does anyone have any keen suggestions? >>>>>> >>>>>> Thanks, >>>>>> >>>>>> Callum >>>>>> >>>>>> ______________________________ >>>>>> >>>>>> Callum Guy >>>>>> >>>>>> X-on >>>>>> Framlingham Technology Centre >>>>>> Station Road, Framlingham, >>>>>> Suffolk, IP13 9EZ >>>>>> >>>>>> T 0333 332 0116 >>>>>> E callum.guy at x-on.co.uk >>>>>> >>>>>> >>>>>> X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales >>>>>> Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD >>>>>> Company Registration No. 2578478 >>>>>> >>>>>> This email has been sent from X-on.The contents and attachments are confidential to the sender and the intended addressees.If the message >>>>>> is received by anyone other than the addressee please return the message to the sender by replying to it and then delete the message from >>>>>> your computer without copying or disclosing the contents to anyone.Opinions, conclusions and statements of intent in this email are those of >>>>>> the sender and do not bind X-on unless confirmed by authorised representatives independently of this message.While best endeavours have >>>>>> been taken to avoid transmission of viruses, it is the responsibility of the recipient to scan for these.Please note emails sent to and from X-on >>>>>> are routinely monitored for record keeping and quality control, to ensure regulatory compliance and prevent unauthorised use of our systems. >>>>>> Please consider the environment before printing this email. >>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://confluence.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>> >>>>> >>>>> -- >>>>> ?talo Rossi >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> -- >> >> *Brian West* >> brian at freeswitch.org >> >> >> *Twitter: @FreeSWITCH , @briankwest* >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was 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URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141218/f251c6b8/attachment-0001.html From krice at freeswitch.org Fri Dec 19 08:41:09 2014 From: krice at freeswitch.org (Ken Rice) Date: Fri, 19 Dec 2014 05:41:09 +0000 Subject: [Freeswitch-users] FreeSWITCH Week in Review (Master Branch) December 7th-13th Message-ID: <5493ba752600a_162757f31435741@ip-10-99-166-230.mail> New Post on freeswitch.org from kathleen check it out at http://ift.tt/13iKNPZ FreeSWITCH Week in Review (Master Branch) December 7th-13th Hello, again. This week in the FreeSWITCH master branch we had 27 commits. The features for this week are: allowing the jitter buffer to start at 10ms and arbitrary MRCP headers can now be sent to unimrcp input components in mod_rayo. New features that were added: f024ea3 FS-7047 Arbitrary MRCP headers can now be sent to unimrcp input components in mod_rayo [Jira: http://ift.tt/1sNiaHd] e783999 Some changes to webrtc to make it work with iDoubs in rtcweb profile mode d189e98 Allow 10ms jb In terms of stability these were the use cases that were fixed: 392c687 FS-7055 Fix for a stability race condition in FS [Jira: http://ift.tt/1zCVZCS] Improvements in performance: 4bcf1d8 Use cached time to save cpu Additional documentation: f63f868 FS-7049 ? Documentation for state optional paramenter in callcenter_config queue list and count [Jira: http://ift.tt/1sNiaXt] The following bugs were squashed: 99a5b50 FS-7063 Fix for media delay issue [Jira: http://ift.tt/1zCVZCU] 21458f8 FS-7062 On redirect, when uri are passed in without <> with multiple uris, automatically add the q= header param in decending order in mod_sofia. [Jira: http://ift.tt/1sNieqa] 5376e82 FS-6688 This will fix the normal case of record route from a proxy without breaking normal changing of a contact in mod_sofia [Jira: http://ift.tt/1yzQEuo] 06c241a FS-6891 FS-7002 FS-7059 FS-7072 FS-7073 FS-7076 #close #comment All of these bugs are invalidated due to a botched revert [Jira: http://ift.tt/1yJc22F] 922fd81 FS-7015 The code was not properly catching the 0.0.0.0 after changing it to work with ICE SDPs because it was looking in the wrong place for the 0.0.0.0 [Jira: http://ift.tt/1wkDp3C] 3d515cf Re-mark cur_payload as negotiated when detected as such by parser or the rtp could stop working on session re-invite 19272dc FS-7078 Fix sip_header_as_string to properly null_terminate on larger header strings [Jira: http://ift.tt/1sNiaXv] e268a72 FS-6994 Fix for Codec OPUS decoder error in mod_opus [Jira: http://ift.tt/1zCVZTf] 6dbb416 FS-7086 FS-6798 Fix for invalid codec tearing down the call request [Jira: http://ift.tt/1sNieqf] -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141219/e5776418/attachment.html From akhilgarg7 at gmail.com Fri Dec 19 10:19:49 2014 From: akhilgarg7 at gmail.com (akhil garg) Date: Fri, 19 Dec 2014 12:49:49 +0530 Subject: [Freeswitch-users] fs_cli hangs Message-ID: Bug FS-7097 has been created on JIRA < https://freeswitch.org/jira/browse/FS-7097> -- regards, akhil -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141219/d80be4cd/attachment.html From rentmycoder at gmail.com Fri Dec 19 11:48:30 2014 From: rentmycoder at gmail.com (rentmycoder rentmycoder) Date: Fri, 19 Dec 2014 09:48:30 +0100 Subject: [Freeswitch-users] Fwd: webrtc chrome->sip bad media jssip error In-Reply-To: References: Message-ID: Hi, Maybe it's a know issue, please direct me to the right direction. Setup: FS 1.4.latest, -nonat, Debian64 7.8, http://tryit.jssip.net/ web sip client, Issue: Using Firefox, calls are working great, but with Chrome 39.0.2171.95m calling from browser to SIPendpint fails. The sip client rings, but after answer, the jssip client shows bad media description error.... Ring SDP: v=0 o=FreeSWITCH 1418885182 1418885183 IN IP4 192.168.101.43 s=FreeSWITCH c=IN IP4 192.168.101.43 t=0 0 a=msid-semantic: WMS PLEThV3VtmQWGIk4ICYTkWiv3AtV43mJ m=audio 27200 RTP/SAVPF 0 126 106 a=rtpmap:0 PCMU/8000 a=rtpmap:126 telephone-event/8000 a=rtpmap:106 CN/8000 a=ptime:20 a=sendrecv a=fingerprint:sha-256 E2:56:BC:E3:1E:CA:AD:55:04:E5:94:8F:D6:AD:1B:CA:E0:B8:90:A9:62:35:95:F0:F1:C0:D7:A7:92:35:D9:C8 a=rtcp-mux a=rtcp:27200 IN IP4 192.168.101.43 a=ssrc:1443731118 cname:FQsNBmyVzeI5MT0k a=ssrc:1443731118 msid:PLEThV3VtmQWGIk4ICYTkWiv3AtV43mJ a0 a=ssrc:1443731118 mslabel:PLEThV3VtmQWGIk4ICYTkWiv3AtV43mJ a=ssrc:1443731118 label:PLEThV3VtmQWGIk4ICYTkWiv3AtV43mJa0 a=ice-ufrag:lO4wBcSvCchx5t6B a=ice-pwd:9peBBCn0GWWiFMtRrndSqVeT a=candidate:9761529334 1 udp 659136 192.168.101.43 27200 typ host generation 0 2014-12-18 09:19:46.047277 [DEBUG] sofia.c:6624 Remote SDP: v=0 o=1001 8000 8000 IN IP4 192.168.101.39 s=SIP Call c=IN IP4 192.168.101.39 t=0 0 m=audio 5004 RTP/AVP 0 13 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 Remote SDP: 2014-12-18 09:19:46.067273 [DEBUG] mod_sofia.c:780 Local SDP sofia/internal/ 1002 at 192.168.101.43: v=0 o=FreeSWITCH 1418885182 1418885184 IN IP4 192.168.101.43 s=FreeSWITCH c=IN IP4 192.168.101.43 t=0 0 a=msid-semantic: WMS PLEThV3VtmQWGIk4ICYTkWiv3AtV43mJ m=audio 27200 RTP/SAVPF 0 126 106 a=rtpmap:0 PCMU/8000 a=rtpmap:126 telephone-event/8000 a=rtpmap:106 CN/8000 a=ptime:20 a=sendrecv a=fingerprint:sha-256 E2:56:BC:E3:1E:CA:AD:55:04:E5:94:8F:D6:AD:1B:CA:E0:B8:90:A9:62:35:95:F0:F1:C0:D7:A7:92:35:D9:C8 a=rtcp-mux a=rtcp:27200 IN IP4 192.168.101.43 a=ssrc:1443731118 cname:FQsNBmyVzeI5MT0k a=ssrc:1443731118 msid:PLEThV3VtmQWGIk4ICYTkWiv3AtV43mJ a0 a=ssrc:1443731118 mslabel:PLEThV3VtmQWGIk4ICYTkWiv3AtV43mJ a=ssrc:1443731118 label:PLEThV3VtmQWGIk4ICYTkWiv3AtV43mJa0 a=ice-ufrag:lO4wBcSvCchx5t6B a=ice-pwd:9peBBCn0GWWiFMtRrndSqVeT a=candidate:7802155548 1 udp 659136 192.168.101.43 27200 typ host generation 0 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141219/d22ea116/attachment.html From luke.milbourne at gmail.com Fri Dec 19 12:02:56 2014 From: luke.milbourne at gmail.com (Luke Milbourne) Date: Fri, 19 Dec 2014 09:02:56 +0000 Subject: [Freeswitch-users] RTP NAT No Audio on EC2 Message-ID: These have helped me in the past when dealing with NAT. Chapter 12 'Handling NAT' of the FreeSWITCH 1.2 book http://www.voip-info.org/wiki/view/NAT+and+VOIP https://wiki.freeswitch.org/wiki/NAT On 19 December 2014 at 03:12, 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. Provider Configuration: switch2voip.us (George F. Phelps) > 2. Re: RTP NAT No Audio on EC2 (Brian West) > > > ---------- Forwarded message ---------- > From: "George F. Phelps" > To: > Cc: > Date: Thu, 18 Dec 2014 19:52:01 -0500 > Subject: [Freeswitch-users] Provider Configuration: switch2voip.us > Could someone please provide an example, Freeswitch Provider Configuration > for using a SIP trunk from switch2voip.us? > > I tried to use the tutorial (at the URL below), for the "Digest > Authentication (Account & PIN)", as an example - but I have been > unsuccessful. > > > > http://switch2voip.us/index.php/customer-support/byod/sip-trunking-configura > tion > > Thanks! > > > > > > > ---------- Forwarded message ---------- > From: Brian West > To: FreeSWITCH Users Help > Cc: > Date: Thu, 18 Dec 2014 21:12:19 -0600 > Subject: Re: [Freeswitch-users] RTP NAT No Audio on EC2 > A good understanding of how NAT works is probably the best place to start, > I'm not too sure which materials you should read to start off with... > anyone have a link to a good primer on what NAT is and how the various > versions of NAT work? > > On Thu, Dec 18, 2014 at 4:17 PM, Callum Guy wrote: >> >> Thanks for responding. I've retested from another location and the calls >> are now working as expected without any change so I presume it was an issue >> with the device/network I was calling from. Sorry to waste your time but I >> still struggle with a workflow for determining RTP issues. Does anyone have >> any advice for a good article/book which would help? Cheers >> On 18 Dec 2014 20:44, "Brian West" wrote: >> >>> you need to set ext-sip-ip and ext-rtp-ip, and it looks like your client >>> isn't providing you a valid ip address either. >>> >>> On Thu, Dec 18, 2014 at 11:28 AM, Callum Guy >>> wrote: >>>> >>>> ?OK so i have been changing these options as recommended: >>>> >>>> ? >>>> The SIP option I set to my real external (elastic) IP and it broke the >>>> SIP comms - i removed that because SIP is working correctly. >>>> >>>> The RTP option was set as auto but this doesn't appear to have made any >>>> noticeable difference. >>>> >>>> I haven't found any detailed information about what those fields >>>> actually do so apologies for the shot-in-the-dark approach here. >>>> >>>> Any further ideas? >>>> >>>> ______________________________ >>>> >>>> Callum Guy >>>> >>>> X-on >>>> Framlingham Technology Centre >>>> Station Road, Framlingham, >>>> Suffolk, IP13 9EZ >>>> >>>> T 0333 332 0116 >>>> E callum.guy at x-on.co.uk >>>> >>>> >>>> X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales >>>> Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD >>>> Company Registration No. 2578478 >>>> >>>> This email has been sent from X-on.The contents and attachments are confidential to the sender and the intended addressees.If the message >>>> is received by anyone other than the addressee please return the message to the sender by replying to it and then delete the message from >>>> your computer without copying or disclosing the contents to anyone.Opinions, conclusions and statements of intent in this email are those of >>>> the sender and do not bind X-on unless confirmed by authorised representatives independently of this message.While best endeavours have >>>> been taken to avoid transmission of viruses, it is the responsibility of the recipient to scan for these.Please note emails sent to and from X-on >>>> are routinely monitored for record keeping and quality control, to ensure regulatory compliance and prevent unauthorised use of our systems. >>>> Please consider the environment before printing this email. >>>> >>>> >>>> On 18 December 2014 at 16:58, Callum Guy wrote: >>>>> >>>>> I have not changed those values yet so will look into that immediately >>>>> and get back to you. >>>>> >>>>> Most of my changes are those cited here: >>>>> https://wiki.freeswitch.org/wiki/Amazon_EC2 which include ext- >>>>> prefixed versions of those parameters but i haven't changed those without >>>>> the prefix. >>>>> >>>>> Thanks, back soon >>>>> >>>>> >>>>> >>>>> ______________________________ >>>>> >>>>> Callum Guy >>>>> >>>>> X-on >>>>> Framlingham Technology Centre >>>>> Station Road, Framlingham, >>>>> Suffolk, IP13 9EZ >>>>> >>>>> T 0333 332 0116 >>>>> E callum.guy at x-on.co.uk >>>>> >>>>> >>>>> X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales >>>>> Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD >>>>> Company Registration No. 2578478 >>>>> >>>>> This email has been sent from X-on.The contents and attachments are confidential to the sender and the intended addressees.If the message >>>>> is received by anyone other than the addressee please return the message to the sender by replying to it and then delete the message from >>>>> your computer without copying or disclosing the contents to anyone.Opinions, conclusions and statements of intent in this email are those of >>>>> the sender and do not bind X-on unless confirmed by authorised representatives independently of this message.While best endeavours have >>>>> been taken to avoid transmission of viruses, it is the responsibility of the recipient to scan for these.Please note emails sent to and from X-on >>>>> are routinely monitored for record keeping and quality control, to ensure regulatory compliance and prevent unauthorised use of our systems. >>>>> Please consider the environment before printing this email. >>>>> >>>>> >>>>> On 18 December 2014 at 16:51, ?talo Rossi >>>>> wrote: >>>>>> >>>>>> Did you changed the rtp-ip and sip-ip vars in your profile? >>>>>> >>>>>> Follow https://freeswitch.org/confluence/display/FREESWITCH/NAT >>>>>> >>>>>> On Thu, Dec 18, 2014 at 1:39 PM, Callum Guy >>>>>> wrote: >>>>>> >>>>>>> Hi All, >>>>>>> >>>>>>> Its been a while since i've written to this board so i hope you're >>>>>>> all well. Today i've been trying out FreeSWITCH on AWS for the first time >>>>>>> and it certainly appears to be working as expected however my configuration >>>>>>> must be wrong for the environment so i wondered if someone here could offer >>>>>>> some advice? >>>>>>> >>>>>>> A TCP dump shows correct SIP negotiation between my handset and the >>>>>>> server over the respective public IP's however when viewing the flow in >>>>>>> Wireshark the RTP heads off to my handsets internal IP. >>>>>>> >>>>>>> I see this in the CLI: >>>>>>> >>>>>>> 2014-12-18 16:32:28.531022 [DEBUG] switch_core_media.c:5141 AUDIO >>>>>>> RTP [sofia/internal/1000 at 54.77.115.40] 172.31.6.36 port 30074 -> >>>>>>> 192.168.5.195 port 16468 codec: 0 ms: 20 >>>>>>> >>>>>>> Does anyone have any keen suggestions? >>>>>>> >>>>>>> Thanks, >>>>>>> >>>>>>> Callum >>>>>>> >>>>>>> ______________________________ >>>>>>> >>>>>>> Callum Guy >>>>>>> >>>>>>> X-on >>>>>>> Framlingham Technology Centre >>>>>>> Station Road, Framlingham, >>>>>>> Suffolk, IP13 9EZ >>>>>>> >>>>>>> T 0333 332 0116 >>>>>>> E callum.guy at x-on.co.uk >>>>>>> >>>>>>> >>>>>>> X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales >>>>>>> Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD >>>>>>> Company Registration No. 2578478 >>>>>>> >>>>>>> This email has been sent from X-on.The contents and attachments are confidential to the sender and the intended addressees.If the message >>>>>>> is received by anyone other than the addressee please return the message to the sender by replying to it and then delete the message from >>>>>>> your computer without copying or disclosing the contents to anyone.Opinions, conclusions and statements of intent in this email are those of >>>>>>> the sender and do not bind X-on unless confirmed by authorised representatives independently of this message.While best endeavours have >>>>>>> been taken to avoid transmission of viruses, it is the responsibility of the recipient to scan for these.Please note emails sent to and from X-on >>>>>>> are routinely monitored for record keeping and quality control, to ensure regulatory compliance and prevent unauthorised use of our systems. >>>>>>> Please consider the environment before printing this email. >>>>>>> >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://confluence.freeswitch.org >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> ?talo Rossi >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://confluence.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> -- >>> >>> *Brian West* >>> brian at freeswitch.org >>> >>> >>> *Twitter: @FreeSWITCH , @briankwest* >>> http://www.freeswitchbook.com >>> http://www.freeswitchcookbook.com >>> >>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > -- > > *Brian West* > brian at freeswitch.org > > > *Twitter: @FreeSWITCH , @briankwest* > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Luke Milbourne Tel: 07857154817 Google Talk/Email: luke.milbourne at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141219/771fb0e5/attachment-0001.html From callum.guy at x-on.co.uk Fri Dec 19 12:40:22 2014 From: callum.guy at x-on.co.uk (Callum Guy) Date: Fri, 19 Dec 2014 09:40:22 +0000 Subject: [Freeswitch-users] RTP NAT No Audio on EC2 In-Reply-To: References: Message-ID: Hello Luke - didn't expect you to pop up - hope all is going well for you! Merry Xmas Cool thanks guys, i'll have a read. I do have a general understanding of NAT but i'm not able to recognise whether an issue is due to my box configuration or something natty just from looking at a trace - maybe i'm hoping for too much! You know, if my wireshark trace is showing RTP being initialised to a remote internal IP i'm assuming that means its not correct... Maybe its a case of Jack of all trades! ______________________________ Callum Guy X-on Framlingham Technology Centre Station Road, Framlingham, Suffolk, IP13 9EZ T 0333 332 0116 E callum.guy at x-on.co.uk X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD Company Registration No. 2578478 This email has been sent from X-on.The contents and attachments are confidential to the sender and the intended addressees.If the message is received by anyone other than the addressee please return the message to the sender by replying to it and then delete the message from your computer without copying or disclosing the contents to anyone.Opinions, conclusions and statements of intent in this email are those of the sender and do not bind X-on unless confirmed by authorised representatives independently of this message.While best endeavours have been taken to avoid transmission of viruses, it is the responsibility of the recipient to scan for these.Please note emails sent to and from X-on are routinely monitored for record keeping and quality control, to ensure regulatory compliance and prevent unauthorised use of our systems. Please consider the environment before printing this email. On 19 December 2014 at 09:02, Luke Milbourne wrote: > > These have helped me in the past when dealing with NAT. > > Chapter 12 'Handling NAT' of the FreeSWITCH 1.2 book > > http://www.voip-info.org/wiki/view/NAT+and+VOIP > > https://wiki.freeswitch.org/wiki/NAT > > > On 19 December 2014 at 03:12, < > 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. Provider Configuration: switch2voip.us (George F. Phelps) >> 2. Re: RTP NAT No Audio on EC2 (Brian West) >> >> >> ---------- Forwarded message ---------- >> From: "George F. Phelps" >> To: >> Cc: >> Date: Thu, 18 Dec 2014 19:52:01 -0500 >> Subject: [Freeswitch-users] Provider Configuration: switch2voip.us >> Could someone please provide an example, Freeswitch Provider Configuration >> for using a SIP trunk from switch2voip.us? >> >> I tried to use the tutorial (at the URL below), for the "Digest >> Authentication (Account & PIN)", as an example - but I have been >> unsuccessful. >> >> >> >> http://switch2voip.us/index.php/customer-support/byod/sip-trunking-configura >> tion >> >> Thanks! >> >> >> >> >> >> >> >> ---------- Forwarded message ---------- >> From: Brian West >> To: FreeSWITCH Users Help >> Cc: >> Date: Thu, 18 Dec 2014 21:12:19 -0600 >> Subject: Re: [Freeswitch-users] RTP NAT No Audio on EC2 >> A good understanding of how NAT works is probably the best place to >> start, I'm not too sure which materials you should read to start off >> with... anyone have a link to a good primer on what NAT is and how the >> various versions of NAT work? >> >> On Thu, Dec 18, 2014 at 4:17 PM, Callum Guy >> wrote: >>> >>> Thanks for responding. I've retested from another location and the calls >>> are now working as expected without any change so I presume it was an issue >>> with the device/network I was calling from. Sorry to waste your time but I >>> still struggle with a workflow for determining RTP issues. Does anyone have >>> any advice for a good article/book which would help? Cheers >>> On 18 Dec 2014 20:44, "Brian West" wrote: >>> >>>> you need to set ext-sip-ip and ext-rtp-ip, and it looks like your >>>> client isn't providing you a valid ip address either. >>>> >>>> On Thu, Dec 18, 2014 at 11:28 AM, Callum Guy >>>> wrote: >>>>> >>>>> ?OK so i have been changing these options as recommended: >>>>> >>>>> ? >>>>> The SIP option I set to my real external (elastic) IP and it broke the >>>>> SIP comms - i removed that because SIP is working correctly. >>>>> >>>>> The RTP option was set as auto but this doesn't appear to have made >>>>> any noticeable difference. >>>>> >>>>> I haven't found any detailed information about what those fields >>>>> actually do so apologies for the shot-in-the-dark approach here. >>>>> >>>>> Any further ideas? >>>>> >>>>> ______________________________ >>>>> >>>>> Callum Guy >>>>> >>>>> X-on >>>>> Framlingham Technology Centre >>>>> Station Road, Framlingham, >>>>> Suffolk, IP13 9EZ >>>>> >>>>> T 0333 332 0116 >>>>> E callum.guy at x-on.co.uk >>>>> >>>>> >>>>> X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales >>>>> Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD >>>>> Company Registration No. 2578478 >>>>> >>>>> This email has been sent from X-on.The contents and attachments are confidential to the sender and the intended addressees.If the message >>>>> is received by anyone other than the addressee please return the message to the sender by replying to it and then delete the message from >>>>> your computer without copying or disclosing the contents to anyone.Opinions, conclusions and statements of intent in this email are those of >>>>> the sender and do not bind X-on unless confirmed by authorised representatives independently of this message.While best endeavours have >>>>> been taken to avoid transmission of viruses, it is the responsibility of the recipient to scan for these.Please note emails sent to and from X-on >>>>> are routinely monitored for record keeping and quality control, to ensure regulatory compliance and prevent unauthorised use of our systems. >>>>> Please consider the environment before printing this email. >>>>> >>>>> >>>>> On 18 December 2014 at 16:58, Callum Guy >>>>> wrote: >>>>>> >>>>>> I have not changed those values yet so will look into that >>>>>> immediately and get back to you. >>>>>> >>>>>> Most of my changes are those cited here: >>>>>> https://wiki.freeswitch.org/wiki/Amazon_EC2 which include ext- >>>>>> prefixed versions of those parameters but i haven't changed those without >>>>>> the prefix. >>>>>> >>>>>> Thanks, back soon >>>>>> >>>>>> >>>>>> >>>>>> ______________________________ >>>>>> >>>>>> Callum Guy >>>>>> >>>>>> X-on >>>>>> Framlingham Technology Centre >>>>>> Station Road, Framlingham, >>>>>> Suffolk, IP13 9EZ >>>>>> >>>>>> T 0333 332 0116 >>>>>> E callum.guy at x-on.co.uk >>>>>> >>>>>> >>>>>> X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales >>>>>> Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD >>>>>> Company Registration No. 2578478 >>>>>> >>>>>> This email has been sent from X-on.The contents and attachments are confidential to the sender and the intended addressees.If the message >>>>>> is received by anyone other than the addressee please return the message to the sender by replying to it and then delete the message from >>>>>> your computer without copying or disclosing the contents to anyone.Opinions, conclusions and statements of intent in this email are those of >>>>>> the sender and do not bind X-on unless confirmed by authorised representatives independently of this message.While best endeavours have >>>>>> been taken to avoid transmission of viruses, it is the responsibility of the recipient to scan for these.Please note emails sent to and from X-on >>>>>> are routinely monitored for record keeping and quality control, to ensure regulatory compliance and prevent unauthorised use of our systems. >>>>>> Please consider the environment before printing this email. >>>>>> >>>>>> >>>>>> On 18 December 2014 at 16:51, ?talo Rossi >>>>>> wrote: >>>>>>> >>>>>>> Did you changed the rtp-ip and sip-ip vars in your profile? >>>>>>> >>>>>>> Follow https://freeswitch.org/confluence/display/FREESWITCH/NAT >>>>>>> >>>>>>> On Thu, Dec 18, 2014 at 1:39 PM, Callum Guy >>>>>>> wrote: >>>>>>> >>>>>>>> Hi All, >>>>>>>> >>>>>>>> Its been a while since i've written to this board so i hope you're >>>>>>>> all well. Today i've been trying out FreeSWITCH on AWS for the first time >>>>>>>> and it certainly appears to be working as expected however my configuration >>>>>>>> must be wrong for the environment so i wondered if someone here could offer >>>>>>>> some advice? >>>>>>>> >>>>>>>> A TCP dump shows correct SIP negotiation between my handset and the >>>>>>>> server over the respective public IP's however when viewing the flow in >>>>>>>> Wireshark the RTP heads off to my handsets internal IP. >>>>>>>> >>>>>>>> I see this in the CLI: >>>>>>>> >>>>>>>> 2014-12-18 16:32:28.531022 [DEBUG] switch_core_media.c:5141 AUDIO >>>>>>>> RTP [sofia/internal/1000 at 54.77.115.40] 172.31.6.36 port 30074 -> >>>>>>>> 192.168.5.195 port 16468 codec: 0 ms: 20 >>>>>>>> >>>>>>>> Does anyone have any keen suggestions? >>>>>>>> >>>>>>>> Thanks, >>>>>>>> >>>>>>>> Callum >>>>>>>> >>>>>>>> ______________________________ >>>>>>>> >>>>>>>> Callum Guy >>>>>>>> >>>>>>>> X-on >>>>>>>> Framlingham Technology Centre >>>>>>>> Station Road, Framlingham, >>>>>>>> Suffolk, IP13 9EZ >>>>>>>> >>>>>>>> T 0333 332 0116 >>>>>>>> E callum.guy at x-on.co.uk >>>>>>>> >>>>>>>> >>>>>>>> X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales >>>>>>>> Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD >>>>>>>> Company Registration No. 2578478 >>>>>>>> >>>>>>>> This email has been sent from X-on.The contents and attachments are confidential to the sender and the intended addressees.If the message >>>>>>>> is received by anyone other than the addressee please return the message to the sender by replying to it and then delete the message from >>>>>>>> your computer without copying or disclosing the contents to anyone.Opinions, conclusions and statements of intent in this email are those of >>>>>>>> the sender and do not bind X-on unless confirmed by authorised representatives independently of this message.While best endeavours have >>>>>>>> been taken to avoid transmission of viruses, it is the responsibility of the recipient to scan for these.Please note emails sent to and from X-on >>>>>>>> are routinely monitored for record keeping and quality control, to ensure regulatory compliance and prevent unauthorised use of our systems. >>>>>>>> Please consider the environment before printing this email. >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> _________________________________________________________________________ >>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>> consulting at freeswitch.org >>>>>>>> http://www.freeswitchsolutions.com >>>>>>>> >>>>>>>> Official FreeSWITCH Sites >>>>>>>> http://www.freeswitch.org >>>>>>>> http://confluence.freeswitch.org >>>>>>>> http://www.cluecon.com >>>>>>>> >>>>>>>> FreeSWITCH-users mailing list >>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>> UNSUBSCRIBE: >>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>> http://www.freeswitch.org >>>>>>>> >>>>>>> >>>>>>> >>>>>>> -- >>>>>>> ?talo Rossi >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://confluence.freeswitch.org >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> -- >>>> >>>> *Brian West* >>>> brian at freeswitch.org >>>> >>>> >>>> *Twitter: @FreeSWITCH , @briankwest* >>>> http://www.freeswitchbook.com >>>> http://www.freeswitchcookbook.com >>>> >>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> -- >> >> *Brian West* >> brian at freeswitch.org >> >> >> *Twitter: @FreeSWITCH , @briankwest* >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > Luke Milbourne > > > Tel: 07857154817 > Google Talk/Email: luke.milbourne at gmail.com > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141219/5e15455e/attachment-0001.html From luke.milbourne at gmail.com Fri Dec 19 13:10:48 2014 From: luke.milbourne at gmail.com (Luke Milbourne) Date: Fri, 19 Dec 2014 10:10:48 +0000 Subject: [Freeswitch-users] RTP NAT No Audio on EC2 Message-ID: Yeah still here, haven't shook the FreeSWITCH/VoIP addiction just yet :). Merry Christmas to you too (and everyone else there). Most of the problems I've come across have been solved by ensuring there is no SIP ALG in the way and then use STUN if there is still a problem. Hope it helps, all the best. Luke On 19 December 2014 at 09:40, 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: RTP NAT No Audio on EC2 (Callum Guy) > > > ---------- Forwarded message ---------- > From: Callum Guy > To: FreeSWITCH Users Help > Cc: > Date: Fri, 19 Dec 2014 09:40:22 +0000 > Subject: Re: [Freeswitch-users] RTP NAT No Audio on EC2 > Hello Luke - didn't expect you to pop up - hope all is going well for you! > Merry Xmas > > Cool thanks guys, i'll have a read. I do have a general understanding of > NAT but i'm not able to recognise whether an issue is due to my box > configuration or something natty just from looking at a trace - maybe i'm > hoping for too much! You know, if my wireshark trace is showing RTP being > initialised to a remote internal IP i'm assuming that means its not > correct... Maybe its a case of Jack of all trades! > > ______________________________ > > Callum Guy > > X-on > Framlingham Technology Centre > Station Road, Framlingham, > Suffolk, IP13 9EZ > > T 0333 332 0116 > E callum.guy at x-on.co.uk > > > X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales > Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD > Company Registration No. 2578478 > > This email has been sent from X-on.The contents and attachments are confidential to the sender and the intended addressees.If the message > is received by anyone other than the addressee please return the message to the sender by replying to it and then delete the message from > your computer without copying or disclosing the contents to anyone.Opinions, conclusions and statements of intent in this email are those of > the sender and do not bind X-on unless confirmed by authorised representatives independently of this message.While best endeavours have > been taken to avoid transmission of viruses, it is the responsibility of the recipient to scan for these.Please note emails sent to and from X-on > are routinely monitored for record keeping and quality control, to ensure regulatory compliance and prevent unauthorised use of our systems. > Please consider the environment before printing this email. > > > On 19 December 2014 at 09:02, Luke Milbourne > wrote: >> >> These have helped me in the past when dealing with NAT. >> >> Chapter 12 'Handling NAT' of the FreeSWITCH 1.2 book >> >> http://www.voip-info.org/wiki/view/NAT+and+VOIP >> >> https://wiki.freeswitch.org/wiki/NAT >> >> >> On 19 December 2014 at 03:12, < >> 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. Provider Configuration: switch2voip.us (George F. Phelps) >>> 2. Re: RTP NAT No Audio on EC2 (Brian West) >>> >>> >>> ---------- Forwarded message ---------- >>> From: "George F. Phelps" >>> To: >>> Cc: >>> Date: Thu, 18 Dec 2014 19:52:01 -0500 >>> Subject: [Freeswitch-users] Provider Configuration: switch2voip.us >>> Could someone please provide an example, Freeswitch Provider >>> Configuration >>> for using a SIP trunk from switch2voip.us? >>> >>> I tried to use the tutorial (at the URL below), for the "Digest >>> Authentication (Account & PIN)", as an example - but I have been >>> unsuccessful. >>> >>> >>> >>> http://switch2voip.us/index.php/customer-support/byod/sip-trunking-configura >>> tion >>> >>> Thanks! >>> >>> >>> >>> >>> >>> >>> >>> ---------- Forwarded message ---------- >>> From: Brian West >>> To: FreeSWITCH Users Help >>> Cc: >>> Date: Thu, 18 Dec 2014 21:12:19 -0600 >>> Subject: Re: [Freeswitch-users] RTP NAT No Audio on EC2 >>> A good understanding of how NAT works is probably the best place to >>> start, I'm not too sure which materials you should read to start off >>> with... anyone have a link to a good primer on what NAT is and how the >>> various versions of NAT work? >>> >>> On Thu, Dec 18, 2014 at 4:17 PM, Callum Guy >>> wrote: >>>> >>>> Thanks for responding. I've retested from another location and the >>>> calls are now working as expected without any change so I presume it was an >>>> issue with the device/network I was calling from. Sorry to waste your time >>>> but I still struggle with a workflow for determining RTP issues. Does >>>> anyone have any advice for a good article/book which would help? Cheers >>>> On 18 Dec 2014 20:44, "Brian West" wrote: >>>> >>>>> you need to set ext-sip-ip and ext-rtp-ip, and it looks like your >>>>> client isn't providing you a valid ip address either. >>>>> >>>>> On Thu, Dec 18, 2014 at 11:28 AM, Callum Guy >>>>> wrote: >>>>>> >>>>>> ?OK so i have been changing these options as recommended: >>>>>> >>>>>> ? >>>>>> The SIP option I set to my real external (elastic) IP and it broke >>>>>> the SIP comms - i removed that because SIP is working correctly. >>>>>> >>>>>> The RTP option was set as auto but this doesn't appear to have made >>>>>> any noticeable difference. >>>>>> >>>>>> I haven't found any detailed information about what those fields >>>>>> actually do so apologies for the shot-in-the-dark approach here. >>>>>> >>>>>> Any further ideas? >>>>>> >>>>>> ______________________________ >>>>>> >>>>>> Callum Guy >>>>>> >>>>>> X-on >>>>>> Framlingham Technology Centre >>>>>> Station Road, Framlingham, >>>>>> Suffolk, IP13 9EZ >>>>>> >>>>>> T 0333 332 0116 >>>>>> E callum.guy at x-on.co.uk >>>>>> >>>>>> >>>>>> X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales >>>>>> Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD >>>>>> Company Registration No. 2578478 >>>>>> >>>>>> This email has been sent from X-on.The contents and attachments are confidential to the sender and the intended addressees.If the message >>>>>> is received by anyone other than the addressee please return the message to the sender by replying to it and then delete the message from >>>>>> your computer without copying or disclosing the contents to anyone.Opinions, conclusions and statements of intent in this email are those of >>>>>> the sender and do not bind X-on unless confirmed by authorised representatives independently of this message.While best endeavours have >>>>>> been taken to avoid transmission of viruses, it is the responsibility of the recipient to scan for these.Please note emails sent to and from X-on >>>>>> are routinely monitored for record keeping and quality control, to ensure regulatory compliance and prevent unauthorised use of our systems. >>>>>> Please consider the environment before printing this email. >>>>>> >>>>>> >>>>>> On 18 December 2014 at 16:58, Callum Guy >>>>>> wrote: >>>>>>> >>>>>>> I have not changed those values yet so will look into that >>>>>>> immediately and get back to you. >>>>>>> >>>>>>> Most of my changes are those cited here: >>>>>>> https://wiki.freeswitch.org/wiki/Amazon_EC2 which include ext- >>>>>>> prefixed versions of those parameters but i haven't changed those without >>>>>>> the prefix. >>>>>>> >>>>>>> Thanks, back soon >>>>>>> >>>>>>> >>>>>>> >>>>>>> ______________________________ >>>>>>> >>>>>>> Callum Guy >>>>>>> >>>>>>> X-on >>>>>>> Framlingham Technology Centre >>>>>>> Station Road, Framlingham, >>>>>>> Suffolk, IP13 9EZ >>>>>>> >>>>>>> T 0333 332 0116 >>>>>>> E callum.guy at x-on.co.uk >>>>>>> >>>>>>> >>>>>>> X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales >>>>>>> Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD >>>>>>> Company Registration No. 2578478 >>>>>>> >>>>>>> This email has been sent from X-on.The contents and attachments are confidential to the sender and the intended addressees.If the message >>>>>>> is received by anyone other than the addressee please return the message to the sender by replying to it and then delete the message from >>>>>>> your computer without copying or disclosing the contents to anyone.Opinions, conclusions and statements of intent in this email are those of >>>>>>> the sender and do not bind X-on unless confirmed by authorised representatives independently of this message.While best endeavours have >>>>>>> been taken to avoid transmission of viruses, it is the responsibility of the recipient to scan for these.Please note emails sent to and from X-on >>>>>>> are routinely monitored for record keeping and quality control, to ensure regulatory compliance and prevent unauthorised use of our systems. >>>>>>> Please consider the environment before printing this email. >>>>>>> >>>>>>> >>>>>>> On 18 December 2014 at 16:51, ?talo Rossi >>>>>>> wrote: >>>>>>>> >>>>>>>> Did you changed the rtp-ip and sip-ip vars in your profile? >>>>>>>> >>>>>>>> Follow https://freeswitch.org/confluence/display/FREESWITCH/NAT >>>>>>>> >>>>>>>> On Thu, Dec 18, 2014 at 1:39 PM, Callum Guy >>>>>>>> wrote: >>>>>>>> >>>>>>>>> Hi All, >>>>>>>>> >>>>>>>>> Its been a while since i've written to this board so i hope you're >>>>>>>>> all well. Today i've been trying out FreeSWITCH on AWS for the first time >>>>>>>>> and it certainly appears to be working as expected however my configuration >>>>>>>>> must be wrong for the environment so i wondered if someone here could offer >>>>>>>>> some advice? >>>>>>>>> >>>>>>>>> A TCP dump shows correct SIP negotiation between my handset and >>>>>>>>> the server over the respective public IP's however when viewing the flow in >>>>>>>>> Wireshark the RTP heads off to my handsets internal IP. >>>>>>>>> >>>>>>>>> I see this in the CLI: >>>>>>>>> >>>>>>>>> 2014-12-18 16:32:28.531022 [DEBUG] switch_core_media.c:5141 AUDIO >>>>>>>>> RTP [sofia/internal/1000 at 54.77.115.40] 172.31.6.36 port 30074 -> >>>>>>>>> 192.168.5.195 port 16468 codec: 0 ms: 20 >>>>>>>>> >>>>>>>>> Does anyone have any keen suggestions? >>>>>>>>> >>>>>>>>> Thanks, >>>>>>>>> >>>>>>>>> Callum >>>>>>>>> >>>>>>>>> ______________________________ >>>>>>>>> >>>>>>>>> Callum Guy >>>>>>>>> >>>>>>>>> X-on >>>>>>>>> Framlingham Technology Centre >>>>>>>>> Station Road, Framlingham, >>>>>>>>> Suffolk, IP13 9EZ >>>>>>>>> >>>>>>>>> T 0333 332 0116 >>>>>>>>> E callum.guy at x-on.co.uk >>>>>>>>> >>>>>>>>> >>>>>>>>> X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales >>>>>>>>> Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD >>>>>>>>> Company Registration No. 2578478 >>>>>>>>> >>>>>>>>> This email has been sent from X-on.The contents and attachments are confidential to the sender and the intended addressees.If the message >>>>>>>>> is received by anyone other than the addressee please return the message to the sender by replying to it and then delete the message from >>>>>>>>> your computer without copying or disclosing the contents to anyone.Opinions, conclusions and statements of intent in this email are those of >>>>>>>>> the sender and do not bind X-on unless confirmed by authorised representatives independently of this message.While best endeavours have >>>>>>>>> been taken to avoid transmission of viruses, it is the responsibility of the recipient to scan for these.Please note emails sent to and from X-on >>>>>>>>> are routinely monitored for record keeping and quality control, to ensure regulatory compliance and prevent unauthorised use of our systems. >>>>>>>>> Please consider the environment before printing this email. >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> _________________________________________________________________________ >>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>> consulting at freeswitch.org >>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>> >>>>>>>>> Official FreeSWITCH Sites >>>>>>>>> http://www.freeswitch.org >>>>>>>>> http://confluence.freeswitch.org >>>>>>>>> http://www.cluecon.com >>>>>>>>> >>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>> UNSUBSCRIBE: >>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>> http://www.freeswitch.org >>>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> -- >>>>>>>> ?talo Rossi >>>>>>>> >>>>>>>> >>>>>>>> _________________________________________________________________________ >>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>> consulting at freeswitch.org >>>>>>>> http://www.freeswitchsolutions.com >>>>>>>> >>>>>>>> Official FreeSWITCH Sites >>>>>>>> http://www.freeswitch.org >>>>>>>> http://confluence.freeswitch.org >>>>>>>> http://www.cluecon.com >>>>>>>> >>>>>>>> FreeSWITCH-users mailing list >>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>> UNSUBSCRIBE: >>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>> http://www.freeswitch.org >>>>>>>> >>>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://confluence.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>> >>>>> >>>>> -- >>>>> >>>>> *Brian West* >>>>> brian at freeswitch.org >>>>> >>>>> >>>>> *Twitter: @FreeSWITCH , @briankwest* >>>>> http://www.freeswitchbook.com >>>>> http://www.freeswitchcookbook.com >>>>> >>>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >>>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> -- >>> >>> *Brian West* >>> brian at freeswitch.org >>> >>> >>> *Twitter: @FreeSWITCH , @briankwest* >>> http://www.freeswitchbook.com >>> http://www.freeswitchcookbook.com >>> >>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> -- >> Luke Milbourne >> >> >> Tel: 07857154817 >> Google Talk/Email: luke.milbourne at gmail.com >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 > > -- Luke Milbourne Tel: 07857154817 Google Talk/Email: luke.milbourne at gmail.com -------------- next part -------------- An HTML 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URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141219/52401abb/attachment-0001.html From frederick at targointernet.com Fri Dec 19 17:56:35 2014 From: frederick at targointernet.com (Frederick Pruneau) Date: Fri, 19 Dec 2014 09:56:35 -0500 Subject: [Freeswitch-users] Issue with Freeswitch Behind nat In-Reply-To: References: Message-ID: Did you find something? 