From janne.blom at gmail.com Tue Nov 1 12:03:35 2016 From: janne.blom at gmail.com (Jan Blom) Date: Tue, 1 Nov 2016 10:03:35 +0100 Subject: [Freeswitch-users] RTP timing issue Message-ID: Hi list, I bridge a call with ptime 10 on one side and ptime 20 on the other. PCM codec on both sides, so no transcoding. The issue I face is that when a 20 ms packet arrives it is immediately sent out as two 10ms packets. And then nothing for 20ms until next packet arrives. Delta time between sent our packets are either 0 or 20 ms, never 10ms as one would expect. The result is choppy audio since we introduce jitter. Could this be avoided somehow? I am using Freeswitch 1.6.12 on CentOS 7.2. Best regards, Jan Blom -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161101/d926fa6f/attachment.html From brian at freeswitch.org Tue Nov 1 17:58:48 2016 From: brian at freeswitch.org (Brian West) Date: Tue, 1 Nov 2016 09:58:48 -0500 Subject: [Freeswitch-users] RTP timing issue In-Reply-To: References: Message-ID: Bug reports belong on freeswitch.org/jira Collect the data, logs, pcaps and file a JIRA. /b On Tue, Nov 1, 2016 at 4:03 AM, Jan Blom wrote: > Hi list, > > > > I bridge a call with ptime 10 on one side and ptime 20 on the other. PCM > codec on both sides, so no transcoding. > > > > The issue I face is that when a 20 ms packet arrives it is immediately > sent out as two 10ms packets. And then nothing for 20ms until next packet > arrives. Delta time between sent our packets are either 0 or 20 ms, never > 10ms as one would expect. The result is choppy audio since we introduce > jitter. > > > > Could this be avoided somehow? I am using Freeswitch 1.6.12 on CentOS 7.2. > > > > > > Best regards, > > Jan Blom > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 (50% Discount using code FreeSwitch50) http://www.freeswitchcookbook.com (50% Discount using code FreeSwitch50) https://www.gofundme.com/freeswitch_ubuntu Got Bugs? Report them here ! | Reddit: /r/freeswitch *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/20161101/c629f40b/attachment.html From ssinyagin at gmail.com Tue Nov 1 23:37:42 2016 From: ssinyagin at gmail.com (Stanislav Sinyagin) Date: Tue, 1 Nov 2016 21:37:42 +0100 Subject: [Freeswitch-users] RTP timing issue In-Reply-To: References: Message-ID: As far as I remember, bridged RTP frames are sent out as soon as they are ready, without looking at the timer, so it totally makes sense... kind of :) On 1 Nov 2016 15:59, "Brian West" wrote: > Bug reports belong on freeswitch.org/jira > > Collect the data, logs, pcaps and file a JIRA. > > /b > > > On Tue, Nov 1, 2016 at 4:03 AM, Jan Blom wrote: > >> Hi list, >> >> >> >> I bridge a call with ptime 10 on one side and ptime 20 on the other. PCM >> codec on both sides, so no transcoding. >> >> >> >> The issue I face is that when a 20 ms packet arrives it is immediately >> sent out as two 10ms packets. And then nothing for 20ms until next packet >> arrives. Delta time between sent our packets are either 0 or 20 ms, never >> 10ms as one would expect. The result is choppy audio since we introduce >> jitter. >> >> >> >> Could this be avoided somehow? I am using Freeswitch 1.6.12 on CentOS 7.2. >> >> >> >> >> >> Best regards, >> >> Jan Blom >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 (50% Discount using code FreeSwitch50) > http://www.freeswitchcookbook.com (50% Discount using code FreeSwitch50) > https://www.gofundme.com/freeswitch_ubuntu > > Got Bugs? Report them here ! | Reddit: > /r/freeswitch > > *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/20161101/9f456ad0/attachment.html From freeswitch.opencode at spamgourmet.com Wed Nov 2 02:12:44 2016 From: freeswitch.opencode at spamgourmet.com (freeswitch.opencode at spamgourmet.com) Date: Tue, 1 Nov 2016 23:12:44 +0000 Subject: [Freeswitch-users] mod_h323 and mod_opal Message-ID: I'm working on packaging FreeSWITCH (including all modules) for Arch Linux, and I'm trying to figure out how to handle mod_h323. Following the pointers from src/mod/endpoints/mod_h323/compiling.txt, I see that it consumes code from this project: http://www.h323plus.org/ However, that project appears to be inactive; it hasn't had a release for multiple years (and forum/mailing list activity has nearly ceased as well). According to Wikipedia, the parent OpenH323 project (https://en.wikipedia.org/wiki/OpenH323) forked into H323Plus and Open Phone Abstraction Library (OPAL). >From the FreeSWITCH documentation, it looks like mod_opal consumes the code from one fork and mod_h323 from the other. The documentation also currently says mod_opal is beta quality, but that statement may be outdated (per the bugs file in its module directory). I'm packaging up FreeSWITCH for Arch Linux, which currently doesn't provide H323Plus. I could package up H323Plus for Arch, but if the project is inactive, I'm not sure if that's worthwhile. Is it fair to see the H323Plus project as defunct? If so, do you think FreeSWITCH will end up deprecating/removing mod_h323 in favor of mod_opal? Or is mod_h323 currently favored over mod_opal for production-quality H.323 support and thus important for a complete FreeSWITCH package to provide? Also, I thought this was a more dev-focused question, so I tried posting to freeswitch-dev over a week ago. The auto-reply said it was a moderated list and the message was being held pending moderator approval. It said that either the message would get posted or I would be notified of the moderator's decision, but nothing ever happened. Did this message get stuck in the pending moderator approval list? Thanks, David -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161101/a0b8e179/attachment-0001.html From brian at freeswitch.org Wed Nov 2 02:22:28 2016 From: brian at freeswitch.org (Brian West) Date: Tue, 1 Nov 2016 18:22:28 -0500 Subject: [Freeswitch-users] mod_h323 and mod_opal In-Reply-To: References: Message-ID: David, I've not kept tabs on that, but we do have a module that will probably go in based on ooh323 that is more maintained. Sorry about the moderator on the dev list, I didn't get a notification about it, That list isn't very active currently, I did approve your email to this list. I would say neither of those modules would be something I would use in production at this time, They are not well maintained. /b On Tue, Nov 1, 2016 at 6:12 PM, wrote: > I'm working on packaging FreeSWITCH (including all modules) for Arch > Linux, and I'm trying to figure out how to handle mod_h323. > > Following the pointers from src/mod/endpoints/mod_h323/compiling.txt, I > see that it consumes code from this project: > http://www.h323plus.org/ > > However, that project appears to be inactive; it hasn't had a release for > multiple years (and forum/mailing list activity has nearly ceased as well). > > According to Wikipedia, the parent OpenH323 project ( > https://en.wikipedia.org/wiki/OpenH323) forked into H323Plus and Open > Phone Abstraction Library (OPAL). > > >From the FreeSWITCH documentation, it looks like mod_opal consumes the > code from one fork and mod_h323 from the other. The documentation also > currently says mod_opal is beta quality, but that statement may be outdated > (per the bugs file in its module directory). > > I'm packaging up FreeSWITCH for Arch Linux, which currently doesn't > provide H323Plus. I could package up H323Plus for Arch, but if the project > is inactive, I'm not sure if that's worthwhile. > > Is it fair to see the H323Plus project as defunct? If so, do you think > FreeSWITCH will end up deprecating/removing mod_h323 in favor of mod_opal? > Or is mod_h323 currently favored over mod_opal for production-quality > H.323 support and thus important for a complete FreeSWITCH package to > provide? > > Also, I thought this was a more dev-focused question, so I tried posting > to freeswitch-dev over a week ago. The auto-reply said it was a moderated > list and the message was being held pending moderator approval. It said > that either the message would get posted or I would be notified of the > moderator's decision, but nothing ever happened. Did this message get stuck > in the pending moderator approval list? > > Thanks, > > David > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 (50% Discount using code FreeSwitch50) http://www.freeswitchcookbook.com (50% Discount using code FreeSwitch50) https://www.gofundme.com/freeswitch_ubuntu Got Bugs? Report them here ! | Reddit: /r/freeswitch *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/20161101/0589087c/attachment.html From freeswitch.opencode at spamgourmet.com Wed Nov 2 05:23:19 2016 From: freeswitch.opencode at spamgourmet.com (freeswitch.opencode at spamgourmet.com) Date: Wed, 2 Nov 2016 02:23:19 +0000 Subject: [Freeswitch-users] mod_h323 and mod_opal of 20) In-Reply-To: References: , Message-ID: Thanks, Brian. No problem. Do you think either mod_h323 or mod_opal will be deprecated/removed once the new ooh323-based module is added? On a related note, the other modules I don't yet having fully building on Arch are: applications/mod_av applications/mod_cv applications/mod_osp applications/mod_sms_flowroute asr_tts/mod_cepstral?(due to non-free SDK) asr_tts/mod_flite codecs/mod_bv codecs/mod_codec2 codecs/mod_com_g729 (commercial) codecs/mod_ilbc codecs/mod_sangoma_codec codecs/mod_silk codecs/mod_siren endpoints/mod_gsmopen endpoints/mod_khomp endpoints/mod_skypopen event_handlers/mod_amqp event_handlers/mod_erlang_event event_handlers/mod_kazoo event_handlers/mod_smpp event_handlers/mod_snmp languages/mod_java languages/mod_managed languages/mod_v8 languages/mod_yaml ../../libs/freetdm/mod_freetdm ../../contrib/mod/xml_int/mod_xml_odbc Would you consider any others of these not production-worthy/well-maintained? Also, out of curiosity, where does the source for the last two modules come from? Thanks, David From hardyanto.donny at gmail.com Wed Nov 2 09:50:51 2016 From: hardyanto.donny at gmail.com (Donny Hardyanto) Date: Wed, 2 Nov 2016 13:50:51 +0700 Subject: [Freeswitch-users] JSAPI in Verto In-Reply-To: References: Message-ID: So if no jsonrpc-allowed-jsapi and -fsapi in user config, user cannot send jsapi/fsapi via verto, and only can behave like standard sip user? Donny Pada tanggal 1 Nov 2016 2:14 AM, "Michael Jerris" menulis: > you can secure it on a user by user basis, but limited only to which > commands that user can run the attrs are: > > jsonrpc-allowed-jsapi > jsonrpc-allowed-fsapi > > for fsapi commands and jsapi commands. > > if you look in the default configs you can see similar settings put in at > a global level, the same is possible per user: > > conf/testing/directory/default.xml:5: value="verto"/> > conf/vanilla/directory/default.xml:26: value="verto"/> > conf/vanilla/directory/default.xml:27: > > > > On Oct 31, 2016, at 3:00 AM, Donny Hardyanto > wrote: > > Hi all, > > Is it possible to run JSAPI via verto? Is it as powerfull as Event Socket? > Is there any way to secure it? I like to deploy the verto webrtc to > website, but dont want any one can abuse and control my FS box via > JSAPi/verto. > > Thanks, > > Donny > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > 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/20161102/c46379f8/attachment-0001.html From devang.nathwani31589 at gmail.com Wed Nov 2 09:52:34 2016 From: devang.nathwani31589 at gmail.com (devang nathwani) Date: Wed, 2 Nov 2016 12:22:34 +0530 Subject: [Freeswitch-users] 503- service unavailable In-Reply-To: References: <8521734C-2A6C-4BCE-95C8-B625F91157DD@jerris.com> Message-ID: Hi Brian, Below is the full fs log at debug + sip trace + loglevel 9 https://pastebin.freeswitch.org/view/f2d2606f On Thu, Oct 27, 2016 at 9:54 PM, Brian West wrote: > In addition this log is absent every single log line out of FreeSWITCH. > > /b > > > On Thu, Oct 27, 2016 at 11:24 AM, Brian West wrote: > >> You can use ours pastebin.freeswtich.org >> >> In the future please do not post log files like this to the list. >> >> /b >> >> >> On Thu, Oct 27, 2016 at 9:56 AM, devang nathwani < >> devang.nathwani31589 at gmail.com> wrote: >> >>> Hello, this is requested log, >>> got this from pastebin, >>> You have exceeded the maximum paste size of 512 kilobytes per paste. >>> so attaching the txt file >>> >>> On Thu, Oct 27, 2016 at 8:12 PM, Brian West >>> wrote: >>> >>>> Well this log is somewhat useless to get a full picture, what we need >>>> is the full fs log at debug + sip trace + loglevel 9 >>>> >>>> But at first glance: >>>> >>>> tport_udp_error: No route to host (113) [icmp type=3 code=1] >>>> >>>> That'll cause a 503 for sure. Again guessing because I do not see the >>>> full picture. >>>> >>>> /b >>>> >>>> >>>> On Thu, Oct 27, 2016 at 9:34 AM, devang nathwani < >>>> devang.nathwani31589 at gmail.com> wrote: >>>> >>>>> Thanks Brian, >>>>> >>>>> RAM is upgraded to 16 gigs now >>>>> this is the output of 'status' >>>>> status >>>>> UP 0 years, 0 days, 1 hour, 25 minutes, 47 seconds, 805 milliseconds, >>>>> 279 microseconds >>>>> FreeSWITCH (Version 1.6.11 64bit) is ready >>>>> 251 session(s) since startup >>>>> 0 session(s) - peak 7, last 5min 0 >>>>> 0 session(s) per Sec out of max 300, peak 7, last 5min 0 >>>>> 1000 session(s) max >>>>> min idle cpu 0.00/95.80 >>>>> Current Stack Size/Max 240K/8192K >>>>> >>>>> this sip log is with only INVITE request and nothing back from >>>>> freeswitch >>>>> http://pastebin.com/41AtUAje >>>>> >>>>> this sip log is with INVITE and response of 503 from freeswitch >>>>> http://pastebin.com/PXz277XD >>>>> >>>>> this is the from >>>>> sofia loglevel all 9 >>>>> http://pastebin.com/fBPhfkns >>>>> >>>>> On Thu, Oct 27, 2016 at 7:32 PM, Brian West >>>>> wrote: >>>>> >>>>>> sofia global siptrace on >>>>>> >>>>>> Look at the logs and signaling its usually informative, You can also >>>>>> do sofia loglevel all 9 to really get more details. >>>>>> >>>>>> On Thu, Oct 27, 2016 at 7:38 AM, devang nathwani < >>>>>> devang.nathwani31589 at gmail.com> wrote: >>>>>> >>>>>>> Thanks Michael, >>>>>>> [INFO] mod_dptools.c:3401 Originate Failed. Cause: >>>>>>> NORMAL_TEMPORARY_FAILURE >>>>>>> >>>>>>> This is what freeswitch thinks when i got 503 service unavailable in >>>>>>> sip log >>>>>>> >>>>>>> On Thu, Oct 27, 2016 at 5:57 PM, Michael Jerris >>>>>>> wrote: >>>>>>> >>>>>>>> Have you looked at a debug log when this is happening to see what >>>>>>>> FreeSWITCH thinks is going on? >>>>>>>> >>>>>>>> On Oct 27, 2016, at 7:14 AM, devang nathwani < >>>>>>>> devang.nathwani31589 at gmail.com> wrote: >>>>>>>> >>>>>>>> Hello, >>>>>>>> >>>>>>>> Freeswitch version: 1.6.11 >>>>>>>> Number of registration on my switch: 270+ >>>>>>>> Ram: 4 GB, 1 GB is free; 64 bit debian 8 >>>>>>>> >>>>>>>> I am facing this issue, where some time i am getting continuous >>>>>>>> INVITE registration requests but freeswitch not responding to it >>>>>>>> >>>>>>>> and sometime i am getting 503 service unavailable >>>>>>>> >>>>>>>> I have gone though https://freeswitch.org/jira/si >>>>>>>> /jira.issueviews:issue-html/FS-6589/FS-6589.html as well but not >>>>>>>> finding much helpful. >>>>>>>> >>>>>>>> here is the sip log for the same, if you want to go through. >>>>>>>> http://pastebin.com/PXz277XD >>>>>>>> >>>>>>>> Is RAM upgrade a solution? or any other? >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> ____________________________________________________________ >>>>>>>> _____________ >>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>> consulting at freeswitch.org >>>>>>>> http://www.freeswitchsolutions.com >>>>>>>> >>>>>>>> Official FreeSWITCH Sites >>>>>>>> http://www.freeswitch.org >>>>>>>> http://confluence.freeswitch.org >>>>>>>> http://www.cluecon.com >>>>>>>> >>>>>>>> FreeSWITCH-users mailing list >>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>>>>>> switch-users >>>>>>>> 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/free >>>>>>> switch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> >>>>>> *Brian West* >>>>>> brian at freeswitch.org >>>>>> >>>>>> >>>>>> *Twitter: @FreeSWITCH , @briankwest* >>>>>> http://www.freeswitchbook.com (50% Discount using code FreeSwitch50) >>>>>> http://www.freeswitchcookbook.com (50% Discount using code >>>>>> FreeSwitch50) >>>>>> https://www.gofundme.com/freeswitch_ubuntu >>>>>> >>>>>> Got Bugs? Report them here ! | Reddit: >>>>>> /r/freeswitch >>>>>> >>>>>> *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/free >>>>>> switch-users >>>>>> 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/free >>>>> switch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> >>>> -- >>>> >>>> *Brian West* >>>> brian at freeswitch.org >>>> >>>> >>>> *Twitter: @FreeSWITCH , @briankwest* >>>> http://www.freeswitchbook.com (50% Discount using code FreeSwitch50) >>>> http://www.freeswitchcookbook.com (50% Discount using code >>>> FreeSwitch50) >>>> https://www.gofundme.com/freeswitch_ubuntu >>>> >>>> Got Bugs? Report them here ! | Reddit: >>>> /r/freeswitch >>>> >>>> *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/free >>>> switch-users >>>> 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 (50% Discount using code FreeSwitch50) >> http://www.freeswitchcookbook.com (50% Discount using code FreeSwitch50) >> https://www.gofundme.com/freeswitch_ubuntu >> >> Got Bugs? Report them here ! | Reddit: >> /r/freeswitch >> >> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >> > > > > -- > > *Brian West* > brian at freeswitch.org > > > *Twitter: @FreeSWITCH , @briankwest* > http://www.freeswitchbook.com (50% Discount using code FreeSwitch50) > http://www.freeswitchcookbook.com (50% Discount using code FreeSwitch50) > https://www.gofundme.com/freeswitch_ubuntu > > Got Bugs? Report them here ! | Reddit: > /r/freeswitch > > *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/20161102/96887158/attachment-0001.html From brian at freeswitch.org Wed Nov 2 13:18:43 2016 From: brian at freeswitch.org (Brian West) Date: Wed, 2 Nov 2016 05:18:43 -0500 Subject: [Freeswitch-users] 503- service unavailable In-Reply-To: References: <8521734C-2A6C-4BCE-95C8-B625F91157DD@jerris.com> Message-ID: That log does not contain any 503 Service Unavailable, Sure this is the correct log? On Wed, Nov 2, 2016 at 1:52 AM, devang nathwani < devang.nathwani31589 at gmail.com> wrote: > Hi Brian, > > Below is the full fs log at debug + sip trace + loglevel 9 > https://pastebin.freeswitch.org/view/f2d2606f > > On Thu, Oct 27, 2016 at 9:54 PM, Brian West wrote: > >> In addition this log is absent every single log line out of FreeSWITCH. >> >> /b >> >> >> On Thu, Oct 27, 2016 at 11:24 AM, Brian West >> wrote: >> >>> You can use ours pastebin.freeswtich.org >>> >>> In the future please do not post log files like this to the list. >>> >>> /b >>> >>> >>> On Thu, Oct 27, 2016 at 9:56 AM, devang nathwani < >>> devang.nathwani31589 at gmail.com> wrote: >>> >>>> Hello, this is requested log, >>>> got this from pastebin, >>>> You have exceeded the maximum paste size of 512 kilobytes per paste. >>>> so attaching the txt file >>>> >>>> On Thu, Oct 27, 2016 at 8:12 PM, Brian West >>>> wrote: >>>> >>>>> Well this log is somewhat useless to get a full picture, what we need >>>>> is the full fs log at debug + sip trace + loglevel 9 >>>>> >>>>> But at first glance: >>>>> >>>>> tport_udp_error: No route to host (113) [icmp type=3 code=1] >>>>> >>>>> That'll cause a 503 for sure. Again guessing because I do not see the >>>>> full picture. >>>>> >>>>> /b >>>>> >>>>> >>>>> On Thu, Oct 27, 2016 at 9:34 AM, devang nathwani < >>>>> devang.nathwani31589 at gmail.com> wrote: >>>>> >>>>>> Thanks Brian, >>>>>> >>>>>> RAM is upgraded to 16 gigs now >>>>>> this is the output of 'status' >>>>>> status >>>>>> UP 0 years, 0 days, 1 hour, 25 minutes, 47 seconds, 805 milliseconds, >>>>>> 279 microseconds >>>>>> FreeSWITCH (Version 1.6.11 64bit) is ready >>>>>> 251 session(s) since startup >>>>>> 0 session(s) - peak 7, last 5min 0 >>>>>> 0 session(s) per Sec out of max 300, peak 7, last 5min 0 >>>>>> 1000 session(s) max >>>>>> min idle cpu 0.00/95.80 >>>>>> Current Stack Size/Max 240K/8192K >>>>>> >>>>>> this sip log is with only INVITE request and nothing back from >>>>>> freeswitch >>>>>> http://pastebin.com/41AtUAje >>>>>> >>>>>> this sip log is with INVITE and response of 503 from freeswitch >>>>>> http://pastebin.com/PXz277XD >>>>>> >>>>>> this is the from >>>>>> sofia loglevel all 9 >>>>>> http://pastebin.com/fBPhfkns >>>>>> >>>>>> On Thu, Oct 27, 2016 at 7:32 PM, Brian West >>>>>> wrote: >>>>>> >>>>>>> sofia global siptrace on >>>>>>> >>>>>>> Look at the logs and signaling its usually informative, You can also >>>>>>> do sofia loglevel all 9 to really get more details. >>>>>>> >>>>>>> On Thu, Oct 27, 2016 at 7:38 AM, devang nathwani < >>>>>>> devang.nathwani31589 at gmail.com> wrote: >>>>>>> >>>>>>>> Thanks Michael, >>>>>>>> [INFO] mod_dptools.c:3401 Originate Failed. Cause: >>>>>>>> NORMAL_TEMPORARY_FAILURE >>>>>>>> >>>>>>>> This is what freeswitch thinks when i got 503 service unavailable >>>>>>>> in sip log >>>>>>>> >>>>>>>> On Thu, Oct 27, 2016 at 5:57 PM, Michael Jerris >>>>>>>> wrote: >>>>>>>> >>>>>>>>> Have you looked at a debug log when this is happening to see what >>>>>>>>> FreeSWITCH thinks is going on? >>>>>>>>> >>>>>>>>> On Oct 27, 2016, at 7:14 AM, devang nathwani < >>>>>>>>> devang.nathwani31589 at gmail.com> wrote: >>>>>>>>> >>>>>>>>> Hello, >>>>>>>>> >>>>>>>>> Freeswitch version: 1.6.11 >>>>>>>>> Number of registration on my switch: 270+ >>>>>>>>> Ram: 4 GB, 1 GB is free; 64 bit debian 8 >>>>>>>>> >>>>>>>>> I am facing this issue, where some time i am getting continuous >>>>>>>>> INVITE registration requests but freeswitch not responding to it >>>>>>>>> >>>>>>>>> and sometime i am getting 503 service unavailable >>>>>>>>> >>>>>>>>> I have gone though https://freeswitch.org/jira/si >>>>>>>>> /jira.issueviews:issue-html/FS-6589/FS-6589.html as well but not >>>>>>>>> finding much helpful. >>>>>>>>> >>>>>>>>> here is the sip log for the same, if you want to go through. >>>>>>>>> http://pastebin.com/PXz277XD >>>>>>>>> >>>>>>>>> Is RAM upgrade a solution? or any other? >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> ____________________________________________________________ >>>>>>>>> _____________ >>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>> consulting at freeswitch.org >>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>> >>>>>>>>> Official FreeSWITCH Sites >>>>>>>>> http://www.freeswitch.org >>>>>>>>> http://confluence.freeswitch.org >>>>>>>>> http://www.cluecon.com >>>>>>>>> >>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>>>>>>> switch-users >>>>>>>>> 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/free >>>>>>>> switch-users >>>>>>>> http://www.freeswitch.org >>>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> -- >>>>>>> >>>>>>> *Brian West* >>>>>>> brian at freeswitch.org >>>>>>> >>>>>>> >>>>>>> *Twitter: @FreeSWITCH , @briankwest* >>>>>>> http://www.freeswitchbook.com (50% Discount using code FreeSwitch50) >>>>>>> http://www.freeswitchcookbook.com (50% Discount using code >>>>>>> FreeSwitch50) >>>>>>> https://www.gofundme.com/freeswitch_ubuntu >>>>>>> >>>>>>> Got Bugs? Report them here ! | Reddit: >>>>>>> /r/freeswitch >>>>>>> >>>>>>> *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/free >>>>>>> switch-users >>>>>>> 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/free >>>>>> switch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>> >>>>> >>>>> >>>>> -- >>>>> >>>>> *Brian West* >>>>> brian at freeswitch.org >>>>> >>>>> >>>>> *Twitter: @FreeSWITCH , @briankwest* >>>>> http://www.freeswitchbook.com (50% Discount using code FreeSwitch50) >>>>> http://www.freeswitchcookbook.com (50% Discount using code >>>>> FreeSwitch50) >>>>> https://www.gofundme.com/freeswitch_ubuntu >>>>> >>>>> Got Bugs? Report them here ! | Reddit: >>>>> /r/freeswitch >>>>> >>>>> *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/free >>>>> switch-users >>>>> 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/free >>>> switch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> >>> -- >>> >>> *Brian West* >>> brian at freeswitch.org >>> >>> >>> *Twitter: @FreeSWITCH , @briankwest* >>> http://www.freeswitchbook.com (50% Discount using code FreeSwitch50) >>> http://www.freeswitchcookbook.com (50% Discount using code FreeSwitch50) >>> https://www.gofundme.com/freeswitch_ubuntu >>> >>> Got Bugs? Report them here ! | Reddit: >>> /r/freeswitch >>> >>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >>> >> >> >> >> -- >> >> *Brian West* >> brian at freeswitch.org >> >> >> *Twitter: @FreeSWITCH , @briankwest* >> http://www.freeswitchbook.com (50% Discount using code FreeSwitch50) >> http://www.freeswitchcookbook.com (50% Discount using code FreeSwitch50) >> https://www.gofundme.com/freeswitch_ubuntu >> >> Got Bugs? Report them here ! | Reddit: >> /r/freeswitch >> >> *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 (50% Discount using code FreeSwitch50) http://www.freeswitchcookbook.com (50% Discount using code FreeSwitch50) https://www.gofundme.com/freeswitch_ubuntu Got Bugs? Report them here ! | Reddit: /r/freeswitch *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/20161102/7163eaf0/attachment-0001.html From dimitry.nagorny at gmx.de Wed Nov 2 10:24:34 2016 From: dimitry.nagorny at gmx.de (Dimitry Nagorny) Date: Wed, 2 Nov 2016 08:24:34 +0100 Subject: [Freeswitch-users] mod_event_socket with Dialplan Interaction Message-ID: An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161102/5c20ad69/attachment.html From brian at freeswitch.org Wed Nov 2 17:18:05 2016 From: brian at freeswitch.org (Brian West) Date: Wed, 2 Nov 2016 09:18:05 -0500 Subject: [Freeswitch-users] mod_event_socket with Dialplan Interaction In-Reply-To: References: Message-ID: bgapi originate {variable=foo,baz=bar}sofia/external/user at pbx &socket(...) You set the variables you wish to influence the sip leg inside the {} /b On Wed, Nov 2, 2016 at 2:24 AM, Dimitry Nagorny wrote: > Good morning all, > > I have a quick question: Is it possible that a received command through > mod_event_socket can be send through a (default) dialplan? > > Like ?bgapi originate /sofia/external/user at pbx &socket(...)? goes before > the INVITE through the dialplan to get details from DB. > > Why I want this? I want to enrich the sip of the INVITE with some > additional information. I already searched the wiki/Confluence and couldn?t > find an answer. > > > Best Regards > *Dimitry Nagorny* > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 (50% Discount using code FreeSwitch50) http://www.freeswitchcookbook.com (50% Discount using code FreeSwitch50) https://www.gofundme.com/freeswitch_ubuntu Got Bugs? Report them here ! | Reddit: /r/freeswitch *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/20161102/9661cd0c/attachment.html From treitinger at as-infodienste.de Wed Nov 2 17:22:52 2016 From: treitinger at as-infodienste.de (Melanie Treitinger) Date: Wed, 2 Nov 2016 15:22:52 +0100 Subject: [Freeswitch-users] ESL with PHP - last event is not received In-Reply-To: References: Message-ID: <4427c587-f175-a95f-3c12-462ea042a9cc@as-infodienste.de> Hello, has anyone ever used ESL to receive events? I'd really be glad to get some advice in this matter! Kind Regards, Melanie Am 28.10.2016 um 16:57 schrieb Melanie Treitinger: > Hello, > > I'm using a php event socket listener based on this example (wich is > linked in the freeswitch confluence): > https://bitbucket.org/lrichard/freeswitcheventsocketlistener > > It works well so far - but the last event of an event chaine is always > missing. It is only transfered when the next event takes place. > > For example when a conference ends we have the following events: > floor-change > play-file-done > del-member > conference-destroy > > But conference-destroy is not received - only when there is a next event! > > Is this a bug? Is the example missing something? > > > Thanks a lot! > Melanie > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 service at ivc.nnov.ru Wed Nov 2 17:37:36 2016 From: service at ivc.nnov.ru (Mikhail Demekhov) Date: Wed, 2 Nov 2016 17:37:36 +0300 Subject: [Freeswitch-users] FreeSwitch and CDR-Stats Message-ID: <5819FA30.40407@ivc.nnov.ru> Hello! Anyone using CDR-Stats (http://cdr-stats.readthedocs.io/en/stable/)? There are several questions on the use. Regards, Mikhail Demekhov From mike at jerris.com Wed Nov 2 17:44:54 2016 From: mike at jerris.com (Michael Jerris) Date: Wed, 2 Nov 2016 10:44:54 -0400 Subject: [Freeswitch-users] JSAPI in Verto In-Reply-To: References: Message-ID: They can be in user or domain configuration as i said. Not sure what you mean by behave like a standard sip user. We don't have a capability to execute things like that over sip. On Wednesday, November 2, 2016, Donny Hardyanto wrote: > So if no jsonrpc-allowed-jsapi and -fsapi in user config, user cannot send > jsapi/fsapi via verto, and only can behave like standard sip user? > > Donny > > Pada tanggal 1 Nov 2016 2:14 AM, "Michael Jerris" > menulis: > >> you can secure it on a user by user basis, but limited only to which >> commands that user can run the attrs are: >> >> jsonrpc-allowed-jsapi >> jsonrpc-allowed-fsapi >> >> for fsapi commands and jsapi commands. >> >> if you look in the default configs you can see similar settings put in at >> a global level, the same is possible per user: >> >> conf/testing/directory/default.xml:5: > value="verto"/> >> conf/vanilla/directory/default.xml:26: > value="verto"/> >> conf/vanilla/directory/default.xml:27: >> >> >> >> On Oct 31, 2016, at 3:00 AM, Donny Hardyanto > > wrote: >> >> Hi all, >> >> Is it possible to run JSAPI via verto? Is it as powerfull as Event >> Socket? Is there any way to secure it? I like to deploy the verto webrtc to >> website, but dont want any one can abuse and control my FS box via >> JSAPi/verto. >> >> Thanks, >> >> Donny >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> 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/20161102/e8adc2cc/attachment-0001.html From aqsyounas at gmail.com Wed Nov 2 18:04:43 2016 From: aqsyounas at gmail.com (Aqs Younas) Date: Wed, 2 Nov 2016 20:04:43 +0500 Subject: [Freeswitch-users] ESL with PHP - last event is not received In-Reply-To: <4427c587-f175-a95f-3c12-462ea042a9cc@as-infodienste.de> References: <4427c587-f175-a95f-3c12-462ea042a9cc@as-infodienste.de> Message-ID: In python, I use this. #!/usr/bin/env python ''' events.py - subscribe to all events and print them to stdout ''' import ESL con = ESL.ESLconnection('localhost', '8021', 'ClueCon') if con.connected: con.events('plain', 'all') while 1: e = con.recvEvent() if e: print e.serialize() -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161102/ab8f236c/attachment.html From david.villasmil.work at gmail.com Wed Nov 2 18:08:06 2016 From: david.villasmil.work at gmail.com (David Villasmil) Date: Wed, 02 Nov 2016 15:08:06 +0000 Subject: [Freeswitch-users] ESL with PHP - last event is not received In-Reply-To: References: <4427c587-f175-a95f-3c12-462ea042a9cc@as-infodienste.de> Message-ID: Try removing all logic, and just subscribing to all events and printing them out. I've never used esl with php, but I'd be surprised the last event is missing like that. On Wed, Nov 2, 2016 at 11:05 AM Aqs Younas wrote: > In python, I use this. > > #!