2014-12-17 10:36 GMT-05:00 Frederick Pruneau : > > Here it is: https://pastebin.freeswitch.org/23746 > > It is my freeswitch log. I have followed this guide before: > > https://wiki.freeswitch.org/wiki/Sofia#Debugging_Sofia-SIP > > I have enabled this and make a call: > > sofia global siptrace on > sofia loglevel all 9 > sofia tracelevel alert > console loglevel debug > fsctl debug_level 10 > > This is what you will get in my pastebin > > > 2014-12-17 9:27 GMT-05:00 Brian West : > >> sofia global siptrace on >> >> from fs_cli >> >> On Wed, Dec 17, 2014 at 7:44 AM, Frederick Pruneau < >> frederick at targointernet.com> wrote: >>> >>> Sorry for this noob question but how can I see sip traffic? Is there a >>> specific command to show this? Is it what we find in freeswitch.log? If so, >>> I attached my log file in my first post. >>> >>> Thanks for you help >>> >>> 2014-12-16 15:44 GMT-05:00 Brian West : >>> >>>> have you looked at the signalling? What does the sip traffic show? >>>> Please pastebin that. >>>> >>>> On Tue, Dec 16, 2014 at 2:37 PM, Frederick Pruneau < >>>> frederick at targointernet.com> wrote: >>>>> >>>>> Same problem... >>>>> >>>>> 2014-12-16 13:55 GMT-05:00 Brian West : >>>>> >>>>>> Guessing you don't have UPNP or NAT-PMP on your network, there for >>>>>> that won't work, >>>>>> >>>>>> ext-sip-ip=autonat:x.x.x.x >>>>>> ext-rtp-ip=autonat:x.x.x.x >>>>>> >>>>>> Set local-network-ac to rfc1918.auto >>>>>> >>>>>> On Tue, Dec 16, 2014 at 12:15 PM, Support Technique < >>>>>> support at targointernet.com> wrote: >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> 2014-12-16 12:25 GMT-05:00 Brian West : >>>>>>> >>>>>>>> On the system behind nat what do you have ext-rtp-ip, ext-sip-ip >>>>>>>> and local-network-acl set to? >>>>>>>> >>>>>>>> On Tue, Dec 16, 2014 at 10:41 AM, Frederick Pruneau < >>>>>>>> frederick at targointernet.com> wrote: >>>>>>>> >>>>>>>>> Hi guys, >>>>>>>>> >>>>>>>>> We have an issue with one freeswitch server behind nat. We have a >>>>>>>>> setup like this: >>>>>>>>> >>>>>>>>> -One master Freeswitch server >>>>>>>>> >>>>>>>>> -One freeswitch server connected to the master (Public IP) - >>>>>>>>> Server A >>>>>>>>> >>>>>>>>> -One freeswitch server connected to the master (behind nat) - >>>>>>>>> Server B >>>>>>>>> >>>>>>>>> If server A call server B, nothing happens. There is no sound. >>>>>>>>> After 30 sec, it times out. We have done a tcpdump. From server A to master >>>>>>>>> packets are ok. From Master to server B, we have seen that there is no >>>>>>>>> source and no destination ports for sip invite. >>>>>>>>> >>>>>>>>> If we use our cellphone and we call server B, there is no problem. >>>>>>>>> >>>>>>>>> I have attached the failed call pcap file and freeswitch's log >>>>>>>>> file so you can take a look at them. >>>>>>>>> >>>>>>>>> Master = Freeswitch v1.4.13 >>>>>>>>> Server A = Freeswitch v.1.4.13 >>>>>>>>> Server B = Freeswitch v.1.4.14 (Updated to latest release since we >>>>>>>>> have issues with this server) >>>>>>>>> >>>>>>>>> Thanks in advance. >>>>>>>>> >>>>>>>>> PS: The failed call is from 514-448-0773. >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> _________________________________________________________________________ >>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>> consulting at freeswitch.org >>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>> >>>>>>>>> Official FreeSWITCH Sites >>>>>>>>> http://www.freeswitch.org >>>>>>>>> http://confluence.freeswitch.org >>>>>>>>> http://www.cluecon.com >>>>>>>>> >>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>> UNSUBSCRIBE: >>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>> http://www.freeswitch.org >>>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> -- >>>>>>>> >>>>>>>> *Brian West* >>>>>>>> brian at freeswitch.org >>>>>>>> >>>>>>>> >>>>>>>> *Twitter: @FreeSWITCH , @briankwest* >>>>>>>> http://www.freeswitchbook.com >>>>>>>> http://www.freeswitchcookbook.com >>>>>>>> >>>>>>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >>>>>>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >>>>>>>> >>>>>>>> >>>>>>>> _________________________________________________________________________ >>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>> consulting at freeswitch.org >>>>>>>> http://www.freeswitchsolutions.com >>>>>>>> >>>>>>>> Official FreeSWITCH Sites >>>>>>>> http://www.freeswitch.org >>>>>>>> http://confluence.freeswitch.org >>>>>>>> http://www.cluecon.com >>>>>>>> >>>>>>>> FreeSWITCH-users mailing list >>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>> UNSUBSCRIBE: >>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>> http://www.freeswitch.org >>>>>>>> >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://confluence.freeswitch.org >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> >>>>>> *Brian West* >>>>>> brian at freeswitch.org >>>>>> >>>>>> >>>>>> *Twitter: @FreeSWITCH , @briankwest* >>>>>> http://www.freeswitchbook.com >>>>>> http://www.freeswitchcookbook.com >>>>>> >>>>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >>>>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://confluence.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> -- >>>> >>>> *Brian West* >>>> brian at freeswitch.org >>>> >>>> >>>> *Twitter: @FreeSWITCH , @briankwest* >>>> http://www.freeswitchbook.com >>>> http://www.freeswitchcookbook.com >>>> >>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> -- >> >> *Brian West* >> brian at freeswitch.org >> >> >> *Twitter: @FreeSWITCH , @briankwest* >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141219/95840d13/attachment-0001.html From grcamauer at gmail.com Fri Dec 19 17:58:34 2014 From: grcamauer at gmail.com (Guillermo Ruiz Camauer) Date: Fri, 19 Dec 2014 11:58:34 -0300 Subject: [Freeswitch-users] Hot to determine DTMF negotiation outcome In-Reply-To: <013801d01b10$3bc38380$b34a8a80$@com> References: <013801d01b10$3bc38380$b34a8a80$@com> Message-ID: Thank you for your input. I was wondering if there is any way to not allow the call to be established if a compatible DTMF exchange method cannot be negotiated. Guillermo Sent from my iPhone > On 18/12/2014, at 19:16, Bote Man wrote: > > I believe this is negotiated when the call is set up and is out of your control once it is completed. > > Check in the XML file that handles these calls under conf/sip_profiles/ to see if the following flag is set: > > > > > I don't know what the default value is, but if you uncomment that line it should accept the method negotiated with the far end. > > Hope this helps. > > > John Boteler > Bote Communications > Fort Lauderdale, FL > > > > From: Guillermo Ruiz Camauer > Sent: Thursday, 18 December, 2014 16:43 > Subject: [Freeswitch-users] Hot to determine DTMF negotiation outcome > > How can I determine if a call has successfully negotiated a DTMF transfort mechanism? Case in point: if I dial out from FS, when the call is established, I want to know if DTMF can be successfully interchanged between the dialed party and FS. My FS is setup for RCF-2833, but sometimes I get: > > 2014-12-18 16:59:10.309023 [WARNING] sofia.c:8604 IGNORE INFO DTMF(5) (This channel was not configured to use INFO DTMF!) > > which to me means that the other party is sending DTMF using SIP INFO. > > If I only offer RFC-2833 and the other party does not support this, how can I find out about it? > > Thanks, > > -- > Guillermo Ruiz Camauer > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141219/7d124e03/attachment.html From krice at freeswitch.org Fri Dec 19 18:04:32 2014 From: krice at freeswitch.org (Ken Rice) Date: Fri, 19 Dec 2014 15:04:32 +0000 Subject: [Freeswitch-users] FreeSWITCH Friday FreeForAll Reminder! Message-ID: <54943e80b8c8a_8cf69af330216ab@ip-10-183-6-59.mail> FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888 at conference.freeswitch.org or see http://ift.tt/1prwIZL for access info! -- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141219/51699e30/attachment.html From brian at freeswitch.org Fri Dec 19 20:43:16 2014 From: brian at freeswitch.org (Brian West) Date: Fri, 19 Dec 2014 11:43:16 -0600 Subject: [Freeswitch-users] fs_cli hangs In-Reply-To: References: Message-ID: Now everyone that's had fs_cli hang, start collecting data and lets pile it on this JIRA. On Fri, Dec 19, 2014 at 1:19 AM, akhil garg wrote: > > Bug FS-7097 has been created on JIRA < > https://freeswitch.org/jira/browse/FS-7097> > > > > > -- > regards, > akhil > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141219/b9577afb/attachment.html From aademattia at comcast.net Fri Dec 19 20:49:39 2014 From: aademattia at comcast.net (Andrew) Date: Fri, 19 Dec 2014 12:49:39 -0500 Subject: [Freeswitch-users] SSD Drives vs 15k drives Message-ID: <0ac001d01bb4$2dbd0550$89370ff0$@comcast.net> Hi, We are going to be buying new servers to run FS as a sip server. I know some people say use SSD drives but I wanted to ask. Performance is a big deal so I like to do anything I can to even get one more CPS. We will be running the FS with no SQL and no RTP(media bypass) and in a ram drive. Will the SSD Drives even be hit in this case? Andrew -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141219/ece1a1f2/attachment.html From anthony.minessale at gmail.com Fri Dec 19 20:51:35 2014 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 19 Dec 2014 11:51:35 -0600 Subject: [Freeswitch-users] Huge BLF/NOTIFY delay under load In-Reply-To: <1418917804.4455.9.camel@luna.madrid.commsmundi.com> References: <1418901712.10858.70.camel@luna.madrid.commsmundi.com> <1418917804.4455.9.camel@luna.madrid.commsmundi.com> Message-ID: All you can really do is use some of the features to scatter the expire times and make them longer to avoid too much load. The presence stuff is very heavy and intensive. On Thu, Dec 18, 2014 at 9:50 AM, Fran?ois Delawarde < fdelawarde at wirelessmundi.com> wrote: > > Still not sure if it's a bug and should be added as a JIRA or if it is > an expected behavior for that kind of load. > > PS: I tried tweaking "max-db-handles" but it doesn't seem to make any > difference with sqlite3 core db in tmpfs. > > Regards, > Fran?ois. > > > On Thu, 2014-12-18 at 12:21 +0100, Fran?ois Delawarde wrote: > > Hi all, > > In a decent server (6x core E5-2420 with 16GB RAM) and under a bit of > load, I see that FS sends NOTIFY up to a 5 minutes AFTER the actual event, > so BLFs are useless when it happens. > > Is it a normal/expected behavior? Code suggests that FS can queue up to > 50000 events, and presence does lots and lots of DB stuff. > > This particular site handles around 4000-5000 subscriptions and up to 2-3 > calls per second under load. I use sqlite3 presence.db in tmpfs. > > Is there anything that can be done to reduce that huge NOTIFY delay, or do > I just have to live with it? > > Thanks, > Fran?ois. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.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://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 ? @anthmfs ? @FreeSWITCH ? ? http://freeswitch.org/ ? http://cluecon.com/ ? http://twitter.com/FreeSWITCH ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ * ClueCon Weekly Development Call ? sip:888 at conference.freeswitch.org ? +19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141219/79ffea75/attachment-0001.html From cmrienzo at gmail.com Fri Dec 19 21:02:58 2014 From: cmrienzo at gmail.com (cmrienzo at gmail.com) Date: Fri, 19 Dec 2014 13:02:58 -0500 Subject: [Freeswitch-users] SSD Drives vs 15k drives In-Reply-To: <0ac001d01bb4$2dbd0550$89370ff0$@comcast.net> References: <0ac001d01bb4$2dbd0550$89370ff0$@comcast.net> Message-ID: It won't if your logging and core database goes to ram drive. > On Dec 19, 2014, at 12:49, Andrew wrote: > > Hi, > We are going to be buying new servers to run FS as a sip server. > I know some people say use SSD drives but I wanted to ask. > Performance is a big deal so I like to do anything I can to even get one more CPS. We will be running the > FS with no SQL and no RTP(media bypass) and in a ram drive. Will the SSD Drives even be hit in this case? > > Andrew > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141219/3a6b14e7/attachment.html From brian at freeswitch.org Fri Dec 19 21:05:05 2014 From: brian at freeswitch.org (Brian West) Date: Fri, 19 Dec 2014 12:05:05 -0600 Subject: [Freeswitch-users] Issue with Freeswitch Behind nat In-Reply-To: References: Message-ID: Never able to load your pastebin, it would timeout and not load, what exactly did you paste in there? On Fri, Dec 19, 2014 at 8:56 AM, Frederick Pruneau < frederick at targointernet.com> wrote: > > Did you find something? > > 2014-12-17 10:36 GMT-05:00 Frederick Pruneau > : > >> Here it is: https://pastebin.freeswitch.org/23746 >> >> It is my freeswitch log. I have followed this guide before: >> >> https://wiki.freeswitch.org/wiki/Sofia#Debugging_Sofia-SIP >> >> I have enabled this and make a call: >> >> sofia global siptrace on >> sofia loglevel all 9 >> sofia tracelevel alert >> console loglevel debug >> fsctl debug_level 10 >> >> This is what you will get in my pastebin >> >> >> 2014-12-17 9:27 GMT-05:00 Brian West : >> >>> sofia global siptrace on >>> >>> from fs_cli >>> >>> On Wed, Dec 17, 2014 at 7:44 AM, Frederick Pruneau < >>> frederick at targointernet.com> wrote: >>>> >>>> Sorry for this noob question but how can I see sip traffic? Is there a >>>> specific command to show this? Is it what we find in freeswitch.log? If so, >>>> I attached my log file in my first post. >>>> >>>> Thanks for you help >>>> >>>> 2014-12-16 15:44 GMT-05:00 Brian West : >>>> >>>>> have you looked at the signalling? What does the sip traffic show? >>>>> Please pastebin that. >>>>> >>>>> On Tue, Dec 16, 2014 at 2:37 PM, Frederick Pruneau < >>>>> frederick at targointernet.com> wrote: >>>>>> >>>>>> Same problem... >>>>>> >>>>>> 2014-12-16 13:55 GMT-05:00 Brian West : >>>>>> >>>>>>> Guessing you don't have UPNP or NAT-PMP on your network, there for >>>>>>> that won't work, >>>>>>> >>>>>>> ext-sip-ip=autonat:x.x.x.x >>>>>>> ext-rtp-ip=autonat:x.x.x.x >>>>>>> >>>>>>> Set local-network-ac to rfc1918.auto >>>>>>> >>>>>>> On Tue, Dec 16, 2014 at 12:15 PM, Support Technique < >>>>>>> support at targointernet.com> wrote: >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> 2014-12-16 12:25 GMT-05:00 Brian West : >>>>>>>> >>>>>>>>> On the system behind nat what do you have ext-rtp-ip, ext-sip-ip >>>>>>>>> and local-network-acl set to? >>>>>>>>> >>>>>>>>> On Tue, Dec 16, 2014 at 10:41 AM, Frederick Pruneau < >>>>>>>>> frederick at targointernet.com> wrote: >>>>>>>>> >>>>>>>>>> Hi guys, >>>>>>>>>> >>>>>>>>>> We have an issue with one freeswitch server behind nat. We have a >>>>>>>>>> setup like this: >>>>>>>>>> >>>>>>>>>> -One master Freeswitch server >>>>>>>>>> >>>>>>>>>> -One freeswitch server connected to the master (Public IP) - >>>>>>>>>> Server A >>>>>>>>>> >>>>>>>>>> -One freeswitch server connected to the master (behind nat) - >>>>>>>>>> Server B >>>>>>>>>> >>>>>>>>>> If server A call server B, nothing happens. There is no sound. >>>>>>>>>> After 30 sec, it times out. We have done a tcpdump. From server A to master >>>>>>>>>> packets are ok. From Master to server B, we have seen that there is no >>>>>>>>>> source and no destination ports for sip invite. >>>>>>>>>> >>>>>>>>>> If we use our cellphone and we call server B, there is no problem. >>>>>>>>>> >>>>>>>>>> I have attached the failed call pcap file and freeswitch's log >>>>>>>>>> file so you can take a look at them. >>>>>>>>>> >>>>>>>>>> Master = Freeswitch v1.4.13 >>>>>>>>>> Server A = Freeswitch v.1.4.13 >>>>>>>>>> Server B = Freeswitch v.1.4.14 (Updated to latest release since >>>>>>>>>> we have issues with this server) >>>>>>>>>> >>>>>>>>>> Thanks in advance. >>>>>>>>>> >>>>>>>>>> PS: The failed call is from 514-448-0773. >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> _________________________________________________________________________ >>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>> consulting at freeswitch.org >>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>> >>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>> http://www.freeswitch.org >>>>>>>>>> http://confluence.freeswitch.org >>>>>>>>>> http://www.cluecon.com >>>>>>>>>> >>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>> UNSUBSCRIBE: >>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>>> http://www.freeswitch.org >>>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> -- >>>>>>>>> >>>>>>>>> *Brian West* >>>>>>>>> brian at freeswitch.org >>>>>>>>> >>>>>>>>> >>>>>>>>> *Twitter: @FreeSWITCH , @briankwest* >>>>>>>>> http://www.freeswitchbook.com >>>>>>>>> http://www.freeswitchcookbook.com >>>>>>>>> >>>>>>>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >>>>>>>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >>>>>>>>> >>>>>>>>> >>>>>>>>> _________________________________________________________________________ >>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>> consulting at freeswitch.org >>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>> >>>>>>>>> Official FreeSWITCH Sites >>>>>>>>> http://www.freeswitch.org >>>>>>>>> http://confluence.freeswitch.org >>>>>>>>> http://www.cluecon.com >>>>>>>>> >>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>> UNSUBSCRIBE: >>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>> http://www.freeswitch.org >>>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> _________________________________________________________________________ >>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>> consulting at freeswitch.org >>>>>>>> http://www.freeswitchsolutions.com >>>>>>>> >>>>>>>> Official FreeSWITCH Sites >>>>>>>> http://www.freeswitch.org >>>>>>>> http://confluence.freeswitch.org >>>>>>>> http://www.cluecon.com >>>>>>>> >>>>>>>> FreeSWITCH-users mailing list >>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>> UNSUBSCRIBE: >>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>> http://www.freeswitch.org >>>>>>>> >>>>>>> >>>>>>> >>>>>>> -- >>>>>>> >>>>>>> *Brian West* >>>>>>> brian at freeswitch.org >>>>>>> >>>>>>> >>>>>>> *Twitter: @FreeSWITCH , @briankwest* >>>>>>> http://www.freeswitchbook.com >>>>>>> http://www.freeswitchcookbook.com >>>>>>> >>>>>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >>>>>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://confluence.freeswitch.org >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://confluence.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>> >>>>> >>>>> -- >>>>> >>>>> *Brian West* >>>>> brian at freeswitch.org >>>>> >>>>> >>>>> *Twitter: @FreeSWITCH , @briankwest* >>>>> http://www.freeswitchbook.com >>>>> http://www.freeswitchcookbook.com >>>>> >>>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >>>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> -- >>> >>> *Brian West* >>> brian at freeswitch.org >>> >>> >>> *Twitter: @FreeSWITCH , @briankwest* >>> http://www.freeswitchbook.com >>> http://www.freeswitchcookbook.com >>> >>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141219/8518b215/attachment-0001.html From mike at jerris.com Fri Dec 19 21:24:53 2014 From: mike at jerris.com (Michael Jerris) Date: Fri, 19 Dec 2014 13:24:53 -0500 Subject: [Freeswitch-users] webrtc chrome->sip bad media jssip error In-Reply-To: References: Message-ID: <933B5FCA-CBEF-4F65-ACF4-D27994EA9EEE@jerris.com> Double check with latest master. Also try sipjs, jssip looked like it was bitrot lately > On Dec 19, 2014, at 3:48 AM, rentmycoder rentmycoder wrote: > > Hi, > > Maybe it's a know issue, please direct me to the right direction. > Setup: FS 1.4.latest, -nonat, Debian64 7.8, http://tryit.jssip.net/ web sip client, > > Issue: > Using Firefox, calls are working great, but with Chrome 39.0.2171.95m calling from browser to SIPendpint fails. > The sip client rings, but after answer, the jssip client shows bad media description error.... > > Ring SDP: > v=0 > o=FreeSWITCH 1418885182 1418885183 IN IP4 192.168.101.43 > s=FreeSWITCH > c=IN IP4 192.168.101.43 > t=0 0 > a=msid-semantic: WMS PLEThV3VtmQWGIk4ICYTkWiv3AtV43mJ > m=audio 27200 RTP/SAVPF 0 126 106 > a=rtpmap:0 PCMU/8000 > a=rtpmap:126 telephone-event/8000 > a=rtpmap:106 CN/8000 > a=ptime:20 > a=sendrecv > a=fingerprint:sha-256 E2:56:BC:E3:1E:CA:AD:55:04:E5:94:8F:D6:AD:1B:CA:E0:B8:90:A9:62:35:95:F0:F1:C0:D7:A7:92:35:D9:C8 > a=rtcp-mux > a=rtcp:27200 IN IP4 192.168.101.43 > a=ssrc:1443731118 cname:FQsNBmyVzeI5MT0k > a=ssrc:1443731118 msid:PLEThV3VtmQWGIk4ICYTkWiv3AtV43mJ a0 > a=ssrc:1443731118 mslabel:PLEThV3VtmQWGIk4ICYTkWiv3AtV43mJ > a=ssrc:1443731118 label:PLEThV3VtmQWGIk4ICYTkWiv3AtV43mJa0 > a=ice-ufrag:lO4wBcSvCchx5t6B > a=ice-pwd:9peBBCn0GWWiFMtRrndSqVeT > a=candidate:9761529334 1 udp 659136 192.168.101.43 27200 typ host generation 0 > > > 2014-12-18 09:19:46.047277 [DEBUG] sofia.c:6624 Remote SDP: > v=0 > o=1001 8000 8000 IN IP4 192.168.101.39 > s=SIP Call > c=IN IP4 192.168.101.39 > t=0 0 > m=audio 5004 RTP/AVP 0 13 > a=sendrecv > a=rtpmap:0 PCMU/8000 > a=ptime:20 > > > Remote SDP: > 2014-12-18 09:19:46.067273 [DEBUG] mod_sofia.c:780 Local SDP sofia/internal/1002 at 192.168.101.43 : > v=0 > o=FreeSWITCH 1418885182 1418885184 IN IP4 192.168.101.43 > s=FreeSWITCH > c=IN IP4 192.168.101.43 > t=0 0 > a=msid-semantic: WMS PLEThV3VtmQWGIk4ICYTkWiv3AtV43mJ > m=audio 27200 RTP/SAVPF 0 126 106 > a=rtpmap:0 PCMU/8000 > a=rtpmap:126 telephone-event/8000 > a=rtpmap:106 CN/8000 > a=ptime:20 > a=sendrecv > a=fingerprint:sha-256 E2:56:BC:E3:1E:CA:AD:55:04:E5:94:8F:D6:AD:1B:CA:E0:B8:90:A9:62:35:95:F0:F1:C0:D7:A7:92:35:D9:C8 > a=rtcp-mux > a=rtcp:27200 IN IP4 192.168.101.43 > a=ssrc:1443731118 cname:FQsNBmyVzeI5MT0k > a=ssrc:1443731118 msid:PLEThV3VtmQWGIk4ICYTkWiv3AtV43mJ a0 > a=ssrc:1443731118 mslabel:PLEThV3VtmQWGIk4ICYTkWiv3AtV43mJ > a=ssrc:1443731118 label:PLEThV3VtmQWGIk4ICYTkWiv3AtV43mJa0 > a=ice-ufrag:lO4wBcSvCchx5t6B > a=ice-pwd:9peBBCn0GWWiFMtRrndSqVeT > a=candidate:7802155548 1 udp 659136 192.168.101.43 27200 typ host generation 0 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141219/a1d38355/attachment.html From frederick at targointernet.com Fri Dec 19 23:31:26 2014 From: frederick at targointernet.com (Frederick Pruneau) Date: Fri, 19 Dec 2014 15:31:26 -0500 Subject: [Freeswitch-users] Issue with Freeswitch Behind nat In-Reply-To: References: Message-ID: Sorry, I pasted all my log file. I have a new pastebin: https://pastebin.freeswitch.org/23771 I tested it and I can open it. 2014-12-19 13:05 GMT-05:00 Brian West : > Never able to load your pastebin, it would timeout and not load, what > exactly did you paste in there? > > On Fri, Dec 19, 2014 at 8:56 AM, Frederick Pruneau < > frederick at targointernet.com> wrote: >> >> Did you find something? >> >> 2014-12-17 10:36 GMT-05:00 Frederick Pruneau > >: >> >>> Here it is: https://pastebin.freeswitch.org/23746 >>> >>> It is my freeswitch log. I have followed this guide before: >>> >>> https://wiki.freeswitch.org/wiki/Sofia#Debugging_Sofia-SIP >>> >>> I have enabled this and make a call: >>> >>> sofia global siptrace on >>> sofia loglevel all 9 >>> sofia tracelevel alert >>> console loglevel debug >>> fsctl debug_level 10 >>> >>> This is what you will get in my pastebin >>> >>> >>> 2014-12-17 9:27 GMT-05:00 Brian West : >>> >>>> sofia global siptrace on >>>> >>>> from fs_cli >>>> >>>> On Wed, Dec 17, 2014 at 7:44 AM, Frederick Pruneau < >>>> frederick at targointernet.com> wrote: >>>>> >>>>> Sorry for this noob question but how can I see sip traffic? Is there a >>>>> specific command to show this? Is it what we find in freeswitch.log? If so, >>>>> I attached my log file in my first post. >>>>> >>>>> Thanks for you help >>>>> >>>>> 2014-12-16 15:44 GMT-05:00 Brian West : >>>>> >>>>>> have you looked at the signalling? What does the sip traffic show? >>>>>> Please pastebin that. >>>>>> >>>>>> On Tue, Dec 16, 2014 at 2:37 PM, Frederick Pruneau < >>>>>> frederick at targointernet.com> wrote: >>>>>>> >>>>>>> Same problem... >>>>>>> >>>>>>> 2014-12-16 13:55 GMT-05:00 Brian West : >>>>>>> >>>>>>>> Guessing you don't have UPNP or NAT-PMP on your network, there for >>>>>>>> that won't work, >>>>>>>> >>>>>>>> ext-sip-ip=autonat:x.x.x.x >>>>>>>> ext-rtp-ip=autonat:x.x.x.x >>>>>>>> >>>>>>>> Set local-network-ac to rfc1918.auto >>>>>>>> >>>>>>>> On Tue, Dec 16, 2014 at 12:15 PM, Support Technique < >>>>>>>> support at targointernet.com> wrote: >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> 2014-12-16 12:25 GMT-05:00 Brian West : >>>>>>>>> >>>>>>>>>> On the system behind nat what do you have ext-rtp-ip, ext-sip-ip >>>>>>>>>> and local-network-acl set to? >>>>>>>>>> >>>>>>>>>> On Tue, Dec 16, 2014 at 10:41 AM, Frederick Pruneau < >>>>>>>>>> frederick at targointernet.com> wrote: >>>>>>>>>> >>>>>>>>>>> Hi guys, >>>>>>>>>>> >>>>>>>>>>> We have an issue with one freeswitch server behind nat. We have >>>>>>>>>>> a setup like this: >>>>>>>>>>> >>>>>>>>>>> -One master Freeswitch server >>>>>>>>>>> >>>>>>>>>>> -One freeswitch server connected to the master (Public IP) - >>>>>>>>>>> Server A >>>>>>>>>>> >>>>>>>>>>> -One freeswitch server connected to the master (behind nat) - >>>>>>>>>>> Server B >>>>>>>>>>> >>>>>>>>>>> If server A call server B, nothing happens. There is no sound. >>>>>>>>>>> After 30 sec, it times out. We have done a tcpdump. From server A to master >>>>>>>>>>> packets are ok. From Master to server B, we have seen that there is no >>>>>>>>>>> source and no destination ports for sip invite. >>>>>>>>>>> >>>>>>>>>>> If we use our cellphone and we call server B, there is no >>>>>>>>>>> problem. >>>>>>>>>>> >>>>>>>>>>> I have attached the failed call pcap file and freeswitch's log >>>>>>>>>>> file so you can take a look at them. >>>>>>>>>>> >>>>>>>>>>> Master = Freeswitch v1.4.13 >>>>>>>>>>> Server A = Freeswitch v.1.4.13 >>>>>>>>>>> Server B = Freeswitch v.1.4.14 (Updated to latest release since >>>>>>>>>>> we have issues with this server) >>>>>>>>>>> >>>>>>>>>>> Thanks in advance. >>>>>>>>>>> >>>>>>>>>>> PS: The failed call is from 514-448-0773. >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> _________________________________________________________________________ >>>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>>> consulting at freeswitch.org >>>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>>> >>>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>> http://confluence.freeswitch.org >>>>>>>>>>> http://www.cluecon.com >>>>>>>>>>> >>>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>>> UNSUBSCRIBE: >>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> -- >>>>>>>>>> >>>>>>>>>> *Brian West* >>>>>>>>>> brian at freeswitch.org >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> *Twitter: @FreeSWITCH , @briankwest* >>>>>>>>>> http://www.freeswitchbook.com >>>>>>>>>> http://www.freeswitchcookbook.com >>>>>>>>>> >>>>>>>>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >>>>>>>>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> _________________________________________________________________________ >>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>> consulting at freeswitch.org >>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>> >>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>> http://www.freeswitch.org >>>>>>>>>> http://confluence.freeswitch.org >>>>>>>>>> http://www.cluecon.com >>>>>>>>>> >>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>> UNSUBSCRIBE: >>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>>> http://www.freeswitch.org >>>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> _________________________________________________________________________ >>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>> consulting at freeswitch.org >>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>> >>>>>>>>> Official FreeSWITCH Sites >>>>>>>>> http://www.freeswitch.org >>>>>>>>> http://confluence.freeswitch.org >>>>>>>>> http://www.cluecon.com >>>>>>>>> >>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>> UNSUBSCRIBE: >>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>> http://www.freeswitch.org >>>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> -- >>>>>>>> >>>>>>>> *Brian West* >>>>>>>> brian at freeswitch.org >>>>>>>> >>>>>>>> >>>>>>>> *Twitter: @FreeSWITCH , @briankwest* >>>>>>>> http://www.freeswitchbook.com >>>>>>>> http://www.freeswitchcookbook.com >>>>>>>> >>>>>>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >>>>>>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >>>>>>>> >>>>>>>> >>>>>>>> _________________________________________________________________________ >>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>> consulting at freeswitch.org >>>>>>>> http://www.freeswitchsolutions.com >>>>>>>> >>>>>>>> Official FreeSWITCH Sites >>>>>>>> http://www.freeswitch.org >>>>>>>> http://confluence.freeswitch.org >>>>>>>> http://www.cluecon.com >>>>>>>> >>>>>>>> FreeSWITCH-users mailing list >>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>> UNSUBSCRIBE: >>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>> http://www.freeswitch.org >>>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://confluence.freeswitch.org >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> >>>>>> *Brian West* >>>>>> brian at freeswitch.org >>>>>> >>>>>> >>>>>> *Twitter: @FreeSWITCH , @briankwest* >>>>>> http://www.freeswitchbook.com >>>>>> http://www.freeswitchcookbook.com >>>>>> >>>>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >>>>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://confluence.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>> >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> -- >>>> >>>> *Brian West* >>>> brian at freeswitch.org >>>> >>>> >>>> *Twitter: @FreeSWITCH , @briankwest* >>>> http://www.freeswitchbook.com >>>> http://www.freeswitchcookbook.com >>>> >>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > -- > > *Brian West* > brian at freeswitch.org > > > *Twitter: @FreeSWITCH , @briankwest* > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20141219/2a7e3d70/attachment-0001.html From alipey at gmail.com Fri Dec 19 23:32:49 2014 From: alipey at gmail.com (Ali Pey) Date: Fri, 19 Dec 2014 15:32:49 -0500 Subject: [Freeswitch-users] FAX tone is not detected - sometimes Message-ID: Hello, I have this line in my dial plan: However, the fax is not detected at times. There is one particular fax machine that freeswitch never detects the fax tone. When I listen to the pcap, I do hear the fax tone. Is there anything I can do to improve the fax tone detection? Thanks, Ali Pey -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141219/64f377e7/attachment.html From grcamauer at gmail.com Fri Dec 19 23:53:11 2014 From: grcamauer at gmail.com (Guillermo Ruiz Camauer) Date: Fri, 19 Dec 2014 17:53:11 -0300 Subject: [Freeswitch-users] SSD Drives vs 15k drives In-Reply-To: References: <0ac001d01bb4$2dbd0550$89370ff0$@comcast.net> Message-ID: Putting the CDR on a RAM drive might not be a good idea if you are going to be billing customers with this system. Best to write the CDRs to a database on another machine. Regards, Guillermo On Fri, Dec 19, 2014 at 3:02 PM, wrote: > > It won't if your logging and core database goes to ram drive. > > > On Dec 19, 2014, at 12:49, Andrew wrote: > > Hi, > > We are going to be buying new servers to run FS as a sip server. > > I know some people say use SSD drives but I wanted to ask. > > Performance is a big deal so I like to do anything I can to even get one > more CPS. We will be running the > > FS with no SQL and no RTP(media bypass) and in a ram drive. Will the SSD > Drives even be hit in this case? > > > > Andrew > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Guillermo Ruiz Camauer -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141219/9a8bc83f/attachment.html From nbhatti at gmail.com Sat Dec 20 00:46:43 2014 From: nbhatti at gmail.com (Muhammad Naseer Bhatti) Date: Sat, 20 Dec 2014 00:46:43 +0300 Subject: [Freeswitch-users] FreeSWITCH using outbound proxy but to bypass media Message-ID: I am trying to use an outbound SIP proxy for FreeSWITCH by adding fs_path to the bridge. It?s working fine and I am able to send calls using the proxy. On the proxy (which is also FreeSWITCH) I am receiving the call in a lua script, create a new session with the destination gateway and bridge two sessions (old, incoming and the newly created)? The idea is to have multiple FreeSWITCH servers (all public IP address) originating calls while all calls going to the carrier via the proxy. The problem is that all media is also passing through the proxy. Is there a way I can only send signaling through proxy and media to flow directly between the endpoints? I have tried bypass_media and?bypass_media_after_bridge but it?s not working. Is this even doable? Thanks. --? Muhammad Naseer Bhatti -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141220/71d2467f/attachment.html From support at directvoip.co.uk Sat Dec 20 00:58:21 2014 From: support at directvoip.co.uk (Customer Support) Date: Fri, 19 Dec 2014 21:58:21 +0000 Subject: [Freeswitch-users] SSD Drives vs 15k drives In-Reply-To: References: <0ac001d01bb4$2dbd0550$89370ff0$@comcast.net> Message-ID: <54949F7D.70305@directvoip.co.uk> You may also want to try Freeswitch on either Linux or FreeBSD using ZFS, you should get great IOPS using SSD for log and cache combined with regular HDD. On 19/12/2014 20:53, Guillermo Ruiz Camauer wrote: > Putting the CDR on a RAM drive might not be a good idea if you are > going to be billing customers with this system. Best to write the CDRs > to a database on another machine. > > Regards, > > Guillermo > > On Fri, Dec 19, 2014 at 3:02 PM, > wrote: > > It won't if your logging and core database goes to ram drive. > > > On Dec 19, 2014, at 12:49, Andrew > wrote: > >> Hi, >> >> We are going to be buying new servers to run FS as a sip server. >> >> I know some people say use SSD drives but I wanted to ask. >> >> Performance is a big deal so I like to do anything I can to even >> get one more CPS. We will be running the >> >> FS with no SQL and no RTP(media bypass) and in a ram drive. Will >> the SSD Drives even be hit in this case? >> >> Andrew >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -- > Guillermo Ruiz Camauer > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141219/76e78295/attachment-0001.html From brian at freeswitch.org Sat Dec 20 01:01:13 2014 From: brian at freeswitch.org (Brian West) Date: Fri, 19 Dec 2014 16:01:13 -0600 Subject: [Freeswitch-users] Issue with Freeswitch Behind nat In-Reply-To: References: Message-ID: Describe your topology a little bit. On Fri, Dec 19, 2014 at 2:31 PM, Frederick Pruneau < frederick at targointernet.com> wrote: > > Sorry, I pasted all my log file. I have a new pastebin: > > https://pastebin.freeswitch.org/23771 > > I tested it and I can open it. > > 2014-12-19 13:05 GMT-05:00 Brian West : > > Never able to load your pastebin, it would timeout and not load, what >> exactly did you paste in there? >> >> On Fri, Dec 19, 2014 at 8:56 AM, Frederick Pruneau < >> frederick at targointernet.com> wrote: >>> >>> Did you find something? >>> >>> 2014-12-17 10:36 GMT-05:00 Frederick Pruneau < >>> frederick at targointernet.com>: >>> >>>> Here it is: https://pastebin.freeswitch.org/23746 >>>> >>>> It is my freeswitch log. I have followed this guide before: >>>> >>>> https://wiki.freeswitch.org/wiki/Sofia#Debugging_Sofia-SIP >>>> >>>> I have enabled this and make a call: >>>> >>>> sofia global siptrace on >>>> sofia loglevel all 9 >>>> sofia tracelevel alert >>>> console loglevel debug >>>> fsctl debug_level 10 >>>> >>>> This is what you will get in my pastebin >>>> >>>> >>>> 2014-12-17 9:27 GMT-05:00 Brian West : >>>> >>>>> sofia global siptrace on >>>>> >>>>> from fs_cli >>>>> >>>>> On Wed, Dec 17, 2014 at 7:44 AM, Frederick Pruneau < >>>>> frederick at targointernet.com> wrote: >>>>>> >>>>>> Sorry for this noob question but how can I see sip traffic? Is there >>>>>> a specific command to show this? Is it what we find in freeswitch.log? If >>>>>> so, I attached my log file in my first post. >>>>>> >>>>>> Thanks for you help >>>>>> >>>>>> 2014-12-16 15:44 GMT-05:00 Brian West : >>>>>> >>>>>>> have you looked at the signalling? What does the sip traffic show? >>>>>>> Please pastebin that. >>>>>>> >>>>>>> On Tue, Dec 16, 2014 at 2:37 PM, Frederick Pruneau < >>>>>>> frederick at targointernet.com> wrote: >>>>>>>> >>>>>>>> Same problem... >>>>>>>> >>>>>>>> 2014-12-16 13:55 GMT-05:00 Brian West : >>>>>>>> >>>>>>>>> Guessing you don't have UPNP or NAT-PMP on your network, there for >>>>>>>>> that won't work, >>>>>>>>> >>>>>>>>> ext-sip-ip=autonat:x.x.x.x >>>>>>>>> ext-rtp-ip=autonat:x.x.x.x >>>>>>>>> >>>>>>>>> Set local-network-ac to rfc1918.auto >>>>>>>>> >>>>>>>>> On Tue, Dec 16, 2014 at 12:15 PM, Support Technique < >>>>>>>>> support at targointernet.com> wrote: >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> 2014-12-16 12:25 GMT-05:00 Brian West : >>>>>>>>>> >>>>>>>>>>> On the system behind nat what do you have ext-rtp-ip, ext-sip-ip >>>>>>>>>>> and local-network-acl set to? >>>>>>>>>>> >>>>>>>>>>> On Tue, Dec 16, 2014 at 10:41 AM, Frederick Pruneau < >>>>>>>>>>> frederick at targointernet.com> wrote: >>>>>>>>>>> >>>>>>>>>>>> Hi guys, >>>>>>>>>>>> >>>>>>>>>>>> We have an issue with one freeswitch server behind nat. We have >>>>>>>>>>>> a setup like this: >>>>>>>>>>>> >>>>>>>>>>>> -One master Freeswitch server >>>>>>>>>>>> >>>>>>>>>>>> -One freeswitch server connected to the master (Public IP) - >>>>>>>>>>>> Server A >>>>>>>>>>>> >>>>>>>>>>>> -One freeswitch server connected to the master (behind nat) - >>>>>>>>>>>> Server B >>>>>>>>>>>> >>>>>>>>>>>> If server A call server B, nothing happens. There is no sound. >>>>>>>>>>>> After 30 sec, it times out. We have done a tcpdump. From server A to master >>>>>>>>>>>> packets are ok. From Master to server B, we have seen that there is no >>>>>>>>>>>> source and no destination ports for sip invite. >>>>>>>>>>>> >>>>>>>>>>>> If we use our cellphone and we call server B, there is no >>>>>>>>>>>> problem. >>>>>>>>>>>> >>>>>>>>>>>> I have attached the failed call pcap file and freeswitch's log >>>>>>>>>>>> file so you can take a look at them. >>>>>>>>>>>> >>>>>>>>>>>> Master = Freeswitch v1.4.13 >>>>>>>>>>>> Server A = Freeswitch v.1.4.13 >>>>>>>>>>>> Server B = Freeswitch v.1.4.14 (Updated to latest release since >>>>>>>>>>>> we have issues with this server) >>>>>>>>>>>> >>>>>>>>>>>> Thanks in advance. >>>>>>>>>>>> >>>>>>>>>>>> PS: The failed call is from 514-448-0773. >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> _________________________________________________________________________ >>>>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>>>> consulting at freeswitch.org >>>>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>>>> >>>>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>> http://confluence.freeswitch.org >>>>>>>>>>>> http://www.cluecon.com >>>>>>>>>>>> >>>>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>>>> UNSUBSCRIBE: >>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> -- >>>>>>>>>>> >>>>>>>>>>> *Brian West* >>>>>>>>>>> brian at freeswitch.org >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> *Twitter: @FreeSWITCH , @briankwest* >>>>>>>>>>> http://www.freeswitchbook.com >>>>>>>>>>> http://www.freeswitchcookbook.com >>>>>>>>>>> >>>>>>>>>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >>>>>>>>>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> _________________________________________________________________________ >>>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>>> consulting at freeswitch.org >>>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>>> >>>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>> http://confluence.freeswitch.org >>>>>>>>>>> http://www.cluecon.com >>>>>>>>>>> >>>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>>> UNSUBSCRIBE: >>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> _________________________________________________________________________ >>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>> consulting at freeswitch.org >>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>> >>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>> http://www.freeswitch.org >>>>>>>>>> http://confluence.freeswitch.org >>>>>>>>>> http://www.cluecon.com >>>>>>>>>> >>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>> UNSUBSCRIBE: >>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>>> http://www.freeswitch.org >>>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> -- >>>>>>>>> >>>>>>>>> *Brian West* >>>>>>>>> brian at freeswitch.org >>>>>>>>> >>>>>>>>> >>>>>>>>> *Twitter: @FreeSWITCH , @briankwest* >>>>>>>>> http://www.freeswitchbook.com >>>>>>>>> http://www.freeswitchcookbook.com >>>>>>>>> >>>>>>>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >>>>>>>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >>>>>>>>> >>>>>>>>> >>>>>>>>> _________________________________________________________________________ >>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>> consulting at freeswitch.org >>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>> >>>>>>>>> Official FreeSWITCH Sites >>>>>>>>> http://www.freeswitch.org >>>>>>>>> http://confluence.freeswitch.org >>>>>>>>> http://www.cluecon.com >>>>>>>>> >>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>> UNSUBSCRIBE: >>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>> http://www.freeswitch.org >>>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> _________________________________________________________________________ >>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>> consulting at freeswitch.org >>>>>>>> http://www.freeswitchsolutions.com >>>>>>>> >>>>>>>> Official FreeSWITCH Sites >>>>>>>> http://www.freeswitch.org >>>>>>>> http://confluence.freeswitch.org >>>>>>>> http://www.cluecon.com >>>>>>>> >>>>>>>> FreeSWITCH-users mailing list >>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>> UNSUBSCRIBE: >>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>> http://www.freeswitch.org >>>>>>>> >>>>>>> >>>>>>> >>>>>>> -- >>>>>>> >>>>>>> *Brian West* >>>>>>> brian at freeswitch.org >>>>>>> >>>>>>> >>>>>>> *Twitter: @FreeSWITCH , @briankwest* >>>>>>> http://www.freeswitchbook.com >>>>>>> http://www.freeswitchcookbook.com >>>>>>> >>>>>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >>>>>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://confluence.freeswitch.org >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://confluence.