/usr/bin/env python > ''' > events.py - subscribe to all events and print them to stdout > ''' > import ESL > > con = ESL.ESLconnection('localhost', '8021', 'ClueCon') > > if con.connected: > con.events('plain', 'all') > while 1: > e = con.recvEvent() > if e: > print e.serialize() > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161102/5062a996/attachment.html From blackc2004 at gmail.com Wed Nov 2 19:01:58 2016 From: blackc2004 at gmail.com (Cj B) Date: Wed, 2 Nov 2016 09:01:58 -0700 Subject: [Freeswitch-users] Verto conference and Yealink T49 video Message-ID: Hi all, I have verto working and am able to do video conferences with multiple people. But I was wondering what it takes to get a yealink t49 or other video capable phone to be able to join the conference and share video? I haven?t been able to find anything in documentation or the email list. I can do T49->T49 video right now, but when they join the conference there?s no video. Thanks! Cj B -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161102/da01df03/attachment.html From brian at freeswitch.org Wed Nov 2 19:12:30 2016 From: brian at freeswitch.org (Brian West) Date: Wed, 2 Nov 2016 11:12:30 -0500 Subject: [Freeswitch-users] Verto conference and Yealink T49 video In-Reply-To: References: Message-ID: I'm going to guess your canvas size is larger than the phone can support, You'll probably want to set this variable before you answer video_mirror_input=true /b On Wed, Nov 2, 2016 at 11:01 AM, Cj B wrote: > Hi all, > > I have verto working and am able to do video conferences with multiple > people. But I was wondering what it takes to get a yealink t49 or other > video capable phone to be able to join the conference and share video? I > haven?t been able to find anything in documentation or the email list. > > I can do T49->T49 video right now, but when they join the conference > there?s no video. > > Thanks! > Cj B > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 (50% Discount using code FreeSwitch50) http://www.freeswitchcookbook.com (50% Discount using code FreeSwitch50) https://www.gofundme.com/freeswitch_ubuntu Got Bugs? Report them here ! | Reddit: /r/freeswitch *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/20161102/ca14866d/attachment.html From blackc2004 at gmail.com Wed Nov 2 20:39:18 2016 From: blackc2004 at gmail.com (Cj B) Date: Wed, 2 Nov 2016 10:39:18 -0700 Subject: [Freeswitch-users] Verto conference and Yealink T49 video In-Reply-To: References: Message-ID: <09FB5EEC-246A-4C04-B40E-0319C004CA18@gmail.com> Thanks Brian for the quick reply. I added that to my dialplan and am still not getting video from the T49. Here?s a log of the call: http://pastebin.com/zk9TPYfp Extension 1007 is the Yealink T49. Any other ideas? Thanks for the help! Cj B > On Nov 2, 2016, at 9:12 AM, Brian West wrote: > > I'm going to guess your canvas size is larger than the phone can support, You'll probably want to set this variable before you answer video_mirror_input=true > > /b > > On Wed, Nov 2, 2016 at 11:01 AM, Cj B > wrote: > Hi all, > > I have verto working and am able to do video conferences with multiple people. But I was wondering what it takes to get a yealink t49 or other video capable phone to be able to join the conference and share video? I haven?t been able to find anything in documentation or the email list. > > I can do T49->T49 video right now, but when they join the conference there?s no video. > > Thanks! > Cj B > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 (50% Discount using code FreeSwitch50) > http://www.freeswitchcookbook.com (50% Discount using code FreeSwitch50) > https://www.gofundme.com/freeswitch_ubuntu > Got Bugs? Report them here ! | Reddit: /r/freeswitch > 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/20161102/3b221f70/attachment-0001.html From mandra at gmail.com Wed Nov 2 21:56:55 2016 From: mandra at gmail.com (Chris Mandra) Date: Wed, 2 Nov 2016 14:56:55 -0400 Subject: [Freeswitch-users] linking question Message-ID: Hi Guys - happy Weds! we're writing a module where we need to link in a static library. In the makefile for the module we use: LIBADD+=.a The build goes fine. However, when we load the module we get an error about missing symbols indicating the library was not properly linked. What is the correct syntax in the makefile to link in static libraries? Thanks! chris -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161102/d6168788/attachment.html From nbhatti at gmail.com Wed Nov 2 22:10:59 2016 From: nbhatti at gmail.com (Muhammad Naseer Bhatti) Date: Wed, 02 Nov 2016 22:10:59 +0300 Subject: [Freeswitch-users] How to figure out mod_sofia.c:4391 Gateway 'vgaccount' is down! Message-ID: <1e88f5e7-3b9c-4704-8748-c680a67de771@gmail.com> I am trying to diagnose a gateway down issue, 2016-11-02 17:59:41.304402 [ERR] mod_sofia.c:4391 Gateway 'vgaccount' is down! (Yes it?s a little old version but need to figure out the root cause) I have the gigantic sip trace and the log files but not able to figure out why I am seeing this message. Though at the same time I am able to ping the gateway fine from the same machine. What sort of symptoms should I look for in the log files? I am already looking into network connectivity and other system log files but nothing as of now. Thanks. via Newton Mail [https://cloudmagic.com/k/d/mailapp?ct=dx&cv=9.1.25&pv=10.12.1&source=email_footer_2] -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161102/27003ecb/attachment.html From david.villasmil.work at gmail.com Wed Nov 2 22:14:52 2016 From: david.villasmil.work at gmail.com (David Villasmil) Date: Wed, 02 Nov 2016 19:14:52 +0000 Subject: [Freeswitch-users] How to figure out mod_sofia.c:4391 Gateway 'vgaccount' is down! In-Reply-To: <1e88f5e7-3b9c-4704-8748-c680a67de771@gmail.com> References: <1e88f5e7-3b9c-4704-8748-c680a67de771@gmail.com> Message-ID: Is the gw responding the OPTIONs? On Wed, Nov 2, 2016 at 3:11 PM Muhammad Naseer Bhatti wrote: > I am trying to diagnose a gateway down issue, > > 2016-11-02 17:59:41.304402 [ERR] mod_sofia.c:4391 Gateway 'vgaccount' is > down! > > (Yes it?s a little old version but need to figure out the root cause) > > I have the gigantic sip trace and the log files but not able to figure out > why I am seeing this message. Though at the same time I am able to ping the > gateway fine from the same machine. What sort of symptoms should I look > for in the log files? I am already looking into network connectivity and > other system log files but nothing as of now. > > Thanks. > > via Newton Mail > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161102/6f226543/attachment.html From nbhatti at gmail.com Wed Nov 2 22:17:14 2016 From: nbhatti at gmail.com (Muhammad Naseer Bhatti) Date: Wed, 02 Nov 2016 22:17:14 +0300 Subject: [Freeswitch-users] How to figure out mod_sofia.c:4391 Gateway 'vgaccount' is down! In-Reply-To: References: <1e88f5e7-3b9c-4704-8748-c680a67de771@gmail.com> Message-ID: <320c3eb1-76aa-43d0-a135-c591d73067d4@gmail.com> Yes it is. The gateway works perfectly and out of sudden it goes down. But then FreeSWITCH takes a few seconds to retry and then marks it back up. via Newton Mail [https://cloudmagic.com/k/d/mailapp?ct=dx&cv=9.1.25&pv=10.12.1&source=email_footer_2] On Wed, Nov 2, 2016 at 10:14 PM, David Villasmil wrote: Is the gw responding the OPTIONs? On Wed, Nov 2, 2016 at 3:11 PM Muhammad Naseer Bhatti < nbhatti at gmail.com [nbhatti at gmail.com] > wrote: I am trying to diagnose a gateway down issue, 2016-11-02 17:59:41.304402 [ERR] mod_sofia.c:4391 Gateway 'vgaccount' is down! (Yes it?s a little old version but need to figure out the root cause) I have the gigantic sip trace and the log files but not able to figure out why I am seeing this message. Though at the same time I am able to ping the gateway fine from the same machine. What sort of symptoms should I look for in the log files? I am already looking into network connectivity and other system log files but nothing as of now. Thanks. via Newton Mail [https://cloudmagic.com/k/d/mailapp?ct=dx&cv=9.1.25&pv=10.12.1&source=email_footer_2] _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org [consulting at freeswitch.org] http://www.freeswitchsolutions.com [http://www.freeswitchsolutions.com] Official FreeSWITCH Sites http://www.freeswitch.org [http://www.freeswitch.org] http://confluence.freeswitch.org [http://confluence.freeswitch.org] http://www.cluecon.com [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 [http://lists.freeswitch.org/mailman/listinfo/freeswitch-users] UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users [http://lists.freeswitch.org/mailman/options/freeswitch-users] http://www.freeswitch.org [http://www.freeswitch.org] -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161102/048f0586/attachment.html From david.villasmil.work at gmail.com Wed Nov 2 22:20:53 2016 From: david.villasmil.work at gmail.com (David Villasmil) Date: Wed, 02 Nov 2016 19:20:53 +0000 Subject: [Freeswitch-users] How to figure out mod_sofia.c:4391 Gateway 'vgaccount' is down! In-Reply-To: <320c3eb1-76aa-43d0-a135-c591d73067d4@gmail.com> References: <1e88f5e7-3b9c-4704-8748-c680a67de771@gmail.com> <320c3eb1-76aa-43d0-a135-c591d73067d4@gmail.com> Message-ID: Can you get a trace showing the OPTIONs and the responses, as well as a trace of a call right after the OPTIONs' answer? On Wed, Nov 2, 2016 at 3:18 PM Muhammad Naseer Bhatti wrote: > Yes it is. The gateway works perfectly and out of sudden it goes down. But > then FreeSWITCH takes a few seconds to retry and then marks it back up. > > > via Newton Mail > > > On Wed, Nov 2, 2016 at 10:14 PM, David Villasmil < > david.villasmil.work at gmail.com> wrote: > > Is the gw responding the OPTIONs? > On Wed, Nov 2, 2016 at 3:11 PM Muhammad Naseer Bhatti > wrote: > > I am trying to diagnose a gateway down issue, > > 2016-11-02 17:59:41.304402 [ERR] mod_sofia.c:4391 Gateway 'vgaccount' is > down! > > (Yes it?s a little old version but need to figure out the root cause) > > I have the gigantic sip trace and the log files but not able to figure out > why I am seeing this message. Though at the same time I am able to ping the > gateway fine from the same machine. What sort of symptoms should I look for > in the log files? I am already looking into network connectivity and other > system log files but nothing as of now. > > Thanks. > > via Newton Mail > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > 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/20161102/70705576/attachment-0001.html From mike at jerris.com Wed Nov 2 22:28:03 2016 From: mike at jerris.com (Michael Jerris) Date: Wed, 2 Nov 2016 15:28:03 -0400 Subject: [Freeswitch-users] linking question In-Reply-To: References: Message-ID: LIBADD is the right thing to do in automake. If using automake in the freesiwtch build system you can do: V=1 make and it will provide more verbose output to help you troubleshoot the problem. > On Nov 2, 2016, at 2:56 PM, Chris Mandra wrote: > > Hi Guys - happy Weds! > > we're writing a module where we need to link in a static library. In the makefile for the module we use: LIBADD+=.a > > The build goes fine. However, when we load the module we get an error about missing symbols indicating the library was not properly linked. What is the correct syntax in the makefile to link in static libraries? > > Thanks! > chris From mike at jerris.com Wed Nov 2 22:29:37 2016 From: mike at jerris.com (Michael Jerris) Date: Wed, 2 Nov 2016 15:29:37 -0400 Subject: [Freeswitch-users] How to figure out mod_sofia.c:4391 Gateway 'vgaccount' is down! In-Reply-To: <1e88f5e7-3b9c-4704-8748-c680a67de771@gmail.com> References: <1e88f5e7-3b9c-4704-8748-c680a67de771@gmail.com> Message-ID: How old of a version? I vaguely recall a bug in this quite a while back. > On Nov 2, 2016, at 3:10 PM, Muhammad Naseer Bhatti wrote: > > > I am trying to diagnose a gateway down issue, > > 2016-11-02 17:59:41.304402 [ERR] mod_sofia.c:4391 Gateway 'vgaccount' is down! > > (Yes it?s a little old version but need to figure out the root cause) > > I have the gigantic sip trace and the log files but not able to figure out why I am seeing this message. Though at the same time I am able to ping the gateway fine from the same machine. What sort of symptoms should I look for in the log files? I am already looking into network connectivity and other system log files but nothing as of now. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161102/f95adc65/attachment.html From nbhatti at gmail.com Wed Nov 2 23:11:26 2016 From: nbhatti at gmail.com (Muhammad Naseer Bhatti) Date: Wed, 02 Nov 2016 23:11:26 +0300 Subject: [Freeswitch-users] How to figure out mod_sofia.c:4391 Gateway 'vgaccount' is down! In-Reply-To: References: <1e88f5e7-3b9c-4704-8748-c680a67de771@gmail.com> Message-ID: Shamefully quite old :( FreeSWITCH Version 1.5.15b+git~20150222T001715Z~9df39b8fe4~64bit (git 9df39b8 2015-02-22 00:17:15Z 64bit) I am going to try the latest version to see if I can replicate this. But generally speaking, how to troubleshoot this behavior? via Newton Mail [https://cloudmagic.com/k/d/mailapp?ct=dx&cv=9.1.25&pv=10.12.1&source=email_footer_2] On Wed, Nov 2, 2016 at 10:29 PM, Michael Jerris wrote: How old of a version? I vaguely recall a bug in this quite a while back. On Nov 2, 2016, at 3:10 PM, Muhammad Naseer Bhatti < nbhatti at gmail.com [nbhatti at gmail.com] > wrote: I am trying to diagnose a gateway down issue, 2016-11-02 17:59:41.304402 [ERR] mod_sofia.c:4391 Gateway 'vgaccount' is down! (Yes it?s a little old version but need to figure out the root cause) I have the gigantic sip trace and the log files but not able to figure out why I am seeing this message. Though at the same time I am able to ping the gateway fine from the same machine. What sort of symptoms should I look for in the log files? I am already looking into network connectivity and other system log files but nothing as of now. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161102/39bb6f20/attachment.html From zrothy at monmouth.com Thu Nov 3 00:15:19 2016 From: zrothy at monmouth.com (Zach Rothy) Date: Wed, 02 Nov 2016 17:15:19 -0400 Subject: [Freeswitch-users] Screen sharing with verto.js on firefox Message-ID: <2e8cedee21edaa6505b6d5add0cae534@monmouth.com> Hi, Recently I started a video conferencing project using Freeswitch 1.6.10 running on Centos 7. I had decided to user mod_verto and its javascript library verto.js to create my own client since the verto communicator demo is just a demo and isn't meant for production. I've got all the basics I needed for, video and audio calls on Firefox and chrome, but when it comes to screen sharing it gets a bit confusing. Like the Verto Communicator demo I have the screen sharing using getScreenId.js and in Google Chrome I have it working. Though when it comes to Firefox, getScreenId returns a different object completely than it will for chrome and it seems if you get a mediaStream object from getUserMedia you can't just blindly drop it in the same way as Chrome. I tried asking on the IRC and HipChat, and the only thing I got back was something about adapter.js on the "latest stuff", though when I checked master on the Freeswitch repo I did not see anything regarding this in the Verto Communicator folder. I was wondering if anyone has successfully gotten Firefox to be able to screen share with verto.js and mod_verto? If so how was it done, and if not are there any ideas or clues of where to start looking myself. Thanks in advance, Zach From mike at jerris.com Thu Nov 3 00:22:31 2016 From: mike at jerris.com (Michael Jerris) Date: Wed, 2 Nov 2016 17:22:31 -0400 Subject: [Freeswitch-users] How to figure out mod_sofia.c:4391 Gateway 'vgaccount' is down! In-Reply-To: References: <1e88f5e7-3b9c-4704-8748-c680a67de771@gmail.com> Message-ID: If its a bug in the logic it probably would require a reliable way to reproduce it then adding some logging to track it down? alternatively you could try to look through changes to mod_sofia from the last 18 months and maybe you find a patch in that logic that points to the problem. Either way, tracking down a reliable way to reproduce it is probably the first step. > On Nov 2, 2016, at 4:11 PM, Muhammad Naseer Bhatti wrote: > > > Shamefully quite old :( > > FreeSWITCH Version 1.5.15b+git~20150222T001715Z~9df39b8fe4~64bit (git 9df39b8 2015-02-22 00:17:15Z 64bit) > > I am going to try the latest version to see if I can replicate this. But generally speaking, how to troubleshoot this behavior? > > via Newton Mail > On Wed, Nov 2, 2016 at 10:29 PM, Michael Jerris wrote: > How old of a version? I vaguely recall a bug in this quite a while back. > > On Nov 2, 2016, at 3:10 PM, Muhammad Naseer Bhatti > wrote: > > I am trying to diagnose a gateway down issue, > > 2016-11-02 17:59:41.304402 [ERR] mod_sofia.c:4391 Gateway 'vgaccount' is down! > > (Yes it?s a little old version but need to figure out the root cause) > > I have the gigantic sip trace and the log files but not able to figure out why I am seeing this message. Though at the same time I am able to ping the gateway fine from the same machine. What sort of symptoms should I look for in the log files? I am already looking into network connectivity and other system log files but nothing as of now. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161102/b105ecf3/attachment.html From anthony.minessale at gmail.com Thu Nov 3 00:28:26 2016 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 2 Nov 2016 16:28:26 -0500 Subject: [Freeswitch-users] Screen sharing with verto.js on firefox In-Reply-To: <2e8cedee21edaa6505b6d5add0cae534@monmouth.com> References: <2e8cedee21edaa6505b6d5add0cae534@monmouth.com> Message-ID: No, We currently do not know how to do screen sharing in FireFox and in Chrome its still using a deprecated method because they have not fully implemented (or documented [at least where we can find them]) the constraints necessary to do it the new way. On Wed, Nov 2, 2016 at 4:15 PM, Zach Rothy wrote: > Hi, > > Recently I started a video conferencing project using Freeswitch 1.6.10 > running on Centos 7. I had decided to user mod_verto and its javascript > library verto.js to create my own client since the verto communicator > demo is just a demo and isn't meant for production. > > I've got all the basics I needed for, video and audio calls on Firefox > and chrome, but when it comes to screen sharing it gets a bit confusing. > Like the Verto Communicator demo I have the screen sharing using > getScreenId.js and in Google Chrome I have it working. Though when it > comes to Firefox, getScreenId returns a different object completely than > it will for chrome and it seems if you get a mediaStream object from > getUserMedia you can't just blindly drop it in the same way as Chrome. > > I tried asking on the IRC and HipChat, and the only thing I got back was > something about adapter.js on the "latest stuff", though when I checked > master on the Freeswitch repo I did not see anything regarding this in > the Verto Communicator folder. > > I was wondering if anyone has successfully gotten Firefox to be able to > screen share with verto.js and mod_verto? If so how was it done, and if > not are there any ideas or clues of where to start looking myself. > > Thanks in advance, > Zach > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161102/d1827626/attachment-0001.html From evmakarov at 1cbit.ru Thu Nov 3 00:48:14 2016 From: evmakarov at 1cbit.ru (=?UTF-8?B?0JXQstCz0LXQvdC40Lkg0JzQsNC60LDRgNC+0LI=?=) Date: Thu, 3 Nov 2016 02:48:14 +0500 Subject: [Freeswitch-users] Disable Service Route (RFC3608) Message-ID: <90d765fb-cd21-cfb9-98fb-ebc61b967139@1cbit.ru> Hi all! Friends, can you help me? How can i disable RFC 3608 in sofia profile? My SIP provider sends invalid value in this header. From mike at jerris.com Thu Nov 3 00:58:30 2016 From: mike at jerris.com (Michael Jerris) Date: Wed, 2 Nov 2016 17:58:30 -0400 Subject: [Freeswitch-users] Disable Service Route (RFC3608) In-Reply-To: <90d765fb-cd21-cfb9-98fb-ebc61b967139@1cbit.ru> References: <90d765fb-cd21-cfb9-98fb-ebc61b967139@1cbit.ru> Message-ID: I don?t think we have a way to do that. > On Nov 2, 2016, at 5:48 PM, ??????? ??????? wrote: > > Hi all! Friends, can you help me? > How can i disable RFC 3608 in sofia profile? > My SIP provider sends invalid value in this header. From brian at freeswitch.org Thu Nov 3 01:02:08 2016 From: brian at freeswitch.org (Brian West) Date: Wed, 2 Nov 2016 17:02:08 -0500 Subject: [Freeswitch-users] Disable Service Route (RFC3608) In-Reply-To: References: <90d765fb-cd21-cfb9-98fb-ebc61b967139@1cbit.ru> Message-ID: Can you provide examples of what you're talking about? On Wed, Nov 2, 2016 at 4:58 PM, Michael Jerris wrote: > I don?t think we have a way to do that. > > > On Nov 2, 2016, at 5:48 PM, ??????? ??????? wrote: > > > > Hi all! Friends, can you help me? > > How can i disable RFC 3608 in sofia profile? > > My SIP provider sends invalid value in this header. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 (50% Discount using code FreeSwitch50) http://www.freeswitchcookbook.com (50% Discount using code FreeSwitch50) https://www.gofundme.com/freeswitch_ubuntu Got Bugs? Report them here ! | Reddit: /r/freeswitch *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/20161102/c3c5fc77/attachment.html From evmakarov at 1cbit.ru Thu Nov 3 01:16:17 2016 From: evmakarov at 1cbit.ru (=?UTF-8?B?0JXQstCz0LXQvdC40Lkg0JzQsNC60LDRgNC+0LI=?=) Date: Thu, 3 Nov 2016 03:16:17 +0500 Subject: [Freeswitch-users] Disable Service Route (RFC3608) In-Reply-To: References: <90d765fb-cd21-cfb9-98fb-ebc61b967139@1cbit.ru> Message-ID: <59250614-bc5b-081f-4a02-2f762a4ab61d@1cbit.ru> I am talking about SIP connection to "Multifon". Russina GSM provider - Megafon. SBC sends for me "service-route" header and FreeSWITCH try send all requests to multifon. :-( 03.11.2016 03:02, Brian West ?????: > Can you provide examples of what you're talking about? > > On Wed, Nov 2, 2016 at 4:58 PM, Michael Jerris > wrote: > > I don?t think we have a way to do that. > > > On Nov 2, 2016, at 5:48 PM, ??????? ??????? > wrote: > > > > Hi all! Friends, can you help me? > > How can i disable RFC 3608 in sofia profile? > > My SIP provider sends invalid value in this header. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 (50% Discount using code FreeSwitch50) > http://www.freeswitchcookbook.com (50% Discount using code FreeSwitch50) > https://www.gofundme.com/freeswitch_ubuntu > > Got Bugs? Report them here ! | Reddit: > /r/freeswitch > > *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/20161103/7fb6f2d0/attachment.html From evmakarov at 1cbit.ru Thu Nov 3 01:29:46 2016 From: evmakarov at 1cbit.ru (=?UTF-8?B?0JXQstCz0LXQvdC40Lkg0JzQsNC60LDRgNC+0LI=?=) Date: Thu, 3 Nov 2016 03:29:46 +0500 Subject: [Freeswitch-users] Disable Service Route (RFC3608) In-Reply-To: References: <90d765fb-cd21-cfb9-98fb-ebc61b967139@1cbit.ru> Message-ID: I found something similar here: http://lists.freeswitch.org/pipermail/freeswitch-users/2011-February/069229.html But old thread doesn't have logical end. 03.11.2016 03:02, Brian West ?????: > Can you provide examples of what you're talking about? > > On Wed, Nov 2, 2016 at 4:58 PM, Michael Jerris > wrote: > > I don?t think we have a way to do that. > > > On Nov 2, 2016, at 5:48 PM, ??????? ??????? > wrote: > > > > Hi all! Friends, can you help me? > > How can i disable RFC 3608 in sofia profile? > > My SIP provider sends invalid value in this header. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 (50% Discount using code FreeSwitch50) > http://www.freeswitchcookbook.com (50% Discount using code FreeSwitch50) > https://www.gofundme.com/freeswitch_ubuntu > > Got Bugs? Report them here ! | Reddit: > /r/freeswitch > > *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/20161103/276f9ea3/attachment-0001.html From brian at freeswitch.org Thu Nov 3 01:53:41 2016 From: brian at freeswitch.org (Brian West) Date: Wed, 2 Nov 2016 17:53:41 -0500 Subject: [Freeswitch-users] Disable Service Route (RFC3608) In-Reply-To: <59250614-bc5b-081f-4a02-2f762a4ab61d@1cbit.ru> References: <90d765fb-cd21-cfb9-98fb-ebc61b967139@1cbit.ru> <59250614-bc5b-081f-4a02-2f762a4ab61d@1cbit.ru> Message-ID: I've never seen us add this hear, How is your FreeSWITCH configured? On Wed, Nov 2, 2016 at 5:16 PM, ??????? ??????? wrote: > I am talking about SIP connection to "Multifon". Russina GSM provider - > Megafon. > SBC sends for me "service-route" header and FreeSWITCH try send all > requests to multifon. :-( > 03.11.2016 03:02, Brian West ?????: > > Can you provide examples of what you're talking about? > > On Wed, Nov 2, 2016 at 4:58 PM, Michael Jerris wrote: > >> I don?t think we have a way to do that. >> >> > On Nov 2, 2016, at 5:48 PM, ??????? ??????? wrote: >> > >> > Hi all! Friends, can you help me? >> > How can i disable RFC 3608 in sofia profile? >> > My SIP provider sends invalid value in this header. >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 (50% Discount using code FreeSwitch50) > http://www.freeswitchcookbook.com (50% Discount using code FreeSwitch50) > https://www.gofundme.com/freeswitch_ubuntu > > Got Bugs? Report them here ! | Reddit: > /r/freeswitch > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest > > > _________________________________________________________________________ > 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 > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com (50% Discount using code FreeSwitch50) http://www.freeswitchcookbook.com (50% Discount using code FreeSwitch50) https://www.gofundme.com/freeswitch_ubuntu Got Bugs? Report them here ! | Reddit: /r/freeswitch *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/20161102/8ae70412/attachment.html From evmakarov at 1cbit.ru Thu Nov 3 02:04:51 2016 From: evmakarov at 1cbit.ru (=?UTF-8?B?0JXQstCz0LXQvdC40Lkg0JzQsNC60LDRgNC+0LI=?=) Date: Thu, 3 Nov 2016 04:04:51 +0500 Subject: [Freeswitch-users] Disable Service Route (RFC3608) In-Reply-To: References: <90d765fb-cd21-cfb9-98fb-ebc61b967139@1cbit.ru> <59250614-bc5b-081f-4a02-2f762a4ab61d@1cbit.ru> Message-ID: After update freeswitch from repo - trouble is gone! Magic! o_O ? ?????????, ??????? ??????? ????????? ????????????? ??????????? ?????????? ???????? ?????? ???, ??????????? ???????? ???.: +7 (495) 748-03-28 ???. 3166 VoIP ?????: 4061 ???. ???.: +7 (965) 505-05-71 Skype: patriot-8891 E-mail: EVMakarov at 1cbit.ru www.1cbit.ru 03.11.2016 03:53, Brian West ?????: > I've never seen us add this hear, How is your FreeSWITCH configured? > > On Wed, Nov 2, 2016 at 5:16 PM, ??????? ??????? > wrote: > > I am talking about SIP connection to "Multifon". Russina GSM > provider - Megafon. > SBC sends for me "service-route" header and FreeSWITCH try send > all requests to multifon. :-( > > 03.11.2016 03:02, Brian West ?????: > >> Can you provide examples of what you're talking about? >> >> On Wed, Nov 2, 2016 at 4:58 PM, Michael Jerris > > wrote: >> >> I don?t think we have a way to do that. >> >> > On Nov 2, 2016, at 5:48 PM, ??????? ??????? >> > wrote: >> > >> > Hi all! Friends, can you help me? >> > How can i disable RFC 3608 in sofia profile? >> > My SIP provider sends invalid value in this header. >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 (50% Discount using code FreeSwitch50) >> http://www.freeswitchcookbook.com >> (50% Discount using code >> FreeSwitch50) >> https://www.gofundme.com/freeswitch_ubuntu >> >> >> Got Bugs? Report them here ! | >> Reddit: /r/freeswitch >> >> *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 (50% Discount using code FreeSwitch50) > http://www.freeswitchcookbook.com (50% Discount using code > FreeSwitch50)https://www.gofundme.com/freeswitch_ubuntu > > Got Bugs? Report them here ! | Reddit: > /r/freeswitch > > *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/20161103/6582cd83/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: hpobdambigialmfd.png Type: image/png Size: 9414 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161103/6582cd83/attachment-0001.png From dimitry.nagorny at gmx.de Thu Nov 3 09:46:39 2016 From: dimitry.nagorny at gmx.de (Dimitry Nagorny) Date: Thu, 03 Nov 2016 07:46:39 +0100 Subject: [Freeswitch-users] mod_event_socket with Dialplan Interaction In-Reply-To: References: Message-ID: <908A0B1A-19D0-47ED-980E-5351BEF8F4D2@gmx.de> Am 2. November 2016 15:18:05 MEZ, schrieb Brian West : >bgapi originate {variable=foo,baz=bar}sofia/external/user at pbx >&socket(...) > >You set the variables you wish to influence the sip leg inside the {} > >/b > >On Wed, Nov 2, 2016 at 2:24 AM, Dimitry Nagorny > >wrote: > >> Good morning all, >> >> I have a quick question: Is it possible that a received command >through >> mod_event_socket can be send through a (default) dialplan? >> >> Like ?bgapi originate /sofia/external/user at pbx &socket(...)? goes >before >> the INVITE through the dialplan to get details from DB. >> >> Why I want this? I want to enrich the sip of the INVITE with some >> additional information. I already searched the wiki/Confluence and >couldn?t >> find an answer. >> >> >> Best Regards >> *Dimitry Nagorny* >> >> >_________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 (50% Discount using code FreeSwitch50) >http://www.freeswitchcookbook.com (50% Discount using code >FreeSwitch50) >https://www.gofundme.com/freeswitch_ubuntu > >Got Bugs? Report them here ! | Reddit: >/r/freeswitch > >*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 Thanks you for your answer Brian but maybe my question was wrong. I try to do something like bgapi sofia/default/user &socket (...) but this throws Invalid Profile. So instead of external/gateway I want it to go to dialplan "default". Or something similar that gives me the opperrunity to open a DB connection for a recieved request on the event_socket. I can't grab this info from within the third party system which sends the request. Thanks Dimitry -- Diese Nachricht wurde von meinem Android-Mobiltelefon mit K-9 Mail gesendet. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161103/58dda15b/attachment.html From yu at yu-boot.ru Thu Nov 3 10:29:38 2016 From: yu at yu-boot.ru (Yu Boot) Date: Thu, 3 Nov 2016 10:29:38 +0300 Subject: [Freeswitch-users] SIP accounts Message-ID: <679c193a-3a25-cc1f-4391-46e239aa004a@yu-boot.ru> Is it possible to store SIP clients' login/password in SQL base? "One file - one user" in directory is very uncomfortable, when you need to disable or enable some accounts. From gmaruzz at gmail.com Thu Nov 3 10:46:24 2016 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Thu, 3 Nov 2016 08:46:24 +0100 Subject: [Freeswitch-users] SIP accounts In-Reply-To: <679c193a-3a25-cc1f-4391-46e239aa004a@yu-boot.ru> References: <679c193a-3a25-cc1f-4391-46e239aa004a@yu-boot.ru> Message-ID: Yes it is, check mod-xml-curl and lua xmlhandler sent from mobile cell: +39 347 266 56 18 Giovanni Maruzzelli OpenTelecom.IT On Nov 3, 2016 08:30, "Yu Boot" wrote: > Is it possible to store SIP clients' login/password in SQL base? "One > file - one user" in directory is very uncomfortable, when you need to > disable or enable some accounts. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161103/2f9478a8/attachment.html From tarik at pasifiktelekom.com.tr Thu Nov 3 12:12:47 2016 From: tarik at pasifiktelekom.com.tr (tarik at pasifiktelekom.com.tr) Date: Thu, 3 Nov 2016 12:12:47 +0300 Subject: [Freeswitch-users] FreeSwitch and CDR-Stats Message-ID: An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161103/16a9ad0e/attachment.html From gregor at infomedia.si Thu Nov 3 13:39:03 2016 From: gregor at infomedia.si (Gregor Nanger) Date: Thu, 3 Nov 2016 11:39:03 +0100 Subject: [Freeswitch-users] Windows build v1.6 In-Reply-To: <1FBEEA8F-86E0-4ABF-AEB5-1BD17AF693C7@jerris.com> References: <1FBEEA8F-86E0-4ABF-AEB5-1BD17AF693C7@jerris.com> Message-ID: Just file Jira. Same issues. Latest ver 1.9 builds fine. Best regards, Gregor 2016-10-31 20:09 GMT+01:00 Michael Jerris : > if the autocrlf response doesn?t fix this i need a jira filed with the > full log of building the core, not just the error, so i can see the actual > issue causing the problem. > > > On Oct 30, 2016, at 8:00 PM, Gregor Nanger wrote: > > Need some help. > > Pulled latest -v16 version and on windows build I get error: > > Error C2220 warning treated as error - no 'object' file generated > FreeSwitchCoreLib D:\freeswitch\src\switch_rtp.c 38 > > It is in line 38 of file switch_rtp.c > > Any hints? > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > 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 > -- Gregor Nanger *CTO* t./f.: 00386 (0) 7 6000 308/309 ? m:. 00386 (0)41 756485 ? Infomedia d.o.o. ? Jerebova 3, Novo mesto, Slovenia ? www.infomedia.si -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161103/1dfaddb3/attachment-0001.html From treitinger at as-infodienste.de Thu Nov 3 13:54:26 2016 From: treitinger at as-infodienste.de (Melanie Treitinger) Date: Thu, 3 Nov 2016 11:54:26 +0100 Subject: [Freeswitch-users] ESL with PHP - last event is not received In-Reply-To: References: <4427c587-f175-a95f-3c12-462ea042a9cc@as-infodienste.de> Message-ID: Apparently it has something to do with the log level I had set. "log All" was wrong - when I switched to "log INFO", all events were reveived. https://freeswitch.org/confluence/display/FREESWITCH/mod_event_socket https://freeswitch.org/confluence/display/FREESWITCH/mod_console One more question: is it necessary to use "linger" here? Am 02.11.2016 um 16:08 schrieb David Villasmil: > Try removing all logic, and just subscribing to all events and printing > them out. I've never used esl with php, but I'd be surprised the last > event is missing like that. > On Wed, Nov 2, 2016 at 11:05 AM Aqs Younas > wrote: > > In python, I use this. > > #!/usr/bin/env python > ''' > events.py - subscribe to all events and print them to stdout > ''' > import ESL > > con = ESL.ESLconnection('localhost', '8021', 'ClueCon') > > if con.connected: > con.events('plain', 'all') > while 1: > e = con.recvEvent() > if e: > print e.serialize() > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > 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 steveayre at gmail.com Thu Nov 3 13:57:19 2016 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 3 Nov 2016 10:57:19 +0000 Subject: [Freeswitch-users] How to figure out mod_sofia.c:4391 Gateway 'vgaccount' is down! In-Reply-To: <320c3eb1-76aa-43d0-a135-c591d73067d4@gmail.com> References: <1e88f5e7-3b9c-4704-8748-c680a67de771@gmail.com> <320c3eb1-76aa-43d0-a135-c591d73067d4@gmail.com> Message-ID: Can you get a packet trace of the OPTIONS? This could mean you're getting packet loss, or the gateway isn't responding to some of the OPTIONS packets. If it's not 100% loss it'd let some packets through but occassionally drop others which can make the state flap between up and down. You can tune ping-min and ping-max to handle a small amount of missed OPTIONS without marking it as down. Although if there is loss it's likely going to impact on call quality. -Steve On 2 November 2016 at 19:17, Muhammad Naseer Bhatti wrote: > Yes it is. The gateway works perfectly and out of sudden it goes down. But > then FreeSWITCH takes a few seconds to retry and then marks it back up. > > > via Newton Mail > > > On Wed, Nov 2, 2016 at 10:14 PM, David Villasmil < > david.villasmil.work at gmail.com> wrote: > > Is the gw responding the OPTIONs? > On Wed, Nov 2, 2016 at 3:11 PM Muhammad Naseer Bhatti > wrote: > >> I am trying to diagnose a gateway down issue, >> >> 2016-11-02 17:59:41.304402 [ERR] mod_sofia.c:4391 Gateway 'vgaccount' is >> down! >> >> (Yes it?s a little old version but need to figure out the root cause) >> >> I have the gigantic sip trace and the log files but not able to figure >> out why I am seeing this message. Though at the same time I am able to ping >> the gateway fine from the same machine. What sort of symptoms should I look >> for in the log files? I am already looking into network connectivity and >> other system log files but nothing as of now. >> >> Thanks. >> >> via Newton Mail >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> 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/20161103/f2e03398/attachment.html From gregor at infomedia.si Thu Nov 3 14:28:13 2016 From: gregor at infomedia.si (Gregor Nanger) Date: Thu, 3 Nov 2016 12:28:13 +0100 Subject: [Freeswitch-users] Windows build v1.6 In-Reply-To: References: <1FBEEA8F-86E0-4ABF-AEB5-1BD17AF693C7@jerris.com> Message-ID: Just for info. Latest master 1.9 builds fine.... 2016-11-03 11:39 GMT+01:00 Gregor Nanger : > Just file Jira. > > Same issues. Latest ver 1.9 builds fine. > > Best regards, Gregor > > 2016-10-31 20:09 GMT+01:00 Michael Jerris : > >> if the autocrlf response doesn?t fix this i need a jira filed with the >> full log of building the core, not just the error, so i can see the actual >> issue causing the problem. >> >> >> On Oct 30, 2016, at 8:00 PM, Gregor Nanger wrote: >> >> Need some help. >> >> Pulled latest -v16 version and on windows build I get error: >> >> Error C2220 warning treated as error - no 'object' file generated >> FreeSwitchCoreLib D:\freeswitch\src\switch_rtp.c 38 >> >> It is in line 38 of file switch_rtp.c >> >> Any hints? >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> 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 >> > > > > -- > Gregor Nanger > > *CTO* > t./f.: 00386 (0) 7 6000 308/309 ? m:. 00386 (0)41 756485 > ? Infomedia d.o.o. ? Jerebova 3, Novo mesto, Slovenia > ? www.infomedia.si > -- Gregor Nanger *CTO* t./f.: 00386 (0) 7 6000 308/309 ? m:. 