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>> >>>>> >>>>> -- >>>>> >>>>> *Brian West* >>>>> brian at freeswitch.org >>>>> >>>>> >>>>> *Twitter: @FreeSWITCH , @briankwest* >>>>> http://www.freeswitchbook.com >>>>> http://www.freeswitchcookbook.com >>>>> >>>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >>>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> -- >> >> *Brian West* >> brian at freeswitch.org >> >> >> *Twitter: @FreeSWITCH , @briankwest* >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 <%28514%29%20448-0773> * > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141219/8ff6af8d/attachment-0001.html From Kyle.Haefner at colostate.edu Sat Dec 20 01:29:46 2014 From: Kyle.Haefner at colostate.edu (Kyle Haefner) Date: Fri, 19 Dec 2014 15:29:46 -0700 Subject: [Freeswitch-users] Moving Voicemails In-Reply-To: References: Message-ID: Hi All, I know this treads the line of spam a bit. But we are looking to hire someone who likes open source and is familiar with VoIP and SIP. Colorado State University VoIP Integrator Colorado State University (CSU) seeks applications for a VoIP Integrator who will be responsible for the integration and management of voice communications and unified messaging systems. General Duties and Responsibilities This position will be responsible for: ? Managing the unified communications application, sipXecs ? Performing Linux system administration on underlying servers ? Working closely with Networking and Telecom staff to coordinate the transition to VoIP services ? Developing technical and end user documentation ? Being on call in a rotation schedule For the full position announcement, including required and preferred qualifications as well as instructions for applying, visit: http://www.acns.colostate.edu/jobs. For full consideration, complete applications must be received by 11:59pm MST, January 12, 2015. Colorado State University is an EO/EA/AA employer. Colorado State University conducts background checks on all final candidates. Thanks! Kyle On Mon, Nov 17, 2014 at 1:35 AM, Joel White wrote: > > I initially configured my FreeSWITCH with an IP address, but now for > scalability and high availability I am changing to a domain name. I > successfully changed the system over, but was wondering how do I move the > voicemails to the new domain folder under voicemail? > > The obvious answer is to move the files, I already did that and restarted > the system. The existing voicemails and greetings are not visible to > FreeSWITCH when I call voicemail. > > How do I transfer them over? > > Thank you in advance for any light you may shed on this predicament > -- Kyle Haefner, M.S. Communication Systems Programmer Colorado State University Fort Collins, CO Phone: 970-491-1012 Email: kyle.haefner at colostate.edu -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141219/f1e78c2c/attachment.html From Kyle.Haefner at colostate.edu Sat Dec 20 01:38:06 2014 From: Kyle.Haefner at colostate.edu (Kyle Haefner) Date: Fri, 19 Dec 2014 15:38:06 -0700 Subject: [Freeswitch-users] Job Posting Message-ID: Hi All, I know this treads the line of spam a bit. But we are looking to hire someone who likes open source and is familiar with VoIP and SIP. Colorado State University VoIP Integrator Colorado State University (CSU) seeks applications for a VoIP Integrator who will be responsible for the integration and management of voice communications and unified messaging systems. General Duties and Responsibilities This position will be responsible for: ? Managing the unified communications application, sipXecs ? Performing Linux system administration on underlying servers ? Working closely with Networking and Telecom staff to coordinate the transition to VoIP services ? Developing technical and end user documentation ? Being on call in a rotation schedule For the full position announcement, including required and preferred qualifications as well as instructions for applying, visit: http://www.acns.colostate.edu/jobs. For full consideration, complete applications must be received by 11:59pm MST, January 12, 2015. Colorado State University is an EO/EA/AA employer. Colorado State University conducts background checks on all final candidates. Thanks! Kyle -- Kyle Haefner, M.S. Communication Systems Programmer Colorado State University Fort Collins, CO Phone: 970-491-1012 Email: kyle.haefner at colostate.edu -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141219/baf641dc/attachment.html From mike at jerris.com Sat Dec 20 03:17:31 2014 From: mike at jerris.com (Michael Jerris) Date: Fri, 19 Dec 2014 19:17:31 -0500 Subject: [Freeswitch-users] FreeSWITCH using outbound proxy but to bypass media In-Reply-To: References: Message-ID: bypass_media is the exact way to do this. It would be on the "proxy" box. On Friday, December 19, 2014, Muhammad Naseer Bhatti wrote: > I am trying to use an outbound SIP proxy for FreeSWITCH by adding fs_path > to the bridge. It?s working fine and I am able to send calls using the > proxy. On the proxy (which is also FreeSWITCH) I am receiving the call in a > lua script, create a new session with the destination gateway and bridge > two sessions (old, incoming and the newly created) > > The idea is to have multiple FreeSWITCH servers (all public IP address) > originating calls while all calls going to the carrier via the proxy. The > problem is that all media is also passing through the proxy. Is there a way > I can only send signaling through proxy and media to flow directly between > the endpoints? I have tried bypass_media and bypass_media_after_bridge but > it?s not working. Is this even doable? > > Thanks. > -- > Muhammad Naseer Bhatti > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141219/3debaced/attachment.html From nbhatti at gmail.com Sat Dec 20 03:25:04 2014 From: nbhatti at gmail.com (Muhammad Naseer Bhatti) Date: Sat, 20 Dec 2014 03:25:04 +0300 Subject: [Freeswitch-users] FreeSWITCH using outbound proxy but to bypass media In-Reply-To: References: Message-ID: <28AA5926-009F-4FCE-AE91-3F4AE8AB0615@gmail.com> Inbound-bypass-media is enabled in the profile and also enabled just before the bridge as a session/channel variable. Can't think of a reason why media is still being passed through. :s Sent from my iPad > On Dec 20, 2014, at 03:17, Michael Jerris wrote: > > bypass_media is the exact way to do this. It would be on the "proxy" box. > >> On Friday, December 19, 2014, Muhammad Naseer Bhatti wrote: >> I am trying to use an outbound SIP proxy for FreeSWITCH by adding fs_path to the bridge. It?s working fine and I am able to send calls using the proxy. On the proxy (which is also FreeSWITCH) I am receiving the call in a lua script, create a new session with the destination gateway and bridge two sessions (old, incoming and the newly created) >> >> The idea is to have multiple FreeSWITCH servers (all public IP address) originating calls while all calls going to the carrier via the proxy. The problem is that all media is also passing through the proxy. Is there a way I can only send signaling through proxy and media to flow directly between the endpoints? I have tried bypass_media and bypass_media_after_bridge but it?s not working. Is this even doable? >> >> Thanks. >> -- >> Muhammad Naseer Bhatti > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141220/93a470a8/attachment.html From dujinfang at gmail.com Sat Dec 20 04:15:08 2014 From: dujinfang at gmail.com (Seven Du) Date: Sat, 20 Dec 2014 09:15:08 +0800 Subject: [Freeswitch-users] How to play a stream other than mp3 with mod_shout. In-Reply-To: References: Message-ID: mod_vlc is the best answer to the original question. On Wednesday, December 17, 2014 at 10:26 PM, Brian West wrote: > Don't think mod_rtmp can actually do this, from loading it: > > 2014-12-17 08:25:39.983446 [NOTICE] switch_loadable_module.c:149 Adding Endpoint 'rtmp' > 2014-12-17 08:25:39.983446 [NOTICE] switch_loadable_module.c:315 Adding API Function 'rtmp' > 2014-12-17 08:25:39.983446 [NOTICE] switch_loadable_module.c:315 Adding API Function 'rtmp_contact' > > This is all that gets registered. > > On Sun, Dec 14, 2014 at 1:37 PM, Danny Gershman wrote: > > Also mod_rtmp lets you play from an FMS server. > > > > > > On Friday, December 12, 2014, Aqs Younas wrote: > > > Hi, All > > > > > > How can i play a live stream other than mp3 with mod_shout or any module.? Is there any way to buffer the stream before playing it with mod_shout. > > > > > > > > > Currently i have a list of streams and when i play them with mod_shout some work fine but others give (time out) error. > > > > > > How can i play mostly stream in freeswitch? > > > > > > Thanks > > > > _________________________________________________________________________ > > 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://confluence.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 > > > -- > Brian West > brian at freeswitch.org (mailto:brian at freeswitch.org) > > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378) > iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest > _________________________________________________________________________ > 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://confluence.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/20141220/7b5cfd39/attachment-0001.html From markus.klenk at googlemail.com Sat Dec 20 12:57:49 2014 From: markus.klenk at googlemail.com (Paul Klenk) Date: Sat, 20 Dec 2014 10:57:49 +0100 Subject: [Freeswitch-users] display incoming line Message-ID: Hi, I use several incoming SIP lines connected to freeswitch. How yould I have the incoming line be displayed on the SIP-telephones (Grandstream) besides Caller-ID? e.g.: ------------------ Tom's-Line +17778889999 ----------------- From jobindcruz at gmail.com Sat Dec 20 13:35:19 2014 From: jobindcruz at gmail.com (jobin dcruz) Date: Sat, 20 Dec 2014 16:05:19 +0530 Subject: [Freeswitch-users] Call Conference Time out issue Message-ID: Hi, I had set call conference hang up time,but it is working after 4 seconds as per attendees count. This is dial plan example Ex: i had set one min(60 sec) hangup time,but call conference ended with 64 sec -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141220/f7f9a008/attachment.html From aqsyounas at gmail.com Sat Dec 20 15:02:53 2014 From: aqsyounas at gmail.com (Aqs Younas) Date: Sat, 20 Dec 2014 17:02:53 +0500 Subject: [Freeswitch-users] How to play a stream other than mp3 with mod_shout. In-Reply-To: References: Message-ID: Hi, Seven When i try to play a stream it doesn't give any audio. Also session created when i try to play a stream with mod_vlc does not terminate even with "hupall". [0x7f69b00025d8] main input debug: Creating an input for ' http://s9.voscast.com:7584' [0x7f69b00025d8] main input debug: using timeshift granularity of 50 MiB, in path '/tmp' [0x7f69b00025d8] main input debug: `http://s9.voscast.com:7584' gives access `http' demux `' path `s9.voscast.com:7584' [0x7f69b00025d8] main input debug: creating demux: access='http' demux='' location='s9.voscast.com:7584' file='(null)' [0x26cbd48] main demux debug: looking for access_demux module matching "http": 11 candidates [0x26cbd48] main demux debug: no access_demux modules matched [0x7f69b00025d8] main input debug: creating access 'http' location=' s9.voscast.com:7584' , path='(null)' [0x26cbc68] main access debug: looking for access module matching "http": 19 candidates [0x26cbc68] access_http access debug: querying proxy for http://s9.voscast.com:7584 This is what i see in my logs. Can you provide me with an example or guide me that i am doing wrong? Thanks On 20 December 2014 at 06:15, Seven Du wrote: > mod_vlc is the best answer to the original question. > > On Wednesday, December 17, 2014 at 10:26 PM, Brian West wrote: > > Don't think mod_rtmp can actually do this, from loading it: > > 2014-12-17 08:25:39.983446 [NOTICE] switch_loadable_module.c:149 Adding > Endpoint 'rtmp' > > 2014-12-17 08:25:39.983446 [NOTICE] switch_loadable_module.c:315 Adding > API Function 'rtmp' > > 2014-12-17 08:25:39.983446 [NOTICE] switch_loadable_module.c:315 Adding > API Function 'rtmp_contact' > > > This is all that gets registered. > > On Sun, Dec 14, 2014 at 1:37 PM, Danny Gershman > wrote: > > Also mod_rtmp lets you play from an FMS server. > > > On Friday, December 12, 2014, Aqs Younas wrote: > > Hi, All > > How can i play a live stream other than mp3 with mod_shout or any module.? > Is there any way to buffer the stream before playing it with mod_shout. > > > Currently i have a list of streams and when i play them with mod_shout > some work fine but others give (time out) error. > > How can i play mostly stream in freeswitch? > > Thanks > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -- > > *Brian West* > brian at freeswitch.org > > > *Twitter: @FreeSWITCH , @briankwest* > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141220/025393ab/attachment.html From avi at avimarcus.net Sat Dec 20 19:49:31 2014 From: avi at avimarcus.net (Avi Marcus) Date: Sat, 20 Dec 2014 16:49:31 +0000 Subject: [Freeswitch-users] Job Posting In-Reply-To: References: Message-ID: <0000014a689a9f94-b5049e32-ffac-4f9d-8ad3-202c03be2511-000000@email.amazonses.com> Hi Kyle, freeswitch-biz is where you want to post. -Avi On Sat, Dec 20, 2014 at 12:38 AM, Kyle Haefner wrote: > Hi All, > > I know this treads the line of spam a bit. But we are looking to hire > someone who likes open source and is familiar with VoIP and SIP. > > Colorado State University > > VoIP Integrator > > Colorado State University (CSU) seeks applications for a VoIP Integrator > who will be responsible for the > > integration and management of voice communications and unified messaging > systems. > > General Duties and Responsibilities > > This position will be responsible for: > > ? Managing the unified communications application, sipXecs > > ? Performing Linux system administration on underlying servers > > ? Working closely with Networking and Telecom staff to coordinate the > transition to VoIP services > > ? Developing technical and end user documentation > > ? Being on call in a rotation schedule > > For the full position announcement, including required and preferred > qualifications as well as instructions > > for applying, visit: http://www.acns.colostate.edu/jobs. > > For full consideration, complete applications must be received by 11:59pm > MST, January 12, 2015. > > Colorado State University is an EO/EA/AA employer. > > Colorado State University conducts background checks on all final > candidates. > > Thanks! > > Kyle > > -- > > > Kyle Haefner, M.S. > Communication Systems Programmer > Colorado State University > Fort Collins, CO > Phone: 970-491-1012 > Email: kyle.haefner at colostate.edu > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141220/eb7e1064/attachment.html From anthony.minessale at gmail.com Sat Dec 20 20:45:45 2014 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 20 Dec 2014 11:45:45 -0600 Subject: [Freeswitch-users] How to play a stream other than mp3 with mod_shout. In-Reply-To: References: Message-ID: I think there is an issue playing urls with vlc because vlc has some code in it that uses signal handlers which are blocked by FS. The developer who made this function in vlc is not very receptive to a solution so its being worked on. On Sat, Dec 20, 2014 at 6:02 AM, Aqs Younas wrote: > Hi, Seven > > When i try to play a stream it doesn't give any audio. Also session > created when i try to play a stream with mod_vlc does not terminate even > with "hupall". > > [0x7f69b00025d8] main input debug: Creating an input for ' > http://s9.voscast.com:7584' > [0x7f69b00025d8] main input debug: using timeshift granularity of 50 MiB, > in path '/tmp' > [0x7f69b00025d8] main input debug: `http://s9.voscast.com:7584' gives > access `http' demux `' path `s9.voscast.com:7584' > [0x7f69b00025d8] main input debug: creating demux: access='http' demux='' > location='s9.voscast.com:7584' file='(null)' > [0x26cbd48] main demux debug: looking for access_demux module matching > "http": 11 candidates > [0x26cbd48] main demux debug: no access_demux modules matched > [0x7f69b00025d8] main input debug: creating access 'http' location=' > s9.voscast.com:7584' > , path='(null)' > [0x26cbc68] main access debug: looking for access module matching "http": > 19 candidates > [0x26cbc68] access_http access debug: querying proxy for > http://s9.voscast.com:7584 > > This is what i see in my logs. Can you provide me with an example or guide > me that i am doing wrong? > > Thanks > > > On 20 December 2014 at 06:15, Seven Du wrote: > >> mod_vlc is the best answer to the original question. >> >> On Wednesday, December 17, 2014 at 10:26 PM, Brian West wrote: >> >> Don't think mod_rtmp can actually do this, from loading it: >> >> 2014-12-17 08:25:39.983446 [NOTICE] switch_loadable_module.c:149 Adding >> Endpoint 'rtmp' >> >> 2014-12-17 08:25:39.983446 [NOTICE] switch_loadable_module.c:315 Adding >> API Function 'rtmp' >> >> 2014-12-17 08:25:39.983446 [NOTICE] switch_loadable_module.c:315 Adding >> API Function 'rtmp_contact' >> >> >> This is all that gets registered. >> >> On Sun, Dec 14, 2014 at 1:37 PM, Danny Gershman > > wrote: >> >> Also mod_rtmp lets you play from an FMS server. >> >> >> On Friday, December 12, 2014, Aqs Younas wrote: >> >> Hi, All >> >> How can i play a live stream other than mp3 with mod_shout or any >> module.? Is there any way to buffer the stream before playing it with >> mod_shout. >> >> >> Currently i have a list of streams and when i play them with mod_shout >> some work fine but others give (time out) error. >> >> How can i play mostly stream in freeswitch? >> >> Thanks >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> -- >> >> *Brian West* >> brian at freeswitch.org >> >> >> *Twitter: @FreeSWITCH , @briankwest* >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 ? @anthmfs ? @FreeSWITCH ? ? http://freeswitch.org/ ? http://cluecon.com/ ? http://twitter.com/FreeSWITCH ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ * ClueCon Weekly Development Call ? sip:888 at conference.freeswitch.org ? +19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141220/9369f435/attachment-0001.html From aqsyounas at gmail.com Sat Dec 20 20:50:25 2014 From: aqsyounas at gmail.com (Aqs Younas) Date: Sat, 20 Dec 2014 22:50:25 +0500 Subject: [Freeswitch-users] How to play a stream other than mp3 with mod_shout. In-Reply-To: References: Message-ID: Thanks for your reply and time. Could you tell me how long its gonna take or when this feature will be available.? Many thanks for your reply. On 20 December 2014 at 22:45, Anthony Minessale wrote: > I think there is an issue playing urls with vlc because vlc has some code > in it that uses signal handlers which are blocked by FS. > The developer who made this function in vlc is not very receptive to a > solution so its being worked on. > > > > On Sat, Dec 20, 2014 at 6:02 AM, Aqs Younas wrote: > >> Hi, Seven >> >> When i try to play a stream it doesn't give any audio. Also session >> created when i try to play a stream with mod_vlc does not terminate even >> with "hupall". >> >> [0x7f69b00025d8] main input debug: Creating an input for ' >> http://s9.voscast.com:7584' >> [0x7f69b00025d8] main input debug: using timeshift granularity of 50 MiB, >> in path '/tmp' >> [0x7f69b00025d8] main input debug: `http://s9.voscast.com:7584' gives >> access `http' demux `' path `s9.voscast.com:7584' >> [0x7f69b00025d8] main input debug: creating demux: access='http' demux='' >> location='s9.voscast.com:7584' file='(null)' >> [0x26cbd48] main demux debug: looking for access_demux module matching >> "http": 11 candidates >> [0x26cbd48] main demux debug: no access_demux modules matched >> [0x7f69b00025d8] main input debug: creating access 'http' location=' >> s9.voscast.com:7584' >> , path='(null)' >> [0x26cbc68] main access debug: looking for access module matching "http": >> 19 candidates >> [0x26cbc68] access_http access debug: querying proxy for >> http://s9.voscast.com:7584 >> >> This is what i see in my logs. Can you provide me with an example or >> guide me that i am doing wrong? >> >> Thanks >> >> >> On 20 December 2014 at 06:15, Seven Du wrote: >> >>> mod_vlc is the best answer to the original question. >>> >>> On Wednesday, December 17, 2014 at 10:26 PM, Brian West wrote: >>> >>> Don't think mod_rtmp can actually do this, from loading it: >>> >>> 2014-12-17 08:25:39.983446 [NOTICE] switch_loadable_module.c:149 Adding >>> Endpoint 'rtmp' >>> >>> 2014-12-17 08:25:39.983446 [NOTICE] switch_loadable_module.c:315 Adding >>> API Function 'rtmp' >>> >>> 2014-12-17 08:25:39.983446 [NOTICE] switch_loadable_module.c:315 Adding >>> API Function 'rtmp_contact' >>> >>> >>> This is all that gets registered. >>> >>> On Sun, Dec 14, 2014 at 1:37 PM, Danny Gershman < >>> danny.gershman at gmail.com> wrote: >>> >>> Also mod_rtmp lets you play from an FMS server. >>> >>> >>> On Friday, December 12, 2014, Aqs Younas wrote: >>> >>> Hi, All >>> >>> How can i play a live stream other than mp3 with mod_shout or any >>> module.? Is there any way to buffer the stream before playing it with >>> mod_shout. >>> >>> >>> Currently i have a list of streams and when i play them with mod_shout >>> some work fine but others give (time out) error. >>> >>> How can i play mostly stream in freeswitch? >>> >>> Thanks >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> -- >>> >>> *Brian West* >>> brian at freeswitch.org >>> >>> >>> *Twitter: @FreeSWITCH , @briankwest* >>> http://www.freeswitchbook.com >>> http://www.freeswitchcookbook.com >>> >>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 ? @anthmfs ? @FreeSWITCH ? > > ? http://freeswitch.org/ ? http://cluecon.com/ ? > http://twitter.com/FreeSWITCH > ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ > * > > ClueCon Weekly Development Call > ? sip:888 at conference.freeswitch.org ? +19193869900 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141220/a33bd071/attachment.html From william.king at quentustech.com Sat Dec 20 21:44:24 2014 From: william.king at quentustech.com (William King) Date: Sat, 20 Dec 2014 10:44:24 -0800 Subject: [Freeswitch-users] How to play a stream other than mp3 with mod_shout. In-Reply-To: References: Message-ID: <5495C388.2020902@quentustech.com> I wrote mod_vlc and have been working with the libvlc author who added the patch that is causing the http streaming to hang. Which OS and version of vlc are you using? If it is a debian distro I can provide a patch you can apply while rebuilding libvlc. William King Senior Engineer Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Office: (206) 388-4772 Cell: (253) 686-5518 william.king at quentustech.com On 12/20/14 9:50 AM, Aqs Younas wrote: > Thanks for your reply and time. > > Could you tell me how long its gonna take or when this feature will be > available.? > > Many thanks for your reply. > > On 20 December 2014 at 22:45, Anthony Minessale > > wrote: > > I think there is an issue playing urls with vlc because vlc has some > code in it that uses signal handlers which are blocked by FS. > The developer who made this function in vlc is not very receptive to > a solution so its being worked on. > > > > On Sat, Dec 20, 2014 at 6:02 AM, Aqs Younas > wrote: > > Hi, Seven > > When i try to play a stream it doesn't give any audio. Also > session created when i try to play a stream with mod_vlc does > not terminate even with "hupall". > > [0x7f69b00025d8] main input debug: Creating an input for > 'http://s9.voscast.com:7584' > [0x7f69b00025d8] main input debug: using timeshift granularity > of 50 MiB, in path '/tmp' > [0x7f69b00025d8] main input debug: `http://s9.voscast.com:7584' > gives access `http' demux `' path `s9.voscast.com:7584 > ' > [0x7f69b00025d8] main input debug: creating demux: access='http' > demux='' location='s9.voscast.com:7584 > ' file='(null)' > [0x26cbd48] main demux debug: looking for access_demux module > matching "http": 11 candidates > [0x26cbd48] main demux debug: no access_demux modules matched > [0x7f69b00025d8] main input debug: creating access 'http' > location='s9.voscast.com:7584 ' > , path='(null)' > [0x26cbc68] main access debug: looking for access module > matching "http": 19 candidates > [0x26cbc68] access_http access debug: querying proxy for > http://s9.voscast.com:7584 > > This is what i see in my logs. Can you provide me with an > example or guide me that i am doing wrong? > > Thanks > > > On 20 December 2014 at 06:15, Seven Du > wrote: > > mod_vlc is the best answer to the original question. > > On Wednesday, December 17, 2014 at 10:26 PM, Brian West wrote: > >> Don't think mod_rtmp can actually do this, from loading it: >> >> 2014-12-17 08:25:39.983446 [NOTICE] >> switch_loadable_module.c:149 Adding Endpoint 'rtmp' >> >> 2014-12-17 08:25:39.983446 [NOTICE] >> switch_loadable_module.c:315 Adding API Function 'rtmp' >> >> 2014-12-17 08:25:39.983446 [NOTICE] >> switch_loadable_module.c:315 Adding API Function >> 'rtmp_contact' >> >> >> This is all that gets registered. >> >> >> On Sun, Dec 14, 2014 at 1:37 PM, Danny Gershman >> > > wrote: >>> Also mod_rtmp lets you play from an FMS server. >>> >>> >>> On Friday, December 12, 2014, Aqs Younas >>> > wrote: >>>> Hi, All >>>> >>>> How can i play a live stream other than mp3 with >>>> mod_shout or any module.? Is there any way to buffer the >>>> stream before playing it with mod_shout. >>>> >>>> >>>> Currently i have a list of streams and when i play them >>>> with mod_shout some work fine but others give (time out) >>>> error. >>>> >>>> How can i play mostly stream in freeswitch? >>>> >>>> Thanks >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> -- >> >> */Brian West/* >> brian at freeswitch.org >> >> >> */Twitter: @FreeSWITCH , @briankwest/* >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> *T:*+19184209001 | *F:*+19184209002 >> | *M:*+1918424WEST (9378) >> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 >> | *Skype:*briankwest >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 ? @anthmfs ? @FreeSWITCH ? > > ? http://freeswitch.org/ ? http://cluecon.com/ ? > http://twitter.com/FreeSWITCH > ? irc.freenode.net #freeswitch ? > _http://freeswitch.org/g+_ > > ClueCon Weekly Development Call > ? sip:888 at conference.freeswitch.org > ? +19193869900 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 aqsyounas at gmail.com Sat Dec 20 22:03:59 2014 From: aqsyounas at gmail.com (Aqs Younas) Date: Sun, 21 Dec 2014 00:03:59 +0500 Subject: [Freeswitch-users] How to play a stream other than mp3 with mod_shout. In-Reply-To: <5495C388.2020902@quentustech.com> References: <5495C388.2020902@quentustech.com> Message-ID: I installed vlc 2.1.5 from source code using instructions provided in wiki. https://wiki.freeswitch.org/wiki/Mod_vlc I am currently using Debian 7.6. Thanks for you reply. Would really appreciate your help. On 20 December 2014 at 23:44, William King wrote: > I wrote mod_vlc and have been working with the libvlc author who added > the patch that is causing the http streaming to hang. > > Which OS and version of vlc are you using? If it is a debian distro I > can provide a patch you can apply while rebuilding libvlc. > > William King > Senior Engineer > Quentus Technologies, INC > 1037 NE 65th St Suite 273 > Seattle, WA 98115 > Main: (877) 211-9337 > Office: (206) 388-4772 > Cell: (253) 686-5518 > william.king at quentustech.com > > On 12/20/14 9:50 AM, Aqs Younas wrote: > > Thanks for your reply and time. > > > > Could you tell me how long its gonna take or when this feature will be > > available.? > > > > Many thanks for your reply. > > > > On 20 December 2014 at 22:45, Anthony Minessale > > > > wrote: > > > > I think there is an issue playing urls with vlc because vlc has some > > code in it that uses signal handlers which are blocked by FS. > > The developer who made this function in vlc is not very receptive to > > a solution so its being worked on. > > > > > > > > On Sat, Dec 20, 2014 at 6:02 AM, Aqs Younas > > wrote: > > > > Hi, Seven > > > > When i try to play a stream it doesn't give any audio. Also > > session created when i try to play a stream with mod_vlc does > > not terminate even with "hupall". > > > > [0x7f69b00025d8] main input debug: Creating an input for > > 'http://s9.voscast.com:7584' > > [0x7f69b00025d8] main input debug: using timeshift granularity > > of 50 MiB, in path '/tmp' > > [0x7f69b00025d8] main input debug: `http://s9.voscast.com:7584' > > gives access `http' demux `' path `s9.voscast.com:7584 > > ' > > [0x7f69b00025d8] main input debug: creating demux: access='http' > > demux='' location='s9.voscast.com:7584 > > ' file='(null)' > > [0x26cbd48] main demux debug: looking for access_demux module > > matching "http": 11 candidates > > [0x26cbd48] main demux debug: no access_demux modules matched > > [0x7f69b00025d8] main input debug: creating access 'http' > > location='s9.voscast.com:7584 ' > > , path='(null)' > > [0x26cbc68] main access debug: looking for access module > > matching "http": 19 candidates > > [0x26cbc68] access_http access debug: querying proxy for > > http://s9.voscast.com:7584 > > > > This is what i see in my logs. Can you provide me with an > > example or guide me that i am doing wrong? > > > > Thanks > > > > > > On 20 December 2014 at 06:15, Seven Du > > wrote: > > > > mod_vlc is the best answer to the original question. > > > > On Wednesday, December 17, 2014 at 10:26 PM, Brian West > wrote: > > > >> Don't think mod_rtmp can actually do this, from loading it: > >> > >> 2014-12-17 08:25:39.983446 [NOTICE] > >> switch_loadable_module.c:149 Adding Endpoint 'rtmp' > >> > >> 2014-12-17 08:25:39.983446 [NOTICE] > >> switch_loadable_module.c:315 Adding API Function 'rtmp' > >> > >> 2014-12-17 08:25:39.983446 [NOTICE] > >> switch_loadable_module.c:315 Adding API Function > >> 'rtmp_contact' > >> > >> > >> This is all that gets registered. > >> > >> > >> On Sun, Dec 14, 2014 at 1:37 PM, Danny Gershman > >> >> > wrote: > >>> Also mod_rtmp lets you play from an FMS server. > >>> > >>> > >>> On Friday, December 12, 2014, Aqs Younas > >>> > wrote: > >>>> Hi, All > >>>> > >>>> How can i play a live stream other than mp3 with > >>>> mod_shout or any module.? Is there any way to buffer the > >>>> stream before playing it with mod_shout. > >>>> > >>>> > >>>> Currently i have a list of streams and when i play them > >>>> with mod_shout some work fine but others give (time out) > >>>> error. > >>>> > >>>> How can i play mostly stream in freeswitch? > >>>> > >>>> Thanks > >>> > >>> > _________________________________________________________________________ > >>> Professional FreeSWITCH Consulting Services: > >>> consulting at freeswitch.org consulting at freeswitch.org> > >>> http://www.freeswitchsolutions.com > >>> > >>> Official FreeSWITCH Sites > >>> http://www.freeswitch.org > >>> http://confluence.freeswitch.org > >>> http://www.cluecon.com > >>> > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > >>> > >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >> > >> > >> -- > >> > >> */Brian West/* > >> brian at freeswitch.org > >> > >> > >> */Twitter: @FreeSWITCH , @briankwest/* > >> http://www.freeswitchbook.com > >> http://www.freeswitchcookbook.com > >> > >> *T:*+19184209001 | *F:*+19184209002 > >> | *M:*+1918424WEST (9378) > >> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 > >> | *Skype:*briankwest > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > > >> http://www.freeswitchsolutions.com > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://confluence.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://confluence.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://confluence.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 ? @anthmfs ? @FreeSWITCH ? > > > > ? http://freeswitch.org/ ? http://cluecon.com/ ? > > http://twitter.com/FreeSWITCH > > ? irc.freenode.net #freeswitch ? > > _http://freeswitch.org/g+_ > > > > ClueCon Weekly Development Call > > ? sip:888 at conference.freeswitch.org > > ? +19193869900 > > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://confluence.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://confluence.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141221/9f196787/attachment-0001.html From william.king at quentustech.com Sat Dec 20 23:57:44 2014 From: william.king at quentustech.com (William King) Date: Sat, 20 Dec 2014 12:57:44 -0800 Subject: [Freeswitch-users] How to play a stream other than mp3 with mod_shout. In-Reply-To: References: <5495C388.2020902@quentustech.com> Message-ID: <5495E2C8.40604@quentustech.com> Comment out this line: https://github.com/videolan/vlc/blob/master/modules/access/http.c#L366 once you install the updated http access module it'll no longer hang. William King Senior Engineer Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Office: (206) 388-4772 Cell: (253) 686-5518 william.king at quentustech.com On 12/20/14 11:03 AM, Aqs Younas wrote: > I installed vlc 2.1.5 from source code using instructions provided in > wiki. https://wiki.freeswitch.org/wiki/Mod_vlc > I am currently using Debian 7.6. > > Thanks for you reply. > Would really appreciate your help. > > On 20 December 2014 at 23:44, William King > wrote: > > I wrote mod_vlc and have been working with the libvlc author who added > the patch that is causing the http streaming to hang. > > Which OS and version of vlc are you using? If it is a debian distro I > can provide a patch you can apply while rebuilding libvlc. > > William King > Senior Engineer > Quentus Technologies, INC > 1037 NE 65th St Suite 273 > Seattle, WA 98115 > Main: (877) 211-9337 > Office: (206) 388-4772 > Cell: (253) 686-5518 > william.king at quentustech.com > > On 12/20/14 9:50 AM, Aqs Younas wrote: > > Thanks for your reply and time. > > > > Could you tell me how long its gonna take or when this feature will be > > available.? > > > > Many thanks for your reply. > > > > On 20 December 2014 at 22:45, Anthony Minessale > > > >> wrote: > > > > I think there is an issue playing urls with vlc because vlc has some > > code in it that uses signal handlers which are blocked by FS. > > The developer who made this function in vlc is not very receptive to > > a solution so its being worked on. > > > > > > > > On Sat, Dec 20, 2014 at 6:02 AM, Aqs Younas > > >> wrote: > > > > Hi, Seven > > > > When i try to play a stream it doesn't give any audio. Also > > session created when i try to play a stream with mod_vlc does > > not terminate even with "hupall". > > > > [0x7f69b00025d8] main input debug: Creating an input for > > 'http://s9.voscast.com:7584' > > [0x7f69b00025d8] main input debug: using timeshift granularity > > of 50 MiB, in path '/tmp' > > [0x7f69b00025d8] main input debug: `http://s9.voscast.com:7584' > > gives access `http' demux `' path `s9.voscast.com:7584 > > ' > > [0x7f69b00025d8] main input debug: creating demux: access='http' > > demux='' location='s9.voscast.com:7584 > > ' file='(null)' > > [0x26cbd48] main demux debug: looking for access_demux module > > matching "http": 11 candidates > > [0x26cbd48] main demux debug: no access_demux modules matched > > [0x7f69b00025d8] main input debug: creating access 'http' > > location='s9.voscast.com:7584 > ' > > , path='(null)' > > [0x26cbc68] main access debug: looking for access module > > matching "http": 19 candidates > > [0x26cbc68] access_http access debug: querying proxy for > > http://s9.voscast.com:7584 > > > > This is what i see in my logs. Can you provide me with an > > example or guide me that i am doing wrong? > > > > Thanks > > > > > > On 20 December 2014 at 06:15, Seven Du > > >> wrote: > > > > mod_vlc is the best answer to the original question. > > > > On Wednesday, December 17, 2014 at 10:26 PM, Brian West wrote: > > > >> Don't think mod_rtmp can actually do this, from loading it: > >> > >> 2014-12-17 08:25:39.983446 [NOTICE] > >> switch_loadable_module.c:149 Adding Endpoint 'rtmp' > >> > >> 2014-12-17 08:25:39.983446 [NOTICE] > >> switch_loadable_module.c:315 Adding API Function 'rtmp' > >> > >> 2014-12-17 08:25:39.983446 [NOTICE] > >> switch_loadable_module.c:315 Adding API Function > >> 'rtmp_contact' > >> > >> > >> This is all that gets registered. > >> > >> > >> On Sun, Dec 14, 2014 at 1:37 PM, Danny Gershman > >> > >> >> wrote: > >>> Also mod_rtmp lets you play from an FMS server. > >>> > >>> > >>> On Friday, December 12, 2014, Aqs Younas > >>> > >> wrote: > >>>> Hi, All > >>>> > >>>> How can i play a live stream other than mp3 with > >>>> mod_shout or any module.? Is there any way to buffer the > >>>> stream before playing it with mod_shout. > >>>> > >>>> > >>>> Currently i have a list of streams and when i play them > >>>> with mod_shout some work fine but others give (time out) > >>>> error. > >>>> > >>>> How can i play mostly stream in freeswitch? > >>>> > >>>> Thanks > >>> > >>> _________________________________________________________________________ > >>> Professional FreeSWITCH Consulting Services: > >>> consulting at freeswitch.org > > > >>> http://www.freeswitchsolutions.com > >>> > >>> Official FreeSWITCH Sites > >>> http://www.freeswitch.org > >>> http://confluence.freeswitch.org > >>> http://www.cluecon.com > >>> > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > > >>> > > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >> > >> > >> -- > >> > >> */Brian West/* > >> brian at freeswitch.org > > > >> > >> > >> */Twitter: @FreeSWITCH , @briankwest/* > >> http://www.freeswitchbook.com > >> http://www.freeswitchcookbook.com > >> > >> *T:*+19184209001 > | *F:*+19184209002 > >> | *M:*+1918424WEST (9378) > >> *iNUM:*+883 5100 1420 9001 > | *ISN:*410*543 > >> | *Skype:*briankwest > >> > >> _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > > > >> http://www.freeswitchsolutions.com > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://confluence.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > > >> > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > > > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://confluence.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > > > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://confluence.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 ? @anthmfs ? @FreeSWITCH ? > > > > ? http://freeswitch.org/ ? http://cluecon.com/ ? > > http://twitter.com/FreeSWITCH > > ? irc.freenode.net > #freeswitch ? > > _http://freeswitch.org/g+_ > > > > ClueCon Weekly Development Call > > ? sip:888 at conference.freeswitch.org > > > > ? +19193869900 > > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > > > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://confluence.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://confluence.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 alhakeem at gmail.com Sat Dec 20 01:47:16 2014 From: alhakeem at gmail.com (Abdul Hakeem) Date: Fri, 19 Dec 2014 22:47:16 -0000 Subject: [Freeswitch-users] Issue with Freeswitch Behind nat In-Reply-To: References: Message-ID: From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Friday, December 19, 2014 10:01 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Issue with Freeswitch Behind nat Describe your topology a little bit. On Fri, Dec 19, 2014 at 2:31 PM, Frederick Pruneau wrote: Sorry, I pasted all my log file. I have a new pastebin: https://pastebin.freeswitch.org/23771 I tested it and I can open it. 2014-12-19 13:05 GMT-05:00 Brian West : Never able to load your pastebin, it would timeout and not load, what exactly did you paste in there? On Fri, Dec 19, 2014 at 8:56 AM, Frederick Pruneau wrote: Did you find something? 2014-12-17 10:36 GMT-05:00 Frederick Pruneau : Here it is: https://pastebin.freeswitch.org/23746 It is my freeswitch log. I have followed this guide before: https://wiki.freeswitch.org/wiki/Sofia#Debugging_Sofia-SIP I have enabled this and make a call: sofia global siptrace on sofia loglevel all 9 sofia tracelevel alert console loglevel debug fsctl debug_level 10 This is what you will get in my pastebin 2014-12-17 9:27 GMT-05:00 Brian West : sofia global siptrace on from fs_cli On Wed, Dec 17, 2014 at 7:44 AM, Frederick Pruneau wrote: Sorry for this noob question but how can I see sip traffic? Is there a specific command to show this? Is it what we find in freeswitch.log? If so, I attached my log file in my first post. Thanks for you help 2014-12-16 15:44 GMT-05:00 Brian West : have you looked at the signalling? What does the sip traffic show? Please pastebin that. On Tue, Dec 16, 2014 at 2:37 PM, Frederick Pruneau wrote: Same problem... 2014-12-16 13:55 GMT-05:00 Brian West : Guessing you don't have UPNP or NAT-PMP on your network, there for that won't work, ext-sip-ip=autonat:x.x.x.x ext-rtp-ip=autonat:x.x.x.x Set local-network-ac to rfc1918.auto On Tue, Dec 16, 2014 at 12:15 PM, Support Technique wrote: 2014-12-16 12:25 GMT-05:00 Brian West : On the system behind nat what do you have ext-rtp-ip, ext-sip-ip and local-network-acl set to? On Tue, Dec 16, 2014 at 10:41 AM, Frederick Pruneau wrote: Hi guys, We have an issue with one freeswitch server behind nat. We have a setup like this: -One master Freeswitch server -One freeswitch server connected to the master (Public IP) - Server A -One freeswitch server connected to the master (behind nat) - Server B If server A call server B, nothing happens. There is no sound. After 30 sec, it times out. We have done a tcpdump. From server A to master packets are ok. From Master to server B, we have seen that there is no source and no destination ports for sip invite. If we use our cellphone and we call server B, there is no problem. I have attached the failed call pcap file and freeswitch's log file so you can take a look at them. Master = Freeswitch v1.4.13 Server A = Freeswitch v.1.4.13 Server B = Freeswitch v.1.4.14 (Updated to latest release since we have issues with this server) Thanks in advance. PS: The failed call is from 514-448-0773. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Brian West brian at freeswitch.org Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378) iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Brian West brian at freeswitch.org Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378) iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Brian West brian at freeswitch.org Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378) iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Brian West brian at freeswitch.org Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378) iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Brian West brian at freeswitch.org Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378) iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Brian West brian at freeswitch.org Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378) iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141219/7fbcacde/attachment-0001.html From jaimecm at gmail.com Sat Dec 20 02:22:11 2014 From: jaimecm at gmail.com (Jaime) Date: Fri, 19 Dec 2014 20:22:11 -0300 Subject: [Freeswitch-users] Hoot get variables in a LUA Hook Event. Message-ID: Hi David, Many Thanks for the tip, I?m tested it and work fine in 1.4. Merry Christmas to you and everyone. Jaime From: David Witham Reply-To: FreeSWITCH Users Help Date: martes, 16 de diciembre de 2014, 4:43 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Hoot get variables in a LUA Hook Event. Hi Jamie, What we have done is add these lines in the dialplan: Then in /scripts/myscript.lua: cdr_uuid = session:getVariable("cdr_uuid"); This is on a FreeSWITCH 1.2 box. Hope this helps. regards, David From: freeswitch-users-bounces at lists.freeswitch.org on behalf of Jaime Sent: Tuesday, 16 December 2014 13:04 To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Hoot get variables in a LUA Hook Event. Hello, I?m trying to use the Hook event?s with LUA script?s when the channel Hangup a call, but I?m unable to get the variables, for example uuid, ani or billed seconds for the call, I?m using freeswitch 1.4.13 on Centos 6.6 and my configuration is: In /autoload_configs/lua.conf.xml [?] In my catch-event-cdr1.lua script I wrote: ses = freeswitch.Session(); my_uuid = ses:getVariable("uuid"); freeswitch.consoleLog("notice"," uuid=("..my_uuid..")\n") But I?m get the following error: 2014-12-15 23:51:46.100755 [ERR] switch_cpp.cpp:724 session is not initalized 2014-12-15 23:51:46.100755 [NOTICE] switch_cpp.cpp:1328 uuid=() Also, in the wiki doc I see a reference for the ?env? Object, (https://wiki.freeswitch.org/wiki/Mod_lua#Special_Case:_env_object) but if I call it as the example in the same lua script my result is: How I call the env variable: dat = env:serialize() freeswitch.consoleLog("INFO","Here's everything:\n" .. dat .. "\n?) And I got the following error: 2014-12-16 00:00:17.440782 [ERR] mod_lua.cpp:203 /usr/share/freeswitch/scripts/catch-event-reg6.lua:27: attempt to index global 'env' (a nil value) Your help will be appreciated. ? Jamie Jaimecm at gmail.com _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141219/38177750/attachment.html From alhakeem at gmail.com Sun Dec 21 02:43:00 2014 From: alhakeem at gmail.com (Abdul Hakeem) Date: Sat, 20 Dec 2014 23:43:00 -0000 Subject: [Freeswitch-users] ping Message-ID: From max at nysolutions.com Sun Dec 21 03:54:01 2014 From: max at nysolutions.com (Moishe Grunstein) Date: Sun, 21 Dec 2014 00:54:01 +0000 Subject: [Freeswitch-users] Bookmarking Audio recording Message-ID: I am working on creating a hotline for playback of children stories. One of their requirements are bookmarks, if a child wants to stop in middle of a story and call back the next day, they should be able to resume playback from the location they stopped. I see Freeswitch has the ability to set the start position using @@, so I can resume playback from 3 minutes in to the story for example https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools:+playback I am not clear if there is any method to see the position that the caller is currently at, so I can save his position to a DB when he bookmarks it. I was thinking of using the call duration however that is complicated to calculate as I am planning to use seek & speed options https://wiki.freeswitch.org/wiki/Session_streamFile. Does anyone know of a method to capture the place in a wav or mp3 file currently playing? Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support at nysolutions.com Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [cid:image001.jpg at 01C72F94.9EE45D60] Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141221/e2ce1d4a/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 2424 bytes Desc: image001.jpg Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141221/e2ce1d4a/attachment.jpg From ssinyagin at gmail.com Sun Dec 21 05:00:11 2014 From: ssinyagin at gmail.com (Stanislav Sinyagin) Date: Sun, 21 Dec 2014 03:00:11 +0100 Subject: [Freeswitch-users] Bookmarking Audio recording In-Reply-To: References: Message-ID: I think it's easier to stream from an HTTP source into FreeSWITCH, and to control the actual position and the start of playback at that streaming source. But I don't know much about streaming software that is available. There's also mod_vlc which should support more flexibility in streaming. On Sun, Dec 21, 2014 at 1:54 AM, Moishe Grunstein wrote: > I am working on creating a hotline for playback of children stories. One > of their requirements are bookmarks, if a child wants to stop in middle of > a story and call back the next day, they should be able to resume playback > from the location they stopped. > > > > I see Freeswitch has the ability to set the start position using @@, so I > can resume playback from 3 minutes in to the story for example > https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools:+playback > I am not clear if there is any method to see the position that the caller > is currently at, so I can save his position to a DB when he bookmarks it. > > > > I was thinking of using the call duration however that is complicated to > calculate as I am planning to use seek & speed options > https://wiki.freeswitch.org/wiki/Session_streamFile. > > > > Does anyone know of a method to capture the place in a wav or mp3 file > currently playing? > > > > > > Thanks, > > > > Moishe Grunstein > > Tornado Computer Systems, Inc. > > 212.400.7650 888.IPPBX.US > *Service Request Email: support at nysolutions.com * > > Polycom Certified VAR > Microsoft Small Business Specialist, Cisco SMB Select Certified > > [image: cid:image001.jpg at 01C72F94.9EE45D60] > > Computer Networking * Managed Services * IP Video Surveillance * Network > Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network > Security * Site Surveys * CMS > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141221/d365746b/attachment-0001.html From paul.atreides83 at googlemail.com Sun Dec 21 15:22:53 2014 From: paul.atreides83 at googlemail.com (Paul Atreides) Date: Sun, 21 Dec 2014 13:22:53 +0100 Subject: [Freeswitch-users] Mute without conference Message-ID: How can I implement music on hold without using a conference? I want to turn off the microphone so that the leg B can not hear what leg A is saying. Does anyone know? Thank you -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141221/2167d128/attachment.html From matthewhalos at gmail.com Sun Dec 21 17:13:42 2014 From: matthewhalos at gmail.com (Matthew Halos) Date: Sun, 21 Dec 2014 22:13:42 +0800 Subject: [Freeswitch-users] CISCO IP PHONE FOR FREESWITCH Message-ID: Hello to All! I am newbie to FreeSWITCH. I have a Cisco IP Phone with the Model, CP-7911G. Anybody knows how to use the device with FreeSWITCH? Thanks!? MATTHEW HALOS From max at nysolutions.com Sun Dec 21 17:47:26 2014 From: max at nysolutions.com (Moishe Grunstein) Date: Sun, 21 Dec 2014 14:47:26 +0000 Subject: [Freeswitch-users] CISCO IP PHONE FOR FREESWITCH In-Reply-To: References: Message-ID: Are you running sip firmware or SCCP? Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support at nysolutions.com Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Matthew Halos Sent: Sunday, December 21, 2014 9:14 AM To: freeswitch-users Subject: [Freeswitch-users] CISCO IP PHONE FOR FREESWITCH Hello to All! I am newbie to FreeSWITCH. I have a Cisco IP Phone with the Model, CP-7911G. Anybody knows how to use the device with FreeSWITCH? Thanks!? MATTHEW HALOS _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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 dujinfang at gmail.com Mon Dec 22 02:33:07 2014 From: dujinfang at gmail.com (Seven Du) Date: Mon, 22 Dec 2014 07:33:07 +0800 Subject: [Freeswitch-users] Mute without conference In-Reply-To: References: Message-ID: <102F735DF43C48DAA6C1A59E48E8FFD6@gmail.com> just press the hold button on the aleg phone it should work, or you want to use uuid_hold? -- Seven Du http://about.me/dujinfang http://www.dujinfang.com http://www.freeswitch.org.cn Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Sunday, December 21, 2014 at 8:22 PM, Paul Atreides wrote: > How can I implement music on hold without using a conference? > I want to turn off the microphone so that the leg B can not hear > what leg A is saying. > > Does anyone know? > > Thank you > _________________________________________________________________________ > 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://confluence.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/20141222/b2b384e6/attachment.html From tom at tomlynn.com Mon Dec 22 04:02:41 2014 From: tom at tomlynn.com (Tom Lynn) Date: Sun, 21 Dec 2014 17:02:41 -0800 Subject: [Freeswitch-users] newbie question (first build): FreeSWITCH just "hangs" when I run it In-Reply-To: References: Message-ID: Was this ever resolved? I've just installed and am experiencing similar behavior when I try to start it as a service. When started from the command line, it behaves. On Mon, Jun 23, 2014 at 9:01 AM, Peter Villeneuve wrote: > @Craig No problem. > > Brian, was that skype message intended for me? > > > On Mon, Jun 23, 2014 at 4:02 PM, Craig Stevenson > wrote: > >> Peter, >> >> Thanks. I also saw the symptom of the process consuming massive (nearly >> 100%) of processor and it continued to consume until killing the process. >> >> -- Craig >> >> >> >> On Mon, Jun 23, 2014 at 7:34 AM, Peter Villeneuve >> wrote: >> >>> I don't think you're doing anything wrong at all. My installation was >>> working fine until I decided to update. >>> I also am running Debian 7 on Amd64 like you. >>> >>> In fact I just did a make current and now I have the same problem you're >>> referring. >>> The freeswitch daemon now consumes around 95% of the CPU when starting >>> and just sits there indefinitely. >>> >>> I'm running strace now >>> http://www.cyberciti.biz/tips/linux-strace-command-examples.html to see >>> if I can figure out why the update seems to have broken my freeswitch >>> install. >>> >>> >>> On Mon, Jun 23, 2014 at 5:39 AM, Craig Stevenson >>> wrote: >>> >>>> >>>> Hoping someone can help out a newbie figure out what I'm doing wrong. >>>> This is my first attempt at building and I keep running into same issue >>>> each time. Any pointers would be greatly appreciated. >>>> >>>> My problem is that FreeSWITHC seems to just get stuck when I try to run >>>> the binary. And and if I execute with -ncwait optoin, it keeps spitting >>>> out a line saying that it is waiting for background process to be ready >>>> >>>> This is what happens when I try to run the compiled system on a home >>>> ESXi server in my internal home network. >>>> >>>> # ./freeswitch >>>> 2014-06-22 21:23:06.011792 [INFO] switch_event.c:670 Activate Eventing >>>> Engine. >>>> 2014-06-22 21:23:06.011947 [WARNING] switch_event.c:652 Create >>>> additional event dispatch thread 0 >>>> >>>> >>>> >>>> >>>> BUILD VERSION: >>>> - Debian-7.5.0-amd64-netinst.iso >>>> - FreeSWITCH master (git:git clone git://git.*freeswitch.org/freeswitch.git >>>> )* >>>> >>>> >>>> TEST MACHINE: ESXi 5.1 VM >>>> - Debian GNU/Linus 6 (64 bit) >>>> - 1 CPU >>>> - 1 GB RAM >>>> - 16GB Memory >>>> - 1 NIC = E1000 >>>> >>>> >>>> INSTALLATION STEPS with output piped through 'tee' to capture logs: >>>> >>>> su >>>> (apt-get update 2>&1) | tee ~/apt-get-update.txt >>>> (apt-get upgrade 2>&1) | tee ~/apt-get-upgrade.txt >>>> (dpkg --get-selections 2>&1) | tee ~/dpkg-get-selections.txt >>>> >>>> (apt-get install autoconf automake devscripts gawk g++ git-core >>>> libjpeg-dev libncurses5-dev libtool make python-dev gawk pkg-config >>>> libtiff5-dev libperl-dev libgdbm-dev libdb-dev gettext libssl-dev >>>> libcurl4-openssl-dev libpcre3-dev libspeex-dev libspeexdsp-dev >>>> libsqlite3-dev libedit-dev libldns.dev libpq-dev 2>&1) | tee ~/apt-get.txt >>>> >>>> (dpkg --get-selections 2>&1) | tee >>>> ~/dpkg-get-selections-after-apt-get.txt >>>> >>>> cd /usr/src >>>> (git clone git://git.*freeswitch.org/freeswitch.git >>>> * 2>&1) | tee ~/git.txt >>>> >>>> cd /usr/src/freeswitch >>>> (./bootstrap.sh 2>&1) | tee ~/bootstrap.txt # same result if I add >>>> flag -j >>>> (./configure --enable-core-pgsql-support 2>&1) | tee ~/configure.txt >>>> (make install 2>&1) | tee ~/make-install.txt >>>> (make all cd-sounds-install cd-moh-install 2>&1) | tee ~make-sounds.txt >>>> >>>> cd /usr/local/freeswitch/bin >>>> #./freeswitch >>>> >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at 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 at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> 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 >>> 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 at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE: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 >> 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE: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 > 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141221/9dc4417f/attachment-0001.html From matthewhalos at gmail.com Mon Dec 22 04:41:24 2014 From: matthewhalos at gmail.com (Matthew Halos) Date: Mon, 22 Dec 2014 09:41:24 +0800 Subject: [Freeswitch-users] CISCO IP PHONE FOR FREESWITCH Message-ID: Hi Moishe, I am using SCCP. Thanks, MATTHEW HALOS ---------- Forwarded message ---------- From: Moishe Grunstein To: FreeSWITCH Users Help Cc: Date: Sun, 21 Dec 2014 14:47:26 +0000 Subject: Re: [Freeswitch-users] CISCO IP PHONE FOR FREESWITCH Are you running sip firmware or SCCP? Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support at nysolutions.com Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Matthew Halos Sent: Sunday, December 21, 2014 9:14 AM To: freeswitch-users Subject: [Freeswitch-users] CISCO IP PHONE FOR FREESWITCH Hello to All! I am newbie to FreeSWITCH. I have a Cisco IP Phone with the Model, CP-7911G. Anybody knows how to use the device with FreeSWITCH? Thanks!? MATTHEW HALOS _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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 max at nysolutions.com Mon Dec 22 06:46:12 2014 From: max at nysolutions.com (Moishe Grunstein) Date: Mon, 22 Dec 2014 03:46:12 +0000 Subject: [Freeswitch-users] CISCO IP PHONE FOR FREESWITCH In-Reply-To: References: Message-ID: See https://freeswitch.org/confluence/display/FREESWITCH/mod_skinny https://freeswitch.org/confluence/display/FREESWITCH/Cisco+7960+SIP may also interest you. Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support at nysolutions.com Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Matthew Halos Sent: Sunday, December 21, 2014 8:41 PM To: freeswitch-users Subject: Re: [Freeswitch-users] CISCO IP PHONE FOR FREESWITCH Hi Moishe, I am using SCCP. Thanks, MATTHEW HALOS ---------- Forwarded message ---------- From: Moishe Grunstein To: FreeSWITCH Users Help Cc: Date: Sun, 21 Dec 2014 14:47:26 +0000 Subject: Re: [Freeswitch-users] CISCO IP PHONE FOR FREESWITCH Are you running sip firmware or SCCP? Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support at nysolutions.com Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Matthew Halos Sent: Sunday, December 21, 2014 9:14 AM To: freeswitch-users Subject: [Freeswitch-users] CISCO IP PHONE FOR FREESWITCH Hello to All! I am newbie to FreeSWITCH. I have a Cisco IP Phone with the Model, CP-7911G. Anybody knows how to use the device with FreeSWITCH? Thanks!? MATTHEW HALOS _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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 mvar78 at gmail.com Mon Dec 22 12:37:10 2014 From: mvar78 at gmail.com (Massimo Varriale) Date: Mon, 22 Dec 2014 10:37:10 +0100 Subject: [Freeswitch-users] SQL Lite Different Folder Message-ID: <890A784F-B805-46E1-BA76-8917951EC7E6@gmail.com> Hi Guys! I would like to know if it's possibile to create/manage a SQL Lite Database (cdr_sqlite) into a different folder rather than /usr/local/freeswitch/db because as I followed the wiki guide to optimize a FS installation (https://wiki.freeswitch.org/wiki/Performance_testing_and_configurations#FreeSWITCH.27s_core.db_I.2FO_bottleneck) the /db folder is a RAMDISK and therefore the content is lost on system reboot. I'm using SQLite because FS write directly into a DB and most of all because I would like to save my CDRs within several sources (now the calls are stored into a MongoDB, a CSV and SQLite) Thank you so much Cheers Max From mvar78 at gmail.com Mon Dec 22 12:41:09 2014 From: mvar78 at gmail.com (Massimo Varriale) Date: Mon, 22 Dec 2014 10:41:09 +0100 Subject: [Freeswitch-users] [Mod CDR CSV] Adding Timestamp into the CDR Message-ID: <8D142526-7FDD-411B-BEEB-77B9F6C57FDB@gmail.com> Hi Guys, I would like to write a field into the CSV with the timestamp of the CDR so I can know the writetime. The timestamp should be in this format "2014-12-22 10:32". Is this possible? I digged the wiki but I didn't found any clue regarding this. Thank you Max From mvar78 at gmail.com Mon Dec 22 16:43:06 2014 From: mvar78 at gmail.com (Massimo Varriale) Date: Mon, 22 Dec 2014 14:43:06 +0100 Subject: [Freeswitch-users] Lua Retry Call if call Fail Message-ID: <15F107B8-4D43-4F55-9BB7-DE7A1A9BE8EF@gmail.com> Hi, I'm trying to use Lua to create a callflow for my call. In particular, I would like to try a call to another gateway if the first fail. I found this wiki (https://wiki.freeswitch.org/wiki/Call_retry_based_on_hangup_cause) but doesn't worked for me. What could be the right procedure? Thank you Max From brian at freeswitch.org Mon Dec 22 17:46:28 2014 From: brian at freeswitch.org (Brian West) Date: Mon, 22 Dec 2014 08:46:28 -0600 Subject: [Freeswitch-users] ping In-Reply-To: References: Message-ID: pong! On Sat, Dec 20, 2014 at 5:43 PM, Abdul Hakeem wrote: > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141222/b3b8f8de/attachment.html From amirehsansadeghi at gmail.com Mon Dec 22 14:50:43 2014 From: amirehsansadeghi at gmail.com (Amirehsan Sadeghi) Date: Mon, 22 Dec 2014 15:20:43 +0330 Subject: [Freeswitch-users] Freeswitch Calling Card Solution Message-ID: Dear All , I want install and configure a calling card system with FreeSwitch And FreeRadius i after install and configure this servers in my pc Please Tell me how i do for run a local calling Card system ? -- Best regards Amirehsan Sadeghi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141222/4ae3e010/attachment.html From poliv78 at yahoo.co.uk Mon Dec 22 19:02:46 2014 From: poliv78 at yahoo.co.uk (poliv78 at yahoo.co.uk) Date: Mon, 22 Dec 2014 18:02:46 +0200 Subject: [Freeswitch-users] check for inbound event socket connection In-Reply-To: References: Message-ID: <1391883736.20141222180246@yahoo.co.uk> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141222/f872238e/attachment.html From max at nysolutions.com Mon Dec 22 19:06:49 2014 From: max at nysolutions.com (Moishe Grunstein) Date: Mon, 22 Dec 2014 16:06:49 +0000 Subject: [Freeswitch-users] Bookmarking Audio recording In-Reply-To: References: Message-ID: I don?t see why pulling the stream from a http stream would give me more flexibility than playing back a local sound file. Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support at nysolutions.com Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [cid:image001.jpg at 01C72F94.9EE45D60] Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Stanislav Sinyagin Sent: Saturday, December 20, 2014 9:00 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Bookmarking Audio recording I think it's easier to stream from an HTTP source into FreeSWITCH, and to control the actual position and the start of playback at that streaming source. But I don't know much about streaming software that is available. There's also mod_vlc which should support more flexibility in streaming. On Sun, Dec 21, 2014 at 1:54 AM, Moishe Grunstein > wrote: I am working on creating a hotline for playback of children stories. One of their requirements are bookmarks, if a child wants to stop in middle of a story and call back the next day, they should be able to resume playback from the location they stopped. I see Freeswitch has the ability to set the start position using @@, so I can resume playback from 3 minutes in to the story for example https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools:+playback I am not clear if there is any method to see the position that the caller is currently at, so I can save his position to a DB when he bookmarks it. I was thinking of using the call duration however that is complicated to calculate as I am planning to use seek & speed options https://wiki.freeswitch.org/wiki/Session_streamFile. Does anyone know of a method to capture the place in a wav or mp3 file currently playing? Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support at nysolutions.com Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141222/2cc7a2d4/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 2424 bytes Desc: image001.jpg Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141222/2cc7a2d4/attachment-0001.jpg From aqsyounas at gmail.com Mon Dec 22 19:22:38 2014 From: aqsyounas at gmail.com (Aqs Younas) Date: Mon, 22 Dec 2014 21:22:38 +0500 Subject: [Freeswitch-users] How to play a stream other than mp3 with mod_shout. In-Reply-To: <5495E2C8.40604@quentustech.com> References: <5495C388.2020902@quentustech.com> <5495E2C8.40604@quentustech.com> Message-ID: Many thanks for you reply. I will try this and get back to you if i face any problem. Really Appreciate your Help. On 21 December 2014 at 01:57, William King wrote: > Comment out this line: > https://github.com/videolan/vlc/blob/master/modules/access/http.c#L366 > > once you install the updated http access module it'll no longer hang. > > William King > Senior Engineer > Quentus Technologies, INC > 1037 NE 65th St Suite 273 > Seattle, WA 98115 > Main: (877) 211-9337 > Office: (206) 388-4772 > Cell: (253) 686-5518 > william.king at quentustech.com > > On 12/20/14 11:03 AM, Aqs Younas wrote: > > I installed vlc 2.1.5 from source code using instructions provided in > > wiki. https://wiki.freeswitch.org/wiki/Mod_vlc > > I am currently using Debian 7.6. > > > > Thanks for you reply. > > Would really appreciate your help. > > > > On 20 December 2014 at 23:44, William King > > wrote: > > > > I wrote mod_vlc and have been working with the libvlc author who > added > > the patch that is causing the http streaming to hang. > > > > Which OS and version of vlc are you using? If it is a debian distro I > > can provide a patch you can apply while rebuilding libvlc. > > > > William King > > Senior Engineer > > Quentus Technologies, INC > > 1037 NE 65th St Suite 273 > > Seattle, WA 98115 > > Main: (877) 211-9337 > > Office: (206) 388-4772 > > Cell: (253) 686-5518 > > william.king at quentustech.com > > > > On 12/20/14 9:50 AM, Aqs Younas wrote: > > > Thanks for your reply and time. > > > > > > Could you tell me how long its gonna take or when this feature > will be > > > available.? > > > > > > Many thanks for your reply. > > > > > > On 20 December 2014 at 22:45, Anthony Minessale > > > > > > >> wrote: > > > > > > I think there is an issue playing urls with vlc because vlc > has some > > > code in it that uses signal handlers which are blocked by FS. > > > The developer who made this function in vlc is not very > receptive to > > > a solution so its being worked on. > > > > > > > > > > > > On Sat, Dec 20, 2014 at 6:02 AM, Aqs Younas < > aqsyounas at gmail.com > > > >> > wrote: > > > > > > Hi, Seven > > > > > > When i try to play a stream it doesn't give any audio. Also > > > session created when i try to play a stream with mod_vlc > does > > > not terminate even with "hupall". > > > > > > [0x7f69b00025d8] main input debug: Creating an input for > > > 'http://s9.voscast.com:7584' > > > [0x7f69b00025d8] main input debug: using timeshift > granularity > > > of 50 MiB, in path '/tmp' > > > [0x7f69b00025d8] main input debug: ` > http://s9.voscast.com:7584' > > > gives access `http' demux `' path `s9.voscast.com:7584 < > http://s9.voscast.com:7584> > > > ' > > > [0x7f69b00025d8] main input debug: creating demux: > access='http' > > > demux='' location='s9.voscast.com:7584 < > http://s9.voscast.com:7584> > > > ' file='(null)' > > > [0x26cbd48] main demux debug: looking for access_demux > module > > > matching "http": 11 candidates > > > [0x26cbd48] main demux debug: no access_demux modules > matched > > > [0x7f69b00025d8] main input debug: creating access 'http' > > > location='s9.voscast.com:7584 > > ' > > > , path='(null)' > > > [0x26cbc68] main access debug: looking for access module > > > matching "http": 19 candidates > > > [0x26cbc68] access_http access debug: querying proxy for > > > http://s9.voscast.com:7584 > > > > > > This is what i see in my logs. Can you provide me with an > > > example or guide me that i am doing wrong? > > > > > > Thanks > > > > > > > > > On 20 December 2014 at 06:15, Seven Du < > dujinfang at gmail.com > > > >> > wrote: > > > > > > mod_vlc is the best answer to the original question. > > > > > > On Wednesday, December 17, 2014 at 10:26 PM, Brian > West wrote: > > > > > >> Don't think mod_rtmp can actually do this, from > loading it: > > >> > > >> 2014-12-17 08:25:39.983446 [NOTICE] > > >> switch_loadable_module.c:149 Adding Endpoint 'rtmp' > > >> > > >> 2014-12-17 08:25:39.983446 [NOTICE] > > >> switch_loadable_module.c:315 Adding API Function > 'rtmp' > > >> > > >> 2014-12-17 08:25:39.983446 [NOTICE] > > >> switch_loadable_module.c:315 Adding API Function > > >> 'rtmp_contact' > > >> > > >> > > >> This is all that gets registered. > > >> > > >> > > >> On Sun, Dec 14, 2014 at 1:37 PM, Danny Gershman > > >> danny.gershman at gmail.com> > > >> danny.gershman at gmail.com>>> wrote: > > >>> Also mod_rtmp lets you play from an FMS server. > > >>> > > >>> > > >>> On Friday, December 12, 2014, Aqs Younas > > >>> > > >> wrote: > > >>>> Hi, All > > >>>> > > >>>> How can i play a live stream other than mp3 with > > >>>> mod_shout or any module.? Is there any way to > buffer the > > >>>> stream before playing it with mod_shout. > > >>>> > > >>>> > > >>>> Currently i have a list of streams and when i play > them > > >>>> with mod_shout some work fine but others give (time > out) > > >>>> error. > > >>>> > > >>>> How can i play mostly stream in freeswitch? > > >>>> > > >>>> Thanks > > >>> > > >>> > _________________________________________________________________________ > > >>> Professional FreeSWITCH Consulting Services: > > >>> consulting at freeswitch.org > > > > > > >>> http://www.freeswitchsolutions.com > > >>> > > >>> Official FreeSWITCH Sites > > >>> http://www.freeswitch.org > > >>> http://confluence.freeswitch.org > > >>> http://www.cluecon.com > > >>> > > >>> FreeSWITCH-users mailing list > > >>> FreeSWITCH-users at lists.freeswitch.org > > > > >>> > > > > >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > >>> http://www.freeswitch.org > > >> > > >> > > >> -- > > >> > > >> */Brian West/* > > >> brian at freeswitch.org > > > > > >> > > >> > > >> */Twitter: @FreeSWITCH , @briankwest/* > > >> http://www.freeswitchbook.com > > >> http://www.freeswitchcookbook.com > > >> > > >> *T:*+19184209001 > > | *F:*+19184209002 > > >> | *M:*+1918424WEST (9378) > > >> *iNUM:*+883 5100 1420 9001 > > | *ISN:*410*543 > > >> | *Skype:*briankwest > > >> > > >> > _________________________________________________________________________ > > >> Professional FreeSWITCH Consulting Services: > > >> consulting at freeswitch.org > > > > > > >> http://www.freeswitchsolutions.com > > >> > > >> Official FreeSWITCH Sites > > >> http://www.freeswitch.org > > >> http://confluence.freeswitch.org > > >> http://www.cluecon.com > > >> > > >> FreeSWITCH-users mailing list > > >> FreeSWITCH-users at lists.freeswitch.org > > > > >> > > > > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > >> http://www.freeswitch.org > > > > > > > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org > > > > > > > http://www.freeswitchsolutions.com > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://confluence.freeswitch.org > > > http://www.cluecon.com > > > > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > > > > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > > > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org > > > > > > > http://www.freeswitchsolutions.com > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://confluence.freeswitch.org > > > http://www.cluecon.com > > > > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at 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 ? @anthmfs ? @FreeSWITCH ? > > > > > > ? http://freeswitch.org/ ? http://cluecon.com/ ? > > > http://twitter.com/FreeSWITCH > > > ? irc.freenode.net > > #freeswitch ? > > > _http://freeswitch.org/g+_ > > > > > > ClueCon Weekly Development Call > > > ? sip:888 at conference.freeswitch.org > > > > > > > ? +19193869900 > > > > > > > > > > > > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org > > >> > > > http://www.freeswitchsolutions.com > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://confluence.freeswitch.org > > > http://www.cluecon.com > > > > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > > > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > > > > > > > > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org > > > http://www.freeswitchsolutions.com > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://confluence.freeswitch.org > > > http://www.cluecon.com > > > > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://confluence.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://confluence.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141222/0643e739/attachment-0001.html From adrottenberg at gmail.com Mon Dec 22 19:36:48 2014 From: adrottenberg at gmail.com (Duvid Rottenberg) Date: Mon, 22 Dec 2014 11:36:48 -0500 Subject: [Freeswitch-users] Bookmarking Audio recording In-Reply-To: References: Message-ID: Check out the playback related variables at https://freeswitch.org/confluence/display/FREESWITCH/Channel+Variables#ChannelVariables-Playbackrelatedvariables In the playback_stop event you can store the playback_last_offset_pos in a database, then next time you play the file you append @@ to the end of the filename. On Mon, Dec 22, 2014 at 11:06 AM, Moishe Grunstein wrote: > > I don?t see why pulling the stream from a http stream would give me more > flexibility than playing back a local sound file. > > > > Thanks, > > > > Moishe Grunstein > > Tornado Computer Systems, Inc. > > 212.400.7650 888.IPPBX.US > *Service Request Email: support at nysolutions.com * > > Polycom Certified VAR > Microsoft Small Business Specialist, Cisco SMB Select Certified > > [image: cid:image001.jpg at 01C72F94.9EE45D60] > > Computer Networking * Managed Services * IP Video Surveillance * Network > Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network > Security * Site Surveys * CMS > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Stanislav > Sinyagin > *Sent:* Saturday, December 20, 2014 9:00 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Bookmarking Audio recording > > > > I think it's easier to stream from an HTTP source into FreeSWITCH, and to > control the actual position and the start of playback at that streaming > source. But I don't know much about streaming software that is available. > > There's also mod_vlc which should support more flexibility in streaming. > > > > > > On Sun, Dec 21, 2014 at 1:54 AM, Moishe Grunstein > wrote: > > I am working on creating a hotline for playback of children stories. One > of their requirements are bookmarks, if a child wants to stop in middle of > a story and call back the next day, they should be able to resume playback > from the location they stopped. > > > > I see Freeswitch has the ability to set the start position using @@, so I > can resume playback from 3 minutes in to the story for example > https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools:+playback > I am not clear if there is any method to see the position that the caller > is currently at, so I can save his position to a DB when he bookmarks it. > > > > I was thinking of using the call duration however that is complicated to > calculate as I am planning to use seek & speed options > https://wiki.freeswitch.org/wiki/Session_streamFile. > > > > Does anyone know of a method to capture the place in a wav or mp3 file > currently playing? > > > > > > Thanks, > > > > Moishe Grunstein > > Tornado Computer Systems, Inc. > > 212.400.7650 888.IPPBX.US > *Service Request Email: support at nysolutions.com * > > Polycom Certified VAR > Microsoft Small Business Specialist, Cisco SMB Select Certified > > > > Computer Networking * Managed Services * IP Video Surveillance * Network > Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network > Security * Site Surveys * CMS > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > *consulting at freeswitch.org* > *http://www.freeswitchsolutions.com* > > Official FreeSWITCH Sites > *http://www.freeswitch.org* > *http://confluence.freeswitch.org* > *http://www.cluecon.com* > > FreeSWITCH-users mailing list > *FreeSWITCH-users at lists.freeswitch.org* > *http://lists.freeswitch.org/mailman/listinfo/freeswitch-users* > UNSUBSCRIBE:*http://lists.freeswitch.org/mailman/options/freeswitch-users* > *http://www.freeswitch.org* > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141222/f8f06aab/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 2424 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141222/f8f06aab/attachment.jpg From stesasso at gmail.com Mon Dec 22 21:37:45 2014 From: stesasso at gmail.com (Stefano Sasso) Date: Mon, 22 Dec 2014 19:37:45 +0100 Subject: [Freeswitch-users] strange issue in video call Message-ID: Hi guys, I am encountering a strange issue in video calls. Please note that I cannot use proxy-media or bypass-media, because I need to have audio transcoding. (I tried with version 1.4.14 - but encountered the same problem in 1.2.x) So, in my sofia profile configuration I have as inbound-codec-prefs and outbound-codec-prefs the following string: OPUS,PCMU,PCMA,GSM,H264 I have the late-codec-negotiation and renegotiate-codec-on-reinvite set to true. I am NOT using early audio. (just to clarify) when A starts a call in audio only mode (only m=audio in sdp), on the B-leg I always see also the m=video sdp part. So, after the call is established, if from A I start sending a video, the B UAC receives it. But, if I start sending the video from B, there is no re-invite (I think it's because the B UAC thinks the video channel is already up), so the video ends in freeswitch (A UAC never receives the video - it has only a m=audio channel!). If I start the call with the video from the beginning, everything works fine. I also tried to remove the H264 from the outbound-codec-prefs, but in that case the reinvite with the video does not work. Is there a way to fix that behaviour? (send the m=video in the B-leg only if present in the A-leg, and then accept the re-INVITEs with the m=video from A-leg or B-leg) thanks! bests, stefano -- Stefano Sasso http://stefano.dscnet.org/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141222/9ddb2abf/attachment.html From anthony.minessale at gmail.com Mon Dec 22 22:01:15 2014 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 22 Dec 2014 13:01:15 -0600 Subject: [Freeswitch-users] strange issue in video call In-Reply-To: References: Message-ID: There is a refactor of video going on right now for the next Beta release this spring. There will not be any more changes to 1.4 regarding video support. On Mon, Dec 22, 2014 at 12:37 PM, Stefano Sasso wrote: > Hi guys, > I am encountering a strange issue in video calls. > Please note that I cannot use proxy-media or bypass-media, because I need > to have audio transcoding. > (I tried with version 1.4.14 - but encountered the same problem in 1.2.x) > > So, in my sofia profile configuration I have as inbound-codec-prefs and > outbound-codec-prefs the following string: > OPUS,PCMU,PCMA,GSM,H264 > I have the late-codec-negotiation and renegotiate-codec-on-reinvite set to > true. > I am NOT using early audio. (just to clarify) > > when A starts a call in audio only mode (only m=audio in sdp), on the > B-leg I always see also the m=video sdp part. > So, after the call is established, if from A I start sending a video, the > B UAC receives it. > But, if I start sending the video from B, there is no re-invite (I think > it's because the B UAC thinks the video channel is already up), so the > video ends in freeswitch (A UAC never receives the video - it has only a > m=audio channel!). > > If I start the call with the video from the beginning, everything works > fine. > I also tried to remove the H264 from the outbound-codec-prefs, but in that > case the reinvite with the video does not work. > > Is there a way to fix that behaviour? > (send the m=video in the B-leg only if present in the A-leg, and then > accept the re-INVITEs with the m=video from A-leg or B-leg) > > thanks! > bests, > stefano > > -- > Stefano Sasso > http://stefano.dscnet.org/ > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 ? @anthmfs ? @FreeSWITCH ? ? http://freeswitch.org/ ? http://cluecon.com/ ? http://twitter.com/FreeSWITCH ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ * ClueCon Weekly Development Call ? sip:888 at conference.freeswitch.org ? +19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141222/7854ec84/attachment-0001.html From auge at virtues.net Mon Dec 22 22:26:13 2014 From: auge at virtues.net (Thomas Auge) Date: Mon, 22 Dec 2014 16:26:13 -0300 Subject: [Freeswitch-users] Freeswitch video one to many Message-ID: <54987055.3000006@virtues.net> Hello list :-) does Freeswitch offer functionality to distribute streams to multiple listeners/viewers without conferencing (one way)? Thanks, Thomas From alipey at gmail.com Mon Dec 22 23:41:54 2014 From: alipey at gmail.com (Ali Pey) Date: Mon, 22 Dec 2014 15:41:54 -0500 Subject: [Freeswitch-users] Fax tone detection Message-ID: Hello, Is there anyway to make fax tone detection more sensitive? I use spandsp_start_fax_detect for the first 10 seconds of the call and fax tones are not always detected. It seems that if the tone is not loaud enought, freeswitch doesnt' detect it. Thanks, Ali Pey -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141222/bbae7454/attachment.html From mike at jerris.com Tue Dec 23 00:46:23 2014 From: mike at jerris.com (Michael Jerris) Date: Mon, 22 Dec 2014 16:46:23 -0500 Subject: [Freeswitch-users] Freeswitch video one to many In-Reply-To: <54987055.3000006@virtues.net> References: <54987055.3000006@virtues.net> Message-ID: <21921C27-1328-4743-9110-98E8D4B7BD01@jerris.com> no, currently only in conference. > On Dec 22, 2014, at 2:26 PM, Thomas Auge wrote: > > Hello list :-) > > does Freeswitch offer functionality to distribute streams to multiple listeners/viewers without conferencing (one way)? > > Thanks, > > Thomas From auge at virtues.net Tue Dec 23 01:06:06 2014 From: auge at virtues.net (Thomas Auge) Date: Mon, 22 Dec 2014 19:06:06 -0300 Subject: [Freeswitch-users] Freeswitch video one to many In-Reply-To: <21921C27-1328-4743-9110-98E8D4B7BD01@jerris.com> References: <54987055.3000006@virtues.net> <21921C27-1328-4743-9110-98E8D4B7BD01@jerris.com> Message-ID: <549895CE.5030108@virtues.net> Could simply muting the receiving ends work? Does that "pause" the RTC stream? mod_conference does combine the media streams into one, right? On 22.12.2014 18:46, Michael Jerris wrote: > no, currently only in conference. > > >> On Dec 22, 2014, at 2:26 PM, Thomas Auge wrote: >> >> Hello list :-) >> >> does Freeswitch offer functionality to distribute streams to multiple listeners/viewers without conferencing (one way)? >> >> Thanks, >> >> Thomas > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 lloyd.aloysius at gmail.com Tue Dec 23 01:16:42 2014 From: lloyd.aloysius at gmail.com (Lloyd Aloysius) Date: Mon, 22 Dec 2014 17:16:42 -0500 Subject: [Freeswitch-users] Receives call From Unknown Extensions Message-ID: Hi All I have a multi domain setup. We receive calls from unknown extensions (eg: 100 , 101,1000,1007 etc ).But there is no voice in it. We do not have any default extensions in the system and all default extensions removed from the system. Users are authenticated by alphanumeric (like an email username) Eg: mike@ mydomain.com and passwords are very complicated. How someone can call a user without authentication from these extensions? Please let me know how to solve this issue. Thanks Lloyd -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141222/49402a44/attachment.html From mike at jerris.com Tue Dec 23 01:23:28 2014 From: mike at jerris.com (Michael Jerris) Date: Mon, 22 Dec 2014 17:23:28 -0500 Subject: [Freeswitch-users] Freeswitch video one to many In-Reply-To: <549895CE.5030108@virtues.net> References: <54987055.3000006@virtues.net> <21921C27-1328-4743-9110-98E8D4B7BD01@jerris.com> <549895CE.5030108@virtues.net> Message-ID: For any serious video functionality, you will need to wait for 1.6. We are working on a number of features now, but without them, even the work to do multipoint wont be possible. > On Dec 22, 2014, at 5:06 PM, Thomas Auge wrote: > > Could simply muting the receiving ends work? Does that "pause" the RTC stream? mod_conference does combine the media > streams into one, right? > > On 22.12.2014 18:46, Michael Jerris wrote: >> no, currently only in conference. >> >> >>> On Dec 22, 2014, at 2:26 PM, Thomas Auge wrote: >>> >>> Hello list :-) >>> >>> does Freeswitch offer functionality to distribute streams to multiple listeners/viewers without conferencing (one way)? >>> >>> Thanks, >>> >>> Thomas >> From auge at virtues.net Tue Dec 23 01:35:18 2014 From: auge at virtues.net (Thomas Auge) Date: Mon, 22 Dec 2014 19:35:18 -0300 Subject: [Freeswitch-users] Receives call From Unknown Extensions In-Reply-To: References: Message-ID: <54989CA6.3030900@virtues.net> Do you still have the external domain enabled? I think it routes external calls matching a specific number theme ( ^(10[01][0-9])$ ) to the internal users through the pre-installed dialplan. It listens on different ports (5080/1). Config is in sip_profiles/external.xml and dialplan/public.xml. I see an insane amount of brute force attempts against our PBX', so if there is a way to get anywhere, you can expect people to try it - over and over and over ... I can recommend fail2ban. :-) Just guessing though, if I'm wrong, someone more knowledgeable will probably chime in. :) On 22.12.2014 19:16, Lloyd Aloysius wrote: > Hi All > > I have a multi domain setup. We receive calls from unknown extensions (eg: 100 , 101,1000,1007 etc ).But there is no > voice in it. > > We do not have any default extensions in the system and all default extensions removed from the system. > > Users are authenticated by alphanumeric (like an email username) Eg: mike at mydomain.com and passwords are very > complicated. > > How someone can call a user without authentication from these extensions? > > Please let me know how to solve this issue. > > Thanks Lloyd > > > > > > _________________________________________________________________________ Professional FreeSWITCH Consulting > Services: consulting at freeswitch.org http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com > > FreeSWITCH-users mailing list FreeSWITCH-users at 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 auge at virtues.net Tue Dec 23 01:36:44 2014 From: auge at virtues.net (Thomas Auge) Date: Mon, 22 Dec 2014 19:36:44 -0300 Subject: [Freeswitch-users] Freeswitch video one to many In-Reply-To: References: <54987055.3000006@virtues.net> <21921C27-1328-4743-9110-98E8D4B7BD01@jerris.com> <549895CE.5030108@virtues.net> Message-ID: <54989CFC.5000700@virtues.net> On 22.12.2014 19:23, Michael Jerris wrote: > For any serious video functionality, you will need to wait for 1.6. We are working on a number of features now, but > without them, even the work to do multipoint wont be possible. It's not urgent (for a change :-). Very happy to hear there are more features in the queue. :) From ssinyagin at gmail.com Tue Dec 23 01:39:58 2014 From: ssinyagin at gmail.com (Stanislav Sinyagin) Date: Mon, 22 Dec 2014 23:39:58 +0100 Subject: [Freeswitch-users] Bookmarking Audio recording In-Reply-To: References: Message-ID: because there are multiple open-source streaming solutions, with API and everything you need to control the stream. The built-in streamer in FreeSWITCH is just not designed to offer what you need. On Mon, Dec 22, 2014 at 5:06 PM, Moishe Grunstein wrote: > I don?t see why pulling the stream from a http stream would give me more > flexibility than playing back a local sound file. > > > > Thanks, > > > > Moishe Grunstein > > Tornado Computer Systems, Inc. > > 212.400.7650 888.IPPBX.US > *Service Request Email: support at nysolutions.com * > > Polycom Certified VAR > Microsoft Small Business Specialist, Cisco SMB Select Certified > > [image: cid:image001.jpg at 01C72F94.9EE45D60] > > Computer Networking * Managed Services * IP Video Surveillance * Network > Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network > Security * Site Surveys * CMS > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Stanislav > Sinyagin > *Sent:* Saturday, December 20, 2014 9:00 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Bookmarking Audio recording > > > > I think it's easier to stream from an HTTP source into FreeSWITCH, and to > control the actual position and the start of playback at that streaming > source. But I don't know much about streaming software that is available. > > There's also mod_vlc which should support more flexibility in streaming. > > > > > > On Sun, Dec 21, 2014 at 1:54 AM, Moishe Grunstein > wrote: > > I am working on creating a hotline for playback of children stories. One > of their requirements are bookmarks, if a child wants to stop in middle of > a story and call back the next day, they should be able to resume playback > from the location they stopped. > > > > I see Freeswitch has the ability to set the start position using @@, so I > can resume playback from 3 minutes in to the story for example > https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools:+playback > I am not clear if there is any method to see the position that the caller > is currently at, so I can save his position to a DB when he bookmarks it. > > > > I was thinking of using the call duration however that is complicated to > calculate as I am planning to use seek & speed options > https://wiki.freeswitch.org/wiki/Session_streamFile. > > > > Does anyone know of a method to capture the place in a wav or mp3 file > currently playing? > > > > > > Thanks, > > > > Moishe Grunstein > > Tornado Computer Systems, Inc. > > 212.400.7650 888.IPPBX.US > *Service Request Email: support at nysolutions.com * > > Polycom Certified VAR > Microsoft Small Business Specialist, Cisco SMB Select Certified > > > > Computer Networking * Managed Services * IP Video Surveillance * Network > Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network > Security * Site Surveys * CMS > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > *consulting at freeswitch.org* > *http://www.freeswitchsolutions.com* > > Official FreeSWITCH Sites > *http://www.freeswitch.org* > *http://confluence.freeswitch.org* > *http://www.cluecon.com* > > FreeSWITCH-users mailing list > *FreeSWITCH-users at lists.freeswitch.org* > *http://lists.freeswitch.org/mailman/listinfo/freeswitch-users* > UNSUBSCRIBE:*http://lists.freeswitch.org/mailman/options/freeswitch-users* > *http://www.freeswitch.org* > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141222/916c7b67/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 2424 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141222/916c7b67/attachment-0001.jpg From lloyd.aloysius at gmail.com Tue Dec 23 01:44:57 2014 From: lloyd.aloysius at gmail.com (Lloyd Aloysius) Date: Mon, 22 Dec 2014 17:44:57 -0500 Subject: [Freeswitch-users] Receives call From Unknown Extensions In-Reply-To: <54989CA6.3030900@virtues.net> References: <54989CA6.3030900@virtues.net> Message-ID: Fail2Ban is running in the system I do not have any default dial plans or extensions. On Mon, Dec 22, 2014 at 5:35 PM, Thomas Auge wrote: > Do you still have the external domain enabled? I think it routes external > calls matching a specific number theme ( > ^(10[01][0-9])$ ) to the internal users through the pre-installed > dialplan. It listens on different ports (5080/1). > Config is in sip_profiles/external.xml and dialplan/public.xml. > > I see an insane amount of brute force attempts against our PBX', so if > there is a way to get anywhere, you can expect > people to try it - over and over and over ... I can recommend fail2ban. :-) > > Just guessing though, if I'm wrong, someone more knowledgeable will > probably chime in. :) > > > On 22.12.2014 19:16, Lloyd Aloysius wrote: > > Hi All > > > > I have a multi domain setup. We receive calls from unknown extensions > (eg: 100 , 101,1000,1007 etc ).But there is no > > voice in it. > > > > We do not have any default extensions in the system and all default > extensions removed from the system. > > > > Users are authenticated by alphanumeric (like an email username) Eg: > mike at mydomain.com and passwords are very > > complicated. > > > > How someone can call a user without authentication from these extensions? > > > > Please let me know how to solve this issue. > > > > Thanks Lloyd > > > > > > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting > > Services: consulting at freeswitch.org http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites http://www.freeswitch.org > http://confluence.freeswitch.org http://www.cluecon.com > > > > FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141222/dee4addf/attachment.html From auge at virtues.net Tue Dec 23 01:55:50 2014 From: auge at virtues.net (Thomas Auge) Date: Mon, 22 Dec 2014 19:55:50 -0300 Subject: [Freeswitch-users] Receives call From Unknown Extensions In-Reply-To: References: <54989CA6.3030900@virtues.net> Message-ID: <5498A176.4040009@virtues.net> To eliminate the guessing, check the logs which route the calls took through the system. It should contain the clues you need. You might need to up the log level a bit ... On 22.12.2014 19:44, Lloyd Aloysius wrote: > Fail2Ban is running in the system > > I do not have any default dial plans or extensions. > > > > > > On Mon, Dec 22, 2014 at 5:35 PM, Thomas Auge > wrote: > > Do you still have the external domain enabled? I think it routes external calls matching a specific number theme ( > ^(10[01][0-9])$ ) to the internal users through the pre-installed dialplan. It listens on different ports (5080/1). > Config is in sip_profiles/external.xml and dialplan/public.xml. > > I see an insane amount of brute force attempts against our PBX', so if there is a way to get anywhere, you can expect > people to try it - over and over and over ... I can recommend fail2ban. :-) > > Just guessing though, if I'm wrong, someone more knowledgeable will probably chime in. :) > > > On 22.12.2014 19:16, Lloyd Aloysius wrote: > > Hi All > > > > I have a multi domain setup. We receive calls from unknown extensions (eg: 100 , 101,1000,1007 etc ).But there is no > > voice in it. > > > > We do not have any default extensions in the system and all default extensions removed from the system. > > > > Users are authenticated by alphanumeric (like an email username) Eg: mike at mydomain.com > and passwords are very > > complicated. > > > > How someone can call a user without authentication from these extensions? > > > > Please let me know how to solve this issue. > > > > Thanks Lloyd > > > > > > > > > > > > _________________________________________________________________________ Professional FreeSWITCH Consulting > > Services: consulting at freeswitch.org http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com > > > > FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 mike at jerris.com Tue Dec 23 02:08:30 2014 From: mike at jerris.com (Michael Jerris) Date: Mon, 22 Dec 2014 18:08:30 -0500 Subject: [Freeswitch-users] Freeswitch video one to many In-Reply-To: <54989CFC.5000700@virtues.net> References: <54987055.3000006@virtues.net> <21921C27-1328-4743-9110-98E8D4B7BD01@jerris.com> <549895CE.5030108@virtues.net> <54989CFC.5000700@virtues.net> Message-ID: <939D6966-F6C9-4719-88B4-414224055375@jerris.com> I will say that multipoint work while we have discussed it, is not currently on the roadmap. That being said, its something that at least has a possibility of being looked at in the future. Its certainly an interesting feature to explore, but I can't say at all yet how complicated or not complicated it might be other than to say I expect it will not be a small project. Mike > On Dec 22, 2014, at 5:36 PM, Thomas Auge wrote: > > On 22.12.2014 19:23, Michael Jerris wrote: >> For any serious video functionality, you will need to wait for 1.6. We are working on a number of features now, but >> without them, even the work to do multipoint wont be possible. > > It's not urgent (for a change :-). Very happy to hear there are more features in the queue. :) > From david.villasmil at gmail.com Tue Dec 23 03:32:18 2014 From: david.villasmil at gmail.com (David Villasmil Govea) Date: Tue, 23 Dec 2014 01:32:18 +0100 Subject: [Freeswitch-users] Bleg transfer Message-ID: Hello Guys, I receive a call from side A and send it out to side B. I need to unbridge this call once it is answered and send the B side to a queue to give it moh. How do I do that? I'm doing everything with lua and I've tried: on my dialplan I have: in check_answered-lua, after checking the call was in fact answered I do: session:execute("transfer", "-bleg 9999 XML default"); and on my dialplan I have: queue.lua: session:execute("sched_hangup","+50 alloted_timeout"); session:execute("callcenter","agents_queue"); session:execute("sleep",my_dur); But this doesn't seem to work at all... Can anyone give me a hand? thanks! David -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141223/78f67f0f/attachment.html From telishisheer at gmail.com Tue Dec 23 10:09:24 2014 From: telishisheer at gmail.com (Shisheer Teli) Date: Tue, 23 Dec 2014 12:39:24 +0530 Subject: [Freeswitch-users] Openldap and freeswitch integration problem Message-ID: Hi, I am able to bind with any alise on ldap server except userPassword (MD5) alise. when i bind password with userPassword , authentication fails. I done some following testing Test 1: when i set openldap userPassword in md5 , in freeswitch cli i saw hash password and authentication failed. Test 2: when i set openldap userPassword in plain text, in freeswitch cli i can see plain text password and authentication success. Authentication works with plain text but not for encrypted password. Configuration file: Please reply ASAP... -- Regards, Shisheer T -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141223/e2ba1a7d/attachment-0001.html From paul.atreides83 at googlemail.com Tue Dec 23 10:40:01 2014 From: paul.atreides83 at googlemail.com (Paul Atreides) Date: Tue, 23 Dec 2014 08:40:01 +0100 Subject: [Freeswitch-users] Mute without conference In-Reply-To: <102F735DF43C48DAA6C1A59E48E8FFD6@gmail.com> References: <102F735DF43C48DAA6C1A59E48E8FFD6@gmail.com> Message-ID: But how let I play music while the bleg is on hold? On Mon, Dec 22, 2014 at 12:33 AM, Seven Du wrote: > just press the hold button on the aleg phone it should work, or you want > to use uuid_hold? > > -- > Seven Du > http://about.me/dujinfang > http://www.dujinfang.com > http://www.freeswitch.org.cn > > Sent with Sparrow > > On Sunday, December 21, 2014 at 8:22 PM, Paul Atreides wrote: > > How can I implement music on hold without using a conference? > I want to turn off the microphone so that the leg B can not hear > what leg A is saying. > > Does anyone know? > > Thank you > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141223/136b1371/attachment.html From steveu at coppice.org Tue Dec 23 14:59:01 2014 From: steveu at coppice.org (Steve Underwood) Date: Tue, 23 Dec 2014 19:59:01 +0800 Subject: [Freeswitch-users] Fax tone detection In-Reply-To: References: Message-ID: <54995905.7070501@coppice.org> If your signal is so quiet that the spandsp FAX can't detect it, then nothing else will either. Regards, Steve On 12/23/2014 04:41 AM, Ali Pey wrote: > Hello, > > Is there anyway to make fax tone detection more sensitive? > > I use spandsp_start_fax_detect for the first 10 seconds of the call > and fax tones are not always detected. It seems that if the tone is > not loaud enought, freeswitch doesnt' detect it. > > Thanks, > Ali Pey > > From call.center.morrow at gmail.com Tue Dec 23 14:19:41 2014 From: call.center.morrow at gmail.com (Chris Morrow) Date: Tue, 23 Dec 2014 12:19:41 +0100 Subject: [Freeswitch-users] ASR dialplan.xml example for mod_unimrcp Message-ID: Hello, I am trying make a simple freeswitch->unimrcp speech recognition test with dialplan.xml. I have been experimenting recently with freeswitch and mod_unimrcp. I have checked the documentation on website: - https://freeswitch.org/confluence/display/FREESWITCH/mod_unimrcp and search extensively through the unimrcp and freeswitch lists, and google, but without success. I have successfully installed and tested: - freeswitch [with direct recording via dialplan] - mod_unimrcp - unimrcp [with test unimrcpserver and umc client] There is examples of implementing a dialplan.xml with a tts_engine, and some javascript or other examples for the asr, but I cannot seem to correctly figure out how to make a full, end-to-end example of the dialplan.xml based asr unimrcp call. Ideally I like to just replicate the simple test in unimrcp, but via freeswitch cal [basically the 'run recog']l: - http://www.unimrcp.org/manuals/html/InstallationManual.html#_Toc391758895 I think I can probably figure out the rest if that works. Thanks for your feedback! Chris -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141223/9c9808eb/attachment.html From GeorgePhelps at gfphelps.com Tue Dec 23 16:18:08 2014 From: GeorgePhelps at gfphelps.com (George F. Phelps) Date: Tue, 23 Dec 2014 08:18:08 -0500 Subject: [Freeswitch-users] No package speex-devel available. Message-ID: <022001d01eb2$e564e0f0$b02ea2d0$@gfphelps.com> I am attempting to build Freeswitch, using these instructions: https://freeswitch.org/confluence/display/FREESWITCH/CentOS+6 How do I resolve these two "yum install" errors? No package speex-devel available. No package libedit-devel available. And even though "speex" is installed (yum: Package speex-1.2-0.19.rc1.el7.x86_64 already installed and latest version), I am getting this build error: checking for speex >= 1.2rc1 speexdsp >= 1.2rc1... Package speex was not found in the pkg-config search path. What is the workaround for this error? Thanks! From cmrienzo at gmail.com Tue Dec 23 16:31:13 2014 From: cmrienzo at gmail.com (cmrienzo at gmail.com) Date: Tue, 23 Dec 2014 08:31:13 -0500 Subject: [Freeswitch-users] ASR dialplan.xml example for mod_unimrcp In-Reply-To: References: Message-ID: <4A9E8695-D74D-400C-8D46-2CF3EE69B494@gmail.com> For dialplan, try using play_and_detect_speech. There's an example in the old wiki if you google it. Speech recognition APIs are not very good. Anything more complicated than simple command and control will best be implemented over event socket, embedded scripting, or in adhearsion. Chris > On Dec 23, 2014, at 06:19, Chris Morrow wrote: > > Hello, I am trying make a simple freeswitch->unimrcp speech recognition test with dialplan.xml. I have been experimenting recently with freeswitch and mod_unimrcp. I have checked the documentation on website: > https://freeswitch.org/confluence/display/FREESWITCH/mod_unimrcp > and search extensively through the unimrcp and freeswitch lists, and google, but without success. > > I have successfully installed and tested: > freeswitch [with direct recording via dialplan] > mod_unimrcp > unimrcp [with test unimrcpserver and umc client] > There is examples of implementing a dialplan.xml with a tts_engine, and some javascript or other examples for the asr, but I cannot seem to correctly figure out how to make a full, end-to-end example of the dialplan.xml based asr unimrcp call. > > Ideally I like to just replicate the simple test in unimrcp, but via freeswitch cal [basically the 'run recog']l: > http://www.unimrcp.org/manuals/html/InstallationManual.html#_Toc391758895 > I think I can probably figure out the rest if that works. > > Thanks for your feedback! > > Chris > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141223/5f70dc83/attachment.html From call.center.morrow at gmail.com Tue Dec 23 17:11:08 2014 From: call.center.morrow at gmail.com (Chris Morrow) Date: Tue, 23 Dec 2014 15:11:08 +0100 Subject: [Freeswitch-users] ASR dialplan.xml example for mod_unimrcp In-Reply-To: <4A9E8695-D74D-400C-8D46-2CF3EE69B494@gmail.com> References: <4A9E8695-D74D-400C-8D46-2CF3EE69B494@gmail.com> Message-ID: Hello, Thanks for the advice, googling for 'play_and_detect_speech' worked. I did not hear of Adhearsion before, it looks quite robust. Does it also supports continuous ASR? Thanks. On Tue, Dec 23, 2014 at 2:31 PM, wrote: > For dialplan, try using play_and_detect_speech. There's an example in the > old wiki if you google it. > > Speech recognition APIs are not very good. Anything more complicated than > simple command and control will best be implemented over event socket, > embedded scripting, or in adhearsion. > > Chris > > > > On Dec 23, 2014, at 06:19, Chris Morrow > wrote: > > Hello, I am trying make a simple freeswitch->unimrcp speech recognition > test with dialplan.xml. I have been experimenting recently with freeswitch > and mod_unimrcp. I have checked the documentation on website: > > - https://freeswitch.org/confluence/display/FREESWITCH/mod_unimrcp > > and search extensively through the unimrcp and freeswitch lists, and > google, but without success. > > I have successfully installed and tested: > > - freeswitch [with direct recording via dialplan] > - mod_unimrcp > - unimrcp [with test unimrcpserver and umc client] > > There is examples of implementing a dialplan.xml with a tts_engine, and > some javascript or other examples for the asr, but I cannot seem to > correctly figure out how to make a full, end-to-end example of the > dialplan.xml based asr unimrcp call. > > Ideally I like to just replicate the simple test in unimrcp, but via > freeswitch cal [basically the 'run recog']l: > > - > http://www.unimrcp.org/manuals/html/InstallationManual.html#_Toc391758895 > > I think I can probably figure out the rest if that works. > > Thanks for your feedback! > > Chris > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141223/b5450168/attachment-0001.html From cmrienzo at gmail.com Tue Dec 23 17:43:49 2014 From: cmrienzo at gmail.com (Christopher Rienzo) Date: Tue, 23 Dec 2014 09:43:49 -0500 Subject: [Freeswitch-users] ASR dialplan.xml example for mod_unimrcp In-Reply-To: References: <4A9E8695-D74D-400C-8D46-2CF3EE69B494@gmail.com> Message-ID: I'm not sure what exactly you mean by continuous ASR, but you can run speech detection for a lengthy amount of time using either event socket or adhearsion. Adhearsion is nice in that it gives you a clean framework for writing these kind of apps. However, that comes at a cost of a reduced set of FS features available. Event socket gives you total control at the cost of complexity and tight coupling to FS. On Tue, Dec 23, 2014 at 9:11 AM, Chris Morrow wrote: > Hello, Thanks for the advice, googling for 'play_and_detect_speech' > worked. I did not hear of Adhearsion before, it looks quite robust. Does > it also supports continuous ASR? Thanks. > > On Tue, Dec 23, 2014 at 2:31 PM, wrote: > >> For dialplan, try using play_and_detect_speech. There's an example in >> the old wiki if you google it. >> >> Speech recognition APIs are not very good. Anything more complicated than >> simple command and control will best be implemented over event socket, >> embedded scripting, or in adhearsion. >> >> Chris >> >> >> >> On Dec 23, 2014, at 06:19, Chris Morrow >> wrote: >> >> Hello, I am trying make a simple freeswitch->unimrcp speech recognition >> test with dialplan.xml. I have been experimenting recently with freeswitch >> and mod_unimrcp. I have checked the documentation on website: >> >> - https://freeswitch.org/confluence/display/FREESWITCH/mod_unimrcp >> >> and search extensively through the unimrcp and freeswitch lists, and >> google, but without success. >> >> I have successfully installed and tested: >> >> - freeswitch [with direct recording via dialplan] >> - mod_unimrcp >> - unimrcp [with test unimrcpserver and umc client] >> >> There is examples of implementing a dialplan.xml with a tts_engine, and >> some javascript or other examples for the asr, but I cannot seem to >> correctly figure out how to make a full, end-to-end example of the >> dialplan.xml based asr unimrcp call. >> >> Ideally I like to just replicate the simple test in unimrcp, but via >> freeswitch cal [basically the 'run recog']l: >> >> - >> http://www.unimrcp.org/manuals/html/InstallationManual.html#_Toc391758895 >> >> I think I can probably figure out the rest if that works. >> >> Thanks for your feedback! >> >> Chris >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141223/138193e4/attachment.html From rentmycoder at gmail.com Tue Dec 23 17:51:44 2014 From: rentmycoder at gmail.com (rentmycoder rentmycoder) Date: Tue, 23 Dec 2014 15:51:44 +0100 Subject: [Freeswitch-users] Fwd: webrtc chrome->sip bad media jssip error In-Reply-To: References: Message-ID: I've checked with 1.4 latest git, it works using latest SIPMl5 and SipJs in all directions. But now FireFox latest (34.05) is broken with both js stacks... Browser->SIP is working, but SIP->Browser and Browser->Browser is broken... So FS 1.4.14 does not work with latest Chrome, 1.4.master does not work with latest FF. Maybe SDP issue... On Thu, Dec 18, 2014 at 3:22 PM, rentmycoder rentmycoder < rentmycoder at gmail.com> wrote: > Hi, > > Maybe it's a know issue, please direct me to the right direction. > Setup: FS 1.4.latest, -nonat, Debian64 7.8, http://tryit.jssip.net/ web > sip client, > > Issue: > Using Firefox, calls are working great, but with Chrome 39.0.2171.95m > calling from browser to SIPendpint fails. > The sip client rings, but after answer, the jssip client shows bad media > description error.... > > Ring SDP: > v=0 > o=FreeSWITCH 1418885182 1418885183 IN IP4 192.168.101.43 > s=FreeSWITCH > c=IN IP4 192.168.101.43 > t=0 0 > a=msid-semantic: WMS PLEThV3VtmQWGIk4ICYTkWiv3AtV43mJ > m=audio 27200 RTP/SAVPF 0 126 106 > a=rtpmap:0 PCMU/8000 > a=rtpmap:126 telephone-event/8000 > a=rtpmap:106 CN/8000 > a=ptime:20 > a=sendrecv > a=fingerprint:sha-256 > E2:56:BC:E3:1E:CA:AD:55:04:E5:94:8F:D6:AD:1B:CA:E0:B8:90:A9:62:35:95:F0:F1:C0:D7:A7:92:35:D9:C8 > a=rtcp-mux > a=rtcp:27200 IN IP4 192.168.101.43 > a=ssrc:1443731118 cname:FQsNBmyVzeI5MT0k > a=ssrc:1443731118 msid:PLEThV3VtmQWGIk4ICYTkWiv3AtV43mJ a0 > a=ssrc:1443731118 mslabel:PLEThV3VtmQWGIk4ICYTkWiv3AtV43mJ > a=ssrc:1443731118 label:PLEThV3VtmQWGIk4ICYTkWiv3AtV43mJa0 > a=ice-ufrag:lO4wBcSvCchx5t6B > a=ice-pwd:9peBBCn0GWWiFMtRrndSqVeT > a=candidate:9761529334 1 udp 659136 192.168.101.43 27200 typ host > generation 0 > > > 2014-12-18 09:19:46.047277 [DEBUG] sofia.c:6624 Remote SDP: > v=0 > o=1001 8000 8000 IN IP4 192.168.101.39 > s=SIP Call > c=IN IP4 192.168.101.39 > t=0 0 > m=audio 5004 RTP/AVP 0 13 > a=sendrecv > a=rtpmap:0 PCMU/8000 > a=ptime:20 > > > Remote SDP: > 2014-12-18 09:19:46.067273 [DEBUG] mod_sofia.c:780 Local SDP > sofia/internal/1002 at 192.168.101.43: > v=0 > o=FreeSWITCH 1418885182 1418885184 IN IP4 192.168.101.43 > s=FreeSWITCH > c=IN IP4 192.168.101.43 > t=0 0 > a=msid-semantic: WMS PLEThV3VtmQWGIk4ICYTkWiv3AtV43mJ > m=audio 27200 RTP/SAVPF 0 126 106 > a=rtpmap:0 PCMU/8000 > a=rtpmap:126 telephone-event/8000 > a=rtpmap:106 CN/8000 > a=ptime:20 > a=sendrecv > a=fingerprint:sha-256 > E2:56:BC:E3:1E:CA:AD:55:04:E5:94:8F:D6:AD:1B:CA:E0:B8:90:A9:62:35:95:F0:F1:C0:D7:A7:92:35:D9:C8 > a=rtcp-mux > a=rtcp:27200 IN IP4 192.168.101.43 > a=ssrc:1443731118 cname:FQsNBmyVzeI5MT0k > a=ssrc:1443731118 msid:PLEThV3VtmQWGIk4ICYTkWiv3AtV43mJ a0 > a=ssrc:1443731118 mslabel:PLEThV3VtmQWGIk4ICYTkWiv3AtV43mJ > a=ssrc:1443731118 label:PLEThV3VtmQWGIk4ICYTkWiv3AtV43mJa0 > a=ice-ufrag:lO4wBcSvCchx5t6B > a=ice-pwd:9peBBCn0GWWiFMtRrndSqVeT > a=candidate:7802155548 1 udp 659136 192.168.101.43 27200 typ host > generation 0 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141223/0f9e1ed9/attachment-0001.html From luis.daniel.lucio at gmail.com Tue Dec 23 19:35:10 2014 From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz) Date: Tue, 23 Dec 2014 11:35:10 -0500 Subject: [Freeswitch-users] lua xml-handler-binding and chatplan Message-ID: Hello Just wondering if this is possible to let a custom xml-handler, written in lua to deal with chatplan as it does for dialplan. Any comment? i was reading about hoo event tag, maybe with that? LD From david.villasmil at gmail.com Tue Dec 23 19:48:54 2014 From: david.villasmil at gmail.com (David Villasmil Govea) Date: Tue, 23 Dec 2014 17:48:54 +0100 Subject: [Freeswitch-users] Unbridge and send to queue Message-ID: Hello Guys, I receive a call from side A and send it out to side B. I need to unbridge this call once it is answered and send the B side to a queue to give it moh. How do I do that? I'm doing everything with lua and I've tried: on my dialplan I have: in check_answered-lua, after checking the call was in fact answered I do: session:execute("transfer", "-bleg 9999 XML default"); and on my dialplan I have: queue.lua: session:execute("sched_hangup","+50 alloted_timeout"); session:execute("callcenter","agents_queue"); session:execute("sleep",my_dur); But this doesn't seem to work at all... Can anyone give me a hand? thanks! -- DVG -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141223/e67b0add/attachment.html From david.villasmil at gmail.com Tue Dec 23 20:00:00 2014 From: david.villasmil at gmail.com (David Villasmil Govea) Date: Tue, 23 Dec 2014 18:00:00 +0100 Subject: [Freeswitch-users] lua xml-handler-binding and chatplan In-Reply-To: References: Message-ID: Hello, I'm sure looks possible, have you tried it yet? David On Tue, Dec 23, 2014 at 5:35 PM, Luis Daniel Lucio Quiroz < luis.daniel.lucio at gmail.com> wrote: > Hello > > Just wondering if this is possible > > to let a custom xml-handler, written in lua to deal with chatplan as > it does for dialplan. Any comment? > > i was reading about hoo event tag, maybe with that? > > LD > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- DVG -- Imagination is more important than knowledge Albert Einstein -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141223/f4c6d5b5/attachment.html From david.villasmil at gmail.com Tue Dec 23 20:01:15 2014 From: david.villasmil at gmail.com (David Villasmil Govea) Date: Tue, 23 Dec 2014 18:01:15 +0100 Subject: [Freeswitch-users] lua xml-handler-binding and chatplan In-Reply-To: References: Message-ID: Hello, Programming You can also directly call the various programming language modules from the chatplan: On Tue, Dec 23, 2014 at 6:00 PM, David Villasmil Govea < david.villasmil at gmail.com> wrote: > Hello, > > I'm sure looks possible, have you tried it yet? > > David > > On Tue, Dec 23, 2014 at 5:35 PM, Luis Daniel Lucio Quiroz < > luis.daniel.lucio at gmail.com> wrote: > >> Hello >> >> Just wondering if this is possible >> >> to let a custom xml-handler, written in lua to deal with chatplan as >> it does for dialplan. Any comment? >> >> i was reading about hoo event tag, maybe with that? >> >> LD >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > DVG > > -- > Imagination is more important than knowledge > Albert Einstein > -- DVG -- Imagination is more important than knowledge Albert Einstein -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141223/12b7d1e6/attachment.html From mike at jerris.com Tue Dec 23 20:15:47 2014 From: mike at jerris.com (Michael Jerris) Date: Tue, 23 Dec 2014 12:15:47 -0500 Subject: [Freeswitch-users] Openldap and freeswitch integration problem In-Reply-To: References: Message-ID: <48AD5085-FA78-4CD3-91B4-C056948F861A@jerris.com> Due to the way sip digest auth works, you could not actually validate a password if all you have is the md5 of the password. You can store the a1 hash, which is an md5 of username:realm:password string. For more information on how digest authentication works to help understand why what you are trying is not cryptographically possible, check out: http://en.wikipedia.org/wiki/Digest_access_authentication Mike > On Dec 23, 2014, at 2:09 AM, Shisheer Teli wrote: > > Hi, > > I am able to bind with any alise on ldap server except userPassword (MD5) alise. > > when i bind password with userPassword , authentication fails. > > I done some following testing > > Test 1: > when i set openldap userPassword in md5 , in freeswitch cli i saw hash password and authentication failed. > > Test 2: > when i set openldap userPassword in plain text, in freeswitch cli i can see plain text password and authentication success. > > Authentication works with plain text but not for encrypted password. > > Configuration file: > > > > > > > > > > > > > > > > > > > Please reply ASAP... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141223/2f0ebbc1/attachment-0001.html From mike at jerris.com Tue Dec 23 20:23:46 2014 From: mike at jerris.com (Michael Jerris) Date: Tue, 23 Dec 2014 12:23:46 -0500 Subject: [Freeswitch-users] Mute without conference In-Reply-To: References: <102F735DF43C48DAA6C1A59E48E8FFD6@gmail.com> Message-ID: <798E6C86-5F22-46DF-9D31-2F83F840F89D@jerris.com> https://wiki.freeswitch.org/wiki/Variable_hold_music > On Dec 23, 2014, at 2:40 AM, Paul Atreides wrote: > > But how let I play music while the bleg is on hold? > > On Mon, Dec 22, 2014 at 12:33 AM, Seven Du > wrote: > just press the hold button on the aleg phone it should work, or you want to use uuid_hold? > > -- > Seven Du > http://about.me/dujinfang > http://www.dujinfang.com > http://www.freeswitch.org.cn > > Sent with Sparrow > > On Sunday, December 21, 2014 at 8:22 PM, Paul Atreides wrote: > >> How can I implement music on hold without using a conference? >> I want to turn off the microphone so that the leg B can not hear >> what leg A is saying. >> >> Does anyone know? >> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141223/90318c96/attachment.html From mike at jerris.com Tue Dec 23 20:39:53 2014 From: mike at jerris.com (Michael Jerris) Date: Tue, 23 Dec 2014 12:39:53 -0500 Subject: [Freeswitch-users] webrtc chrome->sip bad media jssip error In-Reply-To: References: Message-ID: If master is not working with one of them, please file a jira. > On Dec 23, 2014, at 9:51 AM, rentmycoder rentmycoder wrote: > > I've checked with 1.4 latest git, it works using latest SIPMl5 and SipJs in all directions. > But now FireFox latest (34.05) is broken with both js stacks... > Browser->SIP is working, but SIP->Browser and Browser->Browser is broken... > > So FS 1.4.14 does not work with latest Chrome, 1.4.master does not work with latest FF. > Maybe SDP issue... > > > On Thu, Dec 18, 2014 at 3:22 PM, rentmycoder rentmycoder > wrote: > Hi, > > Maybe it's a know issue, please direct me to the right direction. > Setup: FS 1.4.latest, -nonat, Debian64 7.8, http://tryit.jssip.net/ web sip client, > > Issue: > Using Firefox, calls are working great, but with Chrome 39.0.2171.95m calling from browser to SIPendpint fails. > The sip client rings, but after answer, the jssip client shows bad media description error.... > > Ring SDP: > v=0 > o=FreeSWITCH 1418885182 1418885183 IN IP4 192.168.101.43 > s=FreeSWITCH > c=IN IP4 192.168.101.43 > t=0 0 > a=msid-semantic: WMS PLEThV3VtmQWGIk4ICYTkWiv3AtV43mJ > m=audio 27200 RTP/SAVPF 0 126 106 > a=rtpmap:0 PCMU/8000 > a=rtpmap:126 telephone-event/8000 > a=rtpmap:106 CN/8000 > a=ptime:20 > a=sendrecv > a=fingerprint:sha-256 E2:56:BC:E3:1E:CA:AD:55:04:E5:94:8F:D6:AD:1B:CA:E0:B8:90:A9:62:35:95:F0:F1:C0:D7:A7:92:35:D9:C8 > a=rtcp-mux > a=rtcp:27200 IN IP4 192.168.101.43 > a=ssrc:1443731118 cname:FQsNBmyVzeI5MT0k > a=ssrc:1443731118 msid:PLEThV3VtmQWGIk4ICYTkWiv3AtV43mJ a0 > a=ssrc:1443731118 mslabel:PLEThV3VtmQWGIk4ICYTkWiv3AtV43mJ > a=ssrc:1443731118 label:PLEThV3VtmQWGIk4ICYTkWiv3AtV43mJa0 > a=ice-ufrag:lO4wBcSvCchx5t6B > a=ice-pwd:9peBBCn0GWWiFMtRrndSqVeT > a=candidate:9761529334 1 udp 659136 192.168.101.43 27200 typ host generation 0 > > > 2014-12-18 09:19:46.047277 [DEBUG] sofia.c:6624 Remote SDP: > v=0 > o=1001 8000 8000 IN IP4 192.168.101.39 > s=SIP Call > c=IN IP4 192.168.101.39 > t=0 0 > m=audio 5004 RTP/AVP 0 13 > a=sendrecv > a=rtpmap:0 PCMU/8000 > a=ptime:20 > > > Remote SDP: > 2014-12-18 09:19:46.067273 [DEBUG] mod_sofia.c:780 Local SDP sofia/internal/1002 at 192.168.101.43 : > v=0 > o=FreeSWITCH 1418885182 1418885184 IN IP4 192.168.101.43 > s=FreeSWITCH > c=IN IP4 192.168.101.43 > t=0 0 > a=msid-semantic: WMS PLEThV3VtmQWGIk4ICYTkWiv3AtV43mJ > m=audio 27200 RTP/SAVPF 0 126 106 > a=rtpmap:0 PCMU/8000 > a=rtpmap:126 telephone-event/8000 > a=rtpmap:106 CN/8000 > a=ptime:20 > a=sendrecv > a=fingerprint:sha-256 E2:56:BC:E3:1E:CA:AD:55:04:E5:94:8F:D6:AD:1B:CA:E0:B8:90:A9:62:35:95:F0:F1:C0:D7:A7:92:35:D9:C8 > a=rtcp-mux > a=rtcp:27200 IN IP4 192.168.101.43 > a=ssrc:1443731118 cname:FQsNBmyVzeI5MT0k > a=ssrc:1443731118 msid:PLEThV3VtmQWGIk4ICYTkWiv3AtV43mJ a0 > a=ssrc:1443731118 mslabel:PLEThV3VtmQWGIk4ICYTkWiv3AtV43mJ > a=ssrc:1443731118 label:PLEThV3VtmQWGIk4ICYTkWiv3AtV43mJa0 > a=ice-ufrag:lO4wBcSvCchx5t6B > a=ice-pwd:9peBBCn0GWWiFMtRrndSqVeT > a=candidate:7802155548 1 udp 659136 192.168.101.43 27200 typ host generation 0 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141223/e1211157/attachment.html From mike at jerris.com Tue Dec 23 20:46:40 2014 From: mike at jerris.com (Michael Jerris) Date: Tue, 23 Dec 2014 12:46:40 -0500 Subject: [Freeswitch-users] webrtc chrome->sip bad media jssip error In-Reply-To: References: Message-ID: Also, take a look if verto works. I know there were changes in firefox that required changes to the verto js. It may be that the issue is now on the js side. > On Dec 23, 2014, at 12:39 PM, Michael Jerris wrote: > > If master is not working with one of them, please file a jira. > >> On Dec 23, 2014, at 9:51 AM, rentmycoder rentmycoder > wrote: >> >> I've checked with 1.4 latest git, it works using latest SIPMl5 and SipJs in all directions. >> But now FireFox latest (34.05) is broken with both js stacks... >> Browser->SIP is working, but SIP->Browser and Browser->Browser is broken... >> >> So FS 1.4.14 does not work with latest Chrome, 1.4.master does not work with latest FF. >> Maybe SDP issue... >> >> >> On Thu, Dec 18, 2014 at 3:22 PM, rentmycoder rentmycoder > wrote: >> Hi, >> >> Maybe it's a know issue, please direct me to the right direction. >> Setup: FS 1.4.latest, -nonat, Debian64 7.8, http://tryit.jssip.net/ web sip client, >> >> Issue: >> Using Firefox, calls are working great, but with Chrome 39.0.2171.95m calling from browser to SIPendpint fails. >> The sip client rings, but after answer, the jssip client shows bad media description error.... >> >> Ring SDP: >> v=0 >> o=FreeSWITCH 1418885182 1418885183 IN IP4 192.168.101.43 >> s=FreeSWITCH >> c=IN IP4 192.168.101.43 >> t=0 0 >> a=msid-semantic: WMS PLEThV3VtmQWGIk4ICYTkWiv3AtV43mJ >> m=audio 27200 RTP/SAVPF 0 126 106 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:126 telephone-event/8000 >> a=rtpmap:106 CN/8000 >> a=ptime:20 >> a=sendrecv >> a=fingerprint:sha-256 E2:56:BC:E3:1E:CA:AD:55:04:E5:94:8F:D6:AD:1B:CA:E0:B8:90:A9:62:35:95:F0:F1:C0:D7:A7:92:35:D9:C8 >> a=rtcp-mux >> a=rtcp:27200 IN IP4 192.168.101.43 >> a=ssrc:1443731118 cname:FQsNBmyVzeI5MT0k >> a=ssrc:1443731118 msid:PLEThV3VtmQWGIk4ICYTkWiv3AtV43mJ a0 >> a=ssrc:1443731118 mslabel:PLEThV3VtmQWGIk4ICYTkWiv3AtV43mJ >> a=ssrc:1443731118 label:PLEThV3VtmQWGIk4ICYTkWiv3AtV43mJa0 >> a=ice-ufrag:lO4wBcSvCchx5t6B >> a=ice-pwd:9peBBCn0GWWiFMtRrndSqVeT >> a=candidate:9761529334 1 udp 659136 192.168.101.43 27200 typ host generation 0 >> >> >> 2014-12-18 09:19:46.047277 [DEBUG] sofia.c:6624 Remote SDP: >> v=0 >> o=1001 8000 8000 IN IP4 192.168.101.39 >> s=SIP Call >> c=IN IP4 192.168.101.39 >> t=0 0 >> m=audio 5004 RTP/AVP 0 13 >> a=sendrecv >> a=rtpmap:0 PCMU/8000 >> a=ptime:20 >> >> >> Remote SDP: >> 2014-12-18 09:19:46.067273 [DEBUG] mod_sofia.c:780 Local SDP sofia/internal/1002 at 192.168.101.43 : >> v=0 >> o=FreeSWITCH 1418885182 1418885184 IN IP4 192.168.101.43 >> s=FreeSWITCH >> c=IN IP4 192.168.101.43 >> t=0 0 >> a=msid-semantic: WMS PLEThV3VtmQWGIk4ICYTkWiv3AtV43mJ >> m=audio 27200 RTP/SAVPF 0 126 106 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:126 telephone-event/8000 >> a=rtpmap:106 CN/8000 >> a=ptime:20 >> a=sendrecv >> a=fingerprint:sha-256 E2:56:BC:E3:1E:CA:AD:55:04:E5:94:8F:D6:AD:1B:CA:E0:B8:90:A9:62:35:95:F0:F1:C0:D7:A7:92:35:D9:C8 >> a=rtcp-mux >> a=rtcp:27200 IN IP4 192.168.101.43 >> a=ssrc:1443731118 cname:FQsNBmyVzeI5MT0k >> a=ssrc:1443731118 msid:PLEThV3VtmQWGIk4ICYTkWiv3AtV43mJ a0 >> a=ssrc:1443731118 mslabel:PLEThV3VtmQWGIk4ICYTkWiv3AtV43mJ >> a=ssrc:1443731118 label:PLEThV3VtmQWGIk4ICYTkWiv3AtV43mJa0 >> a=ice-ufrag:lO4wBcSvCchx5t6B >> a=ice-pwd:9peBBCn0GWWiFMtRrndSqVeT >> a=candidate:7802155548 1 udp 659136 192.168.101.43 27200 typ host generation 0 >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141223/458d5cda/attachment-0001.html From iwada.bassey at gmail.com Tue Dec 23 21:24:57 2014 From: iwada.bassey at gmail.com (Iwada Eja) Date: Tue, 23 Dec 2014 19:24:57 +0100 Subject: [Freeswitch-users] No package speex-devel available. In-Reply-To: <022001d01eb2$e564e0f0$b02ea2d0$@gfphelps.com> References: <022001d01eb2$e564e0f0$b02ea2d0$@gfphelps.com> Message-ID: Hi, You might need the development packages Maybe, yum -y install speex-devel On Tue, Dec 23, 2014 at 2:18 PM, George F. Phelps wrote: > I am attempting to build Freeswitch, using these instructions: > > https://freeswitch.org/confluence/display/FREESWITCH/CentOS+6 > > How do I resolve these two "yum install" errors? > > No package speex-devel available. > > No package libedit-devel available. > > And even though "speex" is installed (yum: Package > speex-1.2-0.19.rc1.el7.x86_64 already installed and latest version), I am > getting this build error: > > checking for speex >= 1.2rc1 speexdsp >= 1.2rc1... Package speex > was > not found in the pkg-config search path. > > What is the workaround for this error? > > Thanks! > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 Iwada -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141223/46e1743b/attachment.html From bilaln018 at gmail.com Tue Dec 23 21:23:20 2014 From: bilaln018 at gmail.com (Bilal Abbasi) Date: Tue, 23 Dec 2014 23:23:20 +0500 Subject: [Freeswitch-users] [ERR][mod_json_cdr.c][Got error [500] posting to web server] Message-ID: Hi All, I am using Mod_json to store cdr in MySQLDB, i have successfully configured that and data is successfully inserted in Database. But i got this error on CLI. I don't know why i get this error and how can i get ride of that. *2014-12-23 10:03:53.157185 [ERR] mod_json_cdr.c:396 Got error [500] posting to web server [http://localhost/cgi-bin/bilal.py ]* *2014-12-23 10:03:53.157185 [ERR] mod_json_cdr.c:403 Retry will be with url [http://localhost/cgi-bin/bilal.py ]* *2014-12-23 10:03:53.157185 [ERR] mod_json_cdr.c:414 Unable to post to web server* Regards Bilal Abbasi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141223/25ac9981/attachment.html From mike at jerris.com Tue Dec 23 21:50:07 2014 From: mike at jerris.com (Michael Jerris) Date: Tue, 23 Dec 2014 13:50:07 -0500 Subject: [Freeswitch-users] [ERR][mod_json_cdr.c][Got error [500] posting to web server] In-Reply-To: References: Message-ID: are you actually getting a 500 posting to the web server? > On Dec 23, 2014, at 1:23 PM, Bilal Abbasi wrote: > > Hi All, > I am using Mod_json to store cdr in MySQLDB, i have successfully configured that and data is successfully inserted in Database. > But i got this error on CLI. > I don't know why i get this error and how can i get ride of that. > > 2014-12-23 10:03:53.157185 [ERR] mod_json_cdr.c:396 Got error [500] posting to web server [http://localhost/cgi-bin/bilal.py ] > 2014-12-23 10:03:53.157185 [ERR] mod_json_cdr.c:403 Retry will be with url [http://localhost/cgi-bin/bilal.py ] > 2014-12-23 10:03:53.157185 [ERR] mod_json_cdr.c:414 Unable to post to web server > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141223/d4fa6864/attachment.html From iskren.hadzhinedev at ikiji.com Tue Dec 23 22:01:06 2014 From: iskren.hadzhinedev at ikiji.com (Iskren Hadzhinedev) Date: Tue, 23 Dec 2014 21:01:06 +0200 Subject: [Freeswitch-users] [ERR][mod_json_cdr.c][Got error [500] posting to web server] In-Reply-To: References: Message-ID: <3ccdb447-a73b-4a70-a386-91fe25781376@getmailbird.com> Hello, the web server expects the script to run without error and get any output from it to push back to the client (in this case - freeswitch). If your script works (I assume it does, because you said that the data gets in the database), then using?a simple 'print "OK"' in the end of the script will fix it. Cheers, Iskren Hadzhinedev On 23.12.2014 ?. 20:40:49, Bilal Abbasi wrote: Hi All, I am using Mod_json to store cdr in MySQLDB, i have successfully configured that and data is successfully inserted in Database. But i got this error on CLI. I don't know why i get this error and how can i get ride of that. 2014-12-23 10:03:53.157185 [ERR] mod_json_cdr.c:396 Got error [500] posting to web server [http://localhost/cgi-bin/bilal.py [http://localhost/cgi-bin/bilal.py]] 2014-12-23 10:03:53.157185 [ERR] mod_json_cdr.c:403 Retry will be with url [http://localhost/cgi-bin/bilal.py [http://localhost/cgi-bin/bilal.py]] 2014-12-23 10:03:53.157185 [ERR] mod_json_cdr.c:414 Unable to post to web server Regards Bilal Abbasi _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141223/b4903ab3/attachment.html From david.villasmil at gmail.com Tue Dec 23 22:03:36 2014 From: david.villasmil at gmail.com (David Villasmil Govea) Date: Tue, 23 Dec 2014 20:03:36 +0100 Subject: [Freeswitch-users] Unbridge and send to queue In-Reply-To: References: Message-ID: Any ideas? On Dec 23, 2014 5:48 PM, "David Villasmil Govea" wrote: > Hello Guys, > > I receive a call from side A and send it out to side B. > I need to unbridge this call once it is answered and send the B side to a > queue to give it moh. How do I do that? > > I'm doing everything with lua and I've tried: > > on my dialplan I have: > > > > in check_answered-lua, after checking the call was in fact answered I do: > > session:execute("transfer", "-bleg 9999 XML default"); > > and on my dialplan I have: > > > > data="/usr/local/freeswitch/scripts/queue.lua"/> > > > > queue.lua: > > session:execute("sched_hangup","+50 alloted_timeout"); > session:execute("callcenter","agents_queue"); > > session:execute("sleep",my_dur); > > But this doesn't seem to work at all... > > Can anyone give me a hand? thanks! > > -- > DVG > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141223/cc522261/attachment.html From GeorgePhelps at gfphelps.com Tue Dec 23 22:47:09 2014 From: GeorgePhelps at gfphelps.com (George F. Phelps) Date: Tue, 23 Dec 2014 14:47:09 -0500 Subject: [Freeswitch-users] No package speex-devel available. In-Reply-To: References: <022001d01eb2$e564e0f0$b02ea2d0$@gfphelps.com> Message-ID: <030601d01ee9$3d7d27d0$b8777770$@gfphelps.com> I tried ?yum -y install speex-devel?. That produces the ?No package speex-devel available? error message. Any other suggestions? Thanks! From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Iwada Eja Sent: Tuesday, December 23, 2014 1:25 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] No package speex-devel available. Hi, You might need the development packages Maybe, yum -y install speex-devel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141223/3195b0d2/attachment-0001.html From david.villasmil at gmail.com Tue Dec 23 22:48:55 2014 From: david.villasmil at gmail.com (David Villasmil Govea) Date: Tue, 23 Dec 2014 20:48:55 +0100 Subject: [Freeswitch-users] No package speex-devel available. In-Reply-To: <030601d01ee9$3d7d27d0$b8777770$@gfphelps.com> References: <022001d01eb2$e564e0f0$b02ea2d0$@gfphelps.com> <030601d01ee9$3d7d27d0$b8777770$@gfphelps.com> Message-ID: Hello, Did you search? Regards, On Dec 23, 2014 8:48 PM, "George F. Phelps" wrote: > I tried ?yum -y install speex-devel?. That produces the ?No package > speex-devel available? error message. > > > > Any other suggestions? > > > > Thanks! > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Iwada Eja > *Sent:* Tuesday, December 23, 2014 1:25 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] No package speex-devel available. > > > > Hi, > > You might need the development packages > > > > Maybe, yum -y install speex-devel > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141223/523d9a9a/attachment.html From GeorgePhelps at gfphelps.com Tue Dec 23 22:59:27 2014 From: GeorgePhelps at gfphelps.com (George F. Phelps) Date: Tue, 23 Dec 2014 14:59:27 -0500 Subject: [Freeswitch-users] No package speex-devel available. In-Reply-To: References: <022001d01eb2$e564e0f0$b02ea2d0$@gfphelps.com> <030601d01ee9$3d7d27d0$b8777770$@gfphelps.com> Message-ID: <032401d01eea$f54932e0$dfdb98a0$@gfphelps.com> Yes, see below: $ yum search speex Loaded plugins: rhui-lb ================================== N/S matched: speex ============================== speex.i686 : A voice compression format (codec) speex.x86_64 : A voice compression format (codec) Name and summary matches only, use "search all" for everything. $ George From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David Villasmil Govea Sent: Tuesday, December 23, 2014 2:49 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] No package speex-devel available. Hello, Did you search? Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141223/deaf64d1/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/pkcs7-signature Size: 6528 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141223/deaf64d1/attachment.bin From shabbirabbasi92 at gmail.com Tue Dec 23 23:01:56 2014 From: shabbirabbasi92 at gmail.com (Shabbir abbasi) Date: Wed, 24 Dec 2014 01:01:56 +0500 Subject: [Freeswitch-users] No package speex-devel available. In-Reply-To: <032401d01eea$f54932e0$dfdb98a0$@gfphelps.com> References: <022001d01eb2$e564e0f0$b02ea2d0$@gfphelps.com> <030601d01ee9$3d7d27d0$b8777770$@gfphelps.com> <032401d01eea$f54932e0$dfdb98a0$@gfphelps.com> Message-ID: please read this may be this will help https://centos.org/forums/viewtopic.php?f=20&t=20877 On Wed, Dec 24, 2014 at 12:59 AM, George F. Phelps < GeorgePhelps at gfphelps.com> wrote: > Yes, see below: > > > > $ yum search speex > > Loaded plugins: rhui-lb > > ================================== N/S matched: speex > ============================== > > speex.i686 : A voice compression format (codec) > > speex.x86_64 : A voice compression format (codec) > > > > Name and summary matches only, use "search all" for everything. > > > > $ > > > > George > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *David > Villasmil Govea > *Sent:* Tuesday, December 23, 2014 2:49 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] No package speex-devel available. > > > > Hello, > > Did you search? > > Regards, > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141224/f20ce489/attachment-0001.html From david.villasmil at gmail.com Tue Dec 23 23:04:26 2014 From: david.villasmil at gmail.com (David Villasmil Govea) Date: Tue, 23 Dec 2014 21:04:26 +0100 Subject: [Freeswitch-users] No package speex-devel available. In-Reply-To: References: <022001d01eb2$e564e0f0$b02ea2d0$@gfphelps.com> <030601d01ee9$3d7d27d0$b8777770$@gfphelps.com> <032401d01eea$f54932e0$dfdb98a0$@gfphelps.com> Message-ID: In voip-info.org it is recommended as: yum -y install speex speex-devel http://www.voip-info.org/wiki/view/Speex Regards, On Dec 23, 2014 9:02 PM, "Shabbir abbasi" wrote: > please read this > > may be this will help > https://centos.org/forums/viewtopic.php?f=20&t=20877 > > On Wed, Dec 24, 2014 at 12:59 AM, George F. Phelps < > GeorgePhelps at gfphelps.com> wrote: > >> Yes, see below: >> >> >> >> $ yum search speex >> >> Loaded plugins: rhui-lb >> >> ================================== N/S matched: speex >> ============================== >> >> speex.i686 : A voice compression format (codec) >> >> speex.x86_64 : A voice compression format (codec) >> >> >> >> Name and summary matches only, use "search all" for everything. >> >> >> >> $ >> >> >> >> George >> >> >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *David >> Villasmil Govea >> *Sent:* Tuesday, December 23, 2014 2:49 PM >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] No package speex-devel available. >> >> >> >> Hello, >> >> Did you search? >> >> Regards, >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141223/2372a01c/attachment.html From matt at williamsmjw.com Tue Dec 23 23:06:23 2014 From: matt at williamsmjw.com (Matt Williams) Date: Tue, 23 Dec 2014 14:06:23 -0600 Subject: [Freeswitch-users] [Mod CDR CSV] Adding Timestamp into the CDR In-Reply-To: <8D142526-7FDD-411B-BEEB-77B9F6C57FDB@gmail.com> References: <8D142526-7FDD-411B-BEEB-77B9F6C57FDB@gmail.com> Message-ID: ${end_stamp} of the call should be pretty close to the write time unless you are holding the call open for some reason. ${strftime()} should give you what you need. Thank You, Matthew Williams IKN Network Operations On Mon, Dec 22, 2014 at 3:41 AM, Massimo Varriale wrote: > Hi Guys, > I would like to write a field into the CSV with the timestamp of the CDR > so I can know the writetime. > The timestamp should be in this format "2014-12-22 10:32". > Is this possible? > I digged the wiki but I didn't found any clue regarding this. > > Thank you > Max > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141223/cf904e90/attachment.html From mike at jerris.com Tue Dec 23 23:08:33 2014 From: mike at jerris.com (Michael Jerris) Date: Tue, 23 Dec 2014 15:08:33 -0500 Subject: [Freeswitch-users] Unbridge and send to queue In-Reply-To: References: Message-ID: If you need to un-bridge it as soon as its answered, why not just send it to the right destination in the first place? > On Dec 23, 2014, at 11:48 AM, David Villasmil Govea wrote: > > Hello Guys, > > I receive a call from side A and send it out to side B. > I need to unbridge this call once it is answered and send the B side to a queue to give it moh. How do I do that? > > I'm doing everything with lua and I've tried: > > on my dialplan I have: > > > > in check_answered-lua, after checking the call was in fact answered I do: > > session:execute("transfer", "-bleg 9999 XML default"); > > and on my dialplan I have: > > > > > > > > queue.lua: > > session:execute("sched_hangup","+50 alloted_timeout"); > session:execute("callcenter","agents_queue"); > > session:execute("sleep",my_dur); > > But this doesn't seem to work at all... > > Can anyone give me a hand? thanks! > > -- > DVG > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141223/9462dccf/attachment.html From mike at jerris.com Tue Dec 23 23:11:03 2014 From: mike at jerris.com (Michael Jerris) Date: Tue, 23 Dec 2014 15:11:03 -0500 Subject: [Freeswitch-users] No package speex-devel available. In-Reply-To: <030601d01ee9$3d7d27d0$b8777770$@gfphelps.com> References: <022001d01eb2$e564e0f0$b02ea2d0$@gfphelps.com> <030601d01ee9$3d7d27d0$b8777770$@gfphelps.com> Message-ID: <4742EA66-2295-4E5E-A2BE-5CFDE75CBF63@jerris.com> speex-devel is in base in centos, so you might not have the regular repositories enabled. you should review your yum configuration as it appears you have broken that. > On Dec 23, 2014, at 2:47 PM, George F. Phelps wrote: > > I tried ?yum -y install speex-devel?. That produces the ?No package speex-devel available? error message. > > Any other suggestions? > > Thanks! > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Iwada Eja > Sent: Tuesday, December 23, 2014 1:25 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] No package speex-devel available. > > Hi, > You might need the development packages > > Maybe, yum -y install speex-devel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141223/4d6d53e8/attachment-0001.html From ab928 at itu.edu.pk Tue Dec 23 23:27:08 2014 From: ab928 at itu.edu.pk (Abdul Mannan Butt) Date: Wed, 24 Dec 2014 01:27:08 +0500 Subject: [Freeswitch-users] [Outbound call using GSMOpen] help needed for research on Maternal Health Message-ID: Hi Freeswitch-Users, We are basically working on a research project on Maternal Health, in which we have to call to women and instruct them about the diet at each stage of their pregnancy to reduce mortality rate due to pregnancy complications. Could anyone please help us in implementing an outbound call? We are trying the following code but we are unable to make an outbound call properly and play a sound file when call get received by the called person. *Code:* -------------------------------------------------------------------------------- audio_file_path = "ivr/MESSAGE_TO_PLAY.wav";freeswitch.consoleLog("INFO","Initiating call\n"); *-- *this will call the person and move to next line of call get accepted/rejected etc *session = freeswitch.Session("{ignore_early_media=true}gsmopen/gsm01/03454329512");* *-- *Checking session status session:consoleLog("info", "checking session status\n"); -- Stream audio if session is ready if(session:ready() == true) then session:consoleLog("info", "ready to stream\n"); session:streamFile(audio_file_path); session:sleep(250); session:hangup();end -------------------------------------------------------------------------------- But the problem is *session = freeswitch.Session("{ignore_early_media=true}gsmopen/gsm01/03454329512"); *is not returning the control to the below code until user disconnect the call. At that time session got invalid. please help, how could we do Call-Out properly. We have posted the following issue n Jira but we remain unable to get our work done. https://freeswitch.org/jira/browse/FS-7101 https://freeswitch.org/jira/browse/FS-7057 Please help us to make an outbound call and play a audio message. We will be very thankful to you. Kind Regards, Amna Batool and Abdul Mannan, Information Technology University, Lahore Pakistan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141224/ef8c64ca/attachment.html From david.villasmil at gmail.com Wed Dec 24 00:16:21 2014 From: david.villasmil at gmail.com (David Villasmil Govea) Date: Tue, 23 Dec 2014 22:16:21 +0100 Subject: [Freeswitch-users] Unbridge and send to queue In-Reply-To: References: Message-ID: Hello, Thanks for answering. Not as soon as it is answered, but after doing some checks I do with lua. regards, David On Tue, Dec 23, 2014 at 9:08 PM, Michael Jerris wrote: > If you need to un-bridge it as soon as its answered, why not just send it > to the right destination in the first place? > > On Dec 23, 2014, at 11:48 AM, David Villasmil Govea < > david.villasmil at gmail.com> wrote: > > Hello Guys, > > I receive a call from side A and send it out to side B. > I need to unbridge this call once it is answered and send the B side to a > queue to give it moh. How do I do that? > > I'm doing everything with lua and I've tried: > > on my dialplan I have: > > > > in check_answered-lua, after checking the call was in fact answered I do: > > session:execute("transfer", "-bleg 9999 XML default"); > > and on my dialplan I have: > > > > data="/usr/local/freeswitch/scripts/queue.lua"/> > > > > queue.lua: > > session:execute("sched_hangup","+50 alloted_timeout"); > session:execute("callcenter","agents_queue"); > > session:execute("sleep",my_dur); > > But this doesn't seem to work at all... > > Can anyone give me a hand? thanks! > > -- > DVG > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- DVG -- Imagination is more important than knowledge Albert Einstein -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141223/e00afd5f/attachment.html From mike at jerris.com Wed Dec 24 00:28:57 2014 From: mike at jerris.com (Michael Jerris) Date: Tue, 23 Dec 2014 16:28:57 -0500 Subject: [Freeswitch-users] [Outbound call using GSMOpen] help needed for research on Maternal Health In-Reply-To: References: Message-ID: <98E3D393-C33F-4210-898F-AC9D1B5DBEDD@jerris.com> I would just do this in dialplan instead of in lua, and just use the api interface originate command to create the call. > On Dec 23, 2014, at 3:27 PM, Abdul Mannan Butt wrote: > > Hi Freeswitch-Users, > > We are basically working on a research project on Maternal Health, in which we have to call to women and instruct them about the diet at each stage of their pregnancy to reduce mortality rate due to pregnancy complications. > > Could anyone please help us in implementing an outbound call? We are trying the following code but we are unable to make an outbound call properly and play a sound file when call get received by the called person. > > Code: > -------------------------------------------------------------------------------- > audio_file_path = "ivr/MESSAGE_TO_PLAY.wav"; > freeswitch.consoleLog("INFO","Initiating call\n"); > -- this will call the person and move to next line of call get accepted/rejected etc > session = freeswitch.Session("{ignore_early_media=true}gsmopen/gsm01/03454329512"); > -- Checking session status > session:consoleLog("info", "checking session status\n"); > -- Stream audio if session is ready > if(session:ready() == true) then > session:consoleLog("info", "ready to stream\n"); > session:streamFile(audio_file_path); > session:sleep(250); > session:hangup(); > end > -------------------------------------------------------------------------------- > > But the problem is > session = freeswitch.Session("{ignore_early_media=true}gsmopen/gsm01/03454329512"); is not returning the control to the below code until user disconnect the call. At that time session got invalid. please help, how could we do Call-Out properly. We have posted the following issue n Jira but we remain unable to get our work done. > https://freeswitch.org/jira/browse/FS-7101 > https://freeswitch.org/jira/browse/FS-7057 > > Please help us to make an outbound call and play a audio message. We will be very thankful to you. > > Kind Regards, > > Amna Batool and Abdul Mannan, > Information Technology University, > Lahore Pakistan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141223/665a3bcf/attachment.html From luis.daniel.lucio at gmail.com Wed Dec 24 04:17:47 2014 From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz) Date: Tue, 23 Dec 2014 20:17:47 -0500 Subject: [Freeswitch-users] Receives call From Unknown Extensions In-Reply-To: <5498A176.4040009@virtues.net> References: <54989CA6.3030900@virtues.net> <5498A176.4040009@virtues.net> Message-ID: Dont worry, your are a target of a kiddy script. As you dont use numeric extensions, they wont authenticate. And as you are using multitenant, they should be targering the IP (as domain, for example 100 at 1.1.1.1) instead 100 at yourdomain. So they wont be able to authenticate (if multidomain is on). CDR will still show the failled call. Its normal, FS is reporting a failed attempt. 2014-12-22 17:55 GMT-05:00 Thomas Auge : > To eliminate the guessing, check the logs which route the calls took through the system. It should contain the clues you > need. You might need to up the log level a bit ... > > > On 22.12.2014 19:44, Lloyd Aloysius wrote: >> Fail2Ban is running in the system >> >> I do not have any default dial plans or extensions. >> >> >> >> >> >> On Mon, Dec 22, 2014 at 5:35 PM, Thomas Auge > wrote: >> >> Do you still have the external domain enabled? I think it routes external calls matching a specific number theme ( >> ^(10[01][0-9])$ ) to the internal users through the pre-installed dialplan. It listens on different ports (5080/1). >> Config is in sip_profiles/external.xml and dialplan/public.xml. >> >> I see an insane amount of brute force attempts against our PBX', so if there is a way to get anywhere, you can expect >> people to try it - over and over and over ... I can recommend fail2ban. :-) >> >> Just guessing though, if I'm wrong, someone more knowledgeable will probably chime in. :) >> >> >> On 22.12.2014 19:16, Lloyd Aloysius wrote: >> > Hi All >> > >> > I have a multi domain setup. We receive calls from unknown extensions (eg: 100 , 101,1000,1007 etc ).But there is no >> > voice in it. >> > >> > We do not have any default extensions in the system and all default extensions removed from the system. >> > >> > Users are authenticated by alphanumeric (like an email username) Eg: mike at mydomain.com >> and passwords are very >> > complicated. >> > >> > How someone can call a user without authentication from these extensions? >> > >> > Please let me know how to solve this issue. >> > >> > Thanks Lloyd >> > >> > >> > >> > >> > >> > _________________________________________________________________________ Professional FreeSWITCH Consulting >> > Services: consulting at freeswitch.org http://www.freeswitchsolutions.com >> > >> > Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org >> > >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 auge at virtues.net Wed Dec 24 04:26:49 2014 From: auge at virtues.net (Thomas Auge) Date: Tue, 23 Dec 2014 22:26:49 -0300 Subject: [Freeswitch-users] Differrent codec parameters (fmtp) per leg Message-ID: <549A1659.8060408@virtues.net> Is it a supported configuration to have different formats for the same codec on each end? I.e. for Opus if one leg sends stereo and the other doesn't, or if they send with different bitrates? I built a webRTC app a while ago, which works like that, and it's not a problem in Chrome. But as I started to work with Freeswitch I realized this is probably not the way it is meant to be done. :-) Would it cause issue or do some codecs not care what they are being fed for decoding? From nbhatti at gmail.com Wed Dec 24 11:04:10 2014 From: nbhatti at gmail.com (Muhammad Naseer Bhatti) Date: Wed, 24 Dec 2014 11:04:10 +0300 Subject: [Freeswitch-users] Receives call From Unknown Extensions In-Reply-To: References: <54989CA6.3030900@virtues.net> <5498A176.4040009@virtues.net> Message-ID: No, he mentioned below ? ...I have a ?multi domain setup. We receive calls from unknown extensions (eg: 100 , 101,1000,1007 etc ).But there is no voice in it? So it look like the call is able to come in. He needs to look into the logs to figure out what?s going on. --? Muhammad Naseer Bhatti From:?Luis Daniel Lucio Quiroz Reply:?FreeSWITCH Users Help > Date:?December 24, 2014 at 4:19:26 AM To:?FreeSWITCH Users Help > Subject:? Re: [Freeswitch-users] Receives call From Unknown Extensions Dont worry, your are a target of a kiddy script. As you dont use numeric extensions, they wont authenticate. And as you are using multitenant, they should be targering the IP (as domain, for example 100 at 1.1.1.1) instead 100 at yourdomain. So they wont be able to authenticate (if multidomain is on). CDR will still show the failled call. Its normal, FS is reporting a failed attempt. 2014-12-22 17:55 GMT-05:00 Thomas Auge : > To eliminate the guessing, check the logs which route the calls took through the system. It should contain the clues you > need. You might need to up the log level a bit ... > > > On 22.12.2014 19:44, Lloyd Aloysius wrote: >> Fail2Ban is running in the system >> >> I do not have any default dial plans or extensions. >> >> >> >> >> >> On Mon, Dec 22, 2014 at 5:35 PM, Thomas Auge > wrote: >> >> Do you still have the external domain enabled? I think it routes external calls matching a specific number theme ( >> ^(10[01][0-9])$ ) to the internal users through the pre-installed dialplan. It listens on different ports (5080/1). >> Config is in sip_profiles/external.xml and dialplan/public.xml. >> >> I see an insane amount of brute force attempts against our PBX', so if there is a way to get anywhere, you can expect >> people to try it - over and over and over ... I can recommend fail2ban. :-) >> >> Just guessing though, if I'm wrong, someone more knowledgeable will probably chime in. :) >> >> >> On 22.12.2014 19:16, Lloyd Aloysius wrote: >> > Hi All >> > >> > I have a multi domain setup. We receive calls from unknown extensions (eg: 100 , 101,1000,1007 etc ).But there is no >> > voice in it. >> > >> > We do not have any default extensions in the system and all default extensions removed from the system. >> > >> > Users are authenticated by alphanumeric (like an email username) Eg: mike at mydomain.com >> and passwords are very >> > complicated. >> > >> > How someone can call a user without authentication from these extensions? >> > >> > Please let me know how to solve this issue. >> > >> > Thanks Lloyd >> > >> > >> > >> > >> > >> > _________________________________________________________________________ Professional FreeSWITCH Consulting >> > Services: consulting at freeswitch.org http://www.freeswitchsolutions.com >> > >> > Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org >> > >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141224/6e8b5653/attachment.html From mvar78 at gmail.com Wed Dec 24 12:33:28 2014 From: mvar78 at gmail.com (Massimo Varriale) Date: Wed, 24 Dec 2014 10:33:28 +0100 Subject: [Freeswitch-users] [Mod CDR CSV] Adding Timestamp into the CDR In-Reply-To: References: <8D142526-7FDD-411B-BEEB-77B9F6C57FDB@gmail.com> Message-ID: HI Matt! Thank you so much for your solution! I will work on strftime because it's really more similar to the write time of the CDR, and also I already have the field end_stamp. Thank you again and as Christmas is approaching....Merry Christmas to you (and to the User List too!) Cheers Max Il giorno 23/dic/2014, alle ore 21:06, Matt Williams ha scritto: > ${end_stamp} of the call should be pretty close to the write time unless you are holding the call open for some reason. ${strftime()} should give you what you need. > > > > Thank You, > Matthew Williams > IKN Network Operations > > > On Mon, Dec 22, 2014 at 3:41 AM, Massimo Varriale wrote: > Hi Guys, > I would like to write a field into the CSV with the timestamp of the CDR so I can know the writetime. > The timestamp should be in this format "2014-12-22 10:32". > Is this possible? > I digged the wiki but I didn't found any clue regarding this. > > Thank you > Max > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141224/945f5a3b/attachment-0001.html From bilaln018 at gmail.com Wed Dec 24 13:17:20 2014 From: bilaln018 at gmail.com (Bilal Abbasi) Date: Wed, 24 Dec 2014 15:17:20 +0500 Subject: [Freeswitch-users] [ERR][mod_json_cdr.c][Got error [500] posting to web server] Message-ID: Hi Iskren Hadzhinedev, Yes i am getting every thing perfectly in my database as required. I have tried print "OK" at end of my script,but no luck. Michael Jerris, Actually freeswitch POST JSON data to python cgi script. Freeswitch is getting error 500 while POST. Regards Bilal Abbasi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141224/34274ee9/attachment.html From david.villasmil at gmail.com Wed Dec 24 13:50:36 2014 From: david.villasmil at gmail.com (David Villasmil Govea) Date: Wed, 24 Dec 2014 11:50:36 +0100 Subject: [Freeswitch-users] Receives call From Unknown Extensions In-Reply-To: References: <54989CA6.3030900@virtues.net> <5498A176.4040009@virtues.net> Message-ID: Hello, At the very least I would say need to install and configure fail2ban urgently, it block ips which try to authenticate and fail. This saves you from brute - force attacks. Regards, David On Dec 24, 2014 2:19 AM, "Luis Daniel Lucio Quiroz" < luis.daniel.lucio at gmail.com> wrote: > Dont worry, your are a target of a kiddy script. As you dont use > numeric extensions, they wont authenticate. And as you are using > multitenant, they should be targering the IP (as domain, for example > 100 at 1.1.1.1) instead 100 at yourdomain. So they wont be able to > authenticate (if multidomain is on). > > CDR will still show the failled call. Its normal, FS is reporting a > failed attempt. > > 2014-12-22 17:55 GMT-05:00 Thomas Auge : > > To eliminate the guessing, check the logs which route the calls took > through the system. It should contain the clues you > > need. You might need to up the log level a bit ... > > > > > > On 22.12.2014 19:44, Lloyd Aloysius wrote: > >> Fail2Ban is running in the system > >> > >> I do not have any default dial plans or extensions. > >> > >> > >> > >> > >> > >> On Mon, Dec 22, 2014 at 5:35 PM, Thomas Auge auge at virtues.net>> wrote: > >> > >> Do you still have the external domain enabled? I think it routes > external calls matching a specific number theme ( > >> ^(10[01][0-9])$ ) to the internal users through the pre-installed > dialplan. It listens on different ports (5080/1). > >> Config is in sip_profiles/external.xml and dialplan/public.xml. > >> > >> I see an insane amount of brute force attempts against our PBX', so > if there is a way to get anywhere, you can expect > >> people to try it - over and over and over ... I can recommend > fail2ban. :-) > >> > >> Just guessing though, if I'm wrong, someone more knowledgeable will > probably chime in. :) > >> > >> > >> On 22.12.2014 19:16, Lloyd Aloysius wrote: > >> > Hi All > >> > > >> > I have a multi domain setup. We receive calls from unknown > extensions (eg: 100 , 101,1000,1007 etc ).But there is no > >> > voice in it. > >> > > >> > We do not have any default extensions in the system and all > default extensions removed from the system. > >> > > >> > Users are authenticated by alphanumeric (like an email username) > Eg: mike at mydomain.com > >> and passwords are very > >> > complicated. > >> > > >> > How someone can call a user without authentication from these > extensions? > >> > > >> > Please let me know how to solve this issue. > >> > > >> > Thanks Lloyd > >> > > >> > > >> > > >> > > >> > > >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting > >> > Services: consulting at freeswitch.org consulting at freeswitch.org> http://www.freeswitchsolutions.com > >> > > >> > Official FreeSWITCH Sites http://www.freeswitch.org > http://confluence.freeswitch.org http://www.cluecon.com > >> > > >> > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org FreeSWITCH-users at lists.freeswitch.org> > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >> > > >> > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://confluence.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org FreeSWITCH-users at lists.freeswitch.org> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > >> > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://confluence.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://confluence.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141224/537833ed/attachment.html From nbhatti at gmail.com Wed Dec 24 13:54:13 2014 From: nbhatti at gmail.com (Muhammad Naseer Bhatti) Date: Wed, 24 Dec 2014 13:54:13 +0300 Subject: [Freeswitch-users] Receives call From Unknown Extensions In-Reply-To: References: <54989CA6.3030900@virtues.net> <5498A176.4040009@virtues.net> Message-ID: The IPs seems not to be failing that?s why he?s able to receive calls. I am not sure if fail2ban would help much except to look into the logs what dial plan and context and from where the calls are coming in. Maybe there is a profile where auth-calls is false and calls are coming in from that profile. But again, without the log file, it?s not easy to help. --? Muhammad Naseer Bhatti From:?David Villasmil Govea Reply:?FreeSWITCH Users Help > Date:?December 24, 2014 at 1:51:32 PM To:?FreeSWITCH Users Help > Subject:? Re: [Freeswitch-users] Receives call From Unknown Extensions Hello, At the very least I would say need to install and configure fail2ban urgently,? it block ips which try to authenticate and fail. This saves you from brute - force attacks. Regards, David On Dec 24, 2014 2:19 AM, "Luis Daniel Lucio Quiroz" wrote: Dont worry, your are a target of a kiddy script. As you dont use numeric extensions, they wont authenticate.? And as you are using multitenant, they should be targering the IP (as domain, for example 100 at 1.1.1.1) instead 100 at yourdomain.? So they wont be able to authenticate (if multidomain is on). CDR will still show the failled call. Its normal, FS is reporting a failed attempt. 2014-12-22 17:55 GMT-05:00 Thomas Auge : > To eliminate the guessing, check the logs which route the calls took through the system. It should contain the clues you > need. You might need to up the log level a bit ... > > > On 22.12.2014 19:44, Lloyd Aloysius wrote: >> Fail2Ban is running in the system >> >> I do not have any default dial plans or extensions. >> >> >> >> >> >> On Mon, Dec 22, 2014 at 5:35 PM, Thomas Auge > wrote: >> >>? ? ?Do you still have the external domain enabled? I think it routes external calls matching a specific number theme ( >>? ? ?^(10[01][0-9])$ ) to the internal users through the pre-installed dialplan. It listens on different ports (5080/1). >>? ? ?Config is in sip_profiles/external.xml and dialplan/public.xml. >> >>? ? ?I see an insane amount of brute force attempts against our PBX', so if there is a way to get anywhere, you can expect >>? ? ?people to try it - over and over and over ... I can recommend fail2ban. :-) >> >>? ? ?Just guessing though, if I'm wrong, someone more knowledgeable will probably chime in. :) >> >> >>? ? ?On 22.12.2014 19:16, Lloyd Aloysius wrote: >>? ? ? > Hi All >>? ? ? > >>? ? ? > I have a? multi domain setup. We receive calls from unknown extensions (eg: 100 , 101,1000,1007 etc ).But there is no >>? ? ? >? voice in it. >>? ? ? > >>? ? ? > We do not have any default extensions in the system and all default extensions removed from the system. >>? ? ? > >>? ? ? > Users are authenticated by alphanumeric (like an email username) Eg: mike at mydomain.com >>? ? ?and passwords are very >>? ? ? > complicated. >>? ? ? > >>? ? ? > How someone can call a user without authentication from these extensions? >>? ? ? > >>? ? ? > Please let me know how to solve this issue. >>? ? ? > >>? ? ? > Thanks Lloyd >>? ? ? > >>? ? ? > >>? ? ? > >>? ? ? > >>? ? ? > >>? ? ? > _________________________________________________________________________ Professional FreeSWITCH Consulting >>? ? ? > Services: consulting at freeswitch.org http://www.freeswitchsolutions.com >>? ? ? > >>? ? ? > Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com >>? ? ? > >>? ? ? > FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org >>? ? ? > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>? ? ? > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org >>? ? ? > >> >> >>? ? ?_________________________________________________________________________ >>? ? ?Professional FreeSWITCH Consulting Services: >>? ? ?consulting at freeswitch.org >>? ? ?http://www.freeswitchsolutions.com >> >>? ? ?Official FreeSWITCH Sites >>? ? ?http://www.freeswitch.org >>? ? ?http://confluence.freeswitch.org >>? ? ?http://www.cluecon.com >> >>? ? ?FreeSWITCH-users mailing list >>? ? ?FreeSWITCH-users at lists.freeswitch.org >>? ? ?http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>? ? ?UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>? ? ?http://www.freeswitch.org >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141224/ac8d3f68/attachment-0001.html From mujahid at ictinnovations.com Wed Dec 24 13:57:31 2014 From: mujahid at ictinnovations.com (Mujahid Ali) Date: Wed, 24 Dec 2014 15:57:31 +0500 Subject: [Freeswitch-users] Zoiper registration on freeswitch Message-ID: hi all, i install freeswitch via yum from http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#YUM_Based_Installation and zoiper on my system (CentOS 6) but when i tried to register Zoiper to freeswitch with defaults user like username: 1001 password: 1001 and click on register i got this on freeswitch cli *freeswitch at internal> 2014-12-24 05:41:18.421648 [CONSOLE] mod_voicemail.c:4066 Event Thread Started* and on zoiper *SIP 403 - Forbidden.* but zoiper not register even i disable SElinux and got same thing. how can i register zoiper with my freeswitch ? thanks in advance. NOTE: all installation is on my local PC freeswitch and zoiper etc. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141224/dcfdde20/attachment.html From shishko69 at gmail.com Wed Dec 24 14:09:38 2014 From: shishko69 at gmail.com (Denis Papes) Date: Wed, 24 Dec 2014 12:09:38 +0100 Subject: [Freeswitch-users] Zoiper registration on freeswitch In-Reply-To: References: Message-ID: Default password should be 1234 On Wed, Dec 24, 2014 at 11:57 AM, Mujahid Ali wrote: > hi all, > i install freeswitch via yum from > > http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#YUM_Based_Installation > and zoiper on my system (CentOS 6) > but when i tried to register Zoiper to freeswitch with defaults user like > username: 1001 > password: 1001 > and click on register i got this on freeswitch cli > *freeswitch at internal> 2014-12-24 05:41:18.421648 [CONSOLE] > mod_voicemail.c:4066 Event Thread Started* > and on zoiper > *SIP 403 - Forbidden.* > but zoiper not register even i disable SElinux and got same thing. > how can i register zoiper with my freeswitch ? > thanks in advance. > NOTE: all installation is on my local PC freeswitch and zoiper etc. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141224/7abfb696/attachment.html From mujahid at ictinnovations.com Wed Dec 24 14:18:22 2014 From: mujahid at ictinnovations.com (Mujahid Ali) Date: Wed, 24 Dec 2014 16:18:22 +0500 Subject: [Freeswitch-users] Zoiper registration on freeswitch In-Reply-To: References: Message-ID: here is 1001 user configuration On Wed, Dec 24, 2014 at 4:09 PM, Denis Papes wrote: > Default password should be 1234 > > On Wed, Dec 24, 2014 at 11:57 AM, Mujahid Ali > wrote: > >> hi all, >> i install freeswitch via yum from >> >> http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#YUM_Based_Installation >> and zoiper on my system (CentOS 6) >> but when i tried to register Zoiper to freeswitch with defaults user like >> username: 1001 >> password: 1001 >> and click on register i got this on freeswitch cli >> *freeswitch at internal> 2014-12-24 05:41:18.421648 [CONSOLE] >> mod_voicemail.c:4066 Event Thread Started* >> and on zoiper >> *SIP 403 - Forbidden.* >> but zoiper not register even i disable SElinux and got same thing. >> how can i register zoiper with my freeswitch ? >> thanks in advance. >> NOTE: all installation is on my local PC freeswitch and zoiper etc. >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141224/abe53cc5/attachment.html From david.villasmil at gmail.com Wed Dec 24 14:21:04 2014 From: david.villasmil at gmail.com (David Villasmil Govea) Date: Wed, 24 Dec 2014 12:21:04 +0100 Subject: [Freeswitch-users] Zoiper registration on freeswitch In-Reply-To: References: Message-ID: The default password is in vars.xml On Dec 24, 2014 12:19 PM, "Mujahid Ali" wrote: > here is 1001 user configuration > > > > > > > > > > > > > > value="$${outbound_caller_name}"/> > value="$${outbound_caller_id}"/> > > > > > > On Wed, Dec 24, 2014 at 4:09 PM, Denis Papes wrote: > >> Default password should be 1234 >> >> On Wed, Dec 24, 2014 at 11:57 AM, Mujahid Ali > > wrote: >> >>> hi all, >>> i install freeswitch via yum from >>> >>> http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#YUM_Based_Installation >>> and zoiper on my system (CentOS 6) >>> but when i tried to register Zoiper to freeswitch with defaults user >>> like >>> username: 1001 >>> password: 1001 >>> and click on register i got this on freeswitch cli >>> *freeswitch at internal> 2014-12-24 05:41:18.421648 [CONSOLE] >>> mod_voicemail.c:4066 Event Thread Started* >>> and on zoiper >>> *SIP 403 - Forbidden.* >>> but zoiper not register even i disable SElinux and got same thing. >>> how can i register zoiper with my freeswitch ? >>> thanks in advance. >>> NOTE: all installation is on my local PC freeswitch and zoiper etc. >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141224/490c3d08/attachment-0001.html From shishko69 at gmail.com Wed Dec 24 14:22:50 2014 From: shishko69 at gmail.com (Denis Papes) Date: Wed, 24 Dec 2014 12:22:50 +0100 Subject: [Freeswitch-users] Zoiper registration on freeswitch In-Reply-To: References: Message-ID: There is line in vars.xml 1001 is voicemail password On Wed, Dec 24, 2014 at 12:21 PM, David Villasmil Govea < david.villasmil at gmail.com> wrote: > The default password is in vars.xml > On Dec 24, 2014 12:19 PM, "Mujahid Ali" > wrote: > >> here is 1001 user configuration >> >> >> >> >> >> >> >> >> >> >> >> >> >> > value="$${outbound_caller_name}"/> >> > value="$${outbound_caller_id}"/> >> >> >> >> >> >> On Wed, Dec 24, 2014 at 4:09 PM, Denis Papes wrote: >> >>> Default password should be 1234 >>> >>> On Wed, Dec 24, 2014 at 11:57 AM, Mujahid Ali < >>> mujahid at ictinnovations.com> wrote: >>> >>>> hi all, >>>> i install freeswitch via yum from >>>> >>>> http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#YUM_Based_Installation >>>> and zoiper on my system (CentOS 6) >>>> but when i tried to register Zoiper to freeswitch with defaults user >>>> like >>>> username: 1001 >>>> password: 1001 >>>> and click on register i got this on freeswitch cli >>>> *freeswitch at internal> 2014-12-24 05:41:18.421648 [CONSOLE] >>>> mod_voicemail.c:4066 Event Thread Started* >>>> and on zoiper >>>> *SIP 403 - Forbidden.* >>>> but zoiper not register even i disable SElinux and got same thing. >>>> how can i register zoiper with my freeswitch ? >>>> thanks in advance. >>>> NOTE: all installation is on my local PC freeswitch and zoiper etc. >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141224/dac779a2/attachment.html From david.villasmil at gmail.com Wed Dec 24 14:26:16 2014 From: david.villasmil at gmail.com (David Villasmil Govea) Date: Wed, 24 Dec 2014 12:26:16 +0100 Subject: [Freeswitch-users] Zoiper registration on freeswitch In-Reply-To: References: Message-ID: So that's the default_password, I would recommend changing it immediately to one of your one. Also setting a custom password on all users, or removing all and just leaving 1001 with a different password, if you're testing. Remember to restart freeswitch after changing vars.xml On Wed, Dec 24, 2014 at 12:22 PM, Denis Papes wrote: > There is line in vars.xml > > > > 1001 is voicemail password > > On Wed, Dec 24, 2014 at 12:21 PM, David Villasmil Govea < > david.villasmil at gmail.com> wrote: > >> The default password is in vars.xml >> On Dec 24, 2014 12:19 PM, "Mujahid Ali" >> wrote: >> >>> here is 1001 user configuration >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >> value="$${outbound_caller_name}"/> >>> >> value="$${outbound_caller_id}"/> >>> >>> >>> >>> >>> >>> On Wed, Dec 24, 2014 at 4:09 PM, Denis Papes >>> wrote: >>> >>>> Default password should be 1234 >>>> >>>> On Wed, Dec 24, 2014 at 11:57 AM, Mujahid Ali < >>>> mujahid at ictinnovations.com> wrote: >>>> >>>>> hi all, >>>>> i install freeswitch via yum from >>>>> >>>>> http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#YUM_Based_Installation >>>>> and zoiper on my system (CentOS 6) >>>>> but when i tried to register Zoiper to freeswitch with defaults user >>>>> like >>>>> username: 1001 >>>>> password: 1001 >>>>> and click on register i got this on freeswitch cli >>>>> *freeswitch at internal> 2014-12-24 05:41:18.421648 [CONSOLE] >>>>> mod_voicemail.c:4066 Event Thread Started* >>>>> and on zoiper >>>>> *SIP 403 - Forbidden.* >>>>> but zoiper not register even i disable SElinux and got same thing. >>>>> how can i register zoiper with my freeswitch ? >>>>> thanks in advance. >>>>> NOTE: all installation is on my local PC freeswitch and zoiper etc. >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- DVG -- Imagination is more important than knowledge Albert Einstein -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141224/e3f8e904/attachment-0001.html From mujahid at ictinnovations.com Wed Dec 24 14:26:56 2014 From: mujahid at ictinnovations.com (Mujahid Ali) Date: Wed, 24 Dec 2014 16:26:56 +0500 Subject: [Freeswitch-users] Zoiper registration on freeswitch In-Reply-To: References: Message-ID: Thanks, problem solved :) On Wed, Dec 24, 2014 at 4:22 PM, Denis Papes wrote: > There is line in vars.xml > > > > 1001 is voicemail password > > On Wed, Dec 24, 2014 at 12:21 PM, David Villasmil Govea < > david.villasmil at gmail.com> wrote: > >> The default password is in vars.xml >> On Dec 24, 2014 12:19 PM, "Mujahid Ali" >> wrote: >> >>> here is 1001 user configuration >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >> value="$${outbound_caller_name}"/> >>> >> value="$${outbound_caller_id}"/> >>> >>> >>> >>> >>> >>> On Wed, Dec 24, 2014 at 4:09 PM, Denis Papes >>> wrote: >>> >>>> Default password should be 1234 >>>> >>>> On Wed, Dec 24, 2014 at 11:57 AM, Mujahid Ali < >>>> mujahid at ictinnovations.com> wrote: >>>> >>>>> hi all, >>>>> i install freeswitch via yum from >>>>> >>>>> http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#YUM_Based_Installation >>>>> and zoiper on my system (CentOS 6) >>>>> but when i tried to register Zoiper to freeswitch with defaults user >>>>> like >>>>> username: 1001 >>>>> password: 1001 >>>>> and click on register i got this on freeswitch cli >>>>> *freeswitch at internal> 2014-12-24 05:41:18.421648 [CONSOLE] >>>>> mod_voicemail.c:4066 Event Thread Started* >>>>> and on zoiper >>>>> *SIP 403 - Forbidden.* >>>>> but zoiper not register even i disable SElinux and got same thing. >>>>> how can i register zoiper with my freeswitch ? >>>>> thanks in advance. >>>>> NOTE: all installation is on my local PC freeswitch and zoiper etc. >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141224/1de6f11d/attachment.html From mujahid at ictinnovations.com Wed Dec 24 15:16:02 2014 From: mujahid at ictinnovations.com (Mujahid Ali) Date: Wed, 24 Dec 2014 17:16:02 +0500 Subject: [Freeswitch-users] Zoiper registration on freeswitch In-Reply-To: References: Message-ID: thanks David Villasmil Govea for these info. On Wed, Dec 24, 2014 at 4:26 PM, David Villasmil Govea < david.villasmil at gmail.com> wrote: > So that's the default_password, I would recommend changing it immediately > to one of your one. > Also setting a custom password on all users, or removing all and just > leaving 1001 with a different password, if you're testing. > > Remember to restart freeswitch after changing vars.xml > > On Wed, Dec 24, 2014 at 12:22 PM, Denis Papes wrote: > >> There is line in vars.xml >> >> >> >> 1001 is voicemail password >> >> On Wed, Dec 24, 2014 at 12:21 PM, David Villasmil Govea < >> david.villasmil at gmail.com> wrote: >> >>> The default password is in vars.xml >>> On Dec 24, 2014 12:19 PM, "Mujahid Ali" >>> wrote: >>> >>>> here is 1001 user configuration >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>> value="$${outbound_caller_name}"/> >>>> >>> value="$${outbound_caller_id}"/> >>>> >>>> >>>> >>>> >>>> >>>> On Wed, Dec 24, 2014 at 4:09 PM, Denis Papes >>>> wrote: >>>> >>>>> Default password should be 1234 >>>>> >>>>> On Wed, Dec 24, 2014 at 11:57 AM, Mujahid Ali < >>>>> mujahid at ictinnovations.com> wrote: >>>>> >>>>>> hi all, >>>>>> i install freeswitch via yum from >>>>>> >>>>>> http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#YUM_Based_Installation >>>>>> and zoiper on my system (CentOS 6) >>>>>> but when i tried to register Zoiper to freeswitch with defaults user >>>>>> like >>>>>> username: 1001 >>>>>> password: 1001 >>>>>> and click on register i got this on freeswitch cli >>>>>> *freeswitch at internal> 2014-12-24 05:41:18.421648 [CONSOLE] >>>>>> mod_voicemail.c:4066 Event Thread Started* >>>>>> and on zoiper >>>>>> *SIP 403 - Forbidden.* >>>>>> but zoiper not register even i disable SElinux and got same thing. >>>>>> how can i register zoiper with my freeswitch ? >>>>>> thanks in advance. >>>>>> NOTE: all installation is on my local PC freeswitch and zoiper etc. >>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://confluence.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > DVG > > -- > Imagination is more important than knowledge > Albert Einstein > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141224/38af55bf/attachment-0001.html From amirehsansadeghi at gmail.com Wed Dec 24 14:33:22 2014 From: amirehsansadeghi at gmail.com (Amirehsan Sadeghi) Date: Wed, 24 Dec 2014 15:03:22 +0330 Subject: [Freeswitch-users] Calling Card Solution With Freeswitch Message-ID: Dear All , I want install and configure a calling card system with FreeSwitch And FreeRadius i after install and configure this servers in my pc Please Tell me how i do for run a local calling Card system ? -- Best regards Amirehsan Sadeghi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141224/2c0e6aa2/attachment.html From luis.daniel.lucio at gmail.com Wed Dec 24 15:46:08 2014 From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz) Date: Wed, 24 Dec 2014 07:46:08 -0500 Subject: [Freeswitch-users] Calling Card Solution With Freeswitch In-Reply-To: References: Message-ID: Contact me offline, I have the Billing & Callingcard scripts for FusionPBX, if that works for you 2014-12-24 6:33 GMT-05:00 Amirehsan Sadeghi : > Dear All , > > I want install and configure a calling card system with FreeSwitch And > FreeRadius > > i after install and configure this servers in my pc > > Please Tell me how i do for run a local calling Card system ? > > > > > > -- > Best regards > Amirehsan Sadeghi > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 max at nysolutions.com Wed Dec 24 17:20:10 2014 From: max at nysolutions.com (Moishe Grunstein) Date: Wed, 24 Dec 2014 14:20:10 +0000 Subject: [Freeswitch-users] Calling Card Solution With Freeswitch In-Reply-To: References: Message-ID: You should use http://lists.freeswitch.org/mailman/listinfo/freeswitch-biz to make business offers. Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support at nysolutions.com Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Luis Daniel Lucio Quiroz Sent: Wednesday, December 24, 2014 7:46 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Calling Card Solution With Freeswitch Contact me offline, I have the Billing & Callingcard scripts for FusionPBX, if that works for you 2014-12-24 6:33 GMT-05:00 Amirehsan Sadeghi : > Dear All , > > I want install and configure a calling card system with FreeSwitch And > FreeRadius > > i after install and configure this servers in my pc > > Please Tell me how i do for run a local calling Card system ? > > > > > > -- > Best regards > Amirehsan Sadeghi > > > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users 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://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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 GeorgePhelps at gfphelps.com Wed Dec 24 19:39:36 2014 From: GeorgePhelps at gfphelps.com (George F. Phelps) Date: Wed, 24 Dec 2014 11:39:36 -0500 Subject: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable Message-ID: <040101d01f98$34ca7d40$9e5f77c0$@gfphelps.com> I am debugging a new/initial Freeswitch configuration. I believe that I have successfully registered with my VoIP provider - "State=REGED". I am able to dial from one extension (x1000) to a different extension (x1001), but after answering, there is NO AUDIO at either end of the call. Problem #1. When I test call to extension x9196, for example, I get an immediate hang-up and SIP response of "SIP/2.0 480 Temporarily Unavailable". Problem #2. Do I have to do anything to enable calling to x9196? And when I attempt to call an external phone number via my VoIP provider, I get the same immediate hang-up and SIP response. Problem #3. I am getting this critical error on startup. Problem #4. 2014-12-24 11:29:09.869357 [CRIT] switch_loadable_module.c:1447 Error Loading module /usr/local/freeswitch/mod/mod_v8.so **/usr/local/freeswitch/mod/mod_v8.so: cannot open shared object file: No such file or directory** Any suggestions as to what configuration might be wrong? Or how I can get additional debug information? Version info: FreeSWITCH Version 1.5.15b+git~20141222T221908Z~067cb0f0f2~64bit (git 067cb0f 2014-12-22 22:19:08Z 64bit) Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141224/15b6e8a9/attachment.html From GeorgePhelps at gfphelps.com Wed Dec 24 19:42:24 2014 From: GeorgePhelps at gfphelps.com (George F. Phelps) Date: Wed, 24 Dec 2014 11:42:24 -0500 Subject: [Freeswitch-users] No package speex-devel available. In-Reply-To: <4742EA66-2295-4E5E-A2BE-5CFDE75CBF63@jerris.com> References: <022001d01eb2$e564e0f0$b02ea2d0$@gfphelps.com> <030601d01ee9$3d7d27d0$b8777770$@gfphelps.com> <4742EA66-2295-4E5E-A2BE-5CFDE75CBF63@jerris.com> Message-ID: <040c01d01f98$98a3dc30$c9eb9490$@gfphelps.com> Problem resolved. By manually locating, downloading, and installing the missing RPM packages. Thanks! From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris Sent: Tuesday, December 23, 2014 3:11 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] No package speex-devel available. speex-devel is in base in centos, so you might not have the regular repositories enabled. you should review your yum configuration as it appears you have broken that. On Dec 23, 2014, at 2:47 PM, George F. Phelps wrote: I tried ?yum -y install speex-devel?. That produces the ?No package speex-devel available? error message. Any other suggestions? Thanks! From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Iwada Eja Sent: Tuesday, December 23, 2014 1:25 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] No package speex-devel available. Hi, You might need the development packages Maybe, yum -y install speex-devel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141224/c0bfc48b/attachment-0001.html From krice at freeswitch.org Wed Dec 24 21:57:20 2014 From: krice at freeswitch.org (Ken Rice) Date: Wed, 24 Dec 2014 18:57:20 +0000 Subject: [Freeswitch-users] FreeSWITCH Week in Review (Master Branch) December 14th-20th Message-ID: <549b0c905cb6d_76d97953247025@ip-10-183-1-212.mail> New Post on freeswitch.org from kathleen check it out at http://ift.tt/1xcyBeB FreeSWITCH Week in Review (Master Branch) December 14th-20th Hello, again. This week in the FreeSWITCH master branch we had 21 commits. The features for this week are: add bert stats to mod_bert::lost_sync event , vs2010 support for recent unimrcp changes , setting rtp_has_crypto for dtls calls, and the creation of uuid_drop_dtmf . New features that were added: 17574a8 Add bert stats to mod_bert::lost_sync event a26e29c vs2010 support for recent unimrcp changes cee8b30 Set rtp_has_crypto for dtls calls 5fcff50 FS-7093 Create uuid_drop_dtmf [Jira: http://ift.tt/1xcyz6f] In terms of stability these were the use cases that were fixed: d5119a7 FS-7091 Removed unnecessary mutex lock inside input component?s cleanup function since the input component won?t be cleaned up unless all references have been released, in mod_rayo [Jira: http://ift.tt/1xcyz6h] Improvements in cross platform build supports: 357ffad Fix windows build error 0b414a8 vs2010 unimrcp working build 0c1e698 Update build deps for debian list 0a0b926 Build fix for gcc 4.9 fixing a variable set but not used error in mod_commands The following bugs were squashed: 4ce2ce3 FS-7092 Fixed bug with Comrex OPUS [Jira: http://ift.tt/1xMpbcg] d786490 Fix timestamps in mod_bert broken by the cpu improvements refactoring ba016c2 FS-7095 Fix for FS sending DTLS HELLO (and STUN binding request) to wrong port [Jira: http://ift.tt/1xcyz6l] e0dcd17 FS-7083 #comment patch to change mod_shout to use lame_encode_buffer_interleaved on stereo channels so we don?t have to mess with the input data [Jira: http://ift.tt/1xMpbci] 326289c FS-7083 This patch adds a dedicated thread for writing to the file and the channel_variable RECORD_USE_THREAD=false will disable it and sync may still be maintained at the cost of dropping more data from the audio signal. [Jira: http://ift.tt/1xMpbci] -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141224/ca03b950/attachment.html From mike at jerris.com Thu Dec 25 04:50:49 2014 From: mike at jerris.com (Michael Jerris) Date: Wed, 24 Dec 2014 20:50:49 -0500 Subject: [Freeswitch-users] [ERR][mod_json_cdr.c][Got error [500] posting to web server] In-Reply-To: References: Message-ID: Okay. so freeswitch is fine, your script is erroring out. You will need to correct the error in your script. > On Dec 24, 2014, at 5:17 AM, Bilal Abbasi wrote: > > Hi Iskren Hadzhinedev, > Yes i am getting every thing perfectly in my database as required. > I have tried print "OK" at end of my script,but no luck. > > Michael Jerris, > Actually freeswitch POST JSON data to python cgi script. > Freeswitch is getting error 500 while POST. > > Regards > Bilal Abbasi From igorolhovskiy at gmail.com Thu Dec 25 14:40:35 2014 From: igorolhovskiy at gmail.com (Igor Olhovskiy) Date: Thu, 25 Dec 2014 13:40:35 +0200 Subject: [Freeswitch-users] mod_verto and FireFox 34 Message-ID: <549BF7B3.20902@gmail.com> Hi! Seems to be mod_verto have no sound with Firefox 34. Even on demo site.... From auge at virtues.net Thu Dec 25 19:07:39 2014 From: auge at virtues.net (Thomas Auge) Date: Thu, 25 Dec 2014 13:07:39 -0300 Subject: [Freeswitch-users] Problem with libspandsp in current master Message-ID: <549C364B.4020300@virtues.net> Hello list, I can't compile the current (date of email) master version: make[5]: Entering directory `/root/freeswitch/src/mod/endpoints/mod_skypopen' CCLD mod_skypopen.la gcc: error: /root/freeswitch/libs/spandsp/src/.libs/libspandsp.a: No such file or directory make[5]: *** [mod_skypopen.la] Error 1 Same for mod_spandsp. I won't pretend to understand what links to what or how, however I'm not sure if it's looking in the right place. The libspandsd-dev package (0.0.6~pre20-3.1) is installed on the system. Shouldn't it use that instead of trying to find it in the freeswitch tree? I ran make distclean, bootstrap, and configure after the pull. Thanks, Thomas From poliv78 at yahoo.co.uk Thu Dec 25 17:51:22 2014 From: poliv78 at yahoo.co.uk (poliv78 at yahoo.co.uk) Date: Thu, 25 Dec 2014 16:51:22 +0200 Subject: [Freeswitch-users] How to execute lua script on register? In-Reply-To: <549BF7B3.20902@gmail.com> References: <549BF7B3.20902@gmail.com> Message-ID: <377656158.20141225165122@yahoo.co.uk> Hi all Is it possible to execute lua script when gateway try to register to know if it successfull or failed? The same question as for internal users. Thanks. -- ? ?????????, Poliv78 mailto:poliv78 at yahoo.co.uk From krice at freeswitch.org Thu Dec 25 21:30:50 2014 From: krice at freeswitch.org (Ken Rice) Date: Thu, 25 Dec 2014 12:30:50 -0600 Subject: [Freeswitch-users] Problem with libspandsp in current master In-Reply-To: <549C364B.4020300@virtues.net> Message-ID: you can not use system spandsp. FreeSWITCH depends on and includes the latest version of SpanDSP. The Author of SpanDSP commits what is required directly to tree. Run "make spandsp-reconf" and them make again if that doesn't work, make distclean probably only did a partial clean. if you have checked out from git, git clean -fdx and start over at bootstrap K On 12/25/14 10:07 AM, "Thomas Auge" wrote: > Hello list, > > I can't compile the current (date of email) master version: > > make[5]: Entering directory `/root/freeswitch/src/mod/endpoints/mod_skypopen' > CCLD mod_skypopen.la > gcc: error: /root/freeswitch/libs/spandsp/src/.libs/libspandsp.a: No such file > or directory > make[5]: *** [mod_skypopen.la] Error 1 > > Same for mod_spandsp. > > I won't pretend to understand what links to what or how, however I'm not sure > if it's looking in the right place. The > libspandsd-dev package (0.0.6~pre20-3.1) is installed on the system. Shouldn't > it use that instead of trying to find it > in the freeswitch tree? > > I ran make distclean, bootstrap, and configure after the pull. > > Thanks, > > Thomas > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 Twitter: @FreeSWITCH From auge at virtues.net Thu Dec 25 21:45:58 2014 From: auge at virtues.net (Thomas Auge) Date: Thu, 25 Dec 2014 15:45:58 -0300 Subject: [Freeswitch-users] Problem with libspandsp in current master In-Reply-To: References: Message-ID: <549C5B66.10208@virtues.net> spandsp-reconf worked, thanks! On 25.12.2014 15:30, Ken Rice wrote: > you can not use system spandsp. FreeSWITCH depends on and includes the > latest version of SpanDSP. The Author of SpanDSP commits what is required > directly to tree. > > Run "make spandsp-reconf" and them make again if that doesn't work, make > distclean probably only did a partial clean. if you have checked out from > git, git clean -fdx and start over at bootstrap > > K > > > On 12/25/14 10:07 AM, "Thomas Auge" wrote: > >> Hello list, >> >> I can't compile the current (date of email) master version: >> >> make[5]: Entering directory `/root/freeswitch/src/mod/endpoints/mod_skypopen' >> CCLD mod_skypopen.la >> gcc: error: /root/freeswitch/libs/spandsp/src/.libs/libspandsp.a: No such file >> or directory >> make[5]: *** [mod_skypopen.la] Error 1 >> >> Same for mod_spandsp. >> >> I won't pretend to understand what links to what or how, however I'm not sure >> if it's looking in the right place. The >> libspandsd-dev package (0.0.6~pre20-3.1) is installed on the system. Shouldn't >> it use that instead of trying to find it >> in the freeswitch tree? >> >> I ran make distclean, bootstrap, and configure after the pull. >> >> Thanks, >> >> Thomas >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 adam.ben.ayoun1 at gmail.com Thu Dec 25 21:55:39 2014 From: adam.ben.ayoun1 at gmail.com (Adam Ben-Ayoun) Date: Thu, 25 Dec 2014 20:55:39 +0200 Subject: [Freeswitch-users] FreeSwitch and WebRTC Message-ID: Hi, We are using FreeSwitch as an MCU for audio-only WebRTC (basically, users connect to conference rooms). We have full control over the client application (it's a mobile app we are developing) and we solely rely on the client to choose the codec and do all the intelligent media related stuff. My question is, how can we minimize the audio processing/transcoding on FreeSwitch/mod_conference and mitigate the related overhead? Can we achieve that with tuning mod_conference's "rate" and "interval" parameters in that sense? Any other recommendations regarding FreeSwitch and WebRTC? Thanks, Adam -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141225/6c6775ad/attachment.html From luis.daniel.lucio at gmail.com Fri Dec 26 00:24:12 2014 From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz) Date: Thu, 25 Dec 2014 16:24:12 -0500 Subject: [Freeswitch-users] How to execute lua script on register? In-Reply-To: <377656158.20141225165122@yahoo.co.uk> References: <549BF7B3.20902@gmail.com> <377656158.20141225165122@yahoo.co.uk> Message-ID: Are you trying to do monitoring? If you are, there are easier ways to do that On Dec 25, 2014 1:21 PM, wrote: > Hi all > > Is it possible to execute lua script when gateway try to register > to know if it successfull or failed? > > The same question as for internal users. > Thanks. > > > -- > ? ?????????, > Poliv78 mailto:poliv78 at yahoo.co.uk > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141225/5c367226/attachment-0001.html From david.villasmil at gmail.com Fri Dec 26 00:33:41 2014 From: david.villasmil at gmail.com (David Villasmil Govea) Date: Thu, 25 Dec 2014 22:33:41 +0100 Subject: [Freeswitch-users] How to execute lua script on register? In-Reply-To: <377656158.20141225165122@yahoo.co.uk> References: <549BF7B3.20902@gmail.com> <377656158.20141225165122@yahoo.co.uk> Message-ID: You could just look at the log for failed attempts. On Dec 25, 2014 1:21 PM, wrote: > Hi all > > Is it possible to execute lua script when gateway try to register > to know if it successfull or failed? > > The same question as for internal users. > Thanks. > > > -- > ? ?????????, > Poliv78 mailto:poliv78 at yahoo.co.uk > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141225/a5fa966d/attachment.html From davidwaf at gmail.com Fri Dec 26 00:52:58 2014 From: davidwaf at gmail.com (David Wafula) Date: Thu, 25 Dec 2014 23:52:58 +0200 Subject: [Freeswitch-users] How to filter chat events Message-ID: Hello all, I wish to filter chat events via ESL but not sure this can be accomplished. I can filter most of other events just fine. Regards, -- David Wafula -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141225/549c77d5/attachment.html From david.villasmil at gmail.com Fri Dec 26 00:58:07 2014 From: david.villasmil at gmail.com (David Villasmil Govea) Date: Thu, 25 Dec 2014 22:58:07 +0100 Subject: [Freeswitch-users] How to filter chat events In-Reply-To: References: Message-ID: Hello, Enable all events and catch a message going by, you can then use its event name to filter. David On Dec 25, 2014 4:54 PM, "David Wafula" wrote: > Hello all, > I wish to filter chat events via ESL but not sure this can be > accomplished. I can filter most of other events just fine. > > Regards, > -- > David Wafula > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141225/5f1d5cb4/attachment.html From davidwaf at gmail.com Fri Dec 26 01:12:59 2014 From: davidwaf at gmail.com (David Wafula) Date: Fri, 26 Dec 2014 00:12:59 +0200 Subject: [Freeswitch-users] How to filter chat events In-Reply-To: References: Message-ID: Thanks. I have tried that, but the strange thing is i dont seem to see anything in event log related to chat messages. Am running: FreeSWITCH Version 1.5.15b+git~20141117T202539Z~424df19083~64bit (git 424df19 2014-11-17 20:25:39Z 64bit) On Thu, Dec 25, 2014 at 11:58 PM, David Villasmil Govea < david.villasmil at gmail.com> wrote: > > Hello, > > Enable all events and catch a message going by, you can then use its event > name to filter. > > David > On Dec 25, 2014 4:54 PM, "David Wafula" wrote: > >> Hello all, >> I wish to filter chat events via ESL but not sure this can be >> accomplished. I can filter most of other events just fine. >> >> Regards, >> -- >> David Wafula >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- David Wafula -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141226/ffca8b04/attachment.html From blasterjr at gmail.com Fri Dec 26 05:24:42 2014 From: blasterjr at gmail.com (Chris Tunbridge) Date: Thu, 25 Dec 2014 19:24:42 -0700 Subject: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable In-Reply-To: <040101d01f98$34ca7d40$9e5f77c0$@gfphelps.com> References: <040101d01f98$34ca7d40$9e5f77c0$@gfphelps.com> Message-ID: 1) Sounds like NAT issue possibly, or incorrect codecs, please elaborate on your topology and configuration 2) If you're using default configs, its configured to look for extensions 10XX, you can see this in conf/dialplan/default.xml (and in conf/dialplan/public.xml for calls coming from the outside) 3) Do you have an outbound route configured that matches your dial string? 4) This just means the module wasn't configured, you can comment out the line in conf/autoload_configs/modules.conf.xml find the line that says mod_v8 and put a 4) The ?mod_v8? issue is now resolved. The module was not being built. I?m not sure why the downloaded default build/install files were not building it, but were attempting to load it. Sounds like a bug to me? Thanks, George From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Chris Tunbridge Sent: Thursday, December 25, 2014 9:25 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable 1) Sounds like NAT issue possibly, or incorrect codecs, please elaborate on your topology and configuration 2) If you're using default configs, its configured to look for extensions 10XX, you can see this in conf/dialplan/default.xml (and in conf/dialplan/public.xml for calls coming from the outside) 3) Do you have an outbound route configured that matches your dial string? 4) This just means the module wasn't configured, you can comment out the line in conf/autoload_configs/modules.conf.xml find the line that says mod_v8 and put a > > > > > > > > > > > > > > > > > > 4) The ?mod_v8? issue is now resolved. The module was not being built. > I?m not sure why the downloaded default build/install files were not > building it, but were attempting to load it. Sounds like a bug to me? > > > > Thanks, > > > > George > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Chris > Tunbridge > *Sent:* Thursday, December 25, 2014 9:25 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable > > > > 1) Sounds like NAT issue possibly, or incorrect codecs, please elaborate > on your topology and configuration > > 2) If you're using default configs, its configured to look for extensions > 10XX, you can see this in conf/dialplan/default.xml (and in > conf/dialplan/public.xml for calls coming from the outside) > > 3) Do you have an outbound route configured that matches your dial string? > > 4) This just means the module wasn't configured, you can comment out the > line in conf/autoload_configs/modules.conf.xml find the line that says > mod_v8 and put a > > > > > > > > > > > > > > > > > > 4) The ?mod_v8? issue is now resolved. The module was not being built. > I?m not sure why the downloaded default build/install files were not > building it, but were attempting to load it. Sounds like a bug to me? > > > > Thanks, > > > > George > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Chris > Tunbridge > *Sent:* Thursday, December 25, 2014 9:25 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable > > > > 1) Sounds like NAT issue possibly, or incorrect codecs, please elaborate > on your topology and configuration > > 2) If you're using default configs, its configured to look for extensions > 10XX, you can see this in conf/dialplan/default.xml (and in > conf/dialplan/public.xml for calls coming from the outside) > > 3) Do you have an outbound route configured that matches your dial string? > > 4) This just means the module wasn't configured, you can comment out the > line in conf/autoload_configs/modules.conf.xml find the line that says > mod_v8 and put a 4) The ?mod_v8? issue is now resolved. The module was not being built. I?m not sure why the downloaded default build/install files were not building it, but were attempting to load it. Sounds like a bug to me? Thanks, George From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Chris Tunbridge Sent: Thursday, December 25, 2014 9:25 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable 1) Sounds like NAT issue possibly, or incorrect codecs, please elaborate on your topology and configuration 2) If you're using default configs, its configured to look for extensions 10XX, you can see this in conf/dialplan/default.xml (and in conf/dialplan/public.xml for calls coming from the outside) 3) Do you have an outbound route configured that matches your dial string? 4) This just means the module wasn't configured, you can comment out the line in conf/autoload_configs/modules.conf.xml find the line that says mod_v8 and put a > > > > > > > > > > > > > > > > > > 4) The ?mod_v8? issue is now resolved. The module was not being built. > I?m not sure why the downloaded default build/install files were not > building it, but were attempting to load it. Sounds like a bug to me? > > > > Thanks, > > > > George > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Chris > Tunbridge > *Sent:* Thursday, December 25, 2014 9:25 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable > > > > 1) Sounds like NAT issue possibly, or incorrect codecs, please elaborate > on your topology and configuration > > 2) If you're using default configs, its configured to look for extensions > 10XX, you can see this in conf/dialplan/default.xml (and in > conf/dialplan/public.xml for calls coming from the outside) > > 3) Do you have an outbound route configured that matches your dial string? > > 4) This just means the module wasn't configured, you can comment out the > line in conf/autoload_configs/modules.conf.xml find the line that says > mod_v8 and put a 4) The ?mod_v8? issue is now resolved. The module was not being built. I?m not sure why the downloaded default build/install files were not building it, but were attempting to load it. Sounds like a bug to me? Thanks, George From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Chris Tunbridge Sent: Thursday, December 25, 2014 9:25 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable 1) Sounds like NAT issue possibly, or incorrect codecs, please elaborate on your topology and configuration 2) If you're using default configs, its configured to look for extensions 10XX, you can see this in conf/dialplan/default.xml (and in conf/dialplan/public.xml for calls coming from the outside) 3) Do you have an outbound route configured that matches your dial string? 4) This just means the module wasn't configured, you can comment out the line in conf/autoload_configs/modules.conf.xml find the line that says mod_v8 and put a > > > > > > > > > > > > > > > > > > 4) The ?mod_v8? issue is now resolved. The module was not being built. > I?m not sure why the downloaded default build/install files were not > building it, but were attempting to load it. Sounds like a bug to me? > > > > Thanks, > > > > George > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Chris > Tunbridge > *Sent:* Thursday, December 25, 2014 9:25 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable > > > > 1) Sounds like NAT issue possibly, or incorrect codecs, please elaborate > on your topology and configuration > > 2) If you're using default configs, its configured to look for extensions > 10XX, you can see this in conf/dialplan/default.xml (and in > conf/dialplan/public.xml for calls coming from the outside) > > 3) Do you have an outbound route configured that matches your dial string? > > 4) This just means the module wasn't configured, you can comment out the > line in conf/autoload_configs/modules.conf.xml find the line that says > mod_v8 and put a My suspicion is that some other dialplan, other than my ?switch2voip.us? dialplan, is being invoked. My SIP Proxy is at 66.33.147.150. IP address ?172.31.33.109? is the local/internal IP address for my AWS virtual cloud server. ?4049392032? is a real phone number ? not an extension. Thanks, George From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David Villasmil Govea Sent: Sunday, December 28, 2014 5:05 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable can you share your dialplan? It looks like you're dialing "To: sip:4049392032 at 172.31.33.109" but have no extension for that... On Sun, Dec 28, 2014 at 10:55 PM, George F. Phelps wrote: New ?pastebin? created: http://pastebin.com/UwmgJGGg George From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David Villasmil Govea Sent: Sunday, December 28, 2014 4:04 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable http://pastebin.com/E4sqTLa4 doesn't show anything. Comes back with "This is a private paste. If you created this paste, please login to view it." On Sun, Dec 28, 2014 at 3:22 PM, George F. Phelps wrote: Chris Tunbridge, 1) I made the updates to my configuration, as suggested in the ?https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2? link. I?m still not able to make a call to an outside number. A call to an extension connects, but there is still no audio. 2) Extension x9161 is one of the default dialplan applications. 3) Call failure log posted at: http://pastebin.com/E4sqTLa4 Thanks, George From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Chris Tunbridge Sent: Saturday, December 27, 2014 2:30 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable 1) This is an issue with the NAT, likely on the freeswitch side, see instructions here: https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2 The important part is the external sip ip and external rtp ip. Without this calls will connect, but audio will not pass. I run dozens of servers on AWS without any issues as long as the external sip and rtp ip's are configured in the sip profile conf/sip_profiles/internal.xml 2) Your issue you said was with extension x9196, is this another sip endpoint or a dialplan application? If this is a sip endpoint, please make some adjustments to the conf/dialplan/default.xml to address extra extensions outside of the 10XX range. 3) Can you post a log here http://pastebin.freeswitch.org of a call attempt? My guess is that something's not matching the request, a complete log of a call attempt would help most here. 4) Glad to hear, its only used if you're using the JavaScript scripting engine for your scripts. On Fri, Dec 26, 2014 at 7:57 AM, George F. Phelps wrote: Chris Tunbridge, et al., 1) Freeswitch is running is running on an Amazon Web Services (AWS) Linux virtual cloud server. I am testing with Bria softphones (both Windows PC and Android smartphone) from my home network (behind a Netgear wireless router). The Freeswitch ?show codecs? command indicates support for ?codec, G.711 ulaw, CORE_PCM_MODULE? ? which is the codec that I am using with Bria. I am able to successfully connect with Bria to my other VoIP services, such as VoIP.ms. 2) I am using mostly a default configuration, i.e., extensions 1000 through 1019 are configured with updated passwords. 3) This is my outbound dialplan. How do I know if this is the dialplan that is actually being used for dialing? It shows up in the ?xml_locate dialplan? output ? but as the very last entry. My guess is that Freeswitch is attempting to us some other (default, example?) gateway instead of my desired (switch2voip.us) gateway. 4) The ?mod_v8? issue is now resolved. The module was not being built. I?m not sure why the downloaded default build/install files were not building it, but were attempting to load it. Sounds like a bug to me? Thanks, George From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Chris Tunbridge Sent: Thursday, December 25, 2014 9:25 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable 1) Sounds like NAT issue possibly, or incorrect codecs, please elaborate on your topology and configuration 2) If you're using default configs, its configured to look for extensions 10XX, you can see this in conf/dialplan/default.xml (and in conf/dialplan/public.xml for calls coming from the outside) 3) Do you have an outbound route configured that matches your dial string? 4) This just means the module wasn't configured, you can comment out the line in conf/autoload_configs/modules.conf.xml find the line that says mod_v8 and put a > > > > > > > > > > > > > > > > > > My suspicion is that some other dialplan, other than my ?switch2voip.us? > dialplan, is being invoked. My SIP Proxy is at 66.33.147.150. IP address > ?172.31.33.109? is the local/internal IP address for my AWS virtual cloud > server. ?4049392032? is a real phone number ? not an extension. > > > > Thanks, > > > > George > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *David > Villasmil Govea > *Sent:* Sunday, December 28, 2014 5:05 PM > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable > > > > can you share your dialplan? It looks like you're dialing > > "To: sip:4049392032 at 172.31.33.109" > > but have no extension for that... > > > > On Sun, Dec 28, 2014 at 10:55 PM, George F. Phelps < > GeorgePhelps at gfphelps.com> wrote: > > New ?pastebin? created: > > > > http://pastebin.com/UwmgJGGg > > > > George > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *David > Villasmil Govea > *Sent:* Sunday, December 28, 2014 4:04 PM > > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable > > > > http://pastebin.com/E4sqTLa4 doesn't show anything. Comes back with "This > is a private paste. If you created this paste, please login > to view it." > > > > On Sun, Dec 28, 2014 at 3:22 PM, George F. Phelps < > GeorgePhelps at gfphelps.com> wrote: > > Chris Tunbridge, > > > > 1) I made the updates to my configuration, as suggested in the ? > https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2? link. > I?m still not able to make a call to an outside number. A call to an > extension connects, but there is still no audio. > > > > 2) Extension x9161 is one of the default dialplan applications. > > > > 3) Call failure log posted at: http://pastebin.com/E4sqTLa4 > > > > Thanks, > > > > George > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Chris > Tunbridge > *Sent:* Saturday, December 27, 2014 2:30 AM > > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable > > > > 1) This is an issue with the NAT, likely on the freeswitch side, see > instructions here: > https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2 The > important part is the external sip ip and external rtp ip. Without this > calls will connect, but audio will not pass. I run dozens of servers on > AWS without any issues as long as the external sip and rtp ip's are > configured in the sip profile conf/sip_profiles/internal.xml > > 2) Your issue you said was with extension x9196, is this another sip > endpoint or a dialplan application? If this is a sip endpoint, please make > some adjustments to the conf/dialplan/default.xml to address extra > extensions outside of the 10XX range. > > 3) Can you post a log here http://pastebin.freeswitch.org of a call > attempt? My guess is that something's not matching the request, a complete > log of a call attempt would help most here. > > 4) Glad to hear, its only used if you're using the JavaScript scripting > engine for your scripts. > > > > On Fri, Dec 26, 2014 at 7:57 AM, George F. Phelps < > GeorgePhelps at gfphelps.com> wrote: > > Chris Tunbridge, et al., > > > > 1) Freeswitch is running is running on an Amazon Web Services (AWS) Linux > virtual cloud server. I am testing with Bria softphones (both Windows PC > and Android smartphone) from my home network (behind a Netgear wireless > router). The Freeswitch ?show codecs? command indicates support for > ?codec, G.711 ulaw, CORE_PCM_MODULE? ? which is the codec that I am using > with Bria. I am able to successfully connect with Bria to my other VoIP > services, such as VoIP.ms. > > > > 2) I am using mostly a default configuration, i.e., extensions 1000 > through 1019 are configured with updated passwords. > > > > 3) This is my outbound dialplan. How do I know if this is the dialplan > that is actually being used for dialing? It shows up in the ?xml_locate > dialplan? output ? but as the very last entry. My guess is that Freeswitch > is attempting to us some other (default, example?) gateway instead of my > desired (switch2voip.us) gateway. > > > > > > > > > > > > > > > > > > > > > > > > 4) The ?mod_v8? issue is now resolved. The module was not being built. > I?m not sure why the downloaded default build/install files were not > building it, but were attempting to load it. Sounds like a bug to me? > > > > Thanks, > > > > George > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Chris > Tunbridge > *Sent:* Thursday, December 25, 2014 9:25 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable > > > > 1) Sounds like NAT issue possibly, or incorrect codecs, please elaborate > on your topology and configuration > > 2) If you're using default configs, its configured to look for extensions > 10XX, you can see this in conf/dialplan/default.xml (and in > conf/dialplan/public.xml for calls coming from the outside) > > 3) Do you have an outbound route configured that matches your dial string? > > 4) This just means the module wasn't configured, you can comment out the > line in conf/autoload_configs/modules.conf.xml find the line that says > mod_v8 and put a >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> My suspicion is that some other dialplan, other than my ?switch2voip.us? >> dialplan, is being invoked. My SIP Proxy is at 66.33.147.150. IP address >> ?172.31.33.109? is the local/internal IP address for my AWS virtual cloud >> server. ?4049392032? is a real phone number ? not an extension. >> >> >> >> Thanks, >> >> >> >> George >> >> >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *David >> Villasmil Govea >> *Sent:* Sunday, December 28, 2014 5:05 PM >> >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable >> >> >> >> can you share your dialplan? It looks like you're dialing >> >> "To: sip:4049392032 at 172.31.33.109" >> >> but have no extension for that... >> >> >> >> On Sun, Dec 28, 2014 at 10:55 PM, George F. Phelps < >> GeorgePhelps at gfphelps.com> wrote: >> >> New ?pastebin? created: >> >> >> >> http://pastebin.com/UwmgJGGg >> >> >> >> George >> >> >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *David >> Villasmil Govea >> *Sent:* Sunday, December 28, 2014 4:04 PM >> >> >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable >> >> >> >> http://pastebin.com/E4sqTLa4 doesn't show anything. Comes back with "This >> is a private paste. If you created this paste, please login >> to view it." >> >> >> >> On Sun, Dec 28, 2014 at 3:22 PM, George F. Phelps < >> GeorgePhelps at gfphelps.com> wrote: >> >> Chris Tunbridge, >> >> >> >> 1) I made the updates to my configuration, as suggested in the ? >> https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2? link. >> I?m still not able to make a call to an outside number. A call to an >> extension connects, but there is still no audio. >> >> >> >> 2) Extension x9161 is one of the default dialplan applications. >> >> >> >> 3) Call failure log posted at: http://pastebin.com/E4sqTLa4 >> >> >> >> Thanks, >> >> >> >> George >> >> >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Chris >> Tunbridge >> *Sent:* Saturday, December 27, 2014 2:30 AM >> >> >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable >> >> >> >> 1) This is an issue with the NAT, likely on the freeswitch side, see >> instructions here: >> https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2 The >> important part is the external sip ip and external rtp ip. Without this >> calls will connect, but audio will not pass. I run dozens of servers on >> AWS without any issues as long as the external sip and rtp ip's are >> configured in the sip profile conf/sip_profiles/internal.xml >> >> 2) Your issue you said was with extension x9196, is this another sip >> endpoint or a dialplan application? If this is a sip endpoint, please make >> some adjustments to the conf/dialplan/default.xml to address extra >> extensions outside of the 10XX range. >> >> 3) Can you post a log here http://pastebin.freeswitch.org of a call >> attempt? My guess is that something's not matching the request, a complete >> log of a call attempt would help most here. >> >> 4) Glad to hear, its only used if you're using the JavaScript scripting >> engine for your scripts. >> >> >> >> On Fri, Dec 26, 2014 at 7:57 AM, George F. Phelps < >> GeorgePhelps at gfphelps.com> wrote: >> >> Chris Tunbridge, et al., >> >> >> >> 1) Freeswitch is running is running on an Amazon Web Services (AWS) >> Linux virtual cloud server. I am testing with Bria softphones (both >> Windows PC and Android smartphone) from my home network (behind a Netgear >> wireless router). The Freeswitch ?show codecs? command indicates support >> for ?codec, G.711 ulaw, CORE_PCM_MODULE? ? which is the codec that I am >> using with Bria. I am able to successfully connect with Bria to my other >> VoIP services, such as VoIP.ms. >> >> >> >> 2) I am using mostly a default configuration, i.e., extensions 1000 >> through 1019 are configured with updated passwords. >> >> >> >> 3) This is my outbound dialplan. How do I know if this is the dialplan >> that is actually being used for dialing? It shows up in the ?xml_locate >> dialplan? output ? but as the very last entry. My guess is that Freeswitch >> is attempting to us some other (default, example?) gateway instead of my >> desired (switch2voip.us) gateway. >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> 4) The ?mod_v8? issue is now resolved. The module was not being built. >> I?m not sure why the downloaded default build/install files were not >> building it, but were attempting to load it. Sounds like a bug to me? >> >> >> >> Thanks, >> >> >> >> George >> >> >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Chris >> Tunbridge >> *Sent:* Thursday, December 25, 2014 9:25 PM >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable >> >> >> >> 1) Sounds like NAT issue possibly, or incorrect codecs, please elaborate >> on your topology and configuration >> >> 2) If you're using default configs, its configured to look for extensions >> 10XX, you can see this in conf/dialplan/default.xml (and in >> conf/dialplan/public.xml for calls coming from the outside) >> >> 3) Do you have an outbound route configured that matches your dial string? >> >> 4) This just means the module wasn't configured, you can comment out the >> line in conf/autoload_configs/modules.conf.xml find the line that says >> mod_v8 and put a My suspicion is that some other dialplan, other than my ?switch2voip.us? dialplan, is being invoked. My SIP Proxy is at 66.33.147.150. IP address ?172.31.33.109? is the local/internal IP address for my AWS virtual cloud server. ?4049392032? is a real phone number ? not an extension. Thanks, George From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David Villasmil Govea Sent: Sunday, December 28, 2014 5:05 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable can you share your dialplan? It looks like you're dialing "To: sip:4049392032 at 172.31.33.109" but have no extension for that... On Sun, Dec 28, 2014 at 10:55 PM, George F. Phelps wrote: New ?pastebin? created: http://pastebin.com/UwmgJGGg George From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David Villasmil Govea Sent: Sunday, December 28, 2014 4:04 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable http://pastebin.com/E4sqTLa4 doesn't show anything. Comes back with "This is a private paste. If you created this paste, please login to view it." On Sun, Dec 28, 2014 at 3:22 PM, George F. Phelps wrote: Chris Tunbridge, 1) I made the updates to my configuration, as suggested in the ?https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2? link. I?m still not able to make a call to an outside number. A call to an extension connects, but there is still no audio. 2) Extension x9161 is one of the default dialplan applications. 3) Call failure log posted at: http://pastebin.com/E4sqTLa4 Thanks, George From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Chris Tunbridge Sent: Saturday, December 27, 2014 2:30 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable 1) This is an issue with the NAT, likely on the freeswitch side, see instructions here: https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2 The important part is the external sip ip and external rtp ip. Without this calls will connect, but audio will not pass. I run dozens of servers on AWS without any issues as long as the external sip and rtp ip's are configured in the sip profile conf/sip_profiles/internal.xml 2) Your issue you said was with extension x9196, is this another sip endpoint or a dialplan application? If this is a sip endpoint, please make some adjustments to the conf/dialplan/default.xml to address extra extensions outside of the 10XX range. 3) Can you post a log here http://pastebin.freeswitch.org of a call attempt? My guess is that something's not matching the request, a complete log of a call attempt would help most here. 4) Glad to hear, its only used if you're using the JavaScript scripting engine for your scripts. On Fri, Dec 26, 2014 at 7:57 AM, George F. Phelps wrote: Chris Tunbridge, et al., 1) Freeswitch is running is running on an Amazon Web Services (AWS) Linux virtual cloud server. I am testing with Bria softphones (both Windows PC and Android smartphone) from my home network (behind a Netgear wireless router). The Freeswitch ?show codecs? command indicates support for ?codec, G.711 ulaw, CORE_PCM_MODULE? ? which is the codec that I am using with Bria. I am able to successfully connect with Bria to my other VoIP services, such as VoIP.ms. 2) I am using mostly a default configuration, i.e., extensions 1000 through 1019 are configured with updated passwords. 3) This is my outbound dialplan. How do I know if this is the dialplan that is actually being used for dialing? It shows up in the ?xml_locate dialplan? output ? but as the very last entry. My guess is that Freeswitch is attempting to us some other (default, example?) gateway instead of my desired (switch2voip.us) gateway. 4) The ?mod_v8? issue is now resolved. The module was not being built. I?m not sure why the downloaded default build/install files were not building it, but were attempting to load it. Sounds like a bug to me? Thanks, George From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Chris Tunbridge Sent: Thursday, December 25, 2014 9:25 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable 1) Sounds like NAT issue possibly, or incorrect codecs, please elaborate on your topology and configuration 2) If you're using default configs, its configured to look for extensions 10XX, you can see this in conf/dialplan/default.xml (and in conf/dialplan/public.xml for calls coming from the outside) 3) Do you have an outbound route configured that matches your dial string? 4) This just means the module wasn't configured, you can comment out the line in conf/autoload_configs/modules.conf.xml find the line that says mod_v8 and put a My suspicion is that some other dialplan, other than my ?switch2voip.us? dialplan, is being invoked. My SIP Proxy is at 66.33.147.150. IP address ?172.31.33.109? is the local/internal IP address for my AWS virtual cloud server. ?4049392032? is a real phone number ? not an extension. Thanks, George From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David Villasmil Govea Sent: Sunday, December 28, 2014 5:05 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable can you share your dialplan? It looks like you're dialing "To: sip:4049392032 at 172.31.33.109" but have no extension for that... On Sun, Dec 28, 2014 at 10:55 PM, George F. Phelps wrote: New ?pastebin? created: http://pastebin.com/UwmgJGGg George From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David Villasmil Govea Sent: Sunday, December 28, 2014 4:04 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable http://pastebin.com/E4sqTLa4 doesn't show anything. Comes back with "This is a private paste. If you created this paste, please login to view it." On Sun, Dec 28, 2014 at 3:22 PM, George F. Phelps wrote: Chris Tunbridge, 1) I made the updates to my configuration, as suggested in the ?https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2? link. I?m still not able to make a call to an outside number. A call to an extension connects, but there is still no audio. 2) Extension x9161 is one of the default dialplan applications. 3) Call failure log posted at: http://pastebin.com/E4sqTLa4 Thanks, George From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Chris Tunbridge Sent: Saturday, December 27, 2014 2:30 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable 1) This is an issue with the NAT, likely on the freeswitch side, see instructions here: https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2 The important part is the external sip ip and external rtp ip. Without this calls will connect, but audio will not pass. I run dozens of servers on AWS without any issues as long as the external sip and rtp ip's are configured in the sip profile conf/sip_profiles/internal.xml 2) Your issue you said was with extension x9196, is this another sip endpoint or a dialplan application? If this is a sip endpoint, please make some adjustments to the conf/dialplan/default.xml to address extra extensions outside of the 10XX range. 3) Can you post a log here http://pastebin.freeswitch.org of a call attempt? My guess is that something's not matching the request, a complete log of a call attempt would help most here. 4) Glad to hear, its only used if you're using the JavaScript scripting engine for your scripts. On Fri, Dec 26, 2014 at 7:57 AM, George F. Phelps wrote: Chris Tunbridge, et al., 1) Freeswitch is running is running on an Amazon Web Services (AWS) Linux virtual cloud server. I am testing with Bria softphones (both Windows PC and Android smartphone) from my home network (behind a Netgear wireless router). The Freeswitch ?show codecs? command indicates support for ?codec, G.711 ulaw, CORE_PCM_MODULE? ? which is the codec that I am using with Bria. I am able to successfully connect with Bria to my other VoIP services, such as VoIP.ms. 2) I am using mostly a default configuration, i.e., extensions 1000 through 1019 are configured with updated passwords. 3) This is my outbound dialplan. How do I know if this is the dialplan that is actually being used for dialing? It shows up in the ?xml_locate dialplan? output ? but as the very last entry. My guess is that Freeswitch is attempting to us some other (default, example?) gateway instead of my desired (switch2voip.us) gateway. 4) The ?mod_v8? issue is now resolved. The module was not being built. I?m not sure why the downloaded default build/install files were not building it, but were attempting to load it. Sounds like a bug to me? Thanks, George From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Chris Tunbridge Sent: Thursday, December 25, 2014 9:25 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable 1) Sounds like NAT issue possibly, or incorrect codecs, please elaborate on your topology and configuration 2) If you're using default configs, its configured to look for extensions 10XX, you can see this in conf/dialplan/default.xml (and in conf/dialplan/public.xml for calls coming from the outside) 3) Do you have an outbound route configured that matches your dial string? 4) This just means the module wasn't configured, you can comment out the line in conf/autoload_configs/modules.conf.xml find the line that says mod_v8 and put a > > > > > > > > > > > > > > > > > > My suspicion is that some other dialplan, other than my ?switch2voip.us? > dialplan, is being invoked. My SIP Proxy is at 66.33.147.150. IP address > ?172.31.33.109? is the local/internal IP address for my AWS virtual cloud > server. ?4049392032? is a real phone number ? not an extension. > > > > Thanks, > > > > George > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *David > Villasmil Govea > *Sent:* Sunday, December 28, 2014 5:05 PM > > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable > > > > can you share your dialplan? It looks like you're dialing > > "To: sip:4049392032 at 172.31.33.109" > > but have no extension for that... > > > > On Sun, Dec 28, 2014 at 10:55 PM, George F. Phelps < > GeorgePhelps at gfphelps.com> wrote: > > New ?pastebin? created: > > > > http://pastebin.com/UwmgJGGg > > > > George > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *David > Villasmil Govea > *Sent:* Sunday, December 28, 2014 4:04 PM > > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable > > > > http://pastebin.com/E4sqTLa4 doesn't show anything. Comes back with "This > is a private paste. If you created this paste, please login > to view it." > > > > On Sun, Dec 28, 2014 at 3:22 PM, George F. Phelps < > GeorgePhelps at gfphelps.com> wrote: > > Chris Tunbridge, > > > > 1) I made the updates to my configuration, as suggested in the ? > https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2? link. > I?m still not able to make a call to an outside number. A call to an > extension connects, but there is still no audio. > > > > 2) Extension x9161 is one of the default dialplan applications. > > > > 3) Call failure log posted at: http://pastebin.com/E4sqTLa4 > > > > Thanks, > > > > George > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Chris > Tunbridge > *Sent:* Saturday, December 27, 2014 2:30 AM > > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable > > > > 1) This is an issue with the NAT, likely on the freeswitch side, see > instructions here: > https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2 The > important part is the external sip ip and external rtp ip. Without this > calls will connect, but audio will not pass. I run dozens of servers on > AWS without any issues as long as the external sip and rtp ip's are > configured in the sip profile conf/sip_profiles/internal.xml > > 2) Your issue you said was with extension x9196, is this another sip > endpoint or a dialplan application? If this is a sip endpoint, please make > some adjustments to the conf/dialplan/default.xml to address extra > extensions outside of the 10XX range. > > 3) Can you post a log here http://pastebin.freeswitch.org of a call > attempt? My guess is that something's not matching the request, a complete > log of a call attempt would help most here. > > 4) Glad to hear, its only used if you're using the JavaScript scripting > engine for your scripts. > > > > On Fri, Dec 26, 2014 at 7:57 AM, George F. Phelps < > GeorgePhelps at gfphelps.com> wrote: > > Chris Tunbridge, et al., > > > > 1) Freeswitch is running is running on an Amazon Web Services (AWS) Linux > virtual cloud server. I am testing with Bria softphones (both Windows PC > and Android smartphone) from my home network (behind a Netgear wireless > router). The Freeswitch ?show codecs? command indicates support for > ?codec, G.711 ulaw, CORE_PCM_MODULE? ? which is the codec that I am using > with Bria. I am able to successfully connect with Bria to my other VoIP > services, such as VoIP.ms. > > > > 2) I am using mostly a default configuration, i.e., extensions 1000 > through 1019 are configured with updated passwords. > > > > 3) This is my outbound dialplan. How do I know if this is the dialplan > that is actually being used for dialing? It shows up in the ?xml_locate > dialplan? output ? but as the very last entry. My guess is that Freeswitch > is attempting to us some other (default, example?) gateway instead of my > desired (switch2voip.us) gateway. > > > > > > > > > > > > > > > > > > > > > > > > 4) The ?mod_v8? issue is now resolved. The module was not being built. > I?m not sure why the downloaded default build/install files were not > building it, but were attempting to load it. Sounds like a bug to me? > > > > Thanks, > > > > George > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Chris > Tunbridge > *Sent:* Thursday, December 25, 2014 9:25 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable > > > > 1) Sounds like NAT issue possibly, or incorrect codecs, please elaborate > on your topology and configuration > > 2) If you're using default configs, its configured to look for extensions > 10XX, you can see this in conf/dialplan/default.xml (and in > conf/dialplan/public.xml for calls coming from the outside) > > 3) Do you have an outbound route configured that matches your dial string? > > 4) This just means the module wasn't configured, you can comment out the > line in conf/autoload_configs/modules.conf.xml find the line that says > mod_v8 and put a My suspicion is that some other dialplan, other than my ?switch2voip.us? dialplan, is being invoked. My SIP Proxy is at 66.33.147.150. IP address ?172.31.33.109? is the local/internal IP address for my AWS virtual cloud server. ?4049392032? is a real phone number ? not an extension. Thanks, George From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David Villasmil Govea Sent: Sunday, December 28, 2014 5:05 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable can you share your dialplan? It looks like you're dialing "To: sip:4049392032 at 172.31.33.109" but have no extension for that... On Sun, Dec 28, 2014 at 10:55 PM, George F. Phelps wrote: New ?pastebin? created: http://pastebin.com/UwmgJGGg George From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David Villasmil Govea Sent: Sunday, December 28, 2014 4:04 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable http://pastebin.com/E4sqTLa4 doesn't show anything. Comes back with "This is a private paste. If you created this paste, please login to view it." On Sun, Dec 28, 2014 at 3:22 PM, George F. Phelps wrote: Chris Tunbridge, 1) I made the updates to my configuration, as suggested in the ?https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2? link. I?m still not able to make a call to an outside number. A call to an extension connects, but there is still no audio. 2) Extension x9161 is one of the default dialplan applications. 3) Call failure log posted at: http://pastebin.com/E4sqTLa4 Thanks, George From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Chris Tunbridge Sent: Saturday, December 27, 2014 2:30 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable 1) This is an issue with the NAT, likely on the freeswitch side, see instructions here: https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2 The important part is the external sip ip and external rtp ip. Without this calls will connect, but audio will not pass. I run dozens of servers on AWS without any issues as long as the external sip and rtp ip's are configured in the sip profile conf/sip_profiles/internal.xml 2) Your issue you said was with extension x9196, is this another sip endpoint or a dialplan application? If this is a sip endpoint, please make some adjustments to the conf/dialplan/default.xml to address extra extensions outside of the 10XX range. 3) Can you post a log here http://pastebin.freeswitch.org of a call attempt? My guess is that something's not matching the request, a complete log of a call attempt would help most here. 4) Glad to hear, its only used if you're using the JavaScript scripting engine for your scripts. On Fri, Dec 26, 2014 at 7:57 AM, George F. Phelps wrote: Chris Tunbridge, et al., 1) Freeswitch is running is running on an Amazon Web Services (AWS) Linux virtual cloud server. I am testing with Bria softphones (both Windows PC and Android smartphone) from my home network (behind a Netgear wireless router). The Freeswitch ?show codecs? command indicates support for ?codec, G.711 ulaw, CORE_PCM_MODULE? ? which is the codec that I am using with Bria. I am able to successfully connect with Bria to my other VoIP services, such as VoIP.ms. 2) I am using mostly a default configuration, i.e., extensions 1000 through 1019 are configured with updated passwords. 3) This is my outbound dialplan. How do I know if this is the dialplan that is actually being used for dialing? It shows up in the ?xml_locate dialplan? output ? but as the very last entry. My guess is that Freeswitch is attempting to us some other (default, example?) gateway instead of my desired (switch2voip.us) gateway. 4) The ?mod_v8? issue is now resolved. The module was not being built. I?m not sure why the downloaded default build/install files were not building it, but were attempting to load it. Sounds like a bug to me? Thanks, George From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Chris Tunbridge Sent: Thursday, December 25, 2014 9:25 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable 1) Sounds like NAT issue possibly, or incorrect codecs, please elaborate on your topology and configuration 2) If you're using default configs, its configured to look for extensions 10XX, you can see this in conf/dialplan/default.xml (and in conf/dialplan/public.xml for calls coming from the outside) 3) Do you have an outbound route configured that matches your dial string? 4) This just means the module wasn't configured, you can comment out the line in conf/autoload_configs/modules.conf.xml find the line that says mod_v8 and put a > > > > > > > > > > > > > > > > > > My suspicion is that some other dialplan, other than my ?switch2voip.us? > dialplan, is being invoked. My SIP Proxy is at 66.33.147.150. IP address > ?172.31.33.109? is the local/internal IP address for my AWS virtual cloud > server. ?4049392032? is a real phone number ? not an extension. > > > > Thanks, > > > > George > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *David > Villasmil Govea > *Sent:* Sunday, December 28, 2014 5:05 PM > > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable > > > > can you share your dialplan? It looks like you're dialing > > "To: sip:4049392032 at 172.31.33.109" > > but have no extension for that... > > > > On Sun, Dec 28, 2014 at 10:55 PM, George F. Phelps < > GeorgePhelps at gfphelps.com> wrote: > > New ?pastebin? created: > > > > http://pastebin.com/UwmgJGGg > > > > George > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *David > Villasmil Govea > *Sent:* Sunday, December 28, 2014 4:04 PM > > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable > > > > http://pastebin.com/E4sqTLa4 doesn't show anything. Comes back with "This > is a private paste. If you created this paste, please login > to view it." > > > > On Sun, Dec 28, 2014 at 3:22 PM, George F. Phelps < > GeorgePhelps at gfphelps.com> wrote: > > Chris Tunbridge, > > > > 1) I made the updates to my configuration, as suggested in the ? > https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2? link. > I?m still not able to make a call to an outside number. A call to an > extension connects, but there is still no audio. > > > > 2) Extension x9161 is one of the default dialplan applications. > > > > 3) Call failure log posted at: http://pastebin.com/E4sqTLa4 > > > > Thanks, > > > > George > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Chris > Tunbridge > *Sent:* Saturday, December 27, 2014 2:30 AM > > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable > > > > 1) This is an issue with the NAT, likely on the freeswitch side, see > instructions here: > https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2 The > important part is the external sip ip and external rtp ip. Without this > calls will connect, but audio will not pass. I run dozens of servers on > AWS without any issues as long as the external sip and rtp ip's are > configured in the sip profile conf/sip_profiles/internal.xml > > 2) Your issue you said was with extension x9196, is this another sip > endpoint or a dialplan application? If this is a sip endpoint, please make > some adjustments to the conf/dialplan/default.xml to address extra > extensions outside of the 10XX range. > > 3) Can you post a log here http://pastebin.freeswitch.org of a call > attempt? My guess is that something's not matching the request, a complete > log of a call attempt would help most here. > > 4) Glad to hear, its only used if you're using the JavaScript scripting > engine for your scripts. > > > > On Fri, Dec 26, 2014 at 7:57 AM, George F. Phelps < > GeorgePhelps at gfphelps.com> wrote: > > Chris Tunbridge, et al., > > > > 1) Freeswitch is running is running on an Amazon Web Services (AWS) Linux > virtual cloud server. I am testing with Bria softphones (both Windows PC > and Android smartphone) from my home network (behind a Netgear wireless > router). The Freeswitch ?show codecs? command indicates support for > ?codec, G.711 ulaw, CORE_PCM_MODULE? ? which is the codec that I am using > with Bria. I am able to successfully connect with Bria to my other VoIP > services, such as VoIP.ms. > > > > 2) I am using mostly a default configuration, i.e., extensions 1000 > through 1019 are configured with updated passwords. > > > > 3) This is my outbound dialplan. How do I know if this is the dialplan > that is actually being used for dialing? It shows up in the ?xml_locate > dialplan? output ? but as the very last entry. My guess is that Freeswitch > is attempting to us some other (default, example?) gateway instead of my > desired (switch2voip.us) gateway. > > > > > > > > > > > > > > > > > > > > > > > > 4) The ?mod_v8? issue is now resolved. The module was not being built. > I?m not sure why the downloaded default build/install files were not > building it, but were attempting to load it. Sounds like a bug to me? > > > > Thanks, > > > > George > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Chris > Tunbridge > *Sent:* Thursday, December 25, 2014 9:25 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable > > > > 1) Sounds like NAT issue possibly, or incorrect codecs, please elaborate > on your topology and configuration > > 2) If you're using default configs, its configured to look for extensions > 10XX, you can see this in conf/dialplan/default.xml (and in > conf/dialplan/public.xml for calls coming from the outside) > > 3) Do you have an outbound route configured that matches your dial string? > > 4) This just means the module wasn't configured, you can comment out the > line in conf/autoload_configs/modules.conf.xml find the line that says > mod_v8 and put a > > > > > > > > > > > > TCP DUMP > 09:31:58.808760 IP (tos 0x0, ttl 64, id 34090, offset 0, flags [none], > proto UDP (17), length 1146) > my.natted.priv.ip.address.5080 > othersipgw.ip.address.5060: SIP, > length: 1118 > INVITE sip:user.name at sip.othersipgw.com;+234diallednumber SIP/2.0 > Via: SIP/2.0/UDP > my.public.ip.address:5080;rport;branch=z9hG4bK8eDcDvZpBm4yr > Max-Forwards: 5 > From: "0diallednumber" > ;tag=6pKBvpXc32yeD > To: > Call-ID: 840abf24-f957-1232-2ca9-525400ecad09 > CSeq: 68677695 INVITE > Contact: > User-Agent: > FreeSWITCH-mod_sofia/1.5.15b+git~20141120T035109Z~79de78a0fb~64bit > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, > UPDATE, REGISTER, REFER, NOTIFY > Supported: timer, path, replaces > Allow-Events: talk, hold, conference, refer > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 247 > X-FS-Support: update_display,send_info > Remote-Party-ID: "0diallednumber" > ;party=calling;screen=yes;privacy=off > > v=0 > o=FreeSWITCH 1418006836 1418006837 IN IP4 my.public.ip.address > s=FreeSWITCH > c=IN IP4 my.public.ip.address > t=0 0 > m=audio 20682 RTP/AVP 8 0 101 13 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > > 09:31:59.080097 IP (tos 0x0, ttl 53, id 5734, offset 0, flags [none], > proto UDP (17), length 384) > othersipgw.ip.address.5060 > my.natted.priv.ip.address.5080: SIP, > length: 356 > SIP/2.0 100 Trying > Via: SIP/2.0/UDP > my.public.ip.address:5080;rport=5080;branch=z9hG4bK8eDcDvZpBm4yr > From: "0diallednumber" > ;tag=6pKBvpXc32yeD > To: > Call-ID: 840abf24-f957-1232-2ca9-525400ecad09 > CSeq: 68677695 INVITE > User-Agent: FreeSWITCH-mod_sofia/1.2.14 > Content-Length: 0 > > > 09:31:59.145629 IP (tos 0x0, ttl 53, id 5735, offset 0, flags [none], > proto UDP (17), length 1015) > othersipgw.ip.address.5060 > my.natted.priv.ip.address.5080: SIP, > length: 987 > SIP/2.0 480 Temporarily Unavailable > Via: SIP/2.0/UDP > my.public.ip.address:5080;rport=5080;branch=z9hG4bK8eDcDvZpBm4yr > Max-Forwards: 4 > From: "0diallednumber" > ;tag=6pKBvpXc32yeD > To: ;tag=Ur840N1DvjvpH > Call-ID: 840abf24-f957-1232-2ca9-525400ecad09 > CSeq: 68677695 INVITE > User-Agent: FreeSWITCH-mod_sofia/1.2.14 > 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=16;text="NORMAL_CLEARING" > Content-Length: 0 > X-FS-Display-Name: user.name > X-FS-Display-Number: sip:user.name at sip.othersipgw.com > Remote-Party-ID: "user.name" > ;party=calling;privacy=off;screen=no > > > 09:31:59.146980 IP (tos 0x0, ttl 64, id 34091, offset 0, flags [none], > proto UDP (17), length 412) > my.natted.priv.ip.address.5080 > othersipgw.ip.address.5060: SIP, > length: 384 > ACK sip:user.name at sip.othersipgw.com;+234diallednumber SIP/2.0 > Via: SIP/2.0/UDP > my.public.ip.address:5080;rport;branch=z9hG4bK8eDcDvZpBm4yr > Max-Forwards: 5 > From: "0diallednumber" > ;tag=6pKBvpXc32yeD > To: ;tag=Ur840N1DvjvpH > Call-ID: 840abf24-f957-1232-2ca9-525400ecad09 > CSeq: 68677695 ACK > Content-Length: 0 From notify.sina at gmail.com Wed Dec 31 12:37:55 2014 From: notify.sina at gmail.com (Sina Owolabi) Date: Wed, 31 Dec 2014 10:37:55 +0100 Subject: [Freeswitch-users] Call back through SIP trunk Message-ID: Hi List! FreeSWITCHNewbie here. Please can I have some guidance on how to setup call back? I would like to be able to dial the DID attached to the SIP trunk Freeswitch is registered to, and then have freeSWITCH hang up the call and dial the caller id number back through any other SIP trunk FreeSWITCH Is registered with, but with the origination number set to the DID that the first call came through in the first place. Please is this possible just through the dial plan? Thanks for any help! From vipkilla at gmail.com Wed Dec 31 16:12:16 2014 From: vipkilla at gmail.com (Vik Killa) Date: Wed, 31 Dec 2014 08:12:16 -0500 Subject: [Freeswitch-users] intercom setup -- page multiple phones in a single endpoint Message-ID: im using "conference_set_auto_outcall" to do an intercom if i set "conference_set_auto_outcall" to "user/1000" and that endpoint points to two physical phones, only one is picked up, the other originate is canceled by FS Example: user/1000 ==> 1000 at 192.168.0.101 and 1000 at 192.168.0.102 does anyone know how to address this so that both phones are placed in the conference? is there a flag to pass to the originate to tell it NOT to cancel the second call? Thanks. /V -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141231/d1c3e36e/attachment.html From max at nysolutions.com Wed Dec 31 16:43:28 2014 From: max at nysolutions.com (Moishe Grunstein) Date: Wed, 31 Dec 2014 13:43:28 +0000 Subject: [Freeswitch-users] Polycom horror story In-Reply-To: References: Message-ID: Web interface on Polycom version 3 is almost useless, version 4 has much improved that. Those phones were designed to be configured using a http/https/tftp/ftp provisioning server. The 501 is long EOL and are not getting any updates for many years. Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support at nysolutions.com Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Luis Daniel Lucio Quiroz Sent: Wednesday, December 31, 2014 5:32 AM To: FreeSWITCH Users Help Subject: [Freeswitch-users] Polycom horror story Hello, Using Freeswitch 1.4.14 with multitenant, I can not make this @#$@%# telephone to register: model: Soundpoint IP501 firmware: 3.1.8.0070 I have done next: NDLB-force-rport with true and safe values No nat involved (switch and phone on same segment) I have tried multiple combinations of configurations on the phone (using web interface). with or without proxy, different sip extensions, different passwords, different ports. I always get 403 error code. however, i can register only if i do next: after telephone is booted, i edit sip password using the phone keypad. And telephone register. This makes me thing it is a bug on the polycom and has nothing to do with Freeswitch. It works on all other phones I have, snom, bria, linphone. But here I am, maybe someone has workarround this somehow. LD _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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 dragic.dusan at gmail.com Wed Dec 31 18:56:30 2014 From: dragic.dusan at gmail.com (=?UTF-8?B?RHXFoWFuIERyYWdpxIc=?=) Date: Wed, 31 Dec 2014 16:56:30 +0100 Subject: [Freeswitch-users] Flash Hook to an FXO port (ata) In-Reply-To: References: Message-ID: I was looking into sending custom sip info messages a while back (but not for fxo hook flash) and found two ways in fs: 1) send_info dialplan app - with this one you can't change the mime type (it's always "freeswitch/data") and you have to set the channel var fs_send_unsupported_info=true beforehand if the other end isn't freeswitch. It's probably useless to you. 2) uuid_send_info api command - this let's you set mime type/subtype so it could work for your case: uuid_send_info application hook-flash some_data On 31 December 2014 at 09:43, davy van de moere wrote: > Gents, > > Does any one have a good setup to send a flash hook to an analogue line? > Preferably with an ATA of some sort. > > So far I've been trying with an Linksys SPA3102, which can do a flash hook > with the specific mime type : application/hook-flash , with Freeswitch I the > send_dtmf didn't get me any further than sending the special flash key F in > application/dtmf-relay. Which is not interpreted by the SPA3102 as a flash > hook. > > Hence, does anyone have a good strategy from a Freeswitch pov to send a > flash hook signal (do I need to think sipsak?), or an ATA/firmware which > does support the F key in dtmf? > > Season grtz! > Davy > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Du?an Dragi? From brian at freeswitch.org Wed Dec 31 19:09:34 2014 From: brian at freeswitch.org (Brian West) Date: Wed, 31 Dec 2014 10:09:34 -0600 Subject: [Freeswitch-users] Polycom horror story In-Reply-To: References: Message-ID: LOL read to Polycom 501 and thought "Yep, make a reservation in the nut house for this one, the insane is strong here", Those things are such a pain in the ass to deal with I just relegated mine to a box in the corner of the room. :P Sadly the VVX's aren't much better, do they even test these things EVER? On Wed, Dec 31, 2014 at 7:43 AM, Moishe Grunstein wrote: > Web interface on Polycom version 3 is almost useless, version 4 has much > improved that. > Those phones were designed to be configured using a http/https/tftp/ftp > provisioning server. > The 501 is long EOL and are not getting any updates for many years. > > Thanks, > > Moishe Grunstein > Tornado Computer Systems, Inc. > 212.400.7650 888.IPPBX.US > Service Request Email: support at nysolutions.com > Polycom Certified VAR > Microsoft Small Business Specialist, Cisco SMB Select Certified > > Computer Networking * Managed Services * IP Video Surveillance * Network > Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network > Security * Site Surveys * CMS > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Luis Daniel > Lucio Quiroz > Sent: Wednesday, December 31, 2014 5:32 AM > To: FreeSWITCH Users Help > Subject: [Freeswitch-users] Polycom horror story > > Hello, Using Freeswitch 1.4.14 with multitenant, I can not make this > @#$@%# telephone to register: > > model: Soundpoint IP501 > firmware: 3.1.8.0070 > > I have done next: > NDLB-force-rport with true and safe values No nat involved (switch and > phone on same segment) > > I have tried multiple combinations of configurations on the phone (using > web interface). with or without proxy, different sip extensions, different > passwords, different ports. I always get 403 error code. > > however, i can register only if i do next: > after telephone is booted, i edit sip password using the phone keypad. > And telephone register. This makes me thing it is a bug on the polycom > and has nothing to do with Freeswitch. > > It works on all other phones I have, snom, bria, linphone. > > But here I am, maybe someone has workarround this somehow. > > LD > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141231/cf296873/attachment-0001.html From dragic.dusan at gmail.com Wed Dec 31 19:23:43 2014 From: dragic.dusan at gmail.com (=?UTF-8?B?RHXFoWFuIERyYWdpxIc=?=) Date: Wed, 31 Dec 2014 17:23:43 +0100 Subject: [Freeswitch-users] How to execute lua script on register? In-Reply-To: References: Message-ID: I agree with Ken, there are better ways to do this. Maybe off topic: I noticed a bug with how the hook event subclass string (in lua.conf) is compared to the event subclass. For example subclass="sofia::register" matches sofia::register, sofia::register_attempt and sofia::register_failure event subclasses. Also something like subclass="ia::register_atte" works (matches sofia::register_attempt) Looks like it's searching for a substring. I should probably open a jira. On 30 December 2014 at 16:40, Ken Rice wrote: > Why would you do this? > > With any kind of registration load this is going to be horrible for CPU > performance... Every time fire this lua script there is no job control is > fire and forget, it spawns a new thread and if it hangs there is nothing you > can do short of restarting freeswitch. > > Now, if you look at mod_sofia, you will notice that you can push all the > internal data tables to an external database such as PostgreSQL... This > allows you to see the state of every registered endpoint. > > Also if you really want a script to monitor things, simply attach to ESL and > only subscribe to the events you want to see. This allows you to use > whatever language you want with job control. > > > > On 12/29/14, 9:06 PM, "Jaime" wrote: > >> Hi, > I use the following line in ?autoload_configs/lua.conf.xml? to monitor >> de > register state of my gateways: > > subclass="sofia::gateway_state" > script="catch-event-gateway.lua"/> > > And put in >> your scripts directory the file catch-event-gateway.lua with > the following >> code to see what happens: > freeswitch.consoleLog("notice",?event ==== >> sofia::gateway_state > =====\n") > freeswitch.consoleLog("notice"," event\n" >> .. event:serialize()) > > > > Also, for internal users register events, you can try >> the same code with > the following ?subclass? events: > sofia::register_failure > >> sofia::register > sofia::expire > sofia::register_attempt > sofia::unregister > >> sofia::pre_register > > > > Regards > JCM > > > From: Ken Rice >> > Reply-To: FreeSWITCH Users Help >> > Date: viernes, 26 de diciembre de >> 2014, 11:30 > To: FreeSWITCH Users Help >> > Subject: Re: [Freeswitch-users] How >> to execute lua script on register? > > > You cant fire a script on register... >> Your options are ESL to monitor it > (which you can write in lua) or luarun and >> have a lua script constantly > running processing register events... The >> external script would probably > > > On 12/26/14 2:39 AM, "poliv78 at yahoo.co.uk" >> wrote: > > > ????????????, David. > Yes I know that, >> thanks. > But I want to monitor all registrations. Simply put in >> database > fail or success. > I know I can use ESL for that. But lua is >> more handy here and I > would like to know if it's possible. > > ?? ?????? 25 >> ??????? 2014 ?., 23:33:41: > > > You could just look at the log for failed >> attempts. > On Dec 25, 2014 1:21 PM, wrote: > Hi all > > Is it >> possible to execute lua script when gateway try to register > to know if it >> successfull or failed? > > The same question as for internal users. > >> Thanks. > > > -- > ? ?????????, > Poliv78 >> mailto:poliv78 at yahoo.co.uk > > > _________________________________________________ >> ________________________ > Professional FreeSWITCH Consulting >> Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Offici >> al FreeSWITCH >> Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cl >> uecon.com > > 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 > Twitter: >> @FreeSWITCH > > > > > ______________________________________________________________ >> ___________ > Professional FreeSWITCH Consulting >> Services: > consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > Officia >> l FreeSWITCH >> Sites > http://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.clue >> con > .com > > FreeSWITCH-users mailing >> list > FreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/ >> li > stinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/opt >> ions/freeswitch-usershtt > p://www.freeswitch.org > > > > ___________________________ >> ______________________________________________ > Professional FreeSWITCH >> Consulting Services: >> > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official >> FreeSWITCH >> Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cl >> uecon.com > > 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 > Twitter: @FreeSWITCH > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Du?an Dragi? From krice at freeswitch.org Wed Dec 31 19:50:13 2014 From: krice at freeswitch.org (Ken Rice) Date: Wed, 31 Dec 2014 10:50:13 -0600 Subject: [Freeswitch-users] How to execute lua script on register? In-Reply-To: Message-ID: Yep you should open a jira on this.... On 12/31/14, 10:23 AM, "Du?an Dragi?" wrote: > I agree with Ken, there are better ways to do this. Maybe off topic: I > noticed a bug with how the hook event subclass string (in lua.conf) is > compared to the event subclass. For example subclass="sofia::register" matches > sofia::register, sofia::register_attempt and sofia::register_failure event > subclasses. Also something like subclass="ia::register_atte" works > (matches sofia::register_attempt) Looks like it's searching for a substring. > I should probably open a jira. On 30 December 2014 at 16:40, Ken Rice > wrote: > Why would you do this? > > With any kind of > registration load this is going to be horrible for CPU > performance... Every > time fire this lua script there is no job control is > fire and forget, it > spawns a new thread and if it hangs there is nothing you > can do short of > restarting freeswitch. > > Now, if you look at mod_sofia, you will notice that > you can push all the > internal data tables to an external database such as > PostgreSQL... This > allows you to see the state of every registered > endpoint. > > Also if you really want a script to monitor things, simply > attach to ESL and > only subscribe to the events you want to see. This allows > you to use > whatever language you want with job control. > > > > On 12/29/14, > 9:06 PM, "Jaime" wrote: > >> Hi, > I use the following > line in ?autoload_configs/lua.conf.xml? to monitor >> de > register state of > my gateways: > > subclass="sofia::gateway_state" > > script="catch-event-gateway.lua"/> > > And put in >> your scripts directory > the file catch-event-gateway.lua with > the following >> code to see what > happens: > freeswitch.consoleLog("notice",?event ==== >> > sofia::gateway_state > =====\n") > freeswitch.consoleLog("notice"," > event\n" >> .. event:serialize()) > > > > Also, for internal users register > events, you can try >> the same code with > the following ?subclass? events: > > sofia::register_failure > >> sofia::register > sofia::expire > > sofia::register_attempt > sofia::unregister > >> sofia::pre_register > > > > > Regards > JCM > > > From: Ken Rice >> > Reply-To: > FreeSWITCH Users Help >> > Date: > viernes, 26 de diciembre de >> 2014, 11:30 > To: FreeSWITCH Users Help >> > > Subject: Re: [Freeswitch-users] > How >> to execute lua script on register? > > > You cant fire a script on > register... >> Your options are ESL to monitor it > (which you can write in > lua) or luarun and >> have a lua script constantly > running processing > register events... The >> external script would probably > > > On 12/26/14 > 2:39 AM, "poliv78 at yahoo.co.uk" >> wrote: > > > > ????????????, David. > Yes I know that, >> thanks. > But I > want to monitor all registrations. Simply put in >> database > fail or > success. > I know I can use ESL for that. But lua is >> more handy > here and I > would like to know if it's possible. > > ?? ?????? 25 >> ??????? > 2014 ?., 23:33:41: > > > You could just look at the log for failed >> > attempts. > On Dec 25, 2014 1:21 PM, wrote: > Hi all > > > Is it >> possible to execute lua script when gateway try to register > to > know if it >> successfull or failed? > > The same question as for internal > users. > >> Thanks. > > > -- > ? ?????????, > Poliv78 >> > mailto:poliv78 at yahoo.co.uk > > > > _________________________________________________ >> > ________________________ > Professional FreeSWITCH Consulting >> Services: > > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Offici >> > al FreeSWITCH >> Sites > http://www.freeswitch.org > > http://confluence.freeswitch.org > http://www.cl >> uecon.com > > > 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 > Twitter: >> @FreeSWITCH > > > > > > ______________________________________________________________ >> > ___________ > Professional FreeSWITCH Consulting >> Services: > > consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > Officia >> l > FreeSWITCH >> Sites > > http://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.clue >> > con > .com > > FreeSWITCH-users mailing >> list > > FreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/>> > li > stinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/opt >> > ions/freeswitch-usershtt > p://www.freeswitch.org > > > > > ___________________________ >> > ______________________________________________ > Professional FreeSWITCH >> > Consulting Services: >> > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > Official >> FreeSWITCH >> Sites > > http://www.freeswitch.org > http://confluence.freeswitch.org > > http://www.cl >> uecon.com > > 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 > > Twitter: @FreeSWITCH > > > > > > _________________________________________________________________________> > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > > http://www.freeswitch.org > http://confluence.freeswitch.org > > http://www.cluecon.com > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org -- Du?an > Dragi? ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org http://www.freeswitchsolutions.com Official > FreeSWITCH > Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cl > uecon.com 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 Twitter: @FreeSWITCH From avi at avimarcus.net Wed Dec 31 20:36:02 2014 From: avi at avimarcus.net (Avi Marcus) Date: Wed, 31 Dec 2014 17:36:02 +0000 Subject: [Freeswitch-users] Call back through SIP trunk In-Reply-To: References: Message-ID: <0000014aa16b2a0b-ee3e05bc-9993-4da0-8da1-2edff71f346b-000000@email.amazonses.com> I've used a lua script to grab the information (and make sure it's a valid callback), hangup, and then run: freeswitch.msleep(2000); --wait 2 seconds to make sure their side will actually have the call over api = freeswitch.API() api:execute("originate", your-dialstring) -Avi On Wed, Dec 31, 2014 at 11:37 AM, Sina Owolabi wrote: > Hi List! > > > FreeSWITCHNewbie here. > Please can I have some guidance on how to setup call back? > > I would like to be able to dial the DID attached to the SIP trunk > Freeswitch is registered to, and then have freeSWITCH hang up the call > and dial the caller id number back through any other SIP trunk > FreeSWITCH Is registered with, but with the origination number set to > the DID that the first call came through in the first place. > > Please is this possible just through the dial plan? > > Thanks for any help! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141231/ba8cd14c/attachment.html From dragic.dusan at gmail.com Wed Dec 31 21:13:19 2014 From: dragic.dusan at gmail.com (=?UTF-8?B?RHXFoWFuIERyYWdpxIc=?=) Date: Wed, 31 Dec 2014 19:13:19 +0100 Subject: [Freeswitch-users] How to execute lua script on register? In-Reply-To: References: Message-ID: FS-7121 .. the problem isn't in mod_lua *** Happy New Year! On 31 December 2014 at 17:50, Ken Rice wrote: > Yep you should open a jira on this.... > > > On 12/31/14, 10:23 AM, "Du?an Dragi?" wrote: > >> I agree with Ken, there are better ways to do this. > > Maybe off topic: > I >> noticed a bug with how the hook event subclass string (in lua.conf) > is >> compared to the event subclass. For example > subclass="sofia::register" matches >> sofia::register, > sofia::register_attempt and sofia::register_failure event >> subclasses. > Also something like subclass="ia::register_atte" works >> (matches > sofia::register_attempt) > > Looks like it's searching for a substring. >> I should probably open a jira. > > On 30 December 2014 at 16:40, Ken Rice >> wrote: >> Why would you do this? >> >> With any kind of >> registration load this is going to be horrible for CPU >> performance... Every >> time fire this lua script there is no job control is >> fire and forget, it >> spawns a new thread and if it hangs there is nothing you >> can do short of >> restarting freeswitch. >> >> Now, if you look at mod_sofia, you will notice that >> you can push all the >> internal data tables to an external database such as >> PostgreSQL... This >> allows you to see the state of every registered >> endpoint. >> >> Also if you really want a script to monitor things, simply >> attach to ESL and >> only subscribe to the events you want to see. This allows >> you to use >> whatever language you want with job control. >> >> >> >> On 12/29/14, >> 9:06 PM, "Jaime" wrote: >> >>> Hi, >> I use the following >> line in ?autoload_configs/lua.conf.xml? to monitor >>> de >> register state of >> my gateways: >> >> subclass="sofia::gateway_state" >> >> script="catch-event-gateway.lua"/> >> >> And put in >>> your scripts directory >> the file catch-event-gateway.lua with >> the following >>> code to see what >> happens: >> freeswitch.consoleLog("notice",?event ==== >>> >> sofia::gateway_state >> =====\n") >> freeswitch.consoleLog("notice"," >> event\n" >>> .. event:serialize()) >> >> >> >> Also, for internal users register >> events, you can try >>> the same code with >> the following ?subclass? events: >> >> sofia::register_failure >> >>> sofia::register >> sofia::expire >> >> sofia::register_attempt >> sofia::unregister >> >>> sofia::pre_register >> >> >> >> >> Regards >> JCM >> >> >> From: Ken Rice >>> >> Reply-To: >> FreeSWITCH Users Help >>> >> Date: >> viernes, 26 de diciembre de >>> 2014, 11:30 >> To: FreeSWITCH Users Help >>> >> >> Subject: Re: [Freeswitch-users] >> How >>> to execute lua script on register? >> >> >> You cant fire a script on >> register... >>> Your options are ESL to monitor it >> (which you can write in >> lua) or luarun and >>> have a lua script constantly >> running processing >> register events... The >>> external script would probably >> >> >> On 12/26/14 >> 2:39 AM, "poliv78 at yahoo.co.uk" >>> wrote: >> >> >> >> ????????????, David. >> Yes I know that, >>> thanks. >> But I >> want to monitor all registrations. Simply put in >>> database >> fail or >> success. >> I know I can use ESL for that. But lua is >>> more handy >> here and I >> would like to know if it's possible. >> >> ?? ?????? 25 >>> ??????? >> 2014 ?., 23:33:41: >> >> >> You could just look at the log for failed >>> >> attempts. >> On Dec 25, 2014 1:21 PM, wrote: >> Hi all >> >> >> Is it >>> possible to execute lua script when gateway try to register >> to >> know if it >>> successfull or failed? >> >> The same question as for internal >> users. >> >>> Thanks. >> >> >> -- >> ? ?????????, >> Poliv78 >>> >> mailto:poliv78 at yahoo.co.uk >> >> >> >> _________________________________________________ >>> >> ________________________ >> Professional FreeSWITCH Consulting >>> Services: >> >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Offici >>> >> al FreeSWITCH >>> Sites >> http://www.freeswitch.org >> >> http://confluence.freeswitch.org >> http://www.cl >>> uecon.com >> >> >> 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 >> Twitter: >>> @FreeSWITCH >> >> >> >> >> >> ______________________________________________________________ >>> >> ___________ >> Professional FreeSWITCH Consulting >>> Services: >> >> consulting at freeswitch.orghttp://www.freeswitchsolutions.com >> >> Officia >>> l >> FreeSWITCH >>> Sites >> >> http://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.clue >>> >> con >> .com >> >> FreeSWITCH-users mailing >>> list >> >> FreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/>> >> li >> stinfo/freeswitch-users >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/opt >>> >> ions/freeswitch-usershtt >> p://www.freeswitch.org >> >> >> >> >> ___________________________ >>> >> ______________________________________________ >> Professional FreeSWITCH >>> >> Consulting Services: >>> >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> Official >>> FreeSWITCH >>> Sites >> >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> >> http://www.cl >>> uecon.com >> >> 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 >> >> Twitter: @FreeSWITCH >> >> >> >> >> >> _________________________________________________________________________> >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org > > > > -- > Du?an >> Dragi? > > ______________________________________________________________________ >> ___ > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official >> FreeSWITCH >> Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cl >> uecon.com > > 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 > Twitter: @FreeSWITCH > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Du?an Dragi? From david.villasmil at gmail.com Wed Dec 31 22:41:27 2014 From: david.villasmil at gmail.com (David Villasmil Govea) Date: Wed, 31 Dec 2014 20:41:27 +0100 Subject: [Freeswitch-users] Unbridge and send to queue In-Reply-To: References: Message-ID: can anyone help me out with this? On Tue, Dec 23, 2014 at 5:48 PM, David Villasmil Govea < david.villasmil at gmail.com> wrote: > Hello Guys, > > I receive a call from side A and send it out to side B. > I need to unbridge this call once it is answered and send the B side to a > queue to give it moh. How do I do that? > > I'm doing everything with lua and I've tried: > > on my dialplan I have: > > > > in check_answered-lua, after checking the call was in fact answered I do: > > session:execute("transfer", "-bleg 9999 XML default"); > > and on my dialplan I have: > > > > data="/usr/local/freeswitch/scripts/queue.lua"/> > > > > queue.lua: > > session:execute("sched_hangup","+50 alloted_timeout"); > session:execute("callcenter","agents_queue"); > > session:execute("sleep",my_dur); > > But this doesn't seem to work at all... > > Can anyone give me a hand? thanks! > > -- > DVG > -- DVG -- Imagination is more important than knowledge Albert Einstein -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141231/0a53724f/attachment.html From aademattia at comcast.net Wed Dec 31 23:13:46 2014 From: aademattia at comcast.net (Andrew) Date: Wed, 31 Dec 2014 15:13:46 -0500 Subject: [Freeswitch-users] Unbridge and send to queue In-Reply-To: References: Message-ID: <01e301d02536$4c5624c0$e5026e40$@comcast.net> Maybe you can set leg b transfer_after_bridge and hang up on leg a. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David Villasmil Govea Sent: Wednesday, December 31, 2014 2:41 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Unbridge and send to queue can anyone help me out with this? On Tue, Dec 23, 2014 at 5:48 PM, David Villasmil Govea > wrote: Hello Guys, I receive a call from side A and send it out to side B. I need to unbridge this call once it is answered and send the B side to a queue to give it moh. How do I do that? I'm doing everything with lua and I've tried: on my dialplan I have: in check_answered-lua, after checking the call was in fact answered I do: session:execute("transfer", "-bleg 9999 XML default"); and on my dialplan I have: queue.lua: session:execute("sched_hangup","+50 alloted_timeout"); session:execute("callcenter","agents_queue"); session:execute("sleep",my_dur); But this doesn't seem to work at all... Can anyone give me a hand? thanks! -- DVG -- DVG -- Imagination is more important than knowledge Albert Einstein -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141231/df77744a/attachment.html From david.villasmil at gmail.com Wed Dec 31 23:46:26 2014 From: david.villasmil at gmail.com (David Villasmil Govea) Date: Wed, 31 Dec 2014 21:46:26 +0100 Subject: [Freeswitch-users] Unbridge and send to queue In-Reply-To: <01e301d02536$4c5624c0$e5026e40$@comcast.net> References: <01e301d02536$4c5624c0$e5026e40$@comcast.net> Message-ID: Hello, I don't think that would work, I need to launch a lua script before transferring the B-leg. I need to transfer it from the lua script, but session:execute("transfer", "-bleg 9999 XML default"); Doesn't work... On Wed, Dec 31, 2014 at 9:13 PM, Andrew wrote: > Maybe you can set leg b transfer_after_bridge and hang up on leg a. > > > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *David > Villasmil Govea > *Sent:* Wednesday, December 31, 2014 2:41 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Unbridge and send to queue > > > > can anyone help me out with this? > > > > On Tue, Dec 23, 2014 at 5:48 PM, David Villasmil Govea < > david.villasmil at gmail.com> wrote: > > Hello Guys, > > I receive a call from side A and send it out to side B. > I need to unbridge this call once it is answered and send the B side to a > queue to give it moh. How do I do that? > > I'm doing everything with lua and I've tried: > > on my dialplan I have: > > > > in check_answered-lua, after checking the call was in fact answered I do: > > session:execute("transfer", "-bleg 9999 XML default"); > > and on my dialplan I have: > > > > data="/usr/local/freeswitch/scripts/queue.lua"/> > > > > queue.lua: > > session:execute("sched_hangup","+50 alloted_timeout"); > session:execute("callcenter","agents_queue"); > > session:execute("sleep",my_dur); > > But this doesn't seem to work at all... > > Can anyone give me a hand? thanks! > > -- > DVG > > > > > > -- > > DVG > > -- > Imagination is more important than knowledge > Albert Einstein > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- DVG -- Imagination is more important than knowledge Albert Einstein -------------- next part -------------- An HTML attachment was scrubbed... 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