00386 (0)41 756485 ? Infomedia d.o.o. ? Jerebova 3, Novo mesto, Slovenia ? www.infomedia.si -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161103/204e78c1/attachment-0001.html From tarik at pasifiktelekom.com.tr Thu Nov 3 15:08:57 2016 From: tarik at pasifiktelekom.com.tr (tarik at pasifiktelekom.com.tr) Date: Thu, 3 Nov 2016 15:08:57 +0300 Subject: [Freeswitch-users] FW: FreeSwitch and CDR-Stats Message-ID: An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161103/685eff05/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: sqlite_schema.sql Type: application/sql Size: 744 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161103/685eff05/attachment.bin -------------- next part -------------- A non-text attachment was scrubbed... Name: cdr_import.log Type: text/x-log Size: 3945 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161103/685eff05/attachment-0001.bin -------------- next part -------------- A non-text attachment was scrubbed... Name: cdr-pusher.yaml Type: application/x-yaml Size: 3816 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161103/685eff05/attachment-0002.bin -------------- next part -------------- A non-text attachment was scrubbed... Name: cdr_sqlite.conf.xml Type: text/xml Size: 1318 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161103/685eff05/attachment.xml From service at ivc.nnov.ru Thu Nov 3 16:30:45 2016 From: service at ivc.nnov.ru (Mikhail Demekhov) Date: Thu, 3 Nov 2016 16:30:45 +0300 Subject: [Freeswitch-users] FreeSwitch and CDR-Stats In-Reply-To: <3AA99AB9F7B443B2A7BFDC2D9F4A6878.MAI@mx02.garantiserver.com> References: <3AA99AB9F7B443B2A7BFDC2D9F4A6878.MAI@mx02.garantiserver.com> Message-ID: <581B3C05.8040301@ivc.nnov.ru> Hi, > Thanks for the help! Cdr-pusher operates. But the CDR accumulates in the table cdr_import and I do not see CDR in Dashboard. djcelery_error.log: [2016-11-03 15:56:42,241: INFO/MainProcess] Connected to redis://localhost:6379/0 [2016-11-03 15:56:42,280: WARNING/MainProcess] celery at freesw ready. [2016-11-03 15:56:42,292: INFO/MainProcess] Received task: cdr.tasks.run_cdr_import[d85838e5-53b2-42c2-9966-efb9c10dcccd] [2016-11-03 15:56:42,295: INFO/MainProcess] Received task: cdr.tasks.refresh_materialized_views[4e2b2b34-ecf6-404d-add3-fb259142cdd9] [2016-11-03 15:56:42,297: INFO/Worker-11] cdr.tasks.refresh_materialized_views[4e2b2b34-ecf6-404d-add3-fb259142cdd9]: TASK :: refresh_materialized_views [2016-11-03 15:56:42,300: INFO/MainProcess] Received task: cdr.tasks.run_cdr_import[5e08cbbd-b6f5-40cb-9a57-a88a08a08efd] [2016-11-03 15:56:42,301: INFO/Worker-14] cdr.tasks.run_cdr_import[d85838e5-53b2-42c2-9966-efb9c10dcccd]: TASK :: run_cdr_import [2016-11-03 15:56:42,302: INFO/Worker-14] cdr.tasks.run_cdr_import[d85838e5-53b2-42c2-9966-efb9c10dcccd]: in func import_cdr... [2016-11-03 15:56:42,307: INFO/MainProcess] Received task: cdr.tasks.run_cdr_import[3a800491-a4e4-42ff-9da2-b98d0fa1f623] [2016-11-03 15:56:42,312: INFO/MainProcess] Received task: cdr.tasks.run_cdr_import[0a29e731-8692-420d-a05a-951ac7b0981f] [2016-11-03 15:56:42,317: INFO/MainProcess] Received task: cdr.tasks.run_cdr_import[623a625d-be2d-4345-8bd6-cca1d51fbf1b] [2016-11-03 15:56:42,318: INFO/MainProcess] Task cdr.tasks.run_cdr_import[5e08cbbd-b6f5-40cb-9a57-a88a08a08efd] succeeded in 0.00933933118358s: None [2016-11-03 15:56:42,319: INFO/MainProcess] Task cdr.tasks.run_cdr_import[3a800491-a4e4-42ff-9da2-b98d0fa1f623] succeeded in 0.00994342006743s: None [2016-11-03 15:56:42,323: INFO/MainProcess] Received task: cdr.tasks.run_cdr_import[0d430eb2-1e74-404b-a5e7-7d7cb0203445] [2016-11-03 15:56:42,328: INFO/MainProcess] Received task: cdr.tasks.run_cdr_import[ee738776-1c16-41ad-85f1-08ab5a358eaa] Regards, Mikhail Demekhov From mike at jerris.com Thu Nov 3 16:36:30 2016 From: mike at jerris.com (Michael Jerris) Date: Thu, 3 Nov 2016 09:36:30 -0400 Subject: [Freeswitch-users] Windows build v1.6 In-Reply-To: References: <1FBEEA8F-86E0-4ABF-AEB5-1BD17AF693C7@jerris.com> Message-ID: <0B7F5491-D1AA-4540-9127-8ADCF0DDB9EF@jerris.com> Yes, the issue has already been fixed in master. I have not yet done all the back ports to 1.6 branch. I?ll try to get these done before the next release. > On Nov 3, 2016, at 7:28 AM, Gregor Nanger wrote: > > Just for info. Latest master 1.9 builds fine.... > > 2016-11-03 11:39 GMT+01:00 Gregor Nanger >: > Just file Jira. > > Same issues. Latest ver 1.9 builds fine. > > Best regards, Gregor > > 2016-10-31 20:09 GMT+01:00 Michael Jerris >: > if the autocrlf response doesn?t fix this i need a jira filed with the full log of building the core, not just the error, so i can see the actual issue causing the problem. > > >> On Oct 30, 2016, at 8:00 PM, Gregor Nanger > wrote: >> >> Need some help. >> >> Pulled latest -v16 version and on windows build I get error: >> >> Error C2220 warning treated as error - no 'object' file generated FreeSwitchCoreLib D:\freeswitch\src\switch_rtp.c 38 >> >> It is in line 38 of file switch_rtp.c >> >> Any hints? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161103/c29058cd/attachment-0001.html From gregor at infomedia.si Thu Nov 3 16:42:51 2016 From: gregor at infomedia.si (Gregor Nanger) Date: Thu, 3 Nov 2016 14:42:51 +0100 Subject: [Freeswitch-users] Windows build v1.6 In-Reply-To: <0B7F5491-D1AA-4540-9127-8ADCF0DDB9EF@jerris.com> References: <1FBEEA8F-86E0-4ABF-AEB5-1BD17AF693C7@jerris.com> <0B7F5491-D1AA-4540-9127-8ADCF0DDB9EF@jerris.com> Message-ID: Cool, thank you... 2016-11-03 14:36 GMT+01:00 Michael Jerris : > Yes, the issue has already been fixed in master. I have not yet done all > the back ports to 1.6 branch. I?ll try to get these done before the next > release. > > On Nov 3, 2016, at 7:28 AM, Gregor Nanger wrote: > > Just for info. Latest master 1.9 builds fine.... > > 2016-11-03 11:39 GMT+01:00 Gregor Nanger : > >> Just file Jira. >> >> Same issues. Latest ver 1.9 builds fine. >> >> Best regards, Gregor >> >> 2016-10-31 20:09 GMT+01:00 Michael Jerris : >> >>> if the autocrlf response doesn?t fix this i need a jira filed with the >>> full log of building the core, not just the error, so i can see the actual >>> issue causing the problem. >>> >>> >>> On Oct 30, 2016, at 8:00 PM, Gregor Nanger wrote: >>> >>> Need some help. >>> >>> Pulled latest -v16 version and on windows build I get error: >>> >>> Error C2220 warning treated as error - no 'object' file generated >>> FreeSwitchCoreLib D:\freeswitch\src\switch_rtp.c 38 >>> >>> It is in line 38 of file switch_rtp.c >>> >>> Any hints? >>> >>> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Gregor Nanger *CTO* t./f.: 00386 (0) 7 6000 308/309 ? m:. 00386 (0)41 756485 ? Infomedia d.o.o. ? Jerebova 3, Novo mesto, Slovenia ? www.infomedia.si -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161103/981711a1/attachment.html From david.villasmil.work at gmail.com Thu Nov 3 17:31:56 2016 From: david.villasmil.work at gmail.com (David Villasmil) Date: Thu, 03 Nov 2016 14:31:56 +0000 Subject: [Freeswitch-users] How to figure out mod_sofia.c:4391 Gateway 'vgaccount' is down! In-Reply-To: References: <1e88f5e7-3b9c-4704-8748-c680a67de771@gmail.com> <320c3eb1-76aa-43d0-a135-c591d73067d4@gmail.com> Message-ID: In some cases we want to disable the pinging altogether, i haven't found a way to do it, though. On Thu, Nov 3, 2016 at 6:58 AM Steven Ayre wrote: > Can you get a packet trace of the OPTIONS? This could mean you're getting > packet loss, or the gateway isn't responding to some of the OPTIONS > packets. If it's not 100% loss it'd let some packets through but > occassionally drop others which can make the state flap between up and > down. You can tune ping-min and ping-max to handle a small amount of missed > OPTIONS without marking it as down. Although if there is loss it's likely > going to impact on call quality. > > -Steve > > On 2 November 2016 at 19:17, Muhammad Naseer Bhatti > wrote: > > Yes it is. The gateway works perfectly and out of sudden it goes down. But > then FreeSWITCH takes a few seconds to retry and then marks it back up. > > > via Newton Mail > > > On Wed, Nov 2, 2016 at 10:14 PM, David Villasmil < > david.villasmil.work at gmail.com> wrote: > > Is the gw responding the OPTIONs? > On Wed, Nov 2, 2016 at 3:11 PM Muhammad Naseer Bhatti > wrote: > > I am trying to diagnose a gateway down issue, > > 2016-11-02 17:59:41.304402 [ERR] mod_sofia.c:4391 Gateway 'vgaccount' is > down! > > (Yes it?s a little old version but need to figure out the root cause) > > I have the gigantic sip trace and the log files but not able to figure out > why I am seeing this message. Though at the same time I am able to ping the > gateway fine from the same machine. What sort of symptoms should I look for > in the log files? I am already looking into network connectivity and other > system log files but nothing as of now. > > Thanks. > > via Newton Mail > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > 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/20161103/4a04964b/attachment-0001.html From david.villasmil.work at gmail.com Thu Nov 3 17:33:27 2016 From: david.villasmil.work at gmail.com (David Villasmil) Date: Thu, 03 Nov 2016 14:33:27 +0000 Subject: [Freeswitch-users] mod_event_socket with Dialplan Interaction In-Reply-To: <908A0B1A-19D0-47ED-980E-5351BEF8F4D2@gmx.de> References: <908A0B1A-19D0-47ED-980E-5351BEF8F4D2@gmx.de> Message-ID: Try using lua On Thu, Nov 3, 2016 at 2:47 AM Dimitry Nagorny wrote: > Am 2. November 2016 15:18:05 MEZ, schrieb Brian West >: > > bgapi originate {variable=foo,baz=bar}sofia/external/user at pbx &socket(...) > > You set the variables you wish to influence the sip leg inside the {} > > /b > > On Wed, Nov 2, 2016 at 2:24 AM, Dimitry Nagorny > wrote: > > Good morning all, > > I have a quick question: Is it possible that a received command through > mod_event_socket can be send through a (default) dialplan? > > Like ?bgapi originate /sofia/external/user at pbx &socket(...)? goes before > the INVITE through the dialplan to get details from DB. > > Why I want this? I want to enrich the sip of the INVITE with some > additional information. I already searched the wiki/Confluence and couldn?t > find an answer. > > > Best Regards > *Dimitry Nagorny* > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 (50% Discount using code FreeSwitch50) > http://www.freeswitchcookbook.com (50% Discount using code FreeSwitch50) > https://www.gofundme.com/freeswitch_ubuntu > > Got Bugs? Report them here ! | Reddit: > /r/freeswitch > > *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 > > > Thanks you for your answer Brian but maybe my question was wrong. > > I try to do something like > bgapi sofia/default/user &socket (...) > but this throws Invalid Profile. So instead of external/gateway I want it > to go to dialplan "default". > > Or something similar that gives me the opperrunity to open a DB connection > for a recieved request on the event_socket. I can't grab this info from > within the third party system which sends the request. > > > Thanks > > Dimitry > -- > Diese Nachricht wurde von meinem Android-Mobiltelefon mit K-9 Mail > gesendet. > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161103/c95164bb/attachment.html From nbhatti at gmail.com Thu Nov 3 17:47:33 2016 From: nbhatti at gmail.com (Muhammad Naseer Bhatti) Date: Thu, 3 Nov 2016 07:47:33 -0700 Subject: [Freeswitch-users] How to figure out mod_sofia.c:4391 Gateway 'vgaccount' is down! In-Reply-To: References: <1e88f5e7-3b9c-4704-8748-c680a67de771@gmail.com> <320c3eb1-76aa-43d0-a135-c591d73067d4@gmail.com> Message-ID: If I Send a ping OPTIONS and don?t get a reply back, that won?t be shown in the trace. The trace will only show a packet was sent and no ACK was received. This could be network or the gateway both. Either way, how should I capture only the trace? Capture the whole trace and filter OPTIONS only? -- Sent with Airmail From: Steven Ayre Reply: FreeSWITCH Users Help Date: November 3, 2016 at 1:58:33 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to figure out mod_sofia.c:4391 Gateway 'vgaccount' is down! Can you get a packet trace of the OPTIONS? This could mean you're getting packet loss, or the gateway isn't responding to some of the OPTIONS packets. If it's not 100% loss it'd let some packets through but occassionally drop others which can make the state flap between up and down. You can tune ping-min and ping-max to handle a small amount of missed OPTIONS without marking it as down. Although if there is loss it's likely going to impact on call quality. -Steve On 2 November 2016 at 19:17, Muhammad Naseer Bhatti wrote: > Yes it is. The gateway works perfectly and out of sudden it goes down. But > then FreeSWITCH takes a few seconds to retry and then marks it back up. > > > via Newton Mail > > > On Wed, Nov 2, 2016 at 10:14 PM, David Villasmil < > david.villasmil.work at gmail.com> wrote: > > Is the gw responding the OPTIONs? > On Wed, Nov 2, 2016 at 3:11 PM Muhammad Naseer Bhatti > wrote: > >> I am trying to diagnose a gateway down issue, >> >> 2016-11-02 17:59:41.304402 [ERR] mod_sofia.c:4391 Gateway 'vgaccount' is >> down! >> >> (Yes it?s a little old version but need to figure out the root cause) >> >> I have the gigantic sip trace and the log files but not able to figure >> out why I am seeing this message. Though at the same time I am able to ping >> the gateway fine from the same machine. What sort of symptoms should I look >> for in the log files? I am already looking into network connectivity and >> other system log files but nothing as of now. >> >> Thanks. >> >> via Newton Mail >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> 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/20161103/2f7dbc20/attachment-0001.html From david.villasmil.work at gmail.com Thu Nov 3 17:50:21 2016 From: david.villasmil.work at gmail.com (David Villasmil) Date: Thu, 03 Nov 2016 14:50:21 +0000 Subject: [Freeswitch-users] How to figure out mod_sofia.c:4391 Gateway 'vgaccount' is down! In-Reply-To: References: <1e88f5e7-3b9c-4704-8748-c680a67de771@gmail.com> <320c3eb1-76aa-43d0-a135-c591d73067d4@gmail.com> Message-ID: Showing 2 option requests without responses would show the gw is not responding. On Thu, Nov 3, 2016 at 10:48 AM Muhammad Naseer Bhatti wrote: > If I Send a ping OPTIONS and don?t get a reply back, that won?t be shown > in the trace. The trace will only show a packet was sent and no ACK was > received. This could be network or the gateway both. Either way, how should > I capture only the trace? Capture the whole trace and filter OPTIONS only? > > -- > > Sent with Airmail > > From: Steven Ayre > Reply: FreeSWITCH Users Help > > Date: November 3, 2016 at 1:58:33 PM > To: FreeSWITCH Users Help > > Subject: Re: [Freeswitch-users] How to figure out mod_sofia.c:4391 > Gateway 'vgaccount' is down! > > Can you get a packet trace of the OPTIONS? This could mean you're getting > packet loss, or the gateway isn't responding to some of the OPTIONS > packets. If it's not 100% loss it'd let some packets through but > occassionally drop others which can make the state flap between up and > down. You can tune ping-min and ping-max to handle a small amount of missed > OPTIONS without marking it as down. Although if there is loss it's likely > going to impact on call quality. > > -Steve > > On 2 November 2016 at 19:17, Muhammad Naseer Bhatti > wrote: > > Yes it is. The gateway works perfectly and out of sudden it goes down. But > then FreeSWITCH takes a few seconds to retry and then marks it back up. > > > via Newton Mail > > > On Wed, Nov 2, 2016 at 10:14 PM, David Villasmil < > david.villasmil.work at gmail.com> wrote: > > Is the gw responding the OPTIONs? > On Wed, Nov 2, 2016 at 3:11 PM Muhammad Naseer Bhatti > wrote: > > I am trying to diagnose a gateway down issue, > > 2016-11-02 17:59:41.304402 [ERR] mod_sofia.c:4391 Gateway 'vgaccount' is > down! > > (Yes it?s a little old version but need to figure out the root cause) > > I have the gigantic sip trace and the log files but not able to figure out > why I am seeing this message. Though at the same time I am able to ping the > gateway fine from the same machine. What sort of symptoms should I look for > in the log files? I am already looking into network connectivity and other > system log files but nothing as of now. > > Thanks. > > via Newton Mail > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > 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/20161103/ddf8e2b0/attachment.html From covici at ccs.covici.com Thu Nov 3 18:04:53 2016 From: covici at ccs.covici.com (John Covici) Date: Thu, 03 Nov 2016 11:04:53 -0400 Subject: [Freeswitch-users] How to figure out mod_sofia.c:4391 Gateway 'vgaccount' is down! In-Reply-To: References: <1e88f5e7-3b9c-4704-8748-c680a67de771@gmail.com> <320c3eb1-76aa-43d0-a135-c591d73067d4@gmail.com> Message-ID: I have found those options pings to be very unreliable, so I have disabled them for all my trunks, and they seem to be fine. On Thu, 03 Nov 2016 10:31:56 -0400, David Villasmil wrote: > > [1 ] > [1.1 ] > [1.2 ] > In some cases we want to disable the pinging altogether, i haven't found a way to do it, though. > > On Thu, Nov 3, 2016 at 6:58 AM Steven Ayre wrote: > > Can you get a packet trace of the OPTIONS? This could mean you're getting packet loss, or the gateway isn't responding to some of the OPTIONS packets. If it's not 100% loss it'd let some packets through but occassionally drop others > which can make the state flap between up and down. You can tune ping-min and ping-max to handle a small amount of missed OPTIONS without marking it as down. Although if there is loss it's likely going to impact on call quality. > > -Steve > > On 2 November 2016 at 19:17, Muhammad Naseer Bhatti wrote: > > Yes it is. The gateway works perfectly and out of sudden it goes down. But then FreeSWITCH takes a few seconds to retry and then marks it back up. > > via Newton Mail > > On Wed, Nov 2, 2016 at 10:14 PM, David Villasmil wrote: > > Is the gw responding the OPTIONs? > On Wed, Nov 2, 2016 at 3:11 PM Muhammad Naseer Bhatti wrote: > > I am trying to diagnose a gateway down issue, > > 2016-11-02 17:59:41.304402 [ERR] mod_sofia.c:4391 Gateway 'vgaccount' is down! > > (Yes it?s a little old version but need to figure out the root cause) > > I have the gigantic sip trace and the log files but not able to figure out why I am seeing this message. Though at the same time I am able to ping the gateway fine from the same machine. What sort of symptoms should I > look for in the log files? I am already looking into network connectivity and other system log files but nothing as of now. > > Thanks. > > via Newton Mail > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > 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 > [2 ] > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 david.villasmil.work at gmail.com Thu Nov 3 18:11:08 2016 From: david.villasmil.work at gmail.com (David Villasmil) Date: Thu, 03 Nov 2016 15:11:08 +0000 Subject: [Freeswitch-users] How to figure out mod_sofia.c:4391 Gateway 'vgaccount' is down! In-Reply-To: References: <1e88f5e7-3b9c-4704-8748-c680a67de771@gmail.com> <320c3eb1-76aa-43d0-a135-c591d73067d4@gmail.com> Message-ID: How dis you disabled them? On Thu, Nov 3, 2016 at 11:05 AM John Covici wrote: > I have found those options pings to be very unreliable, so I have > disabled them for all my trunks, and they seem to be fine. > > On Thu, 03 Nov 2016 10:31:56 -0400, > David Villasmil wrote: > > > > [1 ] > > [1.1 ] > > [1.2 ] > > In some cases we want to disable the pinging altogether, i haven't found > a way to do it, though. > > > > On Thu, Nov 3, 2016 at 6:58 AM Steven Ayre wrote: > > > > Can you get a packet trace of the OPTIONS? This could mean you're > getting packet loss, or the gateway isn't responding to some of the OPTIONS > packets. If it's not 100% loss it'd let some packets through but > occassionally drop others > > which can make the state flap between up and down. You can tune > ping-min and ping-max to handle a small amount of missed OPTIONS without > marking it as down. Although if there is loss it's likely going to impact > on call quality. > > > > -Steve > > > > On 2 November 2016 at 19:17, Muhammad Naseer Bhatti > wrote: > > > > Yes it is. The gateway works perfectly and out of sudden it goes down. > But then FreeSWITCH takes a few seconds to retry and then marks it back up. > > > > via Newton Mail > > > > On Wed, Nov 2, 2016 at 10:14 PM, David Villasmil < > david.villasmil.work at gmail.com> wrote: > > > > Is the gw responding the OPTIONs? > > On Wed, Nov 2, 2016 at 3:11 PM Muhammad Naseer Bhatti < > nbhatti at gmail.com> wrote: > > > > I am trying to diagnose a gateway down issue, > > > > 2016-11-02 17:59:41.304402 [ERR] mod_sofia.c:4391 Gateway 'vgaccount' > is down! > > > > (Yes it?s a little old version but need to figure out the root cause) > > > > I have the gigantic sip trace and the log files but not able to figure > out why I am seeing this message. Though at the same time I am able to ping > the gateway fine from the same machine. What sort of symptoms should I > > look for in the log files? I am already looking into network > connectivity and other system log files but nothing as of now. > > > > Thanks. > > > > via Newton Mail > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://confluence.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > 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 > > [2 ] > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://confluence.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161103/ccbd3a04/attachment.html From tarik at pasifiktelekom.com.tr Thu Nov 3 18:17:19 2016 From: tarik at pasifiktelekom.com.tr (tarik at pasifiktelekom.com.tr) Date: Thu, 3 Nov 2016 18:17:19 +0300 Subject: [Freeswitch-users] FreeSwitch and CDR-Stats Message-ID: <81168AC5826E496F96D64A3784DBFE1F.MAI@mx02.garantiserver.com> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161103/5f8e8913/attachment.html From Miroslav.Levanic at enghouse.com Thu Nov 3 18:53:02 2016 From: Miroslav.Levanic at enghouse.com (Miroslav Levanic) Date: Thu, 3 Nov 2016 15:53:02 +0000 Subject: [Freeswitch-users] Jitter during on-hold Message-ID: <7165b2066bc84fba8a35daadeb5199ff@UK-MAIL-001.edge.local> Hello, I'm trying to play from file when other party is set on-hold. reINVITE sent by FreeSwitch contains sendonly attribute which opens possibility that FreeSwitch can play something to other end although other party is on-hold. I've used modified switch_ivr_play_file() function which by default prevents sending rtp packets to other side when call is on-hold. I've encountered rtp stream issue causing distorted audio due to dropped packets on the caller end. Codec used is G.711 mulaw 8kHz and ptime is 20ms accepted on both side, FreeSwitch and softphone. Both endpoints are located in the local LAN. Wireshark rtp analysis shows that when call is not on-hold, packets comes in interval of 20ms as expected. But when the call is set on-hold, average delta time between packets is 21ms, causing dropping packet every second (1ms extra time multiplied by 50 packets per second) with jitter buffer set to 50ms in Wireshark. Setting "auto-jitterbuffer-msec" parameter in sofia did not help. Is there a serious reason why I cannot use switch_ivr_play_file() during on-hold and what could cause jitter only during on-hold period? When call is retrieved from hold, jitter disappears and audio is good again. Thanks, Miro -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161103/2ac1c5f9/attachment-0001.html From brian at freeswitch.org Thu Nov 3 22:05:03 2016 From: brian at freeswitch.org (Brian West) Date: Thu, 3 Nov 2016 14:05:03 -0500 Subject: [Freeswitch-users] Jitter during on-hold In-Reply-To: <7165b2066bc84fba8a35daadeb5199ff@UK-MAIL-001.edge.local> References: <7165b2066bc84fba8a35daadeb5199ff@UK-MAIL-001.edge.local> Message-ID: So it sounds like you're writing your own module, How are you doing this hold operation? /b On Thu, Nov 3, 2016 at 10:53 AM, Miroslav Levanic < Miroslav.Levanic at enghouse.com> wrote: > Hello, > > > > I?m trying to play from file when other party is set on?hold. > > reINVITE sent by FreeSwitch contains sendonly attribute which opens > possibility that FreeSwitch can play something to other end although other > party is on-hold. > > I?ve used modified switch_ivr_play_file() function which by default > prevents sending rtp packets to other side when call is on-hold. > > I?ve encountered rtp stream issue causing distorted audio due to dropped > packets on the caller end. > > Codec used is G.711 mulaw 8kHz and ptime is 20ms accepted on both side, > FreeSwitch and softphone. Both endpoints are located in the local LAN. > > Wireshark rtp analysis shows that when call is not on-hold, packets comes > in interval of 20ms as expected. But when the call is set on-hold, average > delta time between packets is 21ms, causing dropping packet every second > (1ms extra time multiplied by 50 packets per second) with jitter buffer set > to 50ms in Wireshark. > > Setting "auto-jitterbuffer-msec" parameter in sofia did not help. > > Is there a serious reason why I cannot use switch_ivr_play_file() during > on-hold and what could cause jitter only during on-hold period? When call > is retrieved from hold, jitter disappears and audio is good again. > > > > Thanks, > > Miro > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 (50% Discount using code FreeSwitch50) http://www.freeswitchcookbook.com (50% Discount using code FreeSwitch50) https://www.gofundme.com/freeswitch_ubuntu Got Bugs? Report them here ! | Reddit: /r/freeswitch *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/20161103/9e0dcc56/attachment.html From hardyanto.donny at gmail.com Fri Nov 4 09:47:26 2016 From: hardyanto.donny at gmail.com (Donny Hardyanto) Date: Fri, 4 Nov 2016 13:47:26 +0700 Subject: [Freeswitch-users] JSAPI in Verto In-Reply-To: References: Message-ID: "We don't have a capability to execute things like that over sip." That exactly we want. We dont want user to execute things using verto. Only can place call, answering call. Donny On Wed, Nov 2, 2016 at 9:44 PM, Michael Jerris wrote: > They can be in user or domain configuration as i said. Not sure what you > mean by behave like a standard sip user. We don't have a capability to > execute things like that over sip. > > > On Wednesday, November 2, 2016, Donny Hardyanto > wrote: > >> So if no jsonrpc-allowed-jsapi and -fsapi in user config, user cannot >> send jsapi/fsapi via verto, and only can behave like standard sip user? >> >> Donny >> >> Pada tanggal 1 Nov 2016 2:14 AM, "Michael Jerris" >> menulis: >> >>> you can secure it on a user by user basis, but limited only to which >>> commands that user can run the attrs are: >>> >>> jsonrpc-allowed-jsapi >>> jsonrpc-allowed-fsapi >>> >>> for fsapi commands and jsapi commands. >>> >>> if you look in the default configs you can see similar settings put in >>> at a global level, the same is possible per user: >>> >>> conf/testing/directory/default.xml:5: >> value="verto"/> >>> conf/vanilla/directory/default.xml:26: >> value="verto"/> >>> conf/vanilla/directory/default.xml:27: >>> >>> >>> >>> On Oct 31, 2016, at 3:00 AM, Donny Hardyanto >>> wrote: >>> >>> Hi all, >>> >>> Is it possible to run JSAPI via verto? Is it as powerfull as Event >>> Socket? Is there any way to secure it? I like to deploy the verto webrtc to >>> website, but dont want any one can abuse and control my FS box via >>> JSAPi/verto. >>> >>> Thanks, >>> >>> Donny >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> 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/20161104/0ba67256/attachment.html From david.villasmil.work at gmail.com Fri Nov 4 10:42:27 2016 From: david.villasmil.work at gmail.com (David Villasmil) Date: Fri, 4 Nov 2016 03:42:27 -0400 Subject: [Freeswitch-users] unrelayed BYE Message-ID: Hello guys, I have this situation where the whole call goes well, except the BYE. Freeswitch receives the BYE, but doesn't relay it, and I can't figure out why, here's a trace: recv 489 bytes from udp/[196.207.243.108]:5080 at 07:22:05.626643: ------------------------------------------------------------------------ BYE sip:gw+gw_108 at 196.207.243.245:5080;transport=udp;gw=gw_108 SIP/2.0 Via: SIP/2.0/UDP 196.207.243.108:5080;branch=z9hG4bK2e5f9502;rport Max-Forwards: 70 From: ;tag=as6c7e3a9d To: "721102372" ;tag=a9B2UttKm9Nyg Call-ID: 16336d2d-1d02-1235-8693-0026b955dc81 CSeq: 102 BYE User-Agent: Asterisk PBX 11.23.0 X-Asterisk-HangupCause: Unallocated (unassigned) number X-Asterisk-HangupCauseCode: 1 Content-Length: 0 ------------------------------------------------------------------------ tport.c:3023 tport_deliver() tport_deliver(0x7f5a04004540): msg 0x7f5a04007eb0 (489 bytes) from udp/196.207.243.108:5080/sip next=(nil) nta.c:2880 agent_recv_request() nta: received BYE sip:gw+gw_108 at 196.207.243.245:5080;transport=udp;gw=gw_108 SIP/2.0 (CSeq 102) nta.c:3248 agent_aliases() nta: canonizing sip:gw+gw_108 at 196.207.243.245:5080 with contact nta.c:3060 agent_recv_request() nta: BYE (102) going to existing leg nua_server.c:102 nua_stack_process_request() nua: nua_stack_process_request: entering nua_stack.c:359 nua_application_event() nua: nua_application_event: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:879 nua_respond() nua: nua_respond: entering nua_stack.c:529 nua_signal() nua(0x7f599c000d40): sent signal r_respond nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering nua.c:921 nua_handle_destroy() nua: nua_handle_destroy: entering nua_stack.c:529 nua_signal() nua(0x7f599c000d40): sent signal r_destroy nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering soa.c:403 soa_set_params() soa_set_params(static::0x7f5a04014610, ...) called tport.c:3257 tport_tsend() tport_tsend(0x7f5a04004540) tpn = UDP/ 196.207.243.108:5080 tport.c:4046 tport_resolve() tport_resolve addrinfo = 196.207.243.108:5080 tport.c:4680 tport_by_addrinfo() tport_by_addrinfo(0x7f5a04004540): not found by name UDP/196.207.243.108:5080 Any ideas? Thanks ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161104/847ddf36/attachment-0001.html From mirkobrankovic at gmail.com Fri Nov 4 11:26:12 2016 From: mirkobrankovic at gmail.com (Mirko Brankovic) Date: Fri, 4 Nov 2016 09:26:12 +0100 Subject: [Freeswitch-users] unrelayed BYE In-Reply-To: References: Message-ID: You need to follow the whole sip flow, maybe you are missing another Via header, or the tag or branch is incorrect.... On Fri, Nov 4, 2016 at 8:42 AM, David Villasmil < david.villasmil.work at gmail.com> wrote: > Hello guys, > > I have this situation where the whole call goes well, except the BYE. > Freeswitch receives the BYE, but doesn't relay it, and I can't figure out > why, here's a trace: > > recv 489 bytes from udp/[196.207.243.108]:5080 at 07:22:05.626643: > ----------------------------------------------------------- > ------------- > BYE sip:gw+gw_108 at 196.207.243.245:5080;transport=udp;gw=gw_108 SIP/2.0 > Via: SIP/2.0/UDP 196.207.243.108:5080;branch=z9hG4bK2e5f9502;rport > Max-Forwards: 70 > From: ;tag=as6c7e3a9d > To: "721102372" ;tag=a9B2UttKm9Nyg > Call-ID: 16336d2d-1d02-1235-8693-0026b955dc81 > CSeq: 102 BYE > User-Agent: Asterisk PBX 11.23.0 > X-Asterisk-HangupCause: Unallocated (unassigned) number > X-Asterisk-HangupCauseCode: 1 > Content-Length: 0 > > ----------------------------------------------------------- > ------------- > tport.c:3023 tport_deliver() tport_deliver(0x7f5a04004540): msg > 0x7f5a04007eb0 (489 bytes) from udp/196.207.243.108:5080/sip next=(nil) > nta.c:2880 agent_recv_request() nta: received BYE > sip:gw+gw_108 at 196.207.243.245:5080;transport=udp;gw=gw_108 SIP/2.0 (CSeq > 102) > nta.c:3248 agent_aliases() nta: canonizing sip:gw+gw_108 at 196.207.243.245: > 5080 with contact > nta.c:3060 agent_recv_request() nta: BYE (102) going to existing leg > nua_server.c:102 nua_stack_process_request() nua: > nua_stack_process_request: entering > nua_stack.c:359 nua_application_event() nua: nua_application_event: > entering > nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering > nua.c:879 nua_respond() nua: nua_respond: entering > nua_stack.c:529 nua_signal() nua(0x7f599c000d40): sent signal r_respond > nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering > nua.c:921 nua_handle_destroy() nua: nua_handle_destroy: entering > nua_stack.c:529 nua_signal() nua(0x7f599c000d40): sent signal r_destroy > nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering > soa.c:403 soa_set_params() soa_set_params(static::0x7f5a04014610, ...) > called > tport.c:3257 tport_tsend() tport_tsend(0x7f5a04004540) tpn = UDP/ > 196.207.243.108:5080 > tport.c:4046 tport_resolve() tport_resolve addrinfo = 196.207.243.108:5080 > tport.c:4680 tport_by_addrinfo() tport_by_addrinfo(0x7f5a04004540): not > found by name UDP/196.207.243.108:5080 > > Any ideas? > > 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 > -- Regards, Mirko -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161104/63698d97/attachment.html From Miroslav.Levanic at enghouse.com Fri Nov 4 13:25:01 2016 From: Miroslav.Levanic at enghouse.com (Miroslav Levanic) Date: Fri, 4 Nov 2016 10:25:01 +0000 Subject: [Freeswitch-users] Jitter during on-hold In-Reply-To: References: <7165b2066bc84fba8a35daadeb5199ff@UK-MAIL-001.edge.local> Message-ID: Hi Brian, Let?s say that I?m rather experimenting then writing a new module. In the switch_ivr_play_file() I?ve just skipped IF condition which checks the status of channel flag CF_HOLD. Hold operation is called from our application layer by switch_api_execute(?uui_hold?,?). Best regards, Miro From: Brian West [mailto:brian at freeswitch.org] Sent: Thursday, November 3, 2016 8:05 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Jitter during on-hold So it sounds like you're writing your own module, How are you doing this hold operation? /b On Thu, Nov 3, 2016 at 10:53 AM, Miroslav Levanic > wrote: Hello, I?m trying to play from file when other party is set on?hold. reINVITE sent by FreeSwitch contains sendonly attribute which opens possibility that FreeSwitch can play something to other end although other party is on-hold. I?ve used modified switch_ivr_play_file() function which by default prevents sending rtp packets to other side when call is on-hold. I?ve encountered rtp stream issue causing distorted audio due to dropped packets on the caller end. Codec used is G.711 mulaw 8kHz and ptime is 20ms accepted on both side, FreeSwitch and softphone. Both endpoints are located in the local LAN. Wireshark rtp analysis shows that when call is not on-hold, packets comes in interval of 20ms as expected. But when the call is set on-hold, average delta time between packets is 21ms, causing dropping packet every second (1ms extra time multiplied by 50 packets per second) with jitter buffer set to 50ms in Wireshark. Setting "auto-jitterbuffer-msec" parameter in sofia did not help. Is there a serious reason why I cannot use switch_ivr_play_file() during on-hold and what could cause jitter only during on-hold period? When call is retrieved from hold, jitter disappears and audio is good again. Thanks, Miro _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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 [Image removed by sender.] Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com (50% Discount using code FreeSwitch50) http://www.freeswitchcookbook.com (50% Discount using code FreeSwitch50) https://www.gofundme.com/freeswitch_ubuntu Got Bugs? Report them here! | Reddit: /r/freeswitch 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/20161104/66d50945/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: ~WRD000.jpg Type: image/jpeg Size: 823 bytes Desc: ~WRD000.jpg Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161104/66d50945/attachment-0001.jpg From shaun.stokes at itec-support.co.uk Fri Nov 4 16:39:58 2016 From: shaun.stokes at itec-support.co.uk (Shaun Stokes) Date: Fri, 4 Nov 2016 13:39:58 +0000 Subject: [Freeswitch-users] Open Source SIP\WebRTC clients compatible with FreeSWITCH Message-ID: <6FD2F8B5BB72834E9939AEDF9FB802A901E860B047@mbx-01.sysconfig.co.uk> Hi All, Does anyone have any recommendations on a good open source SIP\WebRTC client which works on multiple platforms (Windows, Mac, Linux, Mobiles) to provide presence, voice, video, instant messaging, screen sharing and file sharing? This must be capable of integrating with FreeSWITCH for voice and video (presence via FreeSWITCH would be an advantage). Many Thanks, Shaun [http://www.itec-support.co.uk/wp-content/uploads/2016/07/email_logo.jpg] Shaun Stokes - Infrastructure Analyst T : 01453 700713 E : shaun.stokes at itec-support.co.uk W : www.itec-support.co.uk Registered Address :- ITEC Support, Suite 2 Prospect House, Bath Road, Stroud, Gloucestershire GL5 3QF Company No. 06908001 CONFIDENTIALITY NOTICE This communication and the information it contains are intended for the person or organisation to which it is addressed. Its contents are confidential and may be protected in law. Unauthorised use, copying or disclosure of any of it may be unlawful. If you are not the intended recipient, please contact us immediately. The contents of any attachments in this e-mail may contain software viruses, which could damage your own computer system. While ITEC Support has taken every reasonable precaution to minimise this risk, we cannot accept liability for any damage which you sustain as a result of software viruses. You should carry out your own virus checking procedure before opening any attachment. ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161104/8ba8d863/attachment.html From krice at freeswitch.org Fri Nov 4 17:24:13 2016 From: krice at freeswitch.org (Ken Rice) Date: Fri, 4 Nov 2016 09:24:13 -0500 Subject: [Freeswitch-users] Open Source SIP\WebRTC clients compatible with FreeSWITCH In-Reply-To: <6FD2F8B5BB72834E9939AEDF9FB802A901E860B047@mbx-01.sysconfig.co.uk> References: <6FD2F8B5BB72834E9939AEDF9FB802A901E860B047@mbx-01.sysconfig.co.uk> Message-ID: <10c301d236a7$1ea0f8f0$5be2ead0$@freeswitch.org> Verto Works on pretty much any platform that has native webrtc support now... unfortunately things like iOS and don't have native iOS support yet. If you are looking to build something you might contact consulting at freeswitch.org and see if you can work with the FSS Team to develop something From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Shaun Stokes Sent: Friday, November 4, 2016 8:40 AM To: 'FreeSWITCH Users Help' Subject: [Freeswitch-users] Open Source SIP\WebRTC clients compatible with FreeSWITCH Hi All, Does anyone have any recommendations on a good open source SIP\WebRTC client which works on multiple platforms (Windows, Mac, Linux, Mobiles) to provide presence, voice, video, instant messaging, screen sharing and file sharing? This must be capable of integrating with FreeSWITCH for voice and video (presence via FreeSWITCH would be an advantage). Many Thanks, Shaun Shaun Stokes - Infrastructure Analyst T : 01453 700713 E : shaun.stokes at itec-support.co.uk W : www.itec-support.co.uk Registered Address :- ITEC Support, Suite 2 Prospect House, Bath Road, Stroud, Gloucestershire GL5 3QF Company No. 06908001 CONFIDENTIALITY NOTICE This communication and the information it contains are intended for the person or organisation to which it is addressed. Its contents are confidential and may be protected in law. Unauthorised use, copying or disclosure of any of it may be unlawful. If you are not the intended recipient, please contact us immediately. The contents of any attachments in this e-mail may contain software viruses, which could damage your own computer system. While ITEC Support has taken every reasonable precaution to minimise this risk, we cannot accept liability for any damage which you sustain as a result of software viruses. You should carry out your own virus checking procedure before opening any attachment. ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161104/1f1eee88/attachment.html From mouli123 at gmail.com Fri Nov 4 17:29:07 2016 From: mouli123 at gmail.com (Chandramouli P) Date: Fri, 4 Nov 2016 19:59:07 +0530 Subject: [Freeswitch-users] Open Source SIP\WebRTC clients compatible with FreeSWITCH In-Reply-To: <10c301d236a7$1ea0f8f0$5be2ead0$@freeswitch.org> References: <6FD2F8B5BB72834E9939AEDF9FB802A901E860B047@mbx-01.sysconfig.co.uk> <10c301d236a7$1ea0f8f0$5be2ead0$@freeswitch.org> Message-ID: Hello Ken, We worked with Google native WebRTC on Firefox, Chrome, and Opera browsers on Windows OS. I recently noticed that Google has added the WebRTC support for Android, and iOS platforms (webrtc.org). Now, we are planning to develop video calling module using Google native WebRTC on these new platforms. Can anybody give me the information about my below queries: 1) Does Google native WebRTC supports Apple iOS platform (native mobile app)? 2) Does Google native WebRTC supports Apple OS X platform? 3) Is it possible to develop video calling module using native WebRTC on Safari, and Chrome browsers on Apple OS X platform? 4) Does Google native WebRTC supports Android platform (native mobile app)? 5) If it supports, I could not find any documentation for Apple iOS, Apple OS X, and Android platforms specifically. Could you please send some referral links? 6) I could not able to find the referral examples also for Apple iOS, Apple OS X, and Android platforms specifically. Could you please send some referral links? Please do needful. Thank you, Chandramouli. On Fri, Nov 4, 2016 at 7:54 PM, Ken Rice wrote: > Verto Works on pretty much any platform that has native webrtc support > now... unfortunately things like iOS and don?t have native iOS support yet? > > > > If you are looking to build something you might contact > consulting at freeswitch.org and see if you can work with the FSS Team to > develop something > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Shaun Stokes > *Sent:* Friday, November 4, 2016 8:40 AM > *To:* 'FreeSWITCH Users Help' > *Subject:* [Freeswitch-users] Open Source SIP\WebRTC clients compatible > with FreeSWITCH > > > > Hi All, > > > > Does anyone have any recommendations on a good open source SIP\WebRTC > client which works on multiple platforms (Windows, Mac, Linux, Mobiles) to > provide presence, voice, video, instant messaging, screen sharing and file > sharing? This must be capable of integrating with FreeSWITCH for voice and > video (presence via FreeSWITCH would be an advantage). > > > > Many Thanks, > > Shaun > > Shaun Stokes - Infrastructure Analyst > > T : > > 01453 700713 > > E : > > shaun.stokes at itec-support.co.uk > > W : > > www.itec-support.co.uk > > Registered Address :- ITEC Support, Suite 2 Prospect House, Bath Road, > Stroud, Gloucestershire GL5 3QF > Company No. 06908001 > > > CONFIDENTIALITY NOTICE > This communication and the information it contains are intended for the > person or organisation to which it is addressed. Its contents are > confidential and may be protected in law. Unauthorised use, copying or > disclosure of any of it may be unlawful. If you are not the intended > recipient, please contact us immediately. > The contents of any attachments in this e-mail may contain software > viruses, which could damage your own computer system. While ITEC Support > has taken every reasonable precaution to minimise this risk, we cannot > accept liability for any damage which you sustain as a result of software > viruses. You should carry out your own virus checking procedure before > opening any attachment. > > > ______________________________________________________________________ > This message has been checked for all known viruses by MessageLabs Virus > Scanning Service. > ______________________________________________________________________ > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161104/0019fded/attachment-0001.html From krice at freeswitch.org Fri Nov 4 17:32:21 2016 From: krice at freeswitch.org (Ken Rice) Date: Fri, 4 Nov 2016 09:32:21 -0500 Subject: [Freeswitch-users] Open Source SIP\WebRTC clients compatible with FreeSWITCH In-Reply-To: References: <6FD2F8B5BB72834E9939AEDF9FB802A901E860B047@mbx-01.sysconfig.co.uk> <10c301d236a7$1ea0f8f0$5be2ead0$@freeswitch.org> Message-ID: <10cd01d236a8$425136b0$c6f3a410$@freeswitch.org> No one supports Native WebRTC on iOS at this time except for people using their own private SDKs that they are not allowing to get out there? Apple does not have webRTC in webkit (Safari) or iOS at this time. Chrome on iOS is not even really Chrome, its just a wrapper around the WebKIT APIs and is effectively just safari with a few extra functions and built to look like chrome. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Chandramouli P Sent: Friday, November 4, 2016 9:29 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Open Source SIP\WebRTC clients compatible with FreeSWITCH Hello Ken, We worked with Google native WebRTC on Firefox, Chrome, and Opera browsers on Windows OS. I recently noticed that Google has added the WebRTC support for Android, and iOS platforms (webrtc.org ). Now, we are planning to develop video calling module using Google native WebRTC on these new platforms. Can anybody give me the information about my below queries: 1) Does Google native WebRTC supports Apple iOS platform (native mobile app)? 2) Does Google native WebRTC supports Apple OS X platform? 3) Is it possible to develop video calling module using native WebRTC on Safari, and Chrome browsers on Apple OS X platform? 4) Does Google native WebRTC supports Android platform (native mobile app)? 5) If it supports, I could not find any documentation for Apple iOS, Apple OS X, and Android platforms specifically. Could you please send some referral links? 6) I could not able to find the referral examples also for Apple iOS, Apple OS X, and Android platforms specifically. Could you please send some referral links? Please do needful. Thank you, Chandramouli. On Fri, Nov 4, 2016 at 7:54 PM, Ken Rice > wrote: Verto Works on pretty much any platform that has native webrtc support now... unfortunately things like iOS and don?t have native iOS support yet? If you are looking to build something you might contact consulting at freeswitch.org and see if you can work with the FSS Team to develop something From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org ] On Behalf Of Shaun Stokes Sent: Friday, November 4, 2016 8:40 AM To: 'FreeSWITCH Users Help' > Subject: [Freeswitch-users] Open Source SIP\WebRTC clients compatible with FreeSWITCH Hi All, Does anyone have any recommendations on a good open source SIP\WebRTC client which works on multiple platforms (Windows, Mac, Linux, Mobiles) to provide presence, voice, video, instant messaging, screen sharing and file sharing? This must be capable of integrating with FreeSWITCH for voice and video (presence via FreeSWITCH would be an advantage). Many Thanks, Shaun Shaun Stokes - Infrastructure Analyst T : 01453 700713 E : shaun.stokes at itec-support.co.uk W : www.itec-support.co.uk Registered Address :- ITEC Support, Suite 2 Prospect House, Bath Road, Stroud, Gloucestershire GL5 3QF Company No. 06908001 CONFIDENTIALITY NOTICE This communication and the information it contains are intended for the person or organisation to which it is addressed. Its contents are confidential and may be protected in law. Unauthorised use, copying or disclosure of any of it may be unlawful. If you are not the intended recipient, please contact us immediately. The contents of any attachments in this e-mail may contain software viruses, which could damage your own computer system. While ITEC Support has taken every reasonable precaution to minimise this risk, we cannot accept liability for any damage which you sustain as a result of software viruses. You should carry out your own virus checking procedure before opening any attachment. ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161104/dd1e002e/attachment.html From vids.cs at gmail.com Thu Nov 3 07:19:57 2016 From: vids.cs at gmail.com (vidhya sagar dixit) Date: Thu, 3 Nov 2016 09:49:57 +0530 Subject: [Freeswitch-users] DESTINATION_OUT_OF_ORDER calls to mobile and fixed line not working Message-ID: Hi Guys, I am using FreeSWITCH (Version 1.6.12 64bit) I am able to register sip softphone and able to make peer to peer calls without any problem. But I am not able to dial any mobile or fixed line number. External gateway is showing registered. Any help will be highly appreciated. Following is the log i get in cli : 2016-11-03 05:56:54.995540 [DEBUG] switch_core_codec.c:111 sofia/internal/ 3000 at 5.250.179.173:5090 Original read codec set to GSM:3 2016-11-03 05:56:54.995540 [DEBUG] switch_core_media.c:4572 Set telephone-event payload to 101 at 8000 2016-11-03 05:56:54.995540 [DEBUG] switch_core_media.c:4631 sofia/internal/ 3000 at 5.250.179.173:5090 Set 2833 dtmf send payload to 101 recv payload to 101 2016-11-03 05:56:54.995540 [DEBUG] sofia.c:7364 (sofia/internal/ 3000 at 5.250.179.173:5090) State Change CS_NEW -> CS_INIT 2016-11-03 05:56:54.995540 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/3000 at 5.250.179.173:5090) Running State Change CS_INIT 2016-11-03 05:56:54.995540 [DEBUG] switch_core_state_machine.c:627 (sofia/internal/3000 at 5.250.179.173:5090) State INIT 2016-11-03 05:56:54.995540 [DEBUG] mod_sofia.c:90 sofia/internal/ 3000 at 5.250.179.173:5090 SOFIA INIT 2016-11-03 05:56:54.995540 [DEBUG] switch_core_state_machine.c:40 sofia/internal/3000 at 5.250.179.173:5090 Standard INIT 2016-11-03 05:56:54.995540 [DEBUG] switch_core_state_machine.c:48 (sofia/internal/3000 at 5.250.179.173:5090) State Change CS_INIT -> CS_ROUTING 2016-11-03 05:56:54.995540 [DEBUG] switch_core_state_machine.c:627 (sofia/internal/3000 at 5.250.179.173:5090) State INIT going to sleep 2016-11-03 05:56:54.995540 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/3000 at 5.250.179.173:5090) Running State Change CS_ROUTING 2016-11-03 05:56:54.995540 [DEBUG] switch_channel.c:2249 (sofia/internal/ 3000 at 5.250.179.173:5090) Callstate Change DOWN -> RINGING 2016-11-03 05:56:54.995540 [DEBUG] switch_core_state_machine.c:643 (sofia/internal/3000 at 5.250.179.173:5090) State ROUTING 2016-11-03 05:56:54.995540 [DEBUG] mod_sofia.c:143 sofia/internal/ 3000 at 5.250.179.173:5090 SOFIA ROUTING 2016-11-03 05:56:54.995540 [DEBUG] switch_core_state_machine.c:236 sofia/internal/3000 at 5.250.179.173:5090 Standard ROUTING 2016-11-03 05:56:54.995540 [INFO] mod_dialplan_xml.c:637 Processing 3000 <3000>->00919971931166 in context 5.250.179.173 2016-11-03 05:56:55.015522 [DEBUG] freeswitch_lua.cpp:365 DBH handle 0x7f0cc4041120 Connected. 2016-11-03 05:56:55.255553 [DEBUG] freeswitch_lua.cpp:382 DBH handle 0x7f0cc4041120 released. Dialplan: sofia/internal/3000 at 5.250.179.173:5090 parsing [5.250.179.173->gateway1.00d1220] continue=false Dialplan: sofia/internal/3000 at 5.250.179.173:5090 Regex (FAIL) [gateway1.00d1220] destination_number(00919971931166) =~ /00^(\d{12,20})$/ break=on-false Dialplan: sofia/internal/3000 at 5.250.179.173:5090 parsing [5.250.179.173->gateway1.00d1220] continue=false Dialplan: sofia/internal/3000 at 5.250.179.173:5090 Regex (PASS) [gateway1.00d1220] destination_number(00919971931166) =~ /^(\d{12,20})$/ break=on-false Dialplan: sofia/internal/3000 at 5.250.179.173:5090 Action () Dialplan: sofia/internal/3000 at 5.250.179.173:5090 Action set(sip_h_X-accountcode=${accountcode}) Dialplan: sofia/internal/3000 at 5.250.179.173:5090 Action set(call_direction=outbound) Dialplan: sofia/internal/3000 at 5.250.179.173:5090 Action set(hangup_after_bridge=true) Dialplan: sofia/internal/3000 at 5.250.179.173:5090 Action set(effective_caller_id_name=${outbound_caller_id_name}) 2016-11-03 05:56:55.275562 [DEBUG] switch_core_state_machine.c:286 (sofia/internal/3000 at 5.250.179.173:5090) State Change CS_ROUTING -> CS_EXECUTE 2016-11-03 05:56:55.275562 [DEBUG] switch_core_state_machine.c:643 (sofia/internal/3000 at 5.250.179.173:5090) State ROUTING going to sleep 2016-11-03 05:56:55.275562 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/3000 at 5.250.179.173:5090) Running State Change CS_EXECUTE 2016-11-03 05:56:55.275562 [DEBUG] switch_core_state_machine.c:650 (sofia/internal/3000 at 5.250.179.173:5090) State EXECUTE 2016-11-03 05:56:55.275562 [DEBUG] mod_sofia.c:198 sofia/internal/ 3000 at 5.250.179.173:5090 SOFIA EXECUTE 2016-11-03 05:56:55.275562 [DEBUG] switch_core_state_machine.c:328 sofia/internal/3000 at 5.250.179.173:5090 Standard EXECUTE 2016-11-03 05:56:55.275562 [ERR] switch_core_session.c:2604 Invalid Application 2016-11-03 05:56:55.275562 [NOTICE] switch_core_session.c:2605 Hangup sofia/internal/3000 at 5.250.179.173:5090 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER] 2016-11-03 05:56:55.275562 [DEBUG] switch_core_state_machine.c:650 (sofia/internal/3000 at 5.250.179.173:5090) State EXECUTE going to sleep 2016-11-03 05:56:55.275562 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/3000 at 5.250.179.173:5090) Running State Change CS_HANGUP 2016-11-03 05:56:55.275562 [DEBUG] switch_core_state_machine.c:850 (sofia/internal/3000 at 5.250.179.173:5090) Callstate Change RINGING -> HANGUP 2016-11-03 05:56:55.275562 [DEBUG] switch_core_state_machine.c:852 (sofia/internal/3000 at 5.250.179.173:5090) State HANGUP 2016-11-03 05:56:55.275562 [DEBUG] mod_sofia.c:438 Channel sofia/internal/ 3000 at 5.250.179.173:5090 hanging up, cause: DESTINATION_OUT_OF_ORDER 2016-11-03 05:56:55.275562 [DEBUG] mod_sofia.c:577 Responding to INVITE with: 502 2016-11-03 05:56:55.275562 [DEBUG] switch_core_state_machine.c:60 sofia/internal/3000 at 5.250.179.173:5090 Standard HANGUP, cause: DESTINATION_OUT_OF_ORDER 2016-11-03 05:56:55.275562 [DEBUG] switch_core_state_machine.c:852 (sofia/internal/3000 at 5.250.179.173:5090) State HANGUP going to sleep 2016-11-03 05:56:55.275562 [DEBUG] switch_core_state_machine.c:619 (sofia/internal/3000 at 5.250.179.173:5090) State Change CS_HANGUP -> CS_REPORTING 2016-11-03 05:56:55.275562 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/3000 at 5.250.179.173:5090) Running State Change CS_REPORTING 2016-11-03 05:56:55.275562 [DEBUG] switch_core_state_machine.c:938 (sofia/internal/3000 at 5.250.179.173:5090) State REPORTING 2016-11-03 05:56:55.355544 [DEBUG] switch_core_state_machine.c:174 sofia/internal/3000 at 5.250.179.173:5090 Standard REPORTING, cause: DESTINATION_OUT_OF_ORDER 2016-11-03 05:56:55.355544 [DEBUG] switch_core_state_machine.c:938 (sofia/internal/3000 at 5.250.179.173:5090) State REPORTING going to sleep 2016-11-03 05:56:55.355544 [DEBUG] switch_core_state_machine.c:610 (sofia/internal/3000 at 5.250.179.173:5090) State Change CS_REPORTING -> CS_DESTROY 2016-11-03 05:56:55.355544 [DEBUG] switch_core_session.c:1647 Session 8023 (sofia/internal/3000 at 5.250.179.173:5090) Locked, Waiting on external entities 2016-11-03 05:56:55.355544 [NOTICE] switch_core_session.c:1665 Session 8023 (sofia/internal/3000 at 5.250.179.173:5090) Ended 2016-11-03 05:56:55.355544 [NOTICE] switch_core_session.c:1669 Close Channel sofia/internal/3000 at 5.250.179.173:5090 [CS_DESTROY] 2016-11-03 05:56:55.355544 [DEBUG] switch_core_state_machine.c:741 (sofia/internal/3000 at 5.250.179.173:5090) Running State Change CS_DESTROY 2016-11-03 05:56:55.355544 [DEBUG] switch_core_state_machine.c:751 (sofia/internal/3000 at 5.250.179.173:5090) State DESTROY 2016-11-03 05:56:55.355544 [DEBUG] mod_sofia.c:343 sofia/internal/ 3000 at 5.250.179.173:5090 SOFIA DESTROY 2016-11-03 05:56:55.355544 [DEBUG] switch_core_state_machine.c:181 sofia/internal/3000 at 5.250.179.173:5090 Standard DESTROY 2016-11-03 05:56:55.355544 [DEBUG] switch_core_state_machine.c:751 (sofia/internal/3000 at 5.250.179.173:5090) State DESTROY going to sleep Thanks and Regards Vidhya Sagar Dixit -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161103/8f685e49/attachment-0001.html From tomfieldingpersonal at gmail.com Fri Nov 4 09:51:50 2016 From: tomfieldingpersonal at gmail.com (Tom Fielding) Date: Thu, 3 Nov 2016 23:51:50 -0700 Subject: [Freeswitch-users] SIP MESSAGE messages Message-ID: Hi all, I was wondering if I needed to do anything special to route SIP MESSAGE messages from one endpoint to another via Freeswitch. Calls are able to be made between the endpoints but when a SIP MESSAGE is sent, the second leg incorrectly loops back to Freeswitch instead of being sent out to the recipient. Is there any special configuration that needs to be done to send a MESSAGE from one endpoint to another the way an INVITE is routed? Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161103/aa84ddf7/attachment.html From tomfieldingpersonal at gmail.com Fri Nov 4 09:58:49 2016 From: tomfieldingpersonal at gmail.com (Tom Fielding) Date: Thu, 3 Nov 2016 23:58:49 -0700 Subject: [Freeswitch-users] SIP MESSAGE messages In-Reply-To: References: Message-ID: Ah? I need to install mod_sms? On Thu, Nov 3, 2016 at 11:51 PM, Tom Fielding wrote: > Hi all, > > I was wondering if I needed to do anything special to route SIP MESSAGE > messages from one endpoint to another via Freeswitch. > > Calls are able to be made between the endpoints but when a SIP MESSAGE is > sent, the second leg incorrectly loops back to Freeswitch instead of being > sent out to the recipient. > > Is there any special configuration that needs to be done to send a MESSAGE > from one endpoint to another the way an INVITE is routed? > > Thanks! > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161103/9ffdeb75/attachment.html From mike at jerris.com Fri Nov 4 17:40:59 2016 From: mike at jerris.com (Michael Jerris) Date: Fri, 4 Nov 2016 10:40:59 -0400 Subject: [Freeswitch-users] DESTINATION_OUT_OF_ORDER calls to mobile and fixed line not working In-Reply-To: References: Message-ID: > On Nov 3, 2016, at 12:19 AM, vidhya sagar dixit wrote: > > Hi Guys, > > I am using FreeSWITCH (Version 1.6.12 64bit) I am able to register sip softphone and able to make peer to peer calls without any problem. > > But I am not able to dial any mobile or fixed line number. > > External gateway is showing registered. Any help will be highly appreciated. > > Following is the log i get in cli : > > 2016-11-03 05:56:54.995540 [DEBUG] switch_core_codec.c:111 sofia/internal/3000 at 5.250.179.173:5090 Original read codec set to GSM:3 > 2016-11-03 05:56:54.995540 [DEBUG] switch_core_media.c:4572 Set telephone-event payload to 101 at 8000 > 2016-11-03 05:56:54.995540 [DEBUG] switch_core_media.c:4631 sofia/internal/3000 at 5.250.179.173:5090 Set 2833 dtmf send payload to 101 recv payload to 101 > 2016-11-03 05:56:54.995540 [DEBUG] sofia.c:7364 (sofia/internal/3000 at 5.250.179.173:5090 ) State Change CS_NEW -> CS_INIT > 2016-11-03 05:56:54.995540 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/3000 at 5.250.179.173:5090 ) Running State Change CS_INIT > 2016-11-03 05:56:54.995540 [DEBUG] switch_core_state_machine.c:627 (sofia/internal/3000 at 5.250.179.173:5090 ) State INIT > 2016-11-03 05:56:54.995540 [DEBUG] mod_sofia.c:90 sofia/internal/3000 at 5.250.179.173:5090 SOFIA INIT > 2016-11-03 05:56:54.995540 [DEBUG] switch_core_state_machine.c:40 sofia/internal/3000 at 5.250.179.173:5090 Standard INIT > 2016-11-03 05:56:54.995540 [DEBUG] switch_core_state_machine.c:48 (sofia/internal/3000 at 5.250.179.173:5090 ) State Change CS_INIT -> CS_ROUTING > 2016-11-03 05:56:54.995540 [DEBUG] switch_core_state_machine.c:627 (sofia/internal/3000 at 5.250.179.173:5090 ) State INIT going to sleep > 2016-11-03 05:56:54.995540 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/3000 at 5.250.179.173:5090 ) Running State Change CS_ROUTING > 2016-11-03 05:56:54.995540 [DEBUG] switch_channel.c:2249 (sofia/internal/3000 at 5.250.179.173:5090 ) Callstate Change DOWN -> RINGING > 2016-11-03 05:56:54.995540 [DEBUG] switch_core_state_machine.c:643 (sofia/internal/3000 at 5.250.179.173:5090 ) State ROUTING > 2016-11-03 05:56:54.995540 [DEBUG] mod_sofia.c:143 sofia/internal/3000 at 5.250.179.173:5090 SOFIA ROUTING > 2016-11-03 05:56:54.995540 [DEBUG] switch_core_state_machine.c:236 sofia/internal/3000 at 5.250.179.173:5090 Standard ROUTING > 2016-11-03 05:56:54.995540 [INFO] mod_dialplan_xml.c:637 Processing 3000 <3000>->00919971931166 in context 5.250.179.173 > 2016-11-03 05:56:55.015522 [DEBUG] freeswitch_lua.cpp:365 DBH handle 0x7f0cc4041120 Connected. > 2016-11-03 05:56:55.255553 [DEBUG] freeswitch_lua.cpp:382 DBH handle 0x7f0cc4041120 released. > Dialplan: sofia/internal/3000 at 5.250.179.173:5090 parsing [5.250.179.173->gateway1.00d1220] continue=false > Dialplan: sofia/internal/3000 at 5.250.179.173:5090 Regex (FAIL) [gateway1.00d1220] destination_number(00919971931166) =~ /00^(\d{12,20})$/ break=on-false > Dialplan: sofia/internal/3000 at 5.250.179.173:5090 parsing [5.250.179.173->gateway1.00d1220] continue=false > Dialplan: sofia/internal/3000 at 5.250.179.173:5090 Regex (PASS) [gateway1.00d1220] destination_number(00919971931166) =~ /^(\d{12,20})$/ break=on-false > Dialplan: sofia/internal/3000 at 5.250.179.173:5090 Action () This ^^^^^ > Dialplan: sofia/internal/3000 at 5.250.179.173:5090 Action set(sip_h_X-accountcode=${accountcode}) > Dialplan: sofia/internal/3000 at 5.250.179.173:5090 Action set(call_direction=outbound) > Dialplan: sofia/internal/3000 at 5.250.179.173:5090 Action set(hangup_after_bridge=true) > Dialplan: sofia/internal/3000 at 5.250.179.173:5090 Action set(effective_caller_id_name=${outbound_caller_id_name}) > 2016-11-03 05:56:55.275562 [DEBUG] switch_core_state_machine.c:286 (sofia/internal/3000 at 5.250.179.173:5090 ) State Change CS_ROUTING -> CS_EXECUTE > 2016-11-03 05:56:55.275562 [DEBUG] switch_core_state_machine.c:643 (sofia/internal/3000 at 5.250.179.173:5090 ) State ROUTING going to sleep > 2016-11-03 05:56:55.275562 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/3000 at 5.250.179.173:5090 ) Running State Change CS_EXECUTE > 2016-11-03 05:56:55.275562 [DEBUG] switch_core_state_machine.c:650 (sofia/internal/3000 at 5.250.179.173:5090 ) State EXECUTE > 2016-11-03 05:56:55.275562 [DEBUG] mod_sofia.c:198 sofia/internal/3000 at 5.250.179.173:5090 SOFIA EXECUTE > 2016-11-03 05:56:55.275562 [DEBUG] switch_core_state_machine.c:328 sofia/internal/3000 at 5.250.179.173:5090 Standard EXECUTE > 2016-11-03 05:56:55.275562 [ERR] switch_core_session.c:2604 Invalid Application Causes this ^^^^ > 2016-11-03 05:56:55.275562 [NOTICE] switch_core_session.c:2605 Hangup sofia/internal/3000 at 5.250.179.173:5090 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER] > 2016-11-03 05:56:55.275562 [DEBUG] switch_core_state_machine.c:650 (sofia/internal/3000 at 5.250.179.173:5090 ) State EXECUTE going to sleep > 2016-11-03 05:56:55.275562 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/3000 at 5.250.179.173:5090 ) Running State Change CS_HANGUP > 2016-11-03 05:56:55.275562 [DEBUG] switch_core_state_machine.c:850 (sofia/internal/3000 at 5.250.179.173:5090 ) Callstate Change RINGING -> HANGUP > 2016-11-03 05:56:55.275562 [DEBUG] switch_core_state_machine.c:852 (sofia/internal/3000 at 5.250.179.173:5090 ) State HANGUP > 2016-11-03 05:56:55.275562 [DEBUG] mod_sofia.c:438 Channel sofia/internal/3000 at 5.250.179.173:5090 hanging up, cause: DESTINATION_OUT_OF_ORDER > 2016-11-03 05:56:55.275562 [DEBUG] mod_sofia.c:577 Responding to INVITE with: 502 > 2016-11-03 05:56:55.275562 [DEBUG] switch_core_state_machine.c:60 sofia/internal/3000 at 5.250.179.173:5090 Standard HANGUP, cause: DESTINATION_OUT_OF_ORDER > 2016-11-03 05:56:55.275562 [DEBUG] switch_core_state_machine.c:852 (sofia/internal/3000 at 5.250.179.173:5090 ) State HANGUP going to sleep > 2016-11-03 05:56:55.275562 [DEBUG] switch_core_state_machine.c:619 (sofia/internal/3000 at 5.250.179.173:5090 ) State Change CS_HANGUP -> CS_REPORTING > 2016-11-03 05:56:55.275562 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/3000 at 5.250.179.173:5090 ) Running State Change CS_REPORTING > 2016-11-03 05:56:55.275562 [DEBUG] switch_core_state_machine.c:938 (sofia/internal/3000 at 5.250.179.173:5090 ) State REPORTING > 2016-11-03 05:56:55.355544 [DEBUG] switch_core_state_machine.c:174 sofia/internal/3000 at 5.250.179.173:5090 Standard REPORTING, cause: DESTINATION_OUT_OF_ORDER > 2016-11-03 05:56:55.355544 [DEBUG] switch_core_state_machine.c:938 (sofia/internal/3000 at 5.250.179.173:5090 ) State REPORTING going to sleep > 2016-11-03 05:56:55.355544 [DEBUG] switch_core_state_machine.c:610 (sofia/internal/3000 at 5.250.179.173:5090 ) State Change CS_REPORTING -> CS_DESTROY > 2016-11-03 05:56:55.355544 [DEBUG] switch_core_session.c:1647 Session 8023 (sofia/internal/3000 at 5.250.179.173:5090 ) Locked, Waiting on external entities > 2016-11-03 05:56:55.355544 [NOTICE] switch_core_session.c:1665 Session 8023 (sofia/internal/3000 at 5.250.179.173:5090 ) Ended > 2016-11-03 05:56:55.355544 [NOTICE] switch_core_session.c:1669 Close Channel sofia/internal/3000 at 5.250.179.173:5090 [CS_DESTROY] > 2016-11-03 05:56:55.355544 [DEBUG] switch_core_state_machine.c:741 (sofia/internal/3000 at 5.250.179.173:5090 ) Running State Change CS_DESTROY > 2016-11-03 05:56:55.355544 [DEBUG] switch_core_state_machine.c:751 (sofia/internal/3000 at 5.250.179.173:5090 ) State DESTROY > 2016-11-03 05:56:55.355544 [DEBUG] mod_sofia.c:343 sofia/internal/3000 at 5.250.179.173:5090 SOFIA DESTROY > 2016-11-03 05:56:55.355544 [DEBUG] switch_core_state_machine.c:181 sofia/internal/3000 at 5.250.179.173:5090 Standard DESTROY > 2016-11-03 05:56:55.355544 [DEBUG] switch_core_state_machine.c:751 (sofia/internal/3000 at 5.250.179.173:5090 ) State DESTROY going to sleep > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161104/a7b0c8d1/attachment-0001.html From david.villasmil.work at gmail.com Fri Nov 4 17:54:37 2016 From: david.villasmil.work at gmail.com (David Villasmil) Date: Fri, 04 Nov 2016 14:54:37 +0000 Subject: [Freeswitch-users] SIP MESSAGE messages In-Reply-To: References: Message-ID: You can also test it with "chat" on the cli, very handy. Messages work very well, there's no "silo", though. If you send a message to a non-registered endpoint, it will fail and not be retried when the endpoint registers. I was thinking of implementing it, but i haven't got around to do it. On Fri, Nov 4, 2016 at 10:37 AM Tom Fielding wrote: > Ah? I need to install mod_sms? > > On Thu, Nov 3, 2016 at 11:51 PM, Tom Fielding < > tomfieldingpersonal at gmail.com> wrote: > > Hi all, > > I was wondering if I needed to do anything special to route SIP MESSAGE > messages from one endpoint to another via Freeswitch. > > Calls are able to be made between the endpoints but when a SIP MESSAGE is > sent, the second leg incorrectly loops back to Freeswitch instead of being > sent out to the recipient. > > Is there any special configuration that needs to be done to send a MESSAGE > from one endpoint to another the way an INVITE is routed? > > 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/20161104/c4b43ff1/attachment.html From chentom60 at hotmail.com Fri Nov 4 21:40:37 2016 From: chentom60 at hotmail.com (Tom Chen) Date: Fri, 4 Nov 2016 18:40:37 +0000 Subject: [Freeswitch-users] enable RTCP stream In-Reply-To: References: , Message-ID: Hello, It looks like rtcp stream is not available by default, how do I configure freeSwitch to send rtcp stream during a voip call? Tom -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161104/44f2442c/attachment.html From samir.doshi at inextrix.com Fri Nov 4 21:36:13 2016 From: samir.doshi at inextrix.com (Samir Doshi) Date: Sat, 5 Nov 2016 00:06:13 +0530 Subject: [Freeswitch-users] Issue with cc_warning_tone in mod_callcenter module Message-ID: Hi, We are doing impementation using mod_callcenter module by taking reference from documentation ( https://freeswitch.org/confluence/display/FREESWITCH/mod_callcenter). So far everything works well except cc_warning_tone playback. We are never getting tone when some connect to agent. Debug Logs : FS cli Log : https://pastebin.freeswitch.org/view/f9be26b9 Log with sip debug : https://pastebin.freeswitch.org/view/aab8ea85 Log with sip debug + loglevel 9 : https://pastebin.freeswitch.org/view/ff793521 Any hint? Best Regards -- Samir Doshi *iNextrix Technologie**s Pvt. Ltd*. http://www.inextrix.com *Disclaimer:* The information contained in this communication is confidential and may be legally privileged. It is intended solely for the use of the individual or entity to whom it is addressed and others authorised to receive it. If you are not the intended recipient you are hereby notified that any disclosure, copying, distribution or taking action in reliance of the contents of this information is strictly prohibited and may be unlawful. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161105/1bc0600c/attachment.html From lexxua at gmail.com Fri Nov 4 22:30:59 2016 From: lexxua at gmail.com (Volodymyr Fedorov) Date: Fri, 4 Nov 2016 20:30:59 +0100 Subject: [Freeswitch-users] enable RTCP stream In-Reply-To: References: Message-ID: Hi, you have to enable this options in sip profile: Freeswitch will add rtcp attribute to SDP. On Fri, Nov 4, 2016 at 7:40 PM, Tom Chen wrote: > Hello, > > > It looks like rtcp stream is not available by default, how do I configure > freeSwitch to send rtcp stream during a voip call? > > > Tom > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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, Volodymyr -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161104/41a1f9b1/attachment.html From italo at freeswitch.org Fri Nov 4 23:01:39 2016 From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=) Date: Fri, 4 Nov 2016 17:01:39 -0300 Subject: [Freeswitch-users] Issue with cc_warning_tone in mod_callcenter module In-Reply-To: References: Message-ID: mod_callcenter.c:1776 Agent 8003 answered "Outbound Call" from queue agent8003 at default (Recorded) >From this line you're using callback agents and the variable should be cc_outbound_announce instead of cc_warning_tone, this one is specific for uuid-standby agents On Fri, Nov 4, 2016 at 3:36 PM, Samir Doshi wrote: > Hi, > > We are doing impementation using mod_callcenter module by taking reference > from documentation (https://freeswitch.org/confluence/display/FREESWITCH/ > mod_callcenter). So far everything works well except cc_warning_tone > playback. We are never getting tone when some connect to agent. > > Debug Logs : > > FS cli Log : https://pastebin.freeswitch.org/view/f9be26b9 > Log with sip debug : https://pastebin.freeswitch.org/view/aab8ea85 > Log with sip debug + loglevel 9 : https://pastebin.freeswitch. > org/view/ff793521 > > Any hint? > > > Best Regards > -- > Samir Doshi > *iNextrix Technologie**s Pvt. Ltd*. > http://www.inextrix.com > > > > *Disclaimer:* > The information contained in this communication is confidential and may be > legally privileged. It is intended solely for the use of the individual or > entity to whom it is addressed and others authorised to receive it. If you > are not the intended recipient you are hereby notified that any disclosure, > copying, distribution or taking action in reliance of the contents of this > information is strictly prohibited and may be unlawful. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 italo at freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161104/ca5514f7/attachment-0001.html From brian at freeswitch.org Fri Nov 4 23:06:05 2016 From: brian at freeswitch.org (Brian West) Date: Fri, 4 Nov 2016 15:06:05 -0500 Subject: [Freeswitch-users] Jitter during on-hold In-Reply-To: References: <7165b2066bc84fba8a35daadeb5199ff@UK-MAIL-001.edge.local> Message-ID: uuid_hold doesn't do what you think it does, everyone makes this mistake. uuid_hold sends a hold indication to the remote side putting it on hold in the same manner a phone does. /b On Fri, Nov 4, 2016 at 5:25 AM, Miroslav Levanic < Miroslav.Levanic at enghouse.com> wrote: > Hi Brian, > > > > Let?s say that I?m rather experimenting then writing a new module. > > In the switch_ivr_play_file() I?ve just skipped IF condition which checks > the status of channel flag CF_HOLD. > > Hold operation is called from our application layer by > switch_api_execute(?uui_hold?,?). > > > > Best regards, > > Miro > > > > *From:* Brian West [mailto:brian at freeswitch.org] > *Sent:* Thursday, November 3, 2016 8:05 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Jitter during on-hold > > > > So it sounds like you're writing your own module, How are you doing this > hold operation? > > > > /b > > > > > > On Thu, Nov 3, 2016 at 10:53 AM, Miroslav Levanic < > Miroslav.Levanic at enghouse.com> wrote: > > Hello, > > > > I?m trying to play from file when other party is set on?hold. > > reINVITE sent by FreeSwitch contains sendonly attribute which opens > possibility that FreeSwitch can play something to other end although other > party is on-hold. > > I?ve used modified switch_ivr_play_file() function which by default > prevents sending rtp packets to other side when call is on-hold. > > I?ve encountered rtp stream issue causing distorted audio due to dropped > packets on the caller end. > > Codec used is G.711 mulaw 8kHz and ptime is 20ms accepted on both side, > FreeSwitch and softphone. Both endpoints are located in the local LAN. > > Wireshark rtp analysis shows that when call is not on-hold, packets comes > in interval of 20ms as expected. But when the call is set on-hold, average > delta time between packets is 21ms, causing dropping packet every second > (1ms extra time multiplied by 50 packets per second) with jitter buffer set > to 50ms in Wireshark. > > Setting "auto-jitterbuffer-msec" parameter in sofia did not help. > > Is there a serious reason why I cannot use switch_ivr_play_file() during > on-hold and what could cause jitter only during on-hold period? When call > is retrieved from hold, jitter disappears and audio is good again. > > > > Thanks, > > Miro > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 > > [image: Image removed by sender.] > > *Twitter: @FreeSWITCH , @briankwest* > http://www.freeswitchbook.com (50% Discount using code FreeSwitch50) > http://www.freeswitchcookbook.com (50% Discount using code FreeSwitch50) > https://www.gofundme.com/freeswitch_ubuntu > > Got Bugs? Report them here ! | Reddit: > /r/freeswitch > > *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 > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com (50% Discount using code FreeSwitch50) http://www.freeswitchcookbook.com (50% Discount using code FreeSwitch50) https://www.gofundme.com/freeswitch_ubuntu Got Bugs? Report them here ! | Reddit: /r/freeswitch *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/20161104/9a0f03c0/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: ~WRD000.jpg Type: image/jpeg Size: 823 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161104/9a0f03c0/attachment.jpg From gregor at infomedia.si Fri Nov 4 23:54:12 2016 From: gregor at infomedia.si (Gregor Nanger) Date: Fri, 4 Nov 2016 21:54:12 +0100 Subject: [Freeswitch-users] bind_digit_action advice Message-ID: Would need som advice. Just started to implementing bind_digit_action where caller can press 1 to transfer call during active call. Since I am using xml curl, I would like to minimize trip to server to get dialplan actions and make action inline not execute_extension. I would like to bridge call to user with extension 001, but having problem with syntax. Is this correct way? ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161104/bc2cc29b/attachment.html From vids.cs at gmail.com Sat Nov 5 08:33:45 2016 From: vids.cs at gmail.com (vidhya sagar dixit) Date: Sat, 5 Nov 2016 11:03:45 +0530 Subject: [Freeswitch-users] DESTINATION_OUT_OF_ORDER calls to mobile and fixed line not working In-Reply-To: References: Message-ID: Hi Michael, Thank you for replying , can you please explain what I am doing wrong and how can I fix it. Thanks and Regards Vidhya Sagar Dixit On Fri, Nov 4, 2016 at 8:10 PM, Michael Jerris wrote: > > On Nov 3, 2016, at 12:19 AM, vidhya sagar dixit wrote: > > Hi Guys, > > I am using FreeSWITCH (Version 1.6.12 64bit) I am able to register sip > softphone and able to make peer to peer calls without any problem. > > But I am not able to dial any mobile or fixed line number. > > External gateway is showing registered. Any help will be highly > appreciated. > > Following is the log i get in cli : > > 2016-11-03 05:56:54.995540 [DEBUG] switch_core_codec.c:111 sofia/internal/ > 3000 at 5.250.179.173:5090 Original read codec set to GSM:3 > 2016-11-03 05:56:54.995540 [DEBUG] switch_core_media.c:4572 Set > telephone-event payload to 101 at 8000 > 2016-11-03 05:56:54.995540 [DEBUG] switch_core_media.c:4631 sofia/internal/ > 3000 at 5.250.179.173:5090 Set 2833 dtmf send payload to 101 recv payload to > 101 > 2016-11-03 05:56:54.995540 [DEBUG] sofia.c:7364 (sofia/internal/ > 3000 at 5.250.179.173:5090) State Change CS_NEW -> CS_INIT > 2016-11-03 05:56:54.995540 [DEBUG] switch_core_state_machine.c:584 > (sofia/internal/3000 at 5.250.179.173:5090) Running State Change CS_INIT > 2016-11-03 05:56:54.995540 [DEBUG] switch_core_state_machine.c:627 > (sofia/internal/3000 at 5.250.179.173:5090) State INIT > 2016-11-03 05:56:54.995540 [DEBUG] mod_sofia.c:90 sofia/internal/ > 3000 at 5.250.179.173:5090 SOFIA INIT > 2016-11-03 05:56:54.995540 [DEBUG] switch_core_state_machine.c:40 > sofia/internal/3000 at 5.250.179.173:5090 Standard INIT > 2016-11-03 05:56:54.995540 [DEBUG] switch_core_state_machine.c:48 > (sofia/internal/3000 at 5.250.179.173:5090) State Change CS_INIT -> > CS_ROUTING > 2016-11-03 05:56:54.995540 [DEBUG] switch_core_state_machine.c:627 > (sofia/internal/3000 at 5.250.179.173:5090) State INIT going to sleep > 2016-11-03 05:56:54.995540 [DEBUG] switch_core_state_machine.c:584 > (sofia/internal/3000 at 5.250.179.173:5090) Running State Change CS_ROUTING > 2016-11-03 05:56:54.995540 [DEBUG] switch_channel.c:2249 (sofia/internal/ > 3000 at 5.250.179.173:5090) Callstate Change DOWN -> RINGING > 2016-11-03 05:56:54.995540 [DEBUG] switch_core_state_machine.c:643 > (sofia/internal/3000 at 5.250.179.173:5090) State ROUTING > 2016-11-03 05:56:54.995540 [DEBUG] mod_sofia.c:143 sofia/internal/ > 3000 at 5.250.179.173:5090 SOFIA ROUTING > 2016-11-03 05:56:54.995540 [DEBUG] switch_core_state_machine.c:236 > sofia/internal/3000 at 5.250.179.173:5090 Standard ROUTING > 2016-11-03 05:56:54.995540 [INFO] mod_dialplan_xml.c:637 Processing 3000 > <3000>->00919971931166 in context 5.250.179.173 > 2016-11-03 05:56:55.015522 [DEBUG] freeswitch_lua.cpp:365 DBH handle > 0x7f0cc4041120 Connected. > 2016-11-03 05:56:55.255553 [DEBUG] freeswitch_lua.cpp:382 DBH handle > 0x7f0cc4041120 released. > Dialplan: sofia/internal/3000 at 5.250.179.173:5090 parsing > [5.250.179.173->gateway1.00d1220] continue=false > Dialplan: sofia/internal/3000 at 5.250.179.173:5090 Regex (FAIL) > [gateway1.00d1220] destination_number(00919971931166) =~ > /00^(\d{12,20})$/ break=on-false > Dialplan: sofia/internal/3000 at 5.250.179.173:5090 parsing > [5.250.179.173->gateway1.00d1220] continue=false > Dialplan: sofia/internal/3000 at 5.250.179.173:5090 Regex (PASS) > [gateway1.00d1220] destination_number(00919971931166) =~ /^(\d{12,20})$/ > break=on-false > Dialplan: sofia/internal/3000 at 5.250.179.173:5090 Action () > > > This ^^^^^ > > Dialplan: sofia/internal/3000 at 5.250.179.173:5090 Action > set(sip_h_X-accountcode=${accountcode}) > Dialplan: sofia/internal/3000 at 5.250.179.173:5090 Action > set(call_direction=outbound) > Dialplan: sofia/internal/3000 at 5.250.179.173:5090 Action > set(hangup_after_bridge=true) > Dialplan: sofia/internal/3000 at 5.250.179.173:5090 Action > set(effective_caller_id_name=${outbound_caller_id_name}) > 2016-11-03 05:56:55.275562 [DEBUG] switch_core_state_machine.c:286 > (sofia/internal/3000 at 5.250.179.173:5090) State Change CS_ROUTING -> > CS_EXECUTE > 2016-11-03 05:56:55.275562 [DEBUG] switch_core_state_machine.c:643 > (sofia/internal/3000 at 5.250.179.173:5090) State ROUTING going to sleep > 2016-11-03 05:56:55.275562 [DEBUG] switch_core_state_machine.c:584 > (sofia/internal/3000 at 5.250.179.173:5090) Running State Change CS_EXECUTE > 2016-11-03 05:56:55.275562 [DEBUG] switch_core_state_machine.c:650 > (sofia/internal/3000 at 5.250.179.173:5090) State EXECUTE > 2016-11-03 05:56:55.275562 [DEBUG] mod_sofia.c:198 sofia/internal/ > 3000 at 5.250.179.173:5090 SOFIA EXECUTE > 2016-11-03 05:56:55.275562 [DEBUG] switch_core_state_machine.c:328 > sofia/internal/3000 at 5.250.179.173:5090 Standard EXECUTE > 2016-11-03 05:56:55.275562 [ERR] switch_core_session.c:2604 Invalid > Application > > > Causes this ^^^^ > > > 2016-11-03 05:56:55.275562 [NOTICE] switch_core_session.c:2605 Hangup > sofia/internal/3000 at 5.250.179.173:5090 [CS_EXECUTE] > [DESTINATION_OUT_OF_ORDER] > 2016-11-03 05:56:55.275562 [DEBUG] switch_core_state_machine.c:650 > (sofia/internal/3000 at 5.250.179.173:5090) State EXECUTE going to sleep > 2016-11-03 05:56:55.275562 [DEBUG] switch_core_state_machine.c:584 > (sofia/internal/3000 at 5.250.179.173:5090) Running State Change CS_HANGUP > 2016-11-03 05:56:55.275562 [DEBUG] switch_core_state_machine.c:850 > (sofia/internal/3000 at 5.250.179.173:5090) Callstate Change RINGING -> > HANGUP > 2016-11-03 05:56:55.275562 [DEBUG] switch_core_state_machine.c:852 > (sofia/internal/3000 at 5.250.179.173:5090) State HANGUP > 2016-11-03 05:56:55.275562 [DEBUG] mod_sofia.c:438 Channel sofia/internal/ > 3000 at 5.250.179.173:5090 hanging up, cause: DESTINATION_OUT_OF_ORDER > 2016-11-03 05:56:55.275562 [DEBUG] mod_sofia.c:577 Responding to INVITE > with: 502 > 2016-11-03 05:56:55.275562 [DEBUG] switch_core_state_machine.c:60 > sofia/internal/3000 at 5.250.179.173:5090 Standard HANGUP, cause: > DESTINATION_OUT_OF_ORDER > 2016-11-03 05:56:55.275562 [DEBUG] switch_core_state_machine.c:852 > (sofia/internal/3000 at 5.250.179.173:5090) State HANGUP going to sleep > 2016-11-03 05:56:55.275562 [DEBUG] switch_core_state_machine.c:619 > (sofia/internal/3000 at 5.250.179.173:5090) State Change CS_HANGUP -> > CS_REPORTING > 2016-11-03 05:56:55.275562 [DEBUG] switch_core_state_machine.c:584 > (sofia/internal/3000 at 5.250.179.173:5090) Running State Change CS_REPORTING > 2016-11-03 05:56:55.275562 [DEBUG] switch_core_state_machine.c:938 > (sofia/internal/3000 at 5.250.179.173:5090) State REPORTING > 2016-11-03 05:56:55.355544 [DEBUG] switch_core_state_machine.c:174 > sofia/internal/3000 at 5.250.179.173:5090 Standard REPORTING, cause: > DESTINATION_OUT_OF_ORDER > 2016-11-03 05:56:55.355544 [DEBUG] switch_core_state_machine.c:938 > (sofia/internal/3000 at 5.250.179.173:5090) State REPORTING going to sleep > 2016-11-03 05:56:55.355544 [DEBUG] switch_core_state_machine.c:610 > (sofia/internal/3000 at 5.250.179.173:5090) State Change CS_REPORTING -> > CS_DESTROY > 2016-11-03 05:56:55.355544 [DEBUG] switch_core_session.c:1647 Session 8023 > (sofia/internal/3000 at 5.250.179.173:5090) Locked, Waiting on external > entities > 2016-11-03 05:56:55.355544 [NOTICE] switch_core_session.c:1665 Session > 8023 (sofia/internal/3000 at 5.250.179.173:5090) Ended > 2016-11-03 05:56:55.355544 [NOTICE] switch_core_session.c:1669 Close > Channel sofia/internal/3000 at 5.250.179.173:5090 [CS_DESTROY] > 2016-11-03 05:56:55.355544 [DEBUG] switch_core_state_machine.c:741 > (sofia/internal/3000 at 5.250.179.173:5090) Running State Change CS_DESTROY > 2016-11-03 05:56:55.355544 [DEBUG] switch_core_state_machine.c:751 > (sofia/internal/3000 at 5.250.179.173:5090) State DESTROY > 2016-11-03 05:56:55.355544 [DEBUG] mod_sofia.c:343 sofia/internal/ > 3000 at 5.250.179.173:5090 SOFIA DESTROY > 2016-11-03 05:56:55.355544 [DEBUG] switch_core_state_machine.c:181 > sofia/internal/3000 at 5.250.179.173:5090 Standard DESTROY > 2016-11-03 05:56:55.355544 [DEBUG] switch_core_state_machine.c:751 > (sofia/internal/3000 at 5.250.179.173:5090) State DESTROY going to sleep > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161105/6391ba13/attachment-0001.html From vma at 440hz.fr Sat Nov 5 12:47:29 2016 From: vma at 440hz.fr (Vallimamod Abdullah) Date: Sat, 5 Nov 2016 10:47:29 +0100 Subject: [Freeswitch-users] DESTINATION_OUT_OF_ORDER calls to mobile and fixed line not working In-Reply-To: References: Message-ID: <89D60B73-037A-4394-9017-9D52FC05EB36@440hz.fr> Hi, There is an error your dial plan that you should review, something like empty action in extension "gateway1.00d1220". -- Best Regards, Vallimamod . > On 05 Nov 2016, at 06:33, vidhya sagar dixit wrote: > > Hi Michael, > > Thank you for replying , can you please explain what I am doing wrong and how can I fix it. > > Thanks and Regards > Vidhya Sagar Dixit > > > On Fri, Nov 4, 2016 at 8:10 PM, Michael Jerris wrote: > >> On Nov 3, 2016, at 12:19 AM, vidhya sagar dixit wrote: >> >> Hi Guys, >> >> I am using FreeSWITCH (Version 1.6.12 64bit) I am able to register sip softphone and able to make peer to peer calls without any problem. >> >> But I am not able to dial any mobile or fixed line number. >> >> External gateway is showing registered. Any help will be highly appreciated. >> >> Following is the log i get in cli : >> >> 2016-11-03 05:56:54.995540 [DEBUG] switch_core_codec.c:111 sofia/internal/3000 at 5.250.179.173:5090 Original read codec set to GSM:3 >> 2016-11-03 05:56:54.995540 [DEBUG] switch_core_media.c:4572 Set telephone-event payload to 101 at 8000 >> 2016-11-03 05:56:54.995540 [DEBUG] switch_core_media.c:4631 sofia/internal/3000 at 5.250.179.173:5090 Set 2833 dtmf send payload to 101 recv payload to 101 >> 2016-11-03 05:56:54.995540 [DEBUG] sofia.c:7364 (sofia/internal/3000 at 5.250.179.173:5090) State Change CS_NEW -> CS_INIT >> 2016-11-03 05:56:54.995540 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/3000 at 5.250.179.173:5090) Running State Change CS_INIT >> 2016-11-03 05:56:54.995540 [DEBUG] switch_core_state_machine.c:627 (sofia/internal/3000 at 5.250.179.173:5090) State INIT >> 2016-11-03 05:56:54.995540 [DEBUG] mod_sofia.c:90 sofia/internal/3000 at 5.250.179.173:5090 SOFIA INIT >> 2016-11-03 05:56:54.995540 [DEBUG] switch_core_state_machine.c:40 sofia/internal/3000 at 5.250.179.173:5090 Standard INIT >> 2016-11-03 05:56:54.995540 [DEBUG] switch_core_state_machine.c:48 (sofia/internal/3000 at 5.250.179.173:5090) State Change CS_INIT -> CS_ROUTING >> 2016-11-03 05:56:54.995540 [DEBUG] switch_core_state_machine.c:627 (sofia/internal/3000 at 5.250.179.173:5090) State INIT going to sleep >> 2016-11-03 05:56:54.995540 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/3000 at 5.250.179.173:5090) Running State Change CS_ROUTING >> 2016-11-03 05:56:54.995540 [DEBUG] switch_channel.c:2249 (sofia/internal/3000 at 5.250.179.173:5090) Callstate Change DOWN -> RINGING >> 2016-11-03 05:56:54.995540 [DEBUG] switch_core_state_machine.c:643 (sofia/internal/3000 at 5.250.179.173:5090) State ROUTING >> 2016-11-03 05:56:54.995540 [DEBUG] mod_sofia.c:143 sofia/internal/3000 at 5.250.179.173:5090 SOFIA ROUTING >> 2016-11-03 05:56:54.995540 [DEBUG] switch_core_state_machine.c:236 sofia/internal/3000 at 5.250.179.173:5090 Standard ROUTING >> 2016-11-03 05:56:54.995540 [INFO] mod_dialplan_xml.c:637 Processing 3000 <3000>->00919971931166 in context 5.250.179.173 >> 2016-11-03 05:56:55.015522 [DEBUG] freeswitch_lua.cpp:365 DBH handle 0x7f0cc4041120 Connected. >> 2016-11-03 05:56:55.255553 [DEBUG] freeswitch_lua.cpp:382 DBH handle 0x7f0cc4041120 released. >> Dialplan: sofia/internal/3000 at 5.250.179.173:5090 parsing [5.250.179.173->gateway1.00d1220] continue=false >> Dialplan: sofia/internal/3000 at 5.250.179.173:5090 Regex (FAIL) [gateway1.00d1220] destination_number(00919971931166) =~ /00^(\d{12,20})$/ break=on-false >> Dialplan: sofia/internal/3000 at 5.250.179.173:5090 parsing [5.250.179.173->gateway1.00d1220] continue=false >> Dialplan: sofia/internal/3000 at 5.250.179.173:5090 Regex (PASS) [gateway1.00d1220] destination_number(00919971931166) =~ /^(\d{12,20})$/ break=on-false >> Dialplan: sofia/internal/3000 at 5.250.179.173:5090 Action () > > This ^^^^^ > >> Dialplan: sofia/internal/3000 at 5.250.179.173:5090 Action set(sip_h_X-accountcode=${accountcode}) >> Dialplan: sofia/internal/3000 at 5.250.179.173:5090 Action set(call_direction=outbound) >> Dialplan: sofia/internal/3000 at 5.250.179.173:5090 Action set(hangup_after_bridge=true) >> Dialplan: sofia/internal/3000 at 5.250.179.173:5090 Action set(effective_caller_id_name=${outbound_caller_id_name}) >> 2016-11-03 05:56:55.275562 [DEBUG] switch_core_state_machine.c:286 (sofia/internal/3000 at 5.250.179.173:5090) State Change CS_ROUTING -> CS_EXECUTE >> 2016-11-03 05:56:55.275562 [DEBUG] switch_core_state_machine.c:643 (sofia/internal/3000 at 5.250.179.173:5090) State ROUTING going to sleep >> 2016-11-03 05:56:55.275562 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/3000 at 5.250.179.173:5090) Running State Change CS_EXECUTE >> 2016-11-03 05:56:55.275562 [DEBUG] switch_core_state_machine.c:650 (sofia/internal/3000 at 5.250.179.173:5090) State EXECUTE >> 2016-11-03 05:56:55.275562 [DEBUG] mod_sofia.c:198 sofia/internal/3000 at 5.250.179.173:5090 SOFIA EXECUTE >> 2016-11-03 05:56:55.275562 [DEBUG] switch_core_state_machine.c:328 sofia/internal/3000 at 5.250.179.173:5090 Standard EXECUTE >> 2016-11-03 05:56:55.275562 [ERR] switch_core_session.c:2604 Invalid Application > > Causes this ^^^^ > > >> 2016-11-03 05:56:55.275562 [NOTICE] switch_core_session.c:2605 Hangup sofia/internal/3000 at 5.250.179.173:5090 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER] >> 2016-11-03 05:56:55.275562 [DEBUG] switch_core_state_machine.c:650 (sofia/internal/3000 at 5.250.179.173:5090) State EXECUTE going to sleep >> 2016-11-03 05:56:55.275562 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/3000 at 5.250.179.173:5090) Running State Change CS_HANGUP >> 2016-11-03 05:56:55.275562 [DEBUG] switch_core_state_machine.c:850 (sofia/internal/3000 at 5.250.179.173:5090) Callstate Change RINGING -> HANGUP >> 2016-11-03 05:56:55.275562 [DEBUG] switch_core_state_machine.c:852 (sofia/internal/3000 at 5.250.179.173:5090) State HANGUP >> 2016-11-03 05:56:55.275562 [DEBUG] mod_sofia.c:438 Channel sofia/internal/3000 at 5.250.179.173:5090 hanging up, cause: DESTINATION_OUT_OF_ORDER >> 2016-11-03 05:56:55.275562 [DEBUG] mod_sofia.c:577 Responding to INVITE with: 502 >> 2016-11-03 05:56:55.275562 [DEBUG] switch_core_state_machine.c:60 sofia/internal/3000 at 5.250.179.173:5090 Standard HANGUP, cause: DESTINATION_OUT_OF_ORDER >> 2016-11-03 05:56:55.275562 [DEBUG] switch_core_state_machine.c:852 (sofia/internal/3000 at 5.250.179.173:5090) State HANGUP going to sleep >> 2016-11-03 05:56:55.275562 [DEBUG] switch_core_state_machine.c:619 (sofia/internal/3000 at 5.250.179.173:5090) State Change CS_HANGUP -> CS_REPORTING >> 2016-11-03 05:56:55.275562 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/3000 at 5.250.179.173:5090) Running State Change CS_REPORTING >> 2016-11-03 05:56:55.275562 [DEBUG] switch_core_state_machine.c:938 (sofia/internal/3000 at 5.250.179.173:5090) State REPORTING >> 2016-11-03 05:56:55.355544 [DEBUG] switch_core_state_machine.c:174 sofia/internal/3000 at 5.250.179.173:5090 Standard REPORTING, cause: DESTINATION_OUT_OF_ORDER >> 2016-11-03 05:56:55.355544 [DEBUG] switch_core_state_machine.c:938 (sofia/internal/3000 at 5.250.179.173:5090) State REPORTING going to sleep >> 2016-11-03 05:56:55.355544 [DEBUG] switch_core_state_machine.c:610 (sofia/internal/3000 at 5.250.179.173:5090) State Change CS_REPORTING -> CS_DESTROY >> 2016-11-03 05:56:55.355544 [DEBUG] switch_core_session.c:1647 Session 8023 (sofia/internal/3000 at 5.250.179.173:5090) Locked, Waiting on external entities >> 2016-11-03 05:56:55.355544 [NOTICE] switch_core_session.c:1665 Session 8023 (sofia/internal/3000 at 5.250.179.173:5090) Ended >> 2016-11-03 05:56:55.355544 [NOTICE] switch_core_session.c:1669 Close Channel sofia/internal/3000 at 5.250.179.173:5090 [CS_DESTROY] >> 2016-11-03 05:56:55.355544 [DEBUG] switch_core_state_machine.c:741 (sofia/internal/3000 at 5.250.179.173:5090) Running State Change CS_DESTROY >> 2016-11-03 05:56:55.355544 [DEBUG] switch_core_state_machine.c:751 (sofia/internal/3000 at 5.250.179.173:5090) State DESTROY >> 2016-11-03 05:56:55.355544 [DEBUG] mod_sofia.c:343 sofia/internal/3000 at 5.250.179.173:5090 SOFIA DESTROY >> 2016-11-03 05:56:55.355544 [DEBUG] switch_core_state_machine.c:181 sofia/internal/3000 at 5.250.179.173:5090 Standard DESTROY >> 2016-11-03 05:56:55.355544 [DEBUG] switch_core_state_machine.c:751 (sofia/internal/3000 at 5.250.179.173:5090) State DESTROY going to sleep >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > 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 samir.doshi at inextrix.com Sun Nov 6 13:00:24 2016 From: samir.doshi at inextrix.com (Samir Doshi) Date: Sun, 6 Nov 2016 15:30:24 +0530 Subject: [Freeswitch-users] Issue with cc_warning_tone in mod_callcenter module In-Reply-To: References: Message-ID: Thanks. I will give a try and get back to you with result. Best Regards -- Samir Doshi *iNextrix Technologie**s Pvt. Ltd*. http://www.inextrix.com *Disclaimer:* The information contained in this communication is confidential and may be legally privileged. It is intended solely for the use of the individual or entity to whom it is addressed and others authorised to receive it. If you are not the intended recipient you are hereby notified that any disclosure, copying, distribution or taking action in reliance of the contents of this information is strictly prohibited and may be unlawful. On Sat, Nov 5, 2016 at 1:31 AM, ?talo Rossi wrote: > mod_callcenter.c:1776 Agent 8003 answered "Outbound Call" > from queue agent8003 at default (Recorded) > > From this line you're using callback agents and the variable should be > cc_outbound_announce instead of cc_warning_tone, this one is specific for > uuid-standby agents > > On Fri, Nov 4, 2016 at 3:36 PM, Samir Doshi > wrote: > >> Hi, >> >> We are doing impementation using mod_callcenter module by taking >> reference from documentation (https://freeswitch.org/conflu >> ence/display/FREESWITCH/mod_callcenter). So far everything works well >> except cc_warning_tone playback. We are never getting tone when some >> connect to agent. >> >> Debug Logs : >> >> FS cli Log : https://pastebin.freeswitch.org/view/f9be26b9 >> Log with sip debug : https://pastebin.freeswitch.org/view/aab8ea85 >> Log with sip debug + loglevel 9 : https://pastebin.freeswitch.or >> g/view/ff793521 >> >> Any hint? >> >> >> Best Regards >> -- >> Samir Doshi >> *iNextrix Technologie**s Pvt. Ltd*. >> http://www.inextrix.com >> >> >> >> *Disclaimer:* >> The information contained in this communication is confidential and may >> be legally privileged. It is intended solely for the use of the individual >> or entity to whom it is addressed and others authorised to receive it. If >> you are not the intended recipient you are hereby notified that any >> disclosure, copying, distribution or taking action in reliance of the >> contents of this information is strictly prohibited and may be unlawful. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 > italo at freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161106/5a8f5a92/attachment.html From samir.doshi at inextrix.com Sun Nov 6 13:13:03 2016 From: samir.doshi at inextrix.com (Samir Doshi) Date: Sun, 6 Nov 2016 15:43:03 +0530 Subject: [Freeswitch-users] Export CDR on failed tried calls Message-ID: Hi Guys, Wondering if we can post failed tried call cdr to mod_json_cdr. I have below dialplan generated and I want to post cdr any gateway fail to process the call. That means if test1 fail then it should send cdr to mod_json_cdr and then go for test2. If test2 fail then post cdr and then try test3 so on.
Is there any variable or configuration needs to set? Best Regards -- Samir Doshi *iNextrix Technologie**s Pvt. Ltd*. http://www.inextrix.com *Disclaimer:* The information contained in this communication is confidential and may be legally privileged. It is intended solely for the use of the individual or entity to whom it is addressed and others authorised to receive it. If you are not the intended recipient you are hereby notified that any disclosure, copying, distribution or taking action in reliance of the contents of this information is strictly prohibited and may be unlawful. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161106/0857ffdf/attachment-0001.html From colin.morelli at gmail.com Sun Nov 6 21:35:32 2016 From: colin.morelli at gmail.com (Colin Morelli) Date: Sun, 06 Nov 2016 18:35:32 +0000 Subject: [Freeswitch-users] Ignore RTP until Remote Address Confirmed Message-ID: Hello, I'm trying to fix some issues with communication on devices that are on NAT64 networks. In this case, I'm using ICE to generate candidates for the local device before sending an INVITE to FS; however, because the device is on (what it perceives to be) an IPv6-only network, and FS is on an IPv4-only network, the device simply can't generate create a candidate that would be accepted by FS. FS essentially rejects all IPv6 candidates and then rejects the call with a 488. However, if I add a random IPv4 address in the INVITE that is accepted by the apply-candidate-acl in FS, then once the device starts sending RTP data to Freeswitch, auto-adjust kicks in and everything is fine. In other words, the two devices *can* communicate with each other, but based on the ICE candidates provided by the local device, they don't think that they can. So, my question is, is there any "safe" bogus IP I can provide to FS that will essentially allow the call to start, knowing it will just blackhole any RTP data for that client until auto-adjust has found the proper destination? Any other suggestions to solve this? (Besides using STUN, which - even if used - is still only going to generate an IP address with almost certainly an invalid port) Best, Colin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161106/254e5834/attachment.html From jw at vision-gmbh.de Mon Nov 7 09:37:18 2016 From: jw at vision-gmbh.de (=?utf-8?B?SsO8cmdlbiBXZW5kbGVy?=) Date: Mon, 7 Nov 2016 06:37:18 +0000 Subject: [Freeswitch-users] Enable H264 Video In-Reply-To: References: <032a649d78474c319815255e7c8e6037@vision-gmbh.de> <213471c5f51f4bf5a894c60559e2cac6@vision-gmbh.de> <2406A807-7E85-429A-AC9C-E5D0374AB35D@jerris.com> <5ADAAC47-DA2C-41A4-845F-03E19A0E2B18@gmail.com> Message-ID: <8e48560bb8954548878bd940c47c13df@vision-gmbh.de> Hello, i?m very interested too, and i would sponsor your work if we could get it to work. So are there enough people so we can get this working? What do you need from me? Best regards Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Serge Yuriev Gesendet: Dienstag, 27. September 2016 13:36 An: FreeSWITCH Users Help Betreff: Re: [Freeswitch-users] Enable H264 Video Hello, I?m interested but only as my pet project so no sponsoring sorry :( I have made little patch to get network_addr filled and now stuck on passing correct cause codes - it?s always Normal clearing(10). And getting Caller_id_name will be a bonus :) On 27 Sep 2016, at 06:38, Seven Du > wrote: I made that mod and we can make it work if get enough interests/sponsors. On Sep 21, 2016, at 5:52 AM, Anthony Minessale > wrote: Correct, that module is the closest to having video support but needs some major work. The opal one might be possible from the opal authors if you ask/hire them. On Tue, Sep 20, 2016 at 4:35 PM, Serge Yuriev > wrote: AFAIR there is mod_ooh323 in separate branch and it supports video. But module need patches for general usage. On 15 Sep 2016, at 17:06, Brian West > wrote: There is no video support in any H323 modules in tree currently. Nobody has done the work to make them video capable. Maybe someone can sponsor that work. On Thu, Sep 15, 2016 at 12:31 AM, J?rgen Wendler > wrote: And what about mod_h323 ? Video support here included or are the video capabilitys only for SIP? Mit freundlichen Gr??en J?rgen Wendler -Technik- Vision Consulting Deutschland GmbH Bremsstr. 17, D-50969 K?ln (Cologne) Fon: +49-221-995574-20 Fax: +49-221-995574-99 jw at vision-gmbh.de http://www.vision-gmbh.de Gesch?ftsf?hrer: Dipl.-Inf. Stephan Krafft Register: HRB 61562 - Amtsgericht K?ln Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Michael Jerris Gesendet: Mittwoch, 14. September 2016 19:19 An: FreeSWITCH Users Help > Betreff: Re: [Freeswitch-users] Enable H264 Video I can confirm that module does not currently have video support. On Sep 14, 2016, at 11:20 AM, Brian West > wrote: If you code it probably, or find someone to code it. :) On Wed, Sep 14, 2016 at 1:12 AM, J?rgen Wendler > wrote: Well, i thought since Version 1.6 freeswitch is capable of Video support. So there is no chance to get video with h323 listener, mod_opal or mod_h323 ? Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Brian West Gesendet: Dienstag, 13. September 2016 17:22 An: FreeSWITCH Users Help > Betreff: Re: [Freeswitch-users] Enable H264 Video mod_opal doesn't support video in FreeSWITCH last I knew. On Mon, Sep 12, 2016 at 8:18 AM, J?rgen Wendler > wrote: Hello everyone, i?ve found some amazaing articles about FS and so i thought i can give it a try. So I followed instructions in confluenca and got a simple running fs server with mod_verto which is capable of some basic webrtc. Really nice. Now I want to join this room (which is the basic 3500 room with the default video-mcu-profile) with some external h323 device. So I compiled ptlib and opalvoip and finally mod_opal and created a h323 listener. When I call the ip + room number with an external device (polycom, sony) via h323 I get voice only connections, log says the following: 2016-09-12 12:26:17.140849 [WARNING] mod_opal.cpp:750 {Opal Answer:12170,0000000000001} mod_opal Could not match FS codec H263-2000 at 90000 (pt=121) to an OPAL media format. 2016-09-12 12:26:17.140849 [WARNING] mod_opal.cpp:750 {Opal Answer:12170,0000000000001} mod_opal Could not match FS codec H263-1998 at 90000 (pt=115) to an OPAL media format. 2016-09-12 12:26:17.140849 [WARNING] mod_opal.cpp:750 {Opal Answer:12170,0000000000001} mod_opal Could not match FS codec VP8 at 90000 (pt=99) to an OPAL media format. 2016-09-12 12:26:17.140849 [WARNING] mod_opal.cpp:750 {Opal Answer:12170,0000000000001} mod_opal Could not match FS codec VP9 at 90000 (pt=99) to an OPAL media format. 2016-09-12 12:26:17.160836 [WARNING] mod_opal.cpp:750 {Opal Answer:12170,0000000000001} mod_opal Could not match FS codec H263 at 90000 (pt=34) to an OPAL media format. 2016-09-12 12:26:17.160836 [WARNING] mod_opal.cpp:750 {Opal Answer:12170,0000000000001} mod_opal Could not match FS codec LPC at 8000 (pt=7) to an OPAL media format. 2016-09-12 12:26:17.160836 [WARNING] mod_opal.cpp:750 {Opal Answer:12170,0000000000001} mod_opal Could not match FS codec PROXY-VID at 90000 (pt=31) to an OPAL media format. 2016-09-12 12:26:17.160836 [WARNING] mod_opal.cpp:750 {Opal Answer:12170,0000000000001} mod_opal Could not match FS codec H261 at 90000 (pt=31) to an OPAL media format. 2016-09-12 12:26:17.160836 [WARNING] mod_opal.cpp:750 {Opal Answer:12170,0000000000001} mod_opal Could not match FS codec H264 at 90000 (pt=97) to an OPAL media format. So I thought I should have a look about the compiled codecs. I activated mod_opal and mod_h26x in modules.conf but no succes. Log always says: 2016-09-12 12:26:17.160836 [INFO] h323.cxx:3996 {Opal Answer:12170,0000000000001} H323 SetLocalCapabilities: GSM-AMR,G.723.1,G.729,GSM-06.10,G.711-uLaw-64k,G.711-ALaw-64k,T.38,UserInput/hookflash,UserInput/basicString,UserInput/dtmf I?ve tried to change profile or the conference / dialplan, event set ?absolute_codec_string? in dialplan/default.xml but I think I am missing something. Could anyone point me in the right direction where I can ?enable? h263++ or h264 video capabilitys for an external h323 device / call via mod_opal? System is debian 8, FS and ptlib / mod_opal are checked out from git / cvs. Best regards, Juergen _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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 [Das Bild wurde vom Absender entfernt.] Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com (50% Discount using code FreeSwitch50) http://www.freeswitchcookbook.com (50% Discount using code FreeSwitch50) https://www.gofundme.com/freeswitch_ubuntu Got Bugs? Report them here! | Reddit: /r/freeswitch 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 -- Serge S. Yuriev _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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 https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users 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 -- Serge S. Yuriev -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161107/52a2b77f/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: ~WRD000.jpg Type: image/jpeg Size: 823 bytes Desc: ~WRD000.jpg Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161107/52a2b77f/attachment-0001.jpg From s.safarov at gmail.com Mon Nov 7 10:52:20 2016 From: s.safarov at gmail.com (Sergey Safarov) Date: Mon, 07 Nov 2016 07:52:20 +0000 Subject: [Freeswitch-users] Verso installation Message-ID: I has installed verto commucator on my server and point to websocker of FreeSwith team conference server. All works as expected. Then I change websocker URL to my FreeSwith server. Server has trusted certificate. I dial 3000 number, then connects to conference, can send/accept audio/video but cannot see who is connected to conference. Also I cannot send chat messages. I cannot find related error in browser console, FS log also not contain errors. Could you advise what is required to fix? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161107/f3a0e7ab/attachment.html From gmaruzz at gmail.com Mon Nov 7 12:19:29 2016 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Mon, 7 Nov 2016 10:19:29 +0100 Subject: [Freeswitch-users] Verso installation In-Reply-To: References: Message-ID: Sergey, 3000 is a "plain" conference in standard dialplan, with no such features you are looking for. Try calling 3500, in standard dialplan. That is called "stereo-mcu" conference (if i remember correctly) and is what you want. -giovanni sent from mobile cell: +39 347 266 56 18 Giovanni Maruzzelli OpenTelecom.IT On Nov 7, 2016 8:53 AM, "Sergey Safarov" wrote: > I has installed verto commucator on my server and point to websocker of > FreeSwith team conference server. All works as expected. > > Then I change websocker URL to my FreeSwith server. Server has trusted > certificate. I dial 3000 number, then connects to conference, can > send/accept audio/video but cannot see who is connected to conference. Also > I cannot send chat messages. > I cannot find related error in browser console, FS log also not contain > errors. > > Could you advise what is required to fix? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161107/c0ff7b42/attachment.html From manpower13.cse at gmail.com Mon Nov 7 15:03:49 2016 From: manpower13.cse at gmail.com (Murugan Pandian) Date: Mon, 7 Nov 2016 17:33:49 +0530 Subject: [Freeswitch-users] Freeswitch WebRTC with Nginx Message-ID: HI, I am using nginx for load balancing behind my freeswitch ,My WebRTC client SIPJS,i can able to register with my freeswitch and i can make outbound call successfully but when i cant able to receive any incoming call,if i register directly with freeswitch means without nginx i can make and receive call but when i register using nginx i am facing this issue(i cant able to receive call ). -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161107/9633ad5b/attachment.html From Miroslav.Levanic at enghouse.com Mon Nov 7 15:49:06 2016 From: Miroslav.Levanic at enghouse.com (Miroslav Levanic) Date: Mon, 7 Nov 2016 12:49:06 +0000 Subject: [Freeswitch-users] Jitter during on-hold In-Reply-To: References: <7165b2066bc84fba8a35daadeb5199ff@UK-MAIL-001.edge.local> Message-ID: <7581bd86a5c64bd6879cf5057505de54@UK-MAIL-001.edge.local> Hi Brian, I?m little bit confused with your answer. uuid_hold makes exactly what we want to do. At least SIP traffic looks like that ? FreeSwitch sends reINVITE with media attribute sendonly and remote side responds with receiveonly. What is wrong with that scenario? Regards, Miro From: Brian West [mailto:brian at freeswitch.org] Sent: Friday, November 4, 2016 9:06 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Jitter during on-hold uuid_hold doesn't do what you think it does, everyone makes this mistake. uuid_hold sends a hold indication to the remote side putting it on hold in the same manner a phone does. /b On Fri, Nov 4, 2016 at 5:25 AM, Miroslav Levanic > wrote: Hi Brian, Let?s say that I?m rather experimenting then writing a new module. In the switch_ivr_play_file() I?ve just skipped IF condition which checks the status of channel flag CF_HOLD. Hold operation is called from our application layer by switch_api_execute(?uui_hold?,?). Best regards, Miro From: Brian West [mailto:brian at freeswitch.org] Sent: Thursday, November 3, 2016 8:05 PM To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Jitter during on-hold So it sounds like you're writing your own module, How are you doing this hold operation? /b On Thu, Nov 3, 2016 at 10:53 AM, Miroslav Levanic > wrote: Hello, I?m trying to play from file when other party is set on?hold. reINVITE sent by FreeSwitch contains sendonly attribute which opens possibility that FreeSwitch can play something to other end although other party is on-hold. I?ve used modified switch_ivr_play_file() function which by default prevents sending rtp packets to other side when call is on-hold. I?ve encountered rtp stream issue causing distorted audio due to dropped packets on the caller end. Codec used is G.711 mulaw 8kHz and ptime is 20ms accepted on both side, FreeSwitch and softphone. Both endpoints are located in the local LAN. Wireshark rtp analysis shows that when call is not on-hold, packets comes in interval of 20ms as expected. But when the call is set on-hold, average delta time between packets is 21ms, causing dropping packet every second (1ms extra time multiplied by 50 packets per second) with jitter buffer set to 50ms in Wireshark. Setting "auto-jitterbuffer-msec" parameter in sofia did not help. Is there a serious reason why I cannot use switch_ivr_play_file() during on-hold and what could cause jitter only during on-hold period? When call is retrieved from hold, jitter disappears and audio is good again. Thanks, Miro _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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 [Image removed by sender.] Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com (50% Discount using code FreeSwitch50) http://www.freeswitchcookbook.com (50% Discount using code FreeSwitch50) https://www.gofundme.com/freeswitch_ubuntu Got Bugs? Report them here! | Reddit: /r/freeswitch 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 -- Brian West brian at freeswitch.org [Image removed by sender.] Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com (50% Discount using code FreeSwitch50) http://www.freeswitchcookbook.com (50% Discount using code FreeSwitch50) https://www.gofundme.com/freeswitch_ubuntu Got Bugs? Report them here! | Reddit: /r/freeswitch 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/20161107/10e15b3a/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: ~WRD000.jpg Type: image/jpeg Size: 823 bytes Desc: ~WRD000.jpg Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161107/10e15b3a/attachment-0001.jpg From aubalde at presenceco.com Mon Nov 7 18:55:49 2016 From: aubalde at presenceco.com (=?UTF-8?Q?Agust=C3=AD_Ubalde?=) Date: Mon, 7 Nov 2016 16:55:49 +0100 Subject: [Freeswitch-users] Video call failed Message-ID: Hi all, I've just installed FS 1.6 and configured 2 extensions for make video calls. Is this scenario possible without conference the video call? The call failed with the following message: *67cc9edd-bc3a-4b25-bb1c-5dc475a409d6 2016-11-07 15:36:50.676944 [INFO] mod_dptools.c:3401 Originate Failed. Cause: INCOMPATIBLE_DESTINATION* Thanks! *PRESENCE TECHNOLOGY* *Agust? Ubalde Bellot* Chief Developer C/ Comte Urgell 240 3A Barcelona 08036 aubalde at presenceco.com Ph: +34 93 10 10 300 Fx: +34 93 10 10 333 *www.presenceco.com* *Follow us on:* *[image: tw]* *[image: yt]* *[image: in]* *[image: ss]* *[image: fb]* For additional information, please visit our website *www.presenceco.com* -- *Presence Technology - DisclaimerThis message, its content and any file attached thereto is for the intended recipient only and is confidential and /or privileged. If you have received this e-mail in error or had access to it, you should note that the information in it is private and any use thereof is unauthorized. In such an event please notify us by e-mail or by telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by whatsoever means and any transmission or dissemination thereof to other persons is prohibited. It should be deleted immediately from your system. Presence Technology reserves the right to take legal action against any persons unlawfully gaining access to the content of any external message it has emitted.* *For additional information, please visit our website **www.presenceco.com * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161107/ddf7265c/attachment.html From s.safarov at gmail.com Mon Nov 7 18:55:44 2016 From: s.safarov at gmail.com (Sergey Safarov) Date: Mon, 07 Nov 2016 15:55:44 +0000 Subject: [Freeswitch-users] Verso installation In-Reply-To: References: Message-ID: Thank you Giovanni for your response. For conferences with "35\d{2}" numbers chat and conference members is works. This resolves my issues. Also i find that flag "livearray-sync" is responsible for this feature. This flag is already documented and here . I has added on Verto+Communicator page information about test numbers for vanila config. ??, 7 ????. 2016 ?. ? 12:20, Giovanni Maruzzelli : > Sergey, > > 3000 is a "plain" conference in standard dialplan, with no such features > you are looking for. > > Try calling 3500, in standard dialplan. That is called "stereo-mcu" > conference (if i remember correctly) and is what you want. > > -giovanni > > sent from mobile > cell: +39 347 266 56 18 > Giovanni Maruzzelli > OpenTelecom.IT > > On Nov 7, 2016 8:53 AM, "Sergey Safarov" wrote: > > I has installed verto commucator on my server and point to websocker of > FreeSwith team conference server. All works as expected. > > Then I change websocker URL to my FreeSwith server. Server has trusted > certificate. I dial 3000 number, then connects to conference, can > send/accept audio/video but cannot see who is connected to conference. Also > I cannot send chat messages. > I cannot find related error in browser console, FS log also not contain > errors. > > Could you advise what is required to fix? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > 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/20161107/9b84a997/attachment.html From brian at freeswitch.org Mon Nov 7 19:07:03 2016 From: brian at freeswitch.org (Brian West) Date: Mon, 7 Nov 2016 10:07:03 -0600 Subject: [Freeswitch-users] Video call failed In-Reply-To: References: Message-ID: With that line, I can only guess that you do not have any codes in common, You should patebin the entire log for us to see, pastebin.freeswitch.org /b On Mon, Nov 7, 2016 at 9:55 AM, Agust? Ubalde wrote: > Hi all, > > I've just installed FS 1.6 and configured 2 extensions for make video > calls. > Is this scenario possible without conference the video call? > The call failed with the following message: > > *67cc9edd-bc3a-4b25-bb1c-5dc475a409d6 2016-11-07 15:36:50.676944 [INFO] > mod_dptools.c:3401 Originate Failed. Cause: INCOMPATIBLE_DESTINATION* > > > Thanks! > > *PRESENCE TECHNOLOGY* > *Agust? Ubalde Bellot* > Chief Developer > C/ Comte Urgell 240 3A > Barcelona 08036 > aubalde at presenceco.com > > Ph: +34 93 10 10 300 > Fx: +34 93 10 10 333 > > *www.presenceco.com* > > *Follow us on:* > > *[image: tw]* *[image: yt]* > *[image: in]* > *[image: ss]* > *[image: fb]* > > > For additional information, please visit our website *www.presenceco.com* > > > > *Presence Technology - DisclaimerThis message, its content and any file > attached thereto is for the intended recipient only and is confidential and > /or privileged. If you have received this e-mail in error or had access to > it, you should note that the information in it is private and any use > thereof is unauthorized. In such an event please notify us by e-mail or by > telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by > whatsoever means and any transmission or dissemination thereof to other > persons is prohibited. It should be deleted immediately from your system. > Presence Technology reserves the right to take legal action against any > persons unlawfully gaining access to the content of any external message it > has emitted.* > > *For additional information, please visit our website **www.presenceco.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 (50% Discount using code FreeSwitch50) http://www.freeswitchcookbook.com (50% Discount using code FreeSwitch50) https://www.gofundme.com/freeswitch_ubuntu Got Bugs? Report them here ! | Reddit: /r/freeswitch *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/20161107/5fad50fe/attachment-0001.html From gmaruzz at gmail.com Mon Nov 7 20:06:44 2016 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Mon, 7 Nov 2016 18:06:44 +0100 Subject: [Freeswitch-users] Verso installation In-Reply-To: References: Message-ID: Super! Thanks to you for updating documentation! -giovanni On 7 November 2016 at 16:55, Sergey Safarov wrote: > Thank you Giovanni for your response. > For conferences with "35\d{2}" numbers chat and conference members is > works. > This resolves my issues. > Also i find that flag "livearray-sync" is responsible for this feature. > This flag is already documented > and here > . > I has added on Verto+Communicator > > page information about test numbers for vanila config. > > > ??, 7 ????. 2016 ?. ? 12:20, Giovanni Maruzzelli : > >> Sergey, >> >> 3000 is a "plain" conference in standard dialplan, with no such features >> you are looking for. >> >> Try calling 3500, in standard dialplan. That is called "stereo-mcu" >> conference (if i remember correctly) and is what you want. >> >> -giovanni >> >> sent from mobile >> cell: +39 347 266 56 18 >> Giovanni Maruzzelli >> OpenTelecom.IT >> >> On Nov 7, 2016 8:53 AM, "Sergey Safarov" wrote: >> >> I has installed verto commucator on my server and point to websocker of >> FreeSwith team conference server. All works as expected. >> >> Then I change websocker URL to my FreeSwith server. Server has trusted >> certificate. I dial 3000 number, then connects to conference, can >> send/accept audio/video but cannot see who is connected to conference. Also >> I cannot send chat messages. >> I cannot find related error in browser console, FS log also not contain >> errors. >> >> Could you advise what is required to fix? >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> 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 > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161107/c40d178a/attachment.html From kathleen at freeswitch.org Mon Nov 7 20:44:47 2016 From: kathleen at freeswitch.org (Kathleen King) Date: Mon, 7 Nov 2016 09:44:47 -0800 Subject: [Freeswitch-users] Conference DID Testing help Message-ID: Hello, We are doing some testing and want to make sure the DIDs to call into the public 888 conference are working. The team has been able to test the US number, but we need your help to test the international DIDs. Please take a moment and call to see if the DID for your region is working and reply to this email to let us know. Thank-you! Australia +61-7-3188-7519 Thanks to Jay Binks - NetSIP.com.au Canada +1-438-800-0531 Thanks to NG Communications France +33-975-181-606 Thanks to NG Communications Germany +49-228-9293-9009 Thanks to Yiftach at ChooChee Germany +49-2373-913-4009 Thanks to einfachVoIP.de Ireland +353-1-687-9001 Thanks to Ziron Israel +972-2-372-0394 Thanks to Avi Marcus - BestFone.com Netherlands +31-858-880-387 Thanks to NG Communications New Zealand +64-4-887-1401 Thanks to Ziron Portugal +351-300505224 Thanks to Finesource South Africa +27-87-8204656 Thanks to Othos Telecom Spain +34-91-290-12-71 Thanks to SIPtize UK +44-330-320-0105 Thanks to Ziron UK +44-1904-201-313 Thanks to ukddi.com (Routed by Steven Ayre) UK +44-203-298-5931 Thanks to ukddi.com (Routed by Avi Marcus) USA +1-919-386-9900 Kathleen King FreeSWITCH Public Relations Office: +1-213-286-0400 Mobile: +1-703-859-3757 http://freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161107/6e4b64b9/attachment-0001.html From william at williamcollsassoc.ca Mon Nov 7 20:51:24 2016 From: william at williamcollsassoc.ca (William Colls) Date: Mon, 7 Nov 2016 12:51:24 -0500 Subject: [Freeswitch-users] Conference DID Testing help In-Reply-To: References: Message-ID: Seems to be OK from Ottawa, Canada On 2016-11-07 12:44 PM, Kathleen King wrote: > Hello, > > We are doing some testing and want to make sure the DIDs to call into > the public 888 conference are working. The team has been able to test > the US number, but we need your help to test the international DIDs. > Please take a moment and call to see if the DID for your region is > working and reply to this email to let us know. Thank-you! > > > > Australia +61-7-3188-7519 Thanks to Jay Binks - NetSIP.com.au > > Canada +1-438-800-0531 Thanks to NG Communications > > France +33-975-181-606 Thanks to NG Communications > > Germany +49-228-9293-9009 Thanks to Yiftach at ChooChee > Germany +49-2373-913-4009 Thanks to einfachVoIP.de > > Ireland +353-1-687-9001 Thanks to Ziron > Israel +972-2-372-0394 Thanks to Avi Marcus - BestFone.com > > Netherlands +31-858-880-387 Thanks to NG Communications > > New Zealand +64-4-887-1401 Thanks to Ziron > Portugal +351-300505224 Thanks to Finesource > > South Africa +27-87-8204656 Thanks to Othos Telecom > > Spain +34-91-290-12-71 Thanks to SIPtize > UK +44-330-320-0105 Thanks to Ziron > UK +44-1904-201-313 Thanks to ukddi.com > (Routed by Steven Ayre) > UK +44-203-298-5931 Thanks to ukddi.com > (Routed by Avi Marcus) > USA +1-919-386-9900 > > > > > Kathleen King > FreeSWITCH Public Relations > Office: +1-213-286-0400 > Mobile: +1-703-859-3757 > http://freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- William Colls & Associates VoIP Telephone Services Digital PBX and Telephone Systems -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161107/bda76e44/attachment.html From thesipguy at gmail.com Mon Nov 7 20:57:31 2016 From: thesipguy at gmail.com (Schneur Rosenberg) Date: Mon, 7 Nov 2016 19:57:31 +0200 Subject: [Freeswitch-users] Conference DID Testing help In-Reply-To: References: Message-ID: Israeli number works On Nov 7, 2016 19:48, "Kathleen King" wrote: > Hello, > > We are doing some testing and want to make sure the DIDs to call into the > public 888 conference are working. The team has been able to test the US > number, but we need your help to test the international DIDs. Please take a > moment and call to see if the DID for your region is working and reply to > this email to let us know. Thank-you! > > > > Australia +61-7-3188-7519 Thanks to Jay Binks - NetSIP.com.au > > Canada +1-438-800-0531 Thanks to NG Communications > > France +33-975-181-606 Thanks to NG Communications > > Germany +49-228-9293-9009 Thanks to Yiftach at ChooChee > Germany +49-2373-913-4009 Thanks to einfachVoIP.de > > Ireland +353-1-687-9001 Thanks to Ziron > Israel +972-2-372-0394 Thanks to Avi Marcus - BestFone.com > > Netherlands +31-858-880-387 Thanks to NG Communications > > New Zealand +64-4-887-1401 Thanks to Ziron > Portugal +351-300505224 Thanks to Finesource > > South Africa +27-87-8204656 Thanks to Othos Telecom > > Spain +34-91-290-12-71 Thanks to SIPtize > UK +44-330-320-0105 Thanks to Ziron > UK +44-1904-201-313 Thanks to ukddi.com (Routed > by Steven Ayre) > UK +44-203-298-5931 Thanks to ukddi.com (Routed > by Avi Marcus) > USA +1-919-386-9900 > > > > Kathleen King > FreeSWITCH Public Relations > Office: +1-213-286-0400 > Mobile: +1-703-859-3757 > http://freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161107/c1abed04/attachment-0001.html From krice at freeswitch.org Mon Nov 7 21:00:26 2016 From: krice at freeswitch.org (Ken Rice) Date: Mon, 7 Nov 2016 12:00:26 -0600 Subject: [Freeswitch-users] Conference DID Testing help In-Reply-To: References: Message-ID: <1cf301d23920$d36230d0$7a269270$@freeswitch.org> If you are actually responsible for any of these DIDs please make sure they are not pointed to an IP address but pointed to the conference.freeswitch.org hostname so if the IP changes they keep working From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Kathleen King Sent: Monday, November 7, 2016 11:45 AM To: FreeSWITCH-users Subject: [Freeswitch-users] Conference DID Testing help Hello, We are doing some testing and want to make sure the DIDs to call into the public 888 conference are working. The team has been able to test the US number, but we need your help to test the international DIDs. Please take a moment and call to see if the DID for your region is working and reply to this email to let us know. Thank-you! Australia +61-7-3188-7519 Thanks to Jay Binks - NetSIP.com.au Canada +1-438-800-0531 Thanks to NG Communications France +33-975-181-606 Thanks to NG Communications Germany +49-228-9293-9009 Thanks to Yiftach at ChooChee Germany +49-2373-913-4009 Thanks to einfachVoIP.de Ireland +353-1-687-9001 Thanks to Ziron Israel +972-2-372-0394 Thanks to Avi Marcus - BestFone.com Netherlands +31-858-880-387 Thanks to NG Communications New Zealand +64-4-887-1401 Thanks to Ziron Portugal +351-300505224 Thanks to Finesource South Africa +27-87-8204656 Thanks to Othos Telecom Spain +34-91-290-12-71 Thanks to SIPtize UK +44-330-320-0105 Thanks to Ziron UK +44-1904-201-313 Thanks to ukddi.com (Routed by Steven Ayre) UK +44-203-298-5931 Thanks to ukddi.com (Routed by Avi Marcus) USA +1-919-386-9900 Kathleen King FreeSWITCH Public Relations Office: +1-213-286-0400 Mobile: +1-703-859-3757 http://freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161107/1fdee544/attachment.html From avi at avimarcus.net Mon Nov 7 21:01:44 2016 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 7 Nov 2016 18:01:44 +0000 Subject: [Freeswitch-users] Conference DID Testing help In-Reply-To: References: Message-ID: <010001583ff3fd3a-fe603ccc-c26d-4f59-a887-d187466f9e59-000000@email.amazonses.com> Yes, my Israel number works. -Avi Marcus BestFone On Mon, Nov 7, 2016 at 7:44 PM, Kathleen King wrote: > Hello, > > We are doing some testing and want to make sure the DIDs to call into the > public 888 conference are working. The team has been able to test the US > number, but we need your help to test the international DIDs. Please take a > moment and call to see if the DID for your region is working and reply to > this email to let us know. Thank-you! > > > > Australia +61-7-3188-7519 Thanks to Jay Binks - NetSIP.com.au > > Canada +1-438-800-0531 Thanks to NG Communications > > France +33-975-181-606 Thanks to NG Communications > > Germany +49-228-9293-9009 Thanks to Yiftach at ChooChee > Germany +49-2373-913-4009 Thanks to einfachVoIP.de > > Ireland +353-1-687-9001 Thanks to Ziron > Israel +972-2-372-0394 Thanks to Avi Marcus - BestFone.com > > Netherlands +31-858-880-387 Thanks to NG Communications > > New Zealand +64-4-887-1401 Thanks to Ziron > Portugal +351-300505224 Thanks to Finesource > > South Africa +27-87-8204656 Thanks to Othos Telecom > > Spain +34-91-290-12-71 Thanks to SIPtize > UK +44-330-320-0105 Thanks to Ziron > UK +44-1904-201-313 Thanks to ukddi.com (Routed > by Steven Ayre) > UK +44-203-298-5931 Thanks to ukddi.com (Routed > by Avi Marcus) > USA +1-919-386-9900 > > > > Kathleen King > FreeSWITCH Public Relations > Office: +1-213-286-0400 > Mobile: +1-703-859-3757 > http://freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161107/a11312a8/attachment-0001.html From danb.lists at gmail.com Mon Nov 7 21:08:34 2016 From: danb.lists at gmail.com (DanB) Date: Mon, 7 Nov 2016 19:08:34 +0100 Subject: [Freeswitch-users] Conference DID Testing help In-Reply-To: References: Message-ID: <21d4b293-7230-8f8b-b0fa-ed511e518d2c@gmail.com> Kathleen, Germany numbers appear to be both down, first pointed towards some other IVR, second reporting not available. BR DanB Germany +49-228-9293-9009 Thanks to Yiftach at ChooChee Germany +49-2373-913-4009 Thanks to einfachVoIP.de -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161107/8e68fdd7/attachment.html From andrew at cassidywebservices.co.uk Mon Nov 7 21:34:04 2016 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Mon, 7 Nov 2016 18:34:04 +0000 Subject: [Freeswitch-users] Conference DID Testing help In-Reply-To: References: Message-ID: All three UK ones work, however I've heard through the grapevine that ukddi.com are no longer providing their free service. On 7 November 2016 at 17:44, Kathleen King wrote: > Hello, > > We are doing some testing and want to make sure the DIDs to call into the > public 888 conference are working. The team has been able to test the US > number, but we need your help to test the international DIDs. Please take a > moment and call to see if the DID for your region is working and reply to > this email to let us know. Thank-you! > > > > Australia +61-7-3188-7519 Thanks to Jay Binks - NetSIP.com.au > > Canada +1-438-800-0531 Thanks to NG Communications > > France +33-975-181-606 Thanks to NG Communications > > Germany +49-228-9293-9009 Thanks to Yiftach at ChooChee > Germany +49-2373-913-4009 Thanks to einfachVoIP.de > > Ireland +353-1-687-9001 Thanks to Ziron > Israel +972-2-372-0394 Thanks to Avi Marcus - BestFone.com > > Netherlands +31-858-880-387 Thanks to NG Communications > > New Zealand +64-4-887-1401 Thanks to Ziron > Portugal +351-300505224 Thanks to Finesource > > South Africa +27-87-8204656 Thanks to Othos Telecom > > Spain +34-91-290-12-71 Thanks to SIPtize > UK +44-330-320-0105 Thanks to Ziron > UK +44-1904-201-313 Thanks to ukddi.com (Routed > by Steven Ayre) > UK +44-203-298-5931 Thanks to ukddi.com (Routed > by Avi Marcus) > USA +1-919-386-9900 > > > > Kathleen King > FreeSWITCH Public Relations > Office: +1-213-286-0400 > Mobile: +1-703-859-3757 > http://freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 03303 880 960 andrew at cassidyweb.co.uk www.cassidyweb.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161107/77d2c193/attachment.html From krice at freeswitch.org Mon Nov 7 21:39:54 2016 From: krice at freeswitch.org (Ken Rice) Date: Mon, 7 Nov 2016 12:39:54 -0600 Subject: [Freeswitch-users] Conference DID Testing help In-Reply-To: References: Message-ID: <1d5d01d23926$56ef4eb0$04cdec10$@freeswitch.org> Yeah sounds like David got tired of scammers a while back and stopped doing new numbers after scammers started abusing it From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Andrew Cassidy Sent: Monday, November 7, 2016 12:34 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Conference DID Testing help All three UK ones work, however I've heard through the grapevine that ukddi.com are no longer providing their free service. On 7 November 2016 at 17:44, Kathleen King > wrote: Hello, We are doing some testing and want to make sure the DIDs to call into the public 888 conference are working. The team has been able to test the US number, but we need your help to test the international DIDs. Please take a moment and call to see if the DID for your region is working and reply to this email to let us know. Thank-you! Australia +61-7-3188-7519 Thanks to Jay Binks - NetSIP.com.au Canada +1-438-800-0531 Thanks to NG Communications France +33-975-181-606 Thanks to NG Communications Germany +49-228-9293-9009 Thanks to Yiftach at ChooChee Germany +49-2373-913-4009 Thanks to einfachVoIP.de Ireland +353-1-687-9001 Thanks to Ziron Israel +972-2-372-0394 Thanks to Avi Marcus - BestFone.com Netherlands +31-858-880-387 Thanks to NG Communications New Zealand +64-4-887-1401 Thanks to Ziron Portugal +351-300505224 Thanks to Finesource South Africa +27-87-8204656 Thanks to Othos Telecom Spain +34-91-290-12-71 Thanks to SIPtize UK +44-330-320-0105 Thanks to Ziron UK +44-1904-201-313 Thanks to ukddi.com (Routed by Steven Ayre) UK +44-203-298-5931 Thanks to ukddi.com (Routed by Avi Marcus) USA +1-919-386-9900 Kathleen King FreeSWITCH Public Relations Office: +1-213-286-0400 Mobile: +1-703-859-3757 http://freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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 03303 880 960 andrew at cassidyweb.co.uk www.cassidyweb.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161107/f89326b2/attachment-0001.html From davidwaf at gmail.com Mon Nov 7 21:41:50 2016 From: davidwaf at gmail.com (David Wafula) Date: Mon, 7 Nov 2016 20:41:50 +0200 Subject: [Freeswitch-users] Conference DID Testing help In-Reply-To: References: Message-ID: South Africa works. On Mon, Nov 7, 2016 at 8:34 PM, Andrew Cassidy < andrew at cassidywebservices.co.uk> wrote: > All three UK ones work, however I've heard through the grapevine that > ukddi.com are no longer providing their free service. > > On 7 November 2016 at 17:44, Kathleen King > wrote: > >> Hello, >> >> We are doing some testing and want to make sure the DIDs to call into the >> public 888 conference are working. The team has been able to test the US >> number, but we need your help to test the international DIDs. Please take a >> moment and call to see if the DID for your region is working and reply to >> this email to let us know. Thank-you! >> >> >> >> Australia +61-7-3188-7519 Thanks to Jay Binks - NetSIP.com.au >> >> Canada +1-438-800-0531 Thanks to NG Communications >> >> France +33-975-181-606 Thanks to NG Communications >> >> Germany +49-228-9293-9009 Thanks to Yiftach at ChooChee >> Germany +49-2373-913-4009 Thanks to einfachVoIP.de >> >> Ireland +353-1-687-9001 Thanks to Ziron >> Israel +972-2-372-0394 Thanks to Avi Marcus - BestFone.com >> >> Netherlands +31-858-880-387 Thanks to NG Communications >> >> New Zealand +64-4-887-1401 Thanks to Ziron >> Portugal +351-300505224 Thanks to Finesource >> >> South Africa +27-87-8204656 Thanks to Othos Telecom >> >> Spain +34-91-290-12-71 Thanks to SIPtize >> UK +44-330-320-0105 Thanks to Ziron >> UK +44-1904-201-313 Thanks to ukddi.com (Routed >> by Steven Ayre) >> UK +44-203-298-5931 Thanks to ukddi.com (Routed >> by Avi Marcus) >> USA +1-919-386-9900 >> >> >> >> Kathleen King >> FreeSWITCH Public Relations >> Office: +1-213-286-0400 >> Mobile: +1-703-859-3757 >> http://freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 > > 03303 880 960 andrew at cassidyweb.co.uk ww > w.cassidyweb.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 > -- David W -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161107/cb907ead/attachment.html From hugh at irvine.com.au Tue Nov 8 01:17:31 2016 From: hugh at irvine.com.au (Hugh Irvine) Date: Tue, 8 Nov 2016 09:17:31 +1100 Subject: [Freeswitch-users] Conference DID Testing help In-Reply-To: References: Message-ID: Australia is working. > On 8 Nov. 2016, at 04:44, Kathleen King wrote: > > Hello, > > We are doing some testing and want to make sure the DIDs to call into the public 888 conference are working. The team has been able to test the US number, but we need your help to test the international DIDs. Please take a moment and call to see if the DID for your region is working and reply to this email to let us know. Thank-you! > > > > Australia +61-7-3188-7519 Thanks to Jay Binks - NetSIP.com.au > Canada +1-438-800-0531 Thanks to NG Communications > France +33-975-181-606 Thanks to NG Communications > Germany +49-228-9293-9009 Thanks to Yiftach at ChooChee > Germany +49-2373-913-4009 Thanks to einfachVoIP.de > Ireland +353-1-687-9001 Thanks to Ziron > Israel +972-2-372-0394 Thanks to Avi Marcus - BestFone.com > Netherlands +31-858-880-387 Thanks to NG Communications > New Zealand +64-4-887-1401 Thanks to Ziron > Portugal +351-300505224 Thanks to Finesource > South Africa +27-87-8204656 Thanks to Othos Telecom > Spain +34-91-290-12-71 Thanks to SIPtize > UK +44-330-320-0105 Thanks to Ziron > UK +44-1904-201-313 Thanks to ukddi.com (Routed by Steven Ayre) > UK +44-203-298-5931 Thanks to ukddi.com (Routed by Avi Marcus) > USA +1-919-386-9900 > > > > Kathleen King > FreeSWITCH Public Relations > Office: +1-213-286-0400 > Mobile: +1-703-859-3757 > http://freeswitch.org > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 fs at voice2net.ca Tue Nov 8 01:16:48 2016 From: fs at voice2net.ca (fs at voice2net.ca) Date: Mon, 7 Nov 2016 17:16:48 -0500 Subject: [Freeswitch-users] 486 busy here adding Normal Clearing Message-ID: <01bb01d23944$a2df2ca0$e89d85e0$@ca> Can anyone tell me if it is possible to correct this issue. I am using a freeswitch as a gateway to sereval carriers for routing traffic. When I receive a 486 Busy Here without the Reason Element, when the freeswitch responds to the originator, it adds in a Reason, 16 Normal clearing. Our servers on the other end are deciphering this as the wrong cause, it would be great to either remove the Normal Clearing message or have it reflect 17, user busy. SIP/2.0 486 Busy Here Via: SIP/2.0/UDP xx.xx.xx.xx;received=xx.xx.xx.xx;branch=z9hG4bKSB2rZ17eQ3ryK;rport=5080 From: "temp" ;tag=N5aNrFte8N6SS To: ;tag=SDhv1pb99-400a14cd+1+69160089+a381d68 0 Call-ID: d9d1f48b-1fd7-1235-a9ad-00002d3a2463 CSeq: 98941824 INVITE Server: DC-SIP/2.0 Content-Length: 0 Retry-After: 1 SIP/2.0 486 Busy Here Via: SIP/2.0/UDP xx.xx.xx.xx;rport=5060;branch=z9hG4bKypUUHeUU1gK7j Max-Forwards: 69 From: "temp" ;tag=Z5pjZ2QtN3KpQ To: ;tag=SFZ9Bv2NpD7jK Call-ID: d9d097a2-1fd7-1235-038a-00163eef87f4 CSeq: 98941824 INVITE User-Agent: FreeSWITCH-mod_sofia/1.4.26+git~20160205T175853Z~ca9207aa32~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 X-FS-Display-Name: 14166359251 X-FS-Display-Number: sip:14166359251 at domain.com P-Asserted-Identity: "14166359251" Darcy Primrose octoberlogo3 http://www.voice2net.ca darcy at voice2net.ca 200 Prescott Street Kemptville, ON K0G 1J0 Contact via Phone -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161107/b344bd50/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/png Size: 11737 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161107/b344bd50/attachment-0001.png From fs at voice2net.ca Tue Nov 8 02:44:44 2016 From: fs at voice2net.ca (fs at voice2net.ca) Date: Mon, 7 Nov 2016 18:44:44 -0500 Subject: [Freeswitch-users] 486 busy here adding Normal Clearing In-Reply-To: <01bb01d23944$a2df2ca0$e89d85e0$@ca> References: <01bb01d23944$a2df2ca0$e89d85e0$@ca> Message-ID: <021a01d23950$eba56e20$c2f04a60$@ca> Disregard, I fixed it in the dial plan. Darcy Can anyone tell me if it is possible to correct this issue. I am using a freeswitch as a gateway to sereval carriers for routing traffic. When I receive a 486 Busy Here without the Reason Element, when the freeswitch responds to the originator, it adds in a Reason, 16 Normal clearing. Our servers on the other end are deciphering this as the wrong cause, it would be great to either remove the Normal Clearing message or have it reflect 17, user busy. SIP/2.0 486 Busy Here Via: SIP/2.0/UDP xx.xx.xx.xx;received=xx.xx.xx.xx;branch=z9hG4bKSB2rZ17eQ3ryK;rport=5080 From: "temp" ;tag=N5aNrFte8N6SS To: ;tag=SDhv1pb99-400a14cd+1+69160089+a381d68 0 Call-ID: d9d1f48b-1fd7-1235-a9ad-00002d3a2463 CSeq: 98941824 INVITE Server: DC-SIP/2.0 Content-Length: 0 Retry-After: 1 SIP/2.0 486 Busy Here Via: SIP/2.0/UDP xx.xx.xx.xx;rport=5060;branch=z9hG4bKypUUHeUU1gK7j Max-Forwards: 69 From: "temp" ;tag=Z5pjZ2QtN3KpQ To: ;tag=SFZ9Bv2NpD7jK Call-ID: d9d097a2-1fd7-1235-038a-00163eef87f4 CSeq: 98941824 INVITE User-Agent: FreeSWITCH-mod_sofia/1.4.26+git~20160205T175853Z~ca9207aa32~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 X-FS-Display-Name: 14166359251 X-FS-Display-Number: sip:14166359251 at domain.com P-Asserted-Identity: "14166359251" Darcy Primrose octoberlogo3 http://www.voice2net.ca darcy at voice2net.ca 200 Prescott Street Kemptville, ON K0G 1J0 Contact via Phone -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161107/287e70e1/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/png Size: 11737 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161107/287e70e1/attachment.png From samir.doshi at inextrix.com Tue Nov 8 09:49:51 2016 From: samir.doshi at inextrix.com (Samir Doshi) Date: Tue, 8 Nov 2016 12:19:51 +0530 Subject: [Freeswitch-users] Issue with cc_warning_tone in mod_callcenter module In-Reply-To: References: Message-ID: Hi, I tried with cc_outbound_announce variable but getting same result. Not getting alert tone. Any other things should I look? Best Regards -- Samir Doshi *iNextrix Technologie**s Pvt. Ltd*. http://www.inextrix.com *Disclaimer:* The information contained in this communication is confidential and may be legally privileged. It is intended solely for the use of the individual or entity to whom it is addressed and others authorised to receive it. If you are not the intended recipient you are hereby notified that any disclosure, copying, distribution or taking action in reliance of the contents of this information is strictly prohibited and may be unlawful. On Sun, Nov 6, 2016 at 3:30 PM, Samir Doshi wrote: > Thanks. I will give a try and get back to you with result. > > > Best Regards > -- > Samir Doshi > *iNextrix Technologie**s Pvt. Ltd*. > http://www.inextrix.com > > > > *Disclaimer:* > The information contained in this communication is confidential and may be > legally privileged. It is intended solely for the use of the individual or > entity to whom it is addressed and others authorised to receive it. If you > are not the intended recipient you are hereby notified that any disclosure, > copying, distribution or taking action in reliance of the contents of this > information is strictly prohibited and may be unlawful. > > On Sat, Nov 5, 2016 at 1:31 AM, ?talo Rossi wrote: > >> mod_callcenter.c:1776 Agent 8003 answered "Outbound Call" >> from queue agent8003 at default (Recorded) >> >> From this line you're using callback agents and the variable should be >> cc_outbound_announce instead of cc_warning_tone, this one is specific for >> uuid-standby agents >> >> On Fri, Nov 4, 2016 at 3:36 PM, Samir Doshi >> wrote: >> >>> Hi, >>> >>> We are doing impementation using mod_callcenter module by taking >>> reference from documentation (https://freeswitch.org/conflu >>> ence/display/FREESWITCH/mod_callcenter). So far everything works well >>> except cc_warning_tone playback. We are never getting tone when some >>> connect to agent. >>> >>> Debug Logs : >>> >>> FS cli Log : https://pastebin.freeswitch.org/view/f9be26b9 >>> Log with sip debug : https://pastebin.freeswitch.org/view/aab8ea85 >>> Log with sip debug + loglevel 9 : https://pastebin.freeswitch.or >>> g/view/ff793521 >>> >>> Any hint? >>> >>> >>> Best Regards >>> -- >>> Samir Doshi >>> *iNextrix Technologie**s Pvt. Ltd*. >>> http://www.inextrix.com >>> >>> >>> >>> *Disclaimer:* >>> The information contained in this communication is confidential and may >>> be legally privileged. It is intended solely for the use of the individual >>> or entity to whom it is addressed and others authorised to receive it. If >>> you are not the intended recipient you are hereby notified that any >>> disclosure, copying, distribution or taking action in reliance of the >>> contents of this information is strictly prohibited and may be unlawful. >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 >> italo at freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161108/cc8b9b3a/attachment-0001.html From gascagonzalo at gmail.com Tue Nov 8 10:19:44 2016 From: gascagonzalo at gmail.com (Gonzalo Gasca Meza) Date: Mon, 7 Nov 2016 23:19:44 -0800 Subject: [Freeswitch-users] Freeswitch WebRTC with Nginx In-Reply-To: References: Message-ID: Do you have Ngnix/FS logs. I remember I needed to change URI headers in ngnix. Thanks On Mon, Nov 7, 2016 at 4:03 AM, Murugan Pandian wrote: > HI, > > I am using nginx for load balancing behind my freeswitch ,My WebRTC > client SIPJS,i can able to register with my freeswitch and i can make > outbound call successfully but when i cant able to receive any incoming > call,if i register directly with freeswitch means without nginx i can make > and receive call but when i register using nginx i am facing this issue(i > cant able to receive call ). > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161107/0a5e0eb8/attachment.html From christophe.yann at gmail.com Tue Nov 8 11:22:13 2016 From: christophe.yann at gmail.com (yann christophe) Date: Tue, 8 Nov 2016 09:22:13 +0100 Subject: [Freeswitch-users] Conference DID Testing help In-Reply-To: References: Message-ID: French number works 2016-11-07 23:17 GMT+01:00 Hugh Irvine : > > Australia is working. > > >> On 8 Nov. 2016, at 04:44, Kathleen King wrote: >> >> Hello, >> >> We are doing some testing and want to make sure the DIDs to call into the public 888 conference are working. The team has been able to test the US number, but we need your help to test the international DIDs. Please take a moment and call to see if the DID for your region is working and reply to this email to let us know. Thank-you! >> >> >> >> Australia +61-7-3188-7519 Thanks to Jay Binks - NetSIP.com.au >> Canada +1-438-800-0531 Thanks to NG Communications >> France +33-975-181-606 Thanks to NG Communications >> Germany +49-228-9293-9009 Thanks to Yiftach at ChooChee >> Germany +49-2373-913-4009 Thanks to einfachVoIP.de >> Ireland +353-1-687-9001 Thanks to Ziron >> Israel +972-2-372-0394 Thanks to Avi Marcus - BestFone.com >> Netherlands +31-858-880-387 Thanks to NG Communications >> New Zealand +64-4-887-1401 Thanks to Ziron >> Portugal +351-300505224 Thanks to Finesource >> South Africa +27-87-8204656 Thanks to Othos Telecom >> Spain +34-91-290-12-71 Thanks to SIPtize >> UK +44-330-320-0105 Thanks to Ziron >> UK +44-1904-201-313 Thanks to ukddi.com (Routed by Steven Ayre) >> UK +44-203-298-5931 Thanks to ukddi.com (Routed by Avi Marcus) >> USA +1-919-386-9900 >> >> >> >> Kathleen King >> FreeSWITCH Public Relations >> Office: +1-213-286-0400 >> Mobile: +1-703-859-3757 >> http://freeswitch.org >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> 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 aubalde at presenceco.com Tue Nov 8 11:33:44 2016 From: aubalde at presenceco.com (=?UTF-8?Q?Agust=C3=AD_Ubalde?=) Date: Tue, 8 Nov 2016 09:33:44 +0100 Subject: [Freeswitch-users] Video call failed In-Reply-To: References: Message-ID: Hi Brian, This is the complete log https://pastebin.freeswitch.org/view/9caf57a5. Thanks, *PRESENCE TECHNOLOGY* *Agust? Ubalde Bellot* Chief Developer C/ Comte Urgell 240 3A Barcelona 08036 aubalde at presenceco.com Ph: +34 93 10 10 300 Fx: +34 93 10 10 333 *www.presenceco.com* *Follow us on:* *[image: tw]* *[image: yt]* *[image: in]* *[image: ss]* *[image: fb]* For additional information, please visit our website *www.presenceco.com* 2016-11-07 17:07 GMT+01:00 Brian West : > With that line, I can only guess that you do not have any codes in common, > You should patebin the entire log for us to see, pastebin.freeswitch.org > > /b > > > On Mon, Nov 7, 2016 at 9:55 AM, Agust? Ubalde > wrote: > >> Hi all, >> >> I've just installed FS 1.6 and configured 2 extensions for make video >> calls. >> Is this scenario possible without conference the video call? >> The call failed with the following message: >> >> *67cc9edd-bc3a-4b25-bb1c-5dc475a409d6 2016-11-07 15:36:50.676944 [INFO] >> mod_dptools.c:3401 Originate Failed. Cause: INCOMPATIBLE_DESTINATION* >> >> >> Thanks! >> >> *PRESENCE TECHNOLOGY* >> *Agust? Ubalde Bellot* >> Chief Developer >> C/ Comte Urgell 240 3A >> Barcelona 08036 >> aubalde at presenceco.com >> >> Ph: +34 93 10 10 300 >> Fx: +34 93 10 10 333 >> >> *www.presenceco.com* >> >> *Follow us on:* >> >> *[image: tw]* *[image: yt]* >> *[image: in]* >> *[image: ss]* >> *[image: fb]* >> >> >> For additional information, please visit our website *www.presenceco.com* >> >> >> >> *Presence Technology - DisclaimerThis message, its content and any file >> attached thereto is for the intended recipient only and is confidential and >> /or privileged. If you have received this e-mail in error or had access to >> it, you should note that the information in it is private and any use >> thereof is unauthorized. In such an event please notify us by e-mail or by >> telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by >> whatsoever means and any transmission or dissemination thereof to other >> persons is prohibited. It should be deleted immediately from your system. >> Presence Technology reserves the right to take legal action against any >> persons unlawfully gaining access to the content of any external message it >> has emitted.* >> >> *For additional information, please visit our website **www.presenceco.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 (50% Discount using code FreeSwitch50) > http://www.freeswitchcookbook.com (50% Discount using code FreeSwitch50) > https://www.gofundme.com/freeswitch_ubuntu > > Got Bugs? Report them here ! | Reddit: > /r/freeswitch > > *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 > -- *Presence Technology - DisclaimerThis message, its content and any file attached thereto is for the intended recipient only and is confidential and /or privileged. If you have received this e-mail in error or had access to it, you should note that the information in it is private and any use thereof is unauthorized. In such an event please notify us by e-mail or by telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by whatsoever means and any transmission or dissemination thereof to other persons is prohibited. It should be deleted immediately from your system. Presence Technology reserves the right to take legal action against any persons unlawfully gaining access to the content of any external message it has emitted.* *For additional information, please visit our website **www.presenceco.com * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161108/c2cb6546/attachment-0001.html From me at nevian.org Tue Nov 8 11:51:07 2016 From: me at nevian.org (Serge S. Yuriev) Date: Tue, 08 Nov 2016 11:51:07 +0300 Subject: [Freeswitch-users] 486 busy here adding Normal Clearing In-Reply-To: <021a01d23950$eba56e20$c2f04a60$@ca> References: <021a01d23950$eba56e20$c2f04a60$@ca> <01bb01d23944$a2df2ca0$e89d85e0$@ca> Message-ID: <2287801478595067@web8o.yandex.ru> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161108/0f0a60f5/attachment.html From gascagonzalo at gmail.com Tue Nov 8 12:46:25 2016 From: gascagonzalo at gmail.com (Gonzalo Gasca Meza) Date: Tue, 8 Nov 2016 01:46:25 -0800 Subject: [Freeswitch-users] Video call failed In-Reply-To: References: Message-ID: What type of Browser/client are you using for these calls? Chrome to Chrome..FF to FF ? On Tue, Nov 8, 2016 at 12:33 AM, Agust? Ubalde wrote: > Hi Brian, > > This is the complete log https://pastebin.freeswitch.org/view/9caf57a5. > > > Thanks, > > *PRESENCE TECHNOLOGY* > *Agust? Ubalde Bellot* > Chief Developer > C/ Comte Urgell 240 3A > Barcelona 08036 > aubalde at presenceco.com > > Ph: +34 93 10 10 300 > Fx: +34 93 10 10 333 > > *www.presenceco.com* > > *Follow us on:* > > *[image: tw]* *[image: yt]* > *[image: in]* > *[image: ss]* > *[image: fb]* > > > For additional information, please visit our website *www.presenceco.com* > > > 2016-11-07 17:07 GMT+01:00 Brian West : > >> With that line, I can only guess that you do not have any codes in >> common, You should patebin the entire log for us to see, >> pastebin.freeswitch.org >> >> /b >> >> >> On Mon, Nov 7, 2016 at 9:55 AM, Agust? Ubalde >> wrote: >> >>> Hi all, >>> >>> I've just installed FS 1.6 and configured 2 extensions for make video >>> calls. >>> Is this scenario possible without conference the video call? >>> The call failed with the following message: >>> >>> *67cc9edd-bc3a-4b25-bb1c-5dc475a409d6 2016-11-07 15:36:50.676944 [INFO] >>> mod_dptools.c:3401 Originate Failed. Cause: INCOMPATIBLE_DESTINATION* >>> >>> >>> Thanks! >>> >>> *PRESENCE TECHNOLOGY* >>> *Agust? Ubalde Bellot* >>> Chief Developer >>> C/ Comte Urgell 240 3A >>> Barcelona 08036 >>> aubalde at presenceco.com >>> >>> Ph: +34 93 10 10 300 >>> Fx: +34 93 10 10 333 >>> >>> *www.presenceco.com* >>> >>> *Follow us on:* >>> >>> *[image: tw]* *[image: yt]* >>> *[image: in]* >>> *[image: ss]* >>> *[image: fb]* >>> >>> >>> For additional information, please visit our website >>> *www.presenceco.com* >>> >>> >>> *Presence Technology - DisclaimerThis message, its content and any file >>> attached thereto is for the intended recipient only and is confidential and >>> /or privileged. If you have received this e-mail in error or had access to >>> it, you should note that the information in it is private and any use >>> thereof is unauthorized. In such an event please notify us by e-mail or by >>> telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by >>> whatsoever means and any transmission or dissemination thereof to other >>> persons is prohibited. It should be deleted immediately from your system. >>> Presence Technology reserves the right to take legal action against any >>> persons unlawfully gaining access to the content of any external message it >>> has emitted.* >>> >>> *For additional information, please visit our website **www.presenceco.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 (50% Discount using code FreeSwitch50) >> http://www.freeswitchcookbook.com (50% Discount using code FreeSwitch50) >> https://www.gofundme.com/freeswitch_ubuntu >> >> Got Bugs? Report them here ! | Reddit: >> /r/freeswitch >> >> *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 >> > > > > *Presence Technology - DisclaimerThis message, its content and any file > attached thereto is for the intended recipient only and is confidential and > /or privileged. If you have received this e-mail in error or had access to > it, you should note that the information in it is private and any use > thereof is unauthorized. In such an event please notify us by e-mail or by > telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by > whatsoever means and any transmission or dissemination thereof to other > persons is prohibited. It should be deleted immediately from your system. > Presence Technology reserves the right to take legal action against any > persons unlawfully gaining access to the content of any external message it > has emitted.* > > *For additional information, please visit our website **www.presenceco.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/20161108/0f49373e/attachment-0001.html From aubalde at presenceco.com Tue Nov 8 12:53:50 2016 From: aubalde at presenceco.com (=?UTF-8?Q?Agust=C3=AD_Ubalde?=) Date: Tue, 8 Nov 2016 10:53:50 +0100 Subject: [Freeswitch-users] Video call failed In-Reply-To: References: Message-ID: Hi Gonzalo, Chrome to Chrome, but I've to test all combinations. Thanks, *PRESENCE TECHNOLOGY* An *ENGHOUSE INTERACTIVE* Company *Agust? Ubalde Bellot* Chief Developer C/ Comte Urgell 240 3A Barcelona 08036 *aubalde at presenceco.com * Ph: +34 93 10 10 300 <%2B34%20931%20010%20300> Fax: +34 93 10 10 333 <%2B34%20931%20010%20333> Enghouse is listed on the Toronto Stock Exchange, stock symbol: ENGH 2016-11-08 10:46 GMT+01:00 Gonzalo Gasca Meza : > What type of Browser/client are you using for these calls? Chrome to > Chrome..FF to FF ? > > On Tue, Nov 8, 2016 at 12:33 AM, Agust? Ubalde > wrote: > >> Hi Brian, >> >> This is the complete log https://pastebin.freeswitch.org/view/9caf57a5. >> >> >> Thanks, >> >> *PRESENCE TECHNOLOGY* >> *Agust? Ubalde Bellot* >> Chief Developer >> C/ Comte Urgell 240 3A >> Barcelona 08036 >> aubalde at presenceco.com >> >> Ph: +34 93 10 10 300 >> Fx: +34 93 10 10 333 >> >> *www.presenceco.com* >> >> *Follow us on:* >> >> *[image: tw]* *[image: yt]* >> *[image: in]* >> *[image: ss]* >> *[image: fb]* >> >> >> For additional information, please visit our website *www.presenceco.com* >> >> >> 2016-11-07 17:07 GMT+01:00 Brian West : >> >>> With that line, I can only guess that you do not have any codes in >>> common, You should patebin the entire log for us to see, >>> pastebin.freeswitch.org >>> >>> /b >>> >>> >>> On Mon, Nov 7, 2016 at 9:55 AM, Agust? Ubalde >>> wrote: >>> >>>> Hi all, >>>> >>>> I've just installed FS 1.6 and configured 2 extensions for make video >>>> calls. >>>> Is this scenario possible without conference the video call? >>>> The call failed with the following message: >>>> >>>> *67cc9edd-bc3a-4b25-bb1c-5dc475a409d6 2016-11-07 15:36:50.676944 [INFO] >>>> mod_dptools.c:3401 Originate Failed. Cause: INCOMPATIBLE_DESTINATION* >>>> >>>> >>>> Thanks! >>>> >>>> *PRESENCE TECHNOLOGY* >>>> *Agust? Ubalde Bellot* >>>> Chief Developer >>>> C/ Comte Urgell 240 3A >>>> Barcelona 08036 >>>> aubalde at presenceco.com >>>> >>>> Ph: +34 93 10 10 300 >>>> Fx: +34 93 10 10 333 >>>> >>>> *www.presenceco.com* >>>> >>>> *Follow us on:* >>>> >>>> *[image: tw]* *[image: yt]* >>>> *[image: in]* >>>> *[image: ss]* >>>> *[image: fb]* >>>> >>>> >>>> For additional information, please visit our website >>>> *www.presenceco.com* >>>> >>>> >>>> *Presence Technology - DisclaimerThis message, its content and any file >>>> attached thereto is for the intended recipient only and is confidential and >>>> /or privileged. If you have received this e-mail in error or had access to >>>> it, you should note that the information in it is private and any use >>>> thereof is unauthorized. In such an event please notify us by e-mail or by >>>> telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by >>>> whatsoever means and any transmission or dissemination thereof to other >>>> persons is prohibited. It should be deleted immediately from your system. >>>> Presence Technology reserves the right to take legal action against any >>>> persons unlawfully gaining access to the content of any external message it >>>> has emitted.* >>>> >>>> *For additional information, please visit our website **www.presenceco.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/free >>>> switch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> >>> -- >>> >>> *Brian West* >>> brian at freeswitch.org >>> >>> >>> *Twitter: @FreeSWITCH , @briankwest* >>> http://www.freeswitchbook.com (50% Discount using code FreeSwitch50) >>> http://www.freeswitchcookbook.com (50% Discount using code FreeSwitch50) >>> https://www.gofundme.com/freeswitch_ubuntu >>> >>> Got Bugs? Report them here ! | Reddit: >>> /r/freeswitch >>> >>> *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 >>> >> >> >> >> *Presence Technology - DisclaimerThis message, its content and any file >> attached thereto is for the intended recipient only and is confidential and >> /or privileged. If you have received this e-mail in error or had access to >> it, you should note that the information in it is private and any use >> thereof is unauthorized. In such an event please notify us by e-mail or by >> telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by >> whatsoever means and any transmission or dissemination thereof to other >> persons is prohibited. It should be deleted immediately from your system. >> Presence Technology reserves the right to take legal action against any >> persons unlawfully gaining access to the content of any external message it >> has emitted.* >> >> *For additional information, please visit our website **www.presenceco.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 > -- *Presence Technology - DisclaimerThis message, its content and any file attached thereto is for the intended recipient only and is confidential and /or privileged. If you have received this e-mail in error or had access to it, you should note that the information in it is private and any use thereof is unauthorized. In such an event please notify us by e-mail or by telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by whatsoever means and any transmission or dissemination thereof to other persons is prohibited. It should be deleted immediately from your system. Presence Technology reserves the right to take legal action against any persons unlawfully gaining access to the content of any external message it has emitted.* *For additional information, please visit our website **www.presenceco.com * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161108/626ea303/attachment-0001.html From aubalde at presenceco.com Tue Nov 8 13:20:06 2016 From: aubalde at presenceco.com (=?UTF-8?Q?Agust=C3=AD_Ubalde?=) Date: Tue, 8 Nov 2016 11:20:06 +0100 Subject: [Freeswitch-users] FreeSWICTH console not works on CentOS 7 Message-ID: Hi all, I've just installed FreeSWITCH 1.6 on CentOS 7. Service is running, but I can`t connect with *fs_cli*. Any suggestion? Thanks, *PRESENCE TECHNOLOGY* An *ENGHOUSE INTERACTIVE* Company *Agust? Ubalde Bellot* Chief Developer C/ Comte Urgell 240 3A Barcelona 08036 *aubalde at presenceco.com * Ph: +34 93 10 10 300 <%2B34%20931%20010%20300> Fax: +34 93 10 10 333 <%2B34%20931%20010%20333> Enghouse is listed on the Toronto Stock Exchange, stock symbol: ENGH -- *Presence Technology - DisclaimerThis message, its content and any file attached thereto is for the intended recipient only and is confidential and /or privileged. If you have received this e-mail in error or had access to it, you should note that the information in it is private and any use thereof is unauthorized. In such an event please notify us by e-mail or by telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by whatsoever means and any transmission or dissemination thereof to other persons is prohibited. It should be deleted immediately from your system. Presence Technology reserves the right to take legal action against any persons unlawfully gaining access to the content of any external message it has emitted.* *For additional information, please visit our website **www.presenceco.com * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161108/9ce03790/attachment.html From david.villasmil.work at gmail.com Tue Nov 8 14:03:51 2016 From: david.villasmil.work at gmail.com (David Villasmil) Date: Tue, 8 Nov 2016 06:03:51 -0500 Subject: [Freeswitch-users] FreeSWICTH console not works on CentOS 7 In-Reply-To: References: Message-ID: Hello, Is freeswitch listening on 8021 or maybe you changed the port? could you paste the error? David david.villasmil.work at gmail.com +34669448337 ? On Tue, Nov 8, 2016 at 5:20 AM, Agust? Ubalde wrote: > Hi all, > > I've just installed FreeSWITCH 1.6 on CentOS 7. > Service is running, but I can`t connect with *fs_cli*. > > Any suggestion? > > > Thanks, > > *PRESENCE TECHNOLOGY* > > An *ENGHOUSE INTERACTIVE* Company > > *Agust? Ubalde Bellot* > > Chief Developer > > C/ Comte Urgell 240 3A > > Barcelona 08036 > > *aubalde at presenceco.com * > > Ph: +34 93 10 10 300 <%2B34%20931%20010%20300> > > Fax: +34 93 10 10 333 <%2B34%20931%20010%20333> > > Enghouse is listed on the Toronto Stock Exchange, stock symbol: ENGH > > > > *Presence Technology - DisclaimerThis message, its content and any file > attached thereto is for the intended recipient only and is confidential and > /or privileged. If you have received this e-mail in error or had access to > it, you should note that the information in it is private and any use > thereof is unauthorized. In such an event please notify us by e-mail or by > telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by > whatsoever means and any transmission or dissemination thereof to other > persons is prohibited. It should be deleted immediately from your system. > Presence Technology reserves the right to take legal action against any > persons unlawfully gaining access to the content of any external message it > has emitted.* > > *For additional information, please visit our website **www.presenceco.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/20161108/a4a9a817/attachment.html From aubalde at presenceco.com Tue Nov 8 14:09:00 2016 From: aubalde at presenceco.com (=?UTF-8?Q?Agust=C3=AD_Ubalde?=) Date: Tue, 8 Nov 2016 12:09:00 +0100 Subject: [Freeswitch-users] FreeSWICTH console not works on CentOS 7 In-Reply-To: References: Message-ID: Hi David, I've solved the issue changing the* event_socket.xml* configuration: *- + * Thanks, *PRESENCE TECHNOLOGY* An *ENGHOUSE INTERACTIVE* Company *Agust? Ubalde Bellot* Chief Developer C/ Comte Urgell 240 3A Barcelona 08036 aubalde at presenceco.com Ph: +34 93 10 10 300 <%2B34%20931%20010%20300> Fax: +34 93 10 10 333 <%2B34%20931%20010%20333> Enghouse is listed on the Toronto Stock Exchange, stock symbol: ENGH 2016-11-08 12:03 GMT+01:00 David Villasmil : > Hello, > > Is freeswitch listening on 8021 or maybe you changed the port? > could you paste the error? > > David > > > david.villasmil.work at gmail.com > +34669448337 > ? > > On Tue, Nov 8, 2016 at 5:20 AM, Agust? Ubalde > wrote: > >> Hi all, >> >> I've just installed FreeSWITCH 1.6 on CentOS 7. >> Service is running, but I can`t connect with *fs_cli*. >> >> Any suggestion? >> >> >> Thanks, >> >> *PRESENCE TECHNOLOGY* >> >> An *ENGHOUSE INTERACTIVE* Company >> >> *Agust? Ubalde Bellot* >> >> Chief Developer >> >> C/ Comte Urgell 240 3A >> >> Barcelona 08036 >> >> *aubalde at presenceco.com * >> >> Ph: +34 93 10 10 300 <%2B34%20931%20010%20300> >> >> Fax: +34 93 10 10 333 <%2B34%20931%20010%20333> >> >> Enghouse is listed on the Toronto Stock Exchange, stock symbol: ENGH >> >> >> >> *Presence Technology - DisclaimerThis message, its content and any file >> attached thereto is for the intended recipient only and is confidential and >> /or privileged. If you have received this e-mail in error or had access to >> it, you should note that the information in it is private and any use >> thereof is unauthorized. In such an event please notify us by e-mail or by >> telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by >> whatsoever means and any transmission or dissemination thereof to other >> persons is prohibited. It should be deleted immediately from your system. >> Presence Technology reserves the right to take legal action against any >> persons unlawfully gaining access to the content of any external message it >> has emitted.* >> >> *For additional information, please visit our website **www.presenceco.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 > -- *Presence Technology - DisclaimerThis message, its content and any file attached thereto is for the intended recipient only and is confidential and /or privileged. If you have received this e-mail in error or had access to it, you should note that the information in it is private and any use thereof is unauthorized. In such an event please notify us by e-mail or by telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by whatsoever means and any transmission or dissemination thereof to other persons is prohibited. It should be deleted immediately from your system. Presence Technology reserves the right to take legal action against any persons unlawfully gaining access to the content of any external message it has emitted.* *For additional information, please visit our website **www.presenceco.com * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161108/fb6d918f/attachment-0001.html From david.villasmil.work at gmail.com Tue Nov 8 14:10:06 2016 From: david.villasmil.work at gmail.com (David Villasmil) Date: Tue, 8 Nov 2016 06:10:06 -0500 Subject: [Freeswitch-users] FreeSWICTH console not works on CentOS 7 In-Reply-To: References: Message-ID: good, have fun! ? Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 On Tue, Nov 8, 2016 at 6:09 AM, Agust? Ubalde wrote: > Hi David, > > I've solved the issue changing the* event_socket.xml* configuration: > > > > *- + value="127.0.0.1"/>* > > Thanks, > > *PRESENCE TECHNOLOGY* > > An *ENGHOUSE INTERACTIVE* Company > > *Agust? Ubalde Bellot* > > Chief Developer > > C/ Comte Urgell 240 3A > > Barcelona 08036 > > aubalde at presenceco.com > > Ph: +34 93 10 10 300 <%2B34%20931%20010%20300> > > Fax: +34 93 10 10 333 <%2B34%20931%20010%20333> > > Enghouse is listed on the Toronto Stock Exchange, stock symbol: ENGH > > > 2016-11-08 12:03 GMT+01:00 David Villasmil >: > >> Hello, >> >> Is freeswitch listening on 8021 or maybe you changed the port? >> could you paste the error? >> >> David >> >> >> david.villasmil.work at gmail.com >> +34669448337 >> ? >> >> On Tue, Nov 8, 2016 at 5:20 AM, Agust? Ubalde >> wrote: >> >>> Hi all, >>> >>> I've just installed FreeSWITCH 1.6 on CentOS 7. >>> Service is running, but I can`t connect with *fs_cli*. >>> >>> Any suggestion? >>> >>> >>> Thanks, >>> >>> *PRESENCE TECHNOLOGY* >>> >>> An *ENGHOUSE INTERACTIVE* Company >>> >>> *Agust? Ubalde Bellot* >>> >>> Chief Developer >>> >>> C/ Comte Urgell 240 3A >>> >>> Barcelona 08036 >>> >>> *aubalde at presenceco.com * >>> >>> Ph: +34 93 10 10 300 <%2B34%20931%20010%20300> >>> >>> Fax: +34 93 10 10 333 <%2B34%20931%20010%20333> >>> >>> Enghouse is listed on the Toronto Stock Exchange, stock symbol: ENGH >>> >>> >>> >>> *Presence Technology - DisclaimerThis message, its content and any file >>> attached thereto is for the intended recipient only and is confidential and >>> /or privileged. If you have received this e-mail in error or had access to >>> it, you should note that the information in it is private and any use >>> thereof is unauthorized. In such an event please notify us by e-mail or by >>> telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by >>> whatsoever means and any transmission or dissemination thereof to other >>> persons is prohibited. It should be deleted immediately from your system. >>> Presence Technology reserves the right to take legal action against any >>> persons unlawfully gaining access to the content of any external message it >>> has emitted.* >>> >>> *For additional information, please visit our website **www.presenceco.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 >> > > > > *Presence Technology - DisclaimerThis message, its content and any file > attached thereto is for the intended recipient only and is confidential and > /or privileged. If you have received this e-mail in error or had access to > it, you should note that the information in it is private and any use > thereof is unauthorized. In such an event please notify us by e-mail or by > telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by > whatsoever means and any transmission or dissemination thereof to other > persons is prohibited. It should be deleted immediately from your system. > Presence Technology reserves the right to take legal action against any > persons unlawfully gaining access to the content of any external message it > has emitted.* > > *For additional information, please visit our website **www.presenceco.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/20161108/a3c3e808/attachment-0001.html From italo at freeswitch.org Tue Nov 8 15:15:02 2016 From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=) Date: Tue, 8 Nov 2016 09:15:02 -0300 Subject: [Freeswitch-users] Issue with cc_warning_tone in mod_callcenter module In-Reply-To: References: Message-ID: Fila a JIRA with your configs and debug logs On Tue, Nov 8, 2016 at 3:49 AM, Samir Doshi wrote: > Hi, > > I tried with cc_outbound_announce variable but getting same result. Not > getting alert tone. > Any other things should I look? > > > Best Regards > -- > Samir Doshi > *iNextrix Technologie**s Pvt. Ltd*. > http://www.inextrix.com > > > > *Disclaimer:* > The information contained in this communication is confidential and may be > legally privileged. It is intended solely for the use of the individual or > entity to whom it is addressed and others authorised to receive it. If you > are not the intended recipient you are hereby notified that any disclosure, > copying, distribution or taking action in reliance of the contents of this > information is strictly prohibited and may be unlawful. > > On Sun, Nov 6, 2016 at 3:30 PM, Samir Doshi > wrote: > >> Thanks. I will give a try and get back to you with result. >> >> >> Best Regards >> -- >> Samir Doshi >> *iNextrix Technologie**s Pvt. Ltd*. >> http://www.inextrix.com >> >> >> >> *Disclaimer:* >> The information contained in this communication is confidential and may >> be legally privileged. It is intended solely for the use of the individual >> or entity to whom it is addressed and others authorised to receive it. If >> you are not the intended recipient you are hereby notified that any >> disclosure, copying, distribution or taking action in reliance of the >> contents of this information is strictly prohibited and may be unlawful. >> >> On Sat, Nov 5, 2016 at 1:31 AM, ?talo Rossi wrote: >> >>> mod_callcenter.c:1776 Agent 8003 answered "Outbound Call" >>> from queue agent8003 at default (Recorded) >>> >>> From this line you're using callback agents and the variable should be >>> cc_outbound_announce instead of cc_warning_tone, this one is specific for >>> uuid-standby agents >>> >>> On Fri, Nov 4, 2016 at 3:36 PM, Samir Doshi >>> wrote: >>> >>>> Hi, >>>> >>>> We are doing impementation using mod_callcenter module by taking >>>> reference from documentation (https://freeswitch.org/conflu >>>> ence/display/FREESWITCH/mod_callcenter). So far everything works well >>>> except cc_warning_tone playback. We are never getting tone when some >>>> connect to agent. >>>> >>>> Debug Logs : >>>> >>>> FS cli Log : https://pastebin.freeswitch.org/view/f9be26b9 >>>> Log with sip debug : https://pastebin.freeswitch.org/view/aab8ea85 >>>> Log with sip debug + loglevel 9 : https://pastebin.freeswitch.or >>>> g/view/ff793521 >>>> >>>> Any hint? >>>> >>>> >>>> Best Regards >>>> -- >>>> Samir Doshi >>>> *iNextrix Technologie**s Pvt. Ltd*. >>>> http://www.inextrix.com >>>> >>>> >>>> >>>> *Disclaimer:* >>>> The information contained in this communication is confidential and may >>>> be legally privileged. It is intended solely for the use of the individual >>>> or entity to whom it is addressed and others authorised to receive it. If >>>> you are not the intended recipient you are hereby notified that any >>>> disclosure, copying, distribution or taking action in reliance of the >>>> contents of this information is strictly prohibited and may be unlawful. >>>> >>>> ____________________________________________________________ >>>> _____________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>> switch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> >>> -- >>> ?talo Rossi >>> italo at freeswitch.org >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> 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 italo at freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161108/d4e3d6fe/attachment.html From danny.vandenberg at nepworldwide.nl Tue Nov 8 09:24:02 2016 From: danny.vandenberg at nepworldwide.nl (Danny van den Berg - NEP Worldwide Network) Date: Tue, 8 Nov 2016 07:24:02 +0100 Subject: [Freeswitch-users] Conference DID Testing help In-Reply-To: References: Message-ID: Hi, DID of The Netherlands is working. 2016-11-07 18:44 GMT+01:00 Kathleen King : > Hello, > > We are doing some testing and want to make sure the DIDs to call into the > public 888 conference are working. The team has been able to test the US > number, but we need your help to test the international DIDs. Please take a > moment and call to see if the DID for your region is working and reply to > this email to let us know. Thank-you! > > > > Australia +61-7-3188-7519 Thanks to Jay Binks - NetSIP.com.au > > Canada +1-438-800-0531 Thanks to NG Communications > > France +33-975-181-606 Thanks to NG Communications > > Germany +49-228-9293-9009 Thanks to Yiftach at ChooChee > Germany +49-2373-913-4009 Thanks to einfachVoIP.de > > Ireland +353-1-687-9001 Thanks to Ziron > Israel +972-2-372-0394 Thanks to Avi Marcus - BestFone.com > > Netherlands +31-858-880-387 Thanks to NG Communications > > New Zealand +64-4-887-1401 Thanks to Ziron > Portugal +351-300505224 Thanks to Finesource > > South Africa +27-87-8204656 Thanks to Othos Telecom > > Spain +34-91-290-12-71 Thanks to SIPtize > UK +44-330-320-0105 Thanks to Ziron > UK +44-1904-201-313 Thanks to ukddi.com (Routed > by Steven Ayre) > UK +44-203-298-5931 Thanks to ukddi.com (Routed > by Avi Marcus) > USA +1-919-386-9900 > > > > Kathleen King > FreeSWITCH Public Relations > Office: +1-213-286-0400 > Mobile: +1-703-859-3757 > http://freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Danny van den Berg* SysOps | NEP The Netherlands T: +31 35 67 14 650 | M: +31 6 51 11 16 74 danny.vandenberg at nepworldwide.nl www.nepworldwide.nl *Superior Service & Lasting Relationships* Integrity | One Team | Innovative | Passion CONFIDENTIALITY NOTICE & DISCLAIMER: this e-mail may contain confidential, proprietary, or privileged information. If you are not the intended recipient please contact the sender by reply e-mail and destroy all copies of the original message. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161108/b5efc5de/attachment-0001.html From Joaquin.Alzola at lebara.com Tue Nov 8 13:34:41 2016 From: Joaquin.Alzola at lebara.com (Joaquin Alzola) Date: Tue, 8 Nov 2016 10:34:41 +0000 Subject: [Freeswitch-users] FreeSWICTH console not works on CentOS 7 In-Reply-To: References: Message-ID: >I've just installed FreeSWITCH 1.6 on CentOS 7. >Service is running, but I can`t connect with fs_cli. What error do you get and what did you try to run? https://freeswitch.org/confluence/display/FREESWITCH/Command-Line+Interface+fs_cli Joaquin --- This email is confidential and may be subject to privilege. If you are not the intended recipient, please do not copy or disclose its content but contact the sender immediately upon receipt. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161108/2d13bd2b/attachment.html From kiranravuri04 at gmail.com Tue Nov 8 12:02:53 2016 From: kiranravuri04 at gmail.com (Kiran Ravuri) Date: Tue, 8 Nov 2016 14:32:53 +0530 Subject: [Freeswitch-users] Conference URI Message-ID: Hi All, I could able to create a conference room and dial out from room to participants using ESL. Where can I find the sip URI of the conference room, which is dynamically created from ESL, so that I can give that to the pariticipants who want to dial in to the room ? BRs Kiran -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161108/e749a3a6/attachment.html From nbhatti at gmail.com Tue Nov 8 17:03:00 2016 From: nbhatti at gmail.com (Muhammad Naseer Bhatti) Date: Tue, 8 Nov 2016 09:03:00 -0500 Subject: [Freeswitch-users] Default connection pool size for ODBC Message-ID: Hi, what?s the default connection pool size for ODBC/Postgres is defined? -- Sent with Airmail -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161108/67da2924/attachment.html From brian at freeswitch.org Tue Nov 8 17:35:49 2016 From: brian at freeswitch.org (Brian West) Date: Tue, 8 Nov 2016 08:35:49 -0600 Subject: [Freeswitch-users] Conference URI In-Reply-To: References: Message-ID: What exactly are you doing currently? The conference rooms are usually specified in the dialplan when you first join. /b On Tue, Nov 8, 2016 at 3:02 AM, Kiran Ravuri wrote: > Hi All, > > I could able to create a conference room and dial out from room to > participants using ESL. > Where can I find the sip URI of the conference room, which is dynamically > created from ESL, so that I can give that to the pariticipants who want to > dial in to the room ? > > BRs > Kiran > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 (50% Discount using code FreeSwitch50) http://www.freeswitchcookbook.com (50% Discount using code FreeSwitch50) https://www.gofundme.com/freeswitch_ubuntu Got Bugs? Report them here ! | Reddit: /r/freeswitch *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/20161108/295a73a4/attachment.html From aubalde at presenceco.com Tue Nov 8 17:45:20 2016 From: aubalde at presenceco.com (=?UTF-8?Q?Agust=C3=AD_Ubalde?=) Date: Tue, 8 Nov 2016 15:45:20 +0100 Subject: [Freeswitch-users] FreeSWICTH console not works on CentOS 7 In-Reply-To: References: Message-ID: Hi Joaquin, I get timeout connection because the event_socket has been configured ipv6 and I've disabled the ipv6 configuration on the server. Thanks, *PRESENCE TECHNOLOGY* An *ENGHOUSE INTERACTIVE* Company *Agust? Ubalde Bellot* Chief Developer C/ Comte Urgell 240 3A Barcelona 08036 aubalde at presenceco.com Ph: +34 93 10 10 300 <%2B34%20931%20010%20300> Fax: +34 93 10 10 333 <%2B34%20931%20010%20333> Enghouse is listed on the Toronto Stock Exchange, stock symbol: ENGH 2016-11-08 11:34 GMT+01:00 Joaquin Alzola : > > > >I've just installed FreeSWITCH 1.6 on CentOS 7. > > >Service is running, but I can`t connect with *fs_cli*. > > > > What error do you get and what did you try to run? > > https://freeswitch.org/confluence/display/FREESWITCH/ > Command-Line+Interface+fs_cli > > > > > > Joaquin > > --- > This email is confidential and may be subject to privilege. If you are not > the intended recipient, please do not copy or disclose its content but > contact the sender immediately upon receipt. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Presence Technology - DisclaimerThis message, its content and any file attached thereto is for the intended recipient only and is confidential and /or privileged. If you have received this e-mail in error or had access to it, you should note that the information in it is private and any use thereof is unauthorized. In such an event please notify us by e-mail or by telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by whatsoever means and any transmission or dissemination thereof to other persons is prohibited. It should be deleted immediately from your system. Presence Technology reserves the right to take legal action against any persons unlawfully gaining access to the content of any external message it has emitted.* *For additional information, please visit our website **www.presenceco.com * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161108/b31d5bb2/attachment-0001.html From aubalde at presenceco.com Tue Nov 8 18:21:50 2016 From: aubalde at presenceco.com (=?UTF-8?Q?Agust=C3=AD_Ubalde?=) Date: Tue, 8 Nov 2016 16:21:50 +0100 Subject: [Freeswitch-users] One way audio Message-ID: Hi all, I'm testing 2 video extensions trough FreeSWITCH but I only see the video at the caller browser. Called browser is receiving UDP packets but not show the video. Any sugestion? I'm using *FreesSWITCH 1.6.12* on *CentOS 7*. Thanks, *PRESENCE TECHNOLOGY* An *ENGHOUSE INTERACTIVE* Company *Agust? Ubalde Bellot* Chief Developer C/ Comte Urgell 240 3?-A Barcelona 08036 aubalde at presenceco.com Ph: +34 93 10 10 322 Enghouse is listed on the Toronto Stock Exchange, stock symbol: ENGH -- *Presence Technology - DisclaimerThis message, its content and any file attached thereto is for the intended recipient only and is confidential and /or privileged. If you have received this e-mail in error or had access to it, you should note that the information in it is private and any use thereof is unauthorized. In such an event please notify us by e-mail or by telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by whatsoever means and any transmission or dissemination thereof to other persons is prohibited. It should be deleted immediately from your system. Presence Technology reserves the right to take legal action against any persons unlawfully gaining access to the content of any external message it has emitted.* *For additional information, please visit our website **www.presenceco.com * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161108/715b66c9/attachment.html From krice at freeswitch.org Tue Nov 8 18:29:05 2016 From: krice at freeswitch.org (Ken Rice) Date: Tue, 8 Nov 2016 09:29:05 -0600 Subject: [Freeswitch-users] One way audio In-Reply-To: References: Message-ID: <300f01d239d4$d84e8440$88eb8cc0$@freeswitch.org> You?re probably missing some of the deps for like h264 that centos doesn?t provide From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Agust? Ubalde Sent: Tuesday, November 8, 2016 9:22 AM To: FreeSWITCH Users Help Subject: [Freeswitch-users] One way audio Hi all, I'm testing 2 video extensions trough FreeSWITCH but I only see the video at the caller browser. Called browser is receiving UDP packets but not show the video. Any sugestion? I'm using FreesSWITCH 1.6.12 on CentOS 7. Thanks, PRESENCE TECHNOLOGY An ENGHOUSE INTERACTIVE Company Agust? Ubalde Bellot Chief Developer C/ Comte Urgell 240 3?-A Barcelona 08036 aubalde at presenceco.com Ph: +34 93 10 10 322 Enghouse is listed on the Toronto Stock Exchange, stock symbol: ENGH Presence Technology - Disclaimer This message, its content and any file attached thereto is for the intended recipient only and is confidential and /or privileged. If you have received this e-mail in error or had access to it, you should note that the information in it is private and any use thereof is unauthorized. In such an event please notify us by e-mail or by telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by whatsoever means and any transmission or dissemination thereof to other persons is prohibited. It should be deleted immediately from your system. Presence Technology reserves the right to take legal action against any persons unlawfully gaining access to the content of any external message it has emitted. For additional information, please visit our website www.presenceco.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161108/28c8246c/attachment.html From fanx07 at gmail.com Tue Nov 8 18:31:01 2016 From: fanx07 at gmail.com (Anonim Stefan) Date: Tue, 8 Nov 2016 17:31:01 +0200 Subject: [Freeswitch-users] mod_shout .mp4 Message-ID: Hi, I am trying to use mod_shout to play back the file [1]. I am using freeswitch version [2]. While doing this am getting error [3] and call eventually ends after ~ 10 secons with the error: "2016-11-08 13:31:45.946773 [ERR] mod_shout.c:437 Decoder Error! http://bc05.ajmn.me/665003303001/201605/19/665003303001_4906383897001_20160522-PG014904-V01-1-WIT-5818765.mp4 " I am able to playback the stream using mod_vlc and vlc://http://LINK. Any ideas what might be causing this Decoder Error and how can I fix it? Thank you, Stefan [1] shout:// bc05.ajmn.me/665003303001/201605/19/665003303001_4906383897001_20160522-PG014904-V01-1-WIT-5818765.mp4 [2] FreeSWITCH version: 1.6.12-20-b91a0a6~64bit (-20-b91a0a6 64bit) [3] http://pastebin.com/YRsnVu70 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161108/8f45b175/attachment.html From aubalde at presenceco.com Tue Nov 8 18:38:28 2016 From: aubalde at presenceco.com (=?UTF-8?Q?Agust=C3=AD_Ubalde?=) Date: Tue, 8 Nov 2016 16:38:28 +0100 Subject: [Freeswitch-users] One way audio In-Reply-To: <300f01d239d4$d84e8440$88eb8cc0$@freeswitch.org> References: <300f01d239d4$d84e8440$88eb8cc0$@freeswitch.org> Message-ID: Hi Ken, I just look the codec list and I don't see VP8. How can I enable this codec? H264 is listed as well. Thanks, *PRESENCE TECHNOLOGY* An *ENGHOUSE INTERACTIVE* Company *Agust? Ubalde Bellot* Chief Developer C/ Comte Urgell 240 3?-A Barcelona 08036 aubalde at presenceco.com Ph: +34 93 10 10 322 Enghouse is listed on the Toronto Stock Exchange, stock symbol: ENGH 2016-11-08 16:29 GMT+01:00 Ken Rice : > You?re probably missing some of the deps for like h264 that centos doesn?t > provide > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Agust? > Ubalde > *Sent:* Tuesday, November 8, 2016 9:22 AM > *To:* FreeSWITCH Users Help > *Subject:* [Freeswitch-users] One way audio > > > > Hi all, > > > > I'm testing 2 video extensions trough FreeSWITCH but I only see the video > at the caller browser. > > Called browser is receiving UDP packets but not show the video. > > > > Any sugestion? I'm using *FreesSWITCH 1.6.12* on *CentOS 7*. > > > > > > Thanks, > > *PRESENCE TECHNOLOGY* > An *ENGHOUSE INTERACTIVE* Company > > *Agust? Ubalde Bellot* > Chief Developer > C/ Comte Urgell 240 3?-A > Barcelona 08036 > aubalde at presenceco.com > Ph: +34 93 10 10 322 > Enghouse is listed on the Toronto Stock Exchange, stock symbol: ENGH > > > > > *Presence Technology - DisclaimerThis message, its content and any file > attached thereto is for the intended recipient only and is confidential and > /or privileged. If you have received this e-mail in error or had access to > it, you should note that the information in it is private and any use > thereof is unauthorized. In such an event please notify us by e-mail or by > telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by > whatsoever means and any transmission or dissemination thereof to other > persons is prohibited. It should be deleted immediately from your system. > Presence Technology reserves the right to take legal action against any > persons unlawfully gaining access to the content of any external message it > has emitted.* > > *For additional information, please visit our website **www.presenceco.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 > -- *Presence Technology - DisclaimerThis message, its content and any file attached thereto is for the intended recipient only and is confidential and /or privileged. If you have received this e-mail in error or had access to it, you should note that the information in it is private and any use thereof is unauthorized. In such an event please notify us by e-mail or by telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by whatsoever means and any transmission or dissemination thereof to other persons is prohibited. It should be deleted immediately from your system. Presence Technology reserves the right to take legal action against any persons unlawfully gaining access to the content of any external message it has emitted.* *For additional information, please visit our website **www.presenceco.com * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161108/e662b6fc/attachment-0001.html From krice at freeswitch.org Tue Nov 8 18:39:19 2016 From: krice at freeswitch.org (Ken Rice) Date: Tue, 8 Nov 2016 09:39:19 -0600 Subject: [Freeswitch-users] mod_shout .mp4 In-Reply-To: References: Message-ID: <302601d239d6$460a2420$d21e6c60$@freeswitch.org> Not shout only does MP3s? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anonim Stefan Sent: Tuesday, November 8, 2016 9:31 AM To: Freeswitch-users Subject: [Freeswitch-users] mod_shout .mp4 Hi, I am trying to use mod_shout to play back the file [1]. I am using freeswitch version [2]. While doing this am getting error [3] and call eventually ends after ~ 10 secons with the error: "2016-11-08 13:31:45.946773 [ERR] mod_shout.c:437 Decoder Error! http://bc05.ajmn.me/665003303001/201605/19/665003303001_4906383897001_20160522-PG014904-V01-1-WIT-5818765.mp4" I am able to playback the stream using mod_vlc and vlc://http://LINK. Any ideas what might be causing this Decoder Error and how can I fix it? Thank you, Stefan [1] shout://bc05.ajmn.me/665003303001/201605/19/665003303001_4906383897001_20160522-PG014904-V01-1-WIT-5818765.mp4 [2] FreeSWITCH version: 1.6.12-20-b91a0a6~64bit (-20-b91a0a6 64bit) [3] http://pastebin.com/YRsnVu70 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161108/874a158e/attachment.html From krice at freeswitch.org Tue Nov 8 19:12:52 2016 From: krice at freeswitch.org (Ken Rice) Date: Tue, 8 Nov 2016 10:12:52 -0600 Subject: [Freeswitch-users] One way audio In-Reply-To: References: <300f01d239d4$d84e8440$88eb8cc0$@freeswitch.org> Message-ID: <306401d239da$f5d53490$e17f9db0$@freeswitch.org> Vp8 is built in to the core, and should be enabled in the default configs. H264 may be enabled in the default configs, however centos is missing the required libs in their RPM repos and they do not publish them. You?ll have to look and see what Debian does for this and figure out how to get them imported into your system as FreeSWITCH does not ship a working H264 transcoding module needed for video conferencing with H264. FreeSWITCH depends on the distros to do this. The most painless way to get this all working is on Debian. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Agust? Ubalde Sent: Tuesday, November 8, 2016 9:38 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] One way audio Hi Ken, I just look the codec list and I don't see VP8. How can I enable this codec? H264 is listed as well. Thanks, PRESENCE TECHNOLOGY An ENGHOUSE INTERACTIVE Company Agust? Ubalde Bellot Chief Developer C/ Comte Urgell 240 3?-A Barcelona 08036 aubalde at presenceco.com Ph: +34 93 10 10 322 Enghouse is listed on the Toronto Stock Exchange, stock symbol: ENGH 2016-11-08 16:29 GMT+01:00 Ken Rice >: You?re probably missing some of the deps for like h264 that centos doesn?t provide From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org ] On Behalf Of Agust? Ubalde Sent: Tuesday, November 8, 2016 9:22 AM To: FreeSWITCH Users Help > Subject: [Freeswitch-users] One way audio Hi all, I'm testing 2 video extensions trough FreeSWITCH but I only see the video at the caller browser. Called browser is receiving UDP packets but not show the video. Any sugestion? I'm using FreesSWITCH 1.6.12 on CentOS 7. Thanks, PRESENCE TECHNOLOGY An ENGHOUSE INTERACTIVE Company Agust? Ubalde Bellot Chief Developer C/ Comte Urgell 240 3?-A Barcelona 08036 aubalde at presenceco.com Ph: +34 93 10 10 322 Enghouse is listed on the Toronto Stock Exchange, stock symbol: ENGH Presence Technology - Disclaimer This message, its content and any file attached thereto is for the intended recipient only and is confidential and /or privileged. If you have received this e-mail in error or had access to it, you should note that the information in it is private and any use thereof is unauthorized. In such an event please notify us by e-mail or by telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by whatsoever means and any transmission or dissemination thereof to other persons is prohibited. It should be deleted immediately from your system. Presence Technology reserves the right to take legal action against any persons unlawfully gaining access to the content of any external message it has emitted. For additional information, please visit our website www.presenceco.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 Presence Technology - Disclaimer This message, its content and any file attached thereto is for the intended recipient only and is confidential and /or privileged. If you have received this e-mail in error or had access to it, you should note that the information in it is private and any use thereof is unauthorized. In such an event please notify us by e-mail or by telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by whatsoever means and any transmission or dissemination thereof to other persons is prohibited. It should be deleted immediately from your system. Presence Technology reserves the right to take legal action against any persons unlawfully gaining access to the content of any external message it has emitted. For additional information, please visit our website www.presenceco.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161108/b111ef10/attachment-0001.html From lists at kavun.ch Wed Nov 9 01:02:49 2016 From: lists at kavun.ch (Emrah) Date: Tue, 8 Nov 2016 23:02:49 +0100 Subject: [Freeswitch-users] FS-9113, still experiencing TLS crashes Message-ID: <9D820B3E-294A-49C4-9220-4233FE6994B0@kavun.ch> Hello List, Thanks to the help provided by Stanislav, I learned of issue #9113, https://freeswitch.org/jira/si/jira.issueviews:issue-html/FS-9113/FS-9113.html, which seems to be related to the issues I have been experiencing with FreeSWITCH, TLS and failed call setups. Coincidentally, or not, the fix pushed on that issue was aligned with whole months where I did not experience any TLS issues. Calls were going through fine, until all of a sudden they started failing again. This is on 2 distinct servers running a load balanced FS setup, and using Yealink phones. To sum up, here is what is going on. From the Yealink, calls with TLS work if I don't use SRTP. From the Yealink, calls crash if I use TLS and SRTP. From my laptop softphone, calls only crash sometimes if I use TLS and SRTP. How can I debug the TLS session on the FreeSWITCH side to see what happens with the TLS thread? I don't mean packet capture. I have a feeling that the packet size is too large and doesn't make it to the FS box intact after the 407 Proxy Required is received by the client. Here is the log for the Yealink: http://pastebin.com/smKP286x Your lights would be so appreciated, I'm losing my mind over this. Best, Emrah -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161108/d7502bf8/attachment.html From gregor at infomedia.si Wed Nov 9 01:34:51 2016 From: gregor at infomedia.si (Gregor Nanger) Date: Tue, 8 Nov 2016 23:34:51 +0100 Subject: [Freeswitch-users] Windows build - mod_lua Message-ID: I've compiled FS with visual studio and there is no mod_lua.dll generated. Hence I get error when FS starts that it can not find dll. Windows build went fine (with excluded modv8) Anyone also find this problem? Best regards, Gregor -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161108/dc26bce0/attachment.html From gregor at infomedia.si Wed Nov 9 01:36:04 2016 From: gregor at infomedia.si (Gregor Nanger) Date: Tue, 8 Nov 2016 23:36:04 +0100 Subject: [Freeswitch-users] bind_digit_action advice In-Reply-To: References: Message-ID: yes, this works fine...missed inline in the end :-( 2016-11-04 21:54 GMT+01:00 Gregor Nanger : > Would need som advice. > > Just started to implementing bind_digit_action where caller can press 1 to > transfer call during active call. > > Since I am using xml curl, I would like to minimize trip to server to get > dialplan actions and make action inline not execute_extension. I would like > to bridge call to user with extension 001, but having problem with syntax. > > Is this correct way? > > > ? > -- Gregor Nanger *CTO* t./f.: 00386 (0) 7 6000 308/309 ? m:. 00386 (0)41 756485 ? Infomedia d.o.o. ? Jerebova 3, Novo mesto, Slovenia ? www.infomedia.si -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161108/ea084a06/attachment.html From mike at jerris.com Wed Nov 9 01:41:46 2016 From: mike at jerris.com (Michael Jerris) Date: Tue, 8 Nov 2016 17:41:46 -0500 Subject: [Freeswitch-users] FS-9113, still experiencing TLS crashes In-Reply-To: <9D820B3E-294A-49C4-9220-4233FE6994B0@kavun.ch> References: <9D820B3E-294A-49C4-9220-4233FE6994B0@kavun.ch> Message-ID: Can you confirm if the packet is shown in freeswitch tport_log? > On Nov 8, 2016, at 5:02 PM, Emrah wrote: > > Hello List, > Thanks to the help provided by Stanislav, I learned of issue #9113, https://freeswitch.org/jira/si/jira.issueviews:issue-html/FS-9113/FS-9113.html , which seems to be related to the issues I have been experiencing with FreeSWITCH, TLS and failed call setups. > Coincidentally, or not, the fix pushed on that issue was aligned with whole months where I did not experience any TLS issues. Calls were going through fine, until all of a sudden they started failing again. This is on 2 distinct servers running a load balanced FS setup, and using Yealink phones. > > To sum up, here is what is going on. > From the Yealink, calls with TLS work if I don't use SRTP. > From the Yealink, calls crash if I use TLS and SRTP. > From my laptop softphone, calls only crash sometimes if I use TLS and SRTP. > > How can I debug the TLS session on the FreeSWITCH side to see what happens with the TLS thread? I don't mean packet capture. > > I have a feeling that the packet size is too large and doesn't make it to the FS box intact after the 407 Proxy Required is received by the client. > > Here is the log for the Yealink: > http://pastebin.com/smKP286x > > Your lights would be so appreciated, I'm losing my mind over this. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161108/80fb7f64/attachment.html From fanx07 at gmail.com Wed Nov 9 10:13:08 2016 From: fanx07 at gmail.com (Anonim Stefan) Date: Wed, 9 Nov 2016 09:13:08 +0200 Subject: [Freeswitch-users] mod_shout .mp4 In-Reply-To: <302601d239d6$460a2420$d21e6c60$@freeswitch.org> References: <302601d239d6$460a2420$d21e6c60$@freeswitch.org> Message-ID: Didn't know that. Thank you Ken. On Nov 8, 2016 5:40 PM, "Ken Rice" wrote: > Not shout only does MP3s? > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anonim > Stefan > *Sent:* Tuesday, November 8, 2016 9:31 AM > *To:* Freeswitch-users > *Subject:* [Freeswitch-users] mod_shout .mp4 > > > > Hi, > > > > I am trying to use mod_shout to play back the file [1]. I am using > freeswitch version [2]. While doing this am getting error [3] and call > eventually ends after ~ 10 secons with the error: > > > > "2016-11-08 13:31:45.946773 [ERR] mod_shout.c:437 Decoder Error! > http://bc05.ajmn.me/665003303001/201605/19/665003303001_4906383897001_ > 20160522-PG014904-V01-1-WIT-5818765.mp4" > > > > I am able to playback the stream using mod_vlc and vlc://http://LINK. > > > > Any ideas what might be causing this Decoder Error and how can I fix it? > > > > Thank you, > > Stefan > > > > [1] shout://bc05.ajmn.me/665003303001/201605/19/ > 665003303001_4906383897001_20160522-PG014904-V01-1-WIT-5818765.mp4 > > [2] FreeSWITCH version: 1.6.12-20-b91a0a6~64bit (-20-b91a0a6 64bit) > > [3] http://pastebin.com/YRsnVu70 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161109/0141d356/attachment-0001.html From gregor at infomedia.si Wed Nov 9 11:04:58 2016 From: gregor at infomedia.si (Gregor Nanger) Date: Wed, 9 Nov 2016 09:04:58 +0100 Subject: [Freeswitch-users] Unregister verto user Message-ID: ?I am using verto with web clients without problems. It performs very well. But have one question. Is it possible to unregister verto user? Is there some kind of cli commad like verto xmlstatus? Best regards, Gregor -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161109/819c8de1/attachment.html From lists at kavun.ch Wed Nov 9 11:39:20 2016 From: lists at kavun.ch (Emrah) Date: Wed, 9 Nov 2016 09:39:20 +0100 Subject: [Freeswitch-users] FS-9113, still experiencing TLS crashes In-Reply-To: References: <9D820B3E-294A-49C4-9220-4233FE6994B0@kavun.ch> Message-ID: No Sir, the response packet to the 407 Proxy Authentication Required is never received. So the session then eventually gets abandoned by FS. On the client side, and this is generalized, the packet is sent, except the TLS session breaks. > On Nov 8, 2016, at 11:41 PM, Michael Jerris wrote: > > Can you confirm if the packet is shown in freeswitch tport_log? > >> On Nov 8, 2016, at 5:02 PM, Emrah > wrote: >> >> Hello List, >> Thanks to the help provided by Stanislav, I learned of issue #9113, https://freeswitch.org/jira/si/jira.issueviews:issue-html/FS-9113/FS-9113.html , which seems to be related to the issues I have been experiencing with FreeSWITCH, TLS and failed call setups. >> Coincidentally, or not, the fix pushed on that issue was aligned with whole months where I did not experience any TLS issues. Calls were going through fine, until all of a sudden they started failing again. This is on 2 distinct servers running a load balanced FS setup, and using Yealink phones. >> >> To sum up, here is what is going on. >> From the Yealink, calls with TLS work if I don't use SRTP. >> From the Yealink, calls crash if I use TLS and SRTP. >> From my laptop softphone, calls only crash sometimes if I use TLS and SRTP. >> >> How can I debug the TLS session on the FreeSWITCH side to see what happens with the TLS thread? I don't mean packet capture. >> >> I have a feeling that the packet size is too large and doesn't make it to the FS box intact after the 407 Proxy Required is received by the client. >> >> Here is the log for the Yealink: >> http://pastebin.com/smKP286x >> >> Your lights would be so appreciated, I'm losing my mind over 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/20161109/3d8991f8/attachment.html From nbhatti at gmail.com Wed Nov 9 11:59:59 2016 From: nbhatti at gmail.com (Muhammad Naseer Bhatti) Date: Wed, 9 Nov 2016 03:59:59 -0500 Subject: [Freeswitch-users] Default connection pool size for ODBC In-Reply-To: References: Message-ID: Bump.. -- Sent with Airmail On November 8, 2016 at 5:03:00 PM, Muhammad Naseer Bhatti (nbhatti at gmail.com) wrote: > > Hi, what?s the default connection pool size for ODBC/Postgres is defined? > > > -- > > Sent with Airmail > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161109/caa7f9c0/attachment.html From fanx07 at gmail.com Wed Nov 9 14:24:38 2016 From: fanx07 at gmail.com (Anonim Stefan) Date: Wed, 9 Nov 2016 13:24:38 +0200 Subject: [Freeswitch-users] mod_shout .mp4 In-Reply-To: References: <302601d239d6$460a2420$d21e6c60$@freeswitch.org> Message-ID: Any advice about when/why should someone use mod_shout and not mod_vlc? (I see similar functionalities for both i.e. playback, recording) Thank you, Stefan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161109/4da066f2/attachment-0001.html From udy786 at gmail.com Wed Nov 9 15:32:01 2016 From: udy786 at gmail.com (Uday kumar) Date: Wed, 9 Nov 2016 18:02:01 +0530 Subject: [Freeswitch-users] Need Help on Freeswitch Fail-over (HA) Message-ID: Hi All, Need Help on Freeswitch Fail-over to avoid down time. I have installed two Freeswitch server on two different servers. Both server have web and mysql database. I am accessing my server using Domain name and also using domain name to register softphone. Is possible, if one server will down then automatically shift everything on other server? Right now domain pointed on server A. We are using cron to copy backup like mysql, recording, logs etc from server A to server B. As per my current setup, if server A will down then I need to change domain pointing IP to server B. If this is possible to do automatically then it will be good. Please advice. -- Thanks & Regard Uday Site:- www.shareyourknowledge.in Mobile:- +91-9377579349 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161109/9a4551c4/attachment.html From mike at jerris.com Wed Nov 9 18:25:28 2016 From: mike at jerris.com (Michael Jerris) Date: Wed, 9 Nov 2016 10:25:28 -0500 Subject: [Freeswitch-users] FS-9113, still experiencing TLS crashes In-Reply-To: References: <9D820B3E-294A-49C4-9220-4233FE6994B0@kavun.ch> Message-ID: I need a recipie to reliably reproduce this so I can dig in the code. Is there a way you can put together an environment where this can be reproduced on demand? > On Nov 9, 2016, at 3:39 AM, Emrah wrote: > > No Sir, the response packet to the 407 Proxy Authentication Required is never received. So the session then eventually gets abandoned by FS. On the client side, and this is generalized, the packet is sent, except the TLS session breaks. > >> On Nov 8, 2016, at 11:41 PM, Michael Jerris > wrote: >> >> Can you confirm if the packet is shown in freeswitch tport_log? >> >>> On Nov 8, 2016, at 5:02 PM, Emrah > wrote: >>> >>> Hello List, >>> Thanks to the help provided by Stanislav, I learned of issue #9113, https://freeswitch.org/jira/si/jira.issueviews:issue-html/FS-9113/FS-9113.html , which seems to be related to the issues I have been experiencing with FreeSWITCH, TLS and failed call setups. >>> Coincidentally, or not, the fix pushed on that issue was aligned with whole months where I did not experience any TLS issues. Calls were going through fine, until all of a sudden they started failing again. This is on 2 distinct servers running a load balanced FS setup, and using Yealink phones. >>> >>> To sum up, here is what is going on. >>> From the Yealink, calls with TLS work if I don't use SRTP. >>> From the Yealink, calls crash if I use TLS and SRTP. >>> From my laptop softphone, calls only crash sometimes if I use TLS and SRTP. >>> >>> How can I debug the TLS session on the FreeSWITCH side to see what happens with the TLS thread? I don't mean packet capture. >>> >>> I have a feeling that the packet size is too large and doesn't make it to the FS box intact after the 407 Proxy Required is received by the client. >>> >>> Here is the log for the Yealink: >>> http://pastebin.com/smKP286x >>> >>> Your lights would be so appreciated, I'm losing my mind over this. >> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161109/c052a7ef/attachment.html From mike at jerris.com Wed Nov 9 18:26:51 2016 From: mike at jerris.com (Michael Jerris) Date: Wed, 9 Nov 2016 10:26:51 -0500 Subject: [Freeswitch-users] Default connection pool size for ODBC In-Reply-To: References: Message-ID: <51DA090C-4B29-4E3B-B9F4-AE2B3DCAAB13@jerris.com> Freeswitch does not have a traditional connection pool with a size. What exactly are you referring to? > On Nov 8, 2016, at 9:03 AM, Muhammad Naseer Bhatti wrote: > > > Hi, what?s the default connection pool size for ODBC/Postgres is defined? > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161109/4e8846e7/attachment.html From luis.daniel.lucio at gmail.com Wed Nov 9 15:43:43 2016 From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz) Date: Wed, 9 Nov 2016 07:43:43 -0500 Subject: [Freeswitch-users] Need Help on Freeswitch Fail-over (HA) In-Reply-To: References: Message-ID: Servers in same data center or different ones? Have you google it? Le 9 nov. 2016 7:33 AM, "Uday kumar" a ?crit : > Hi All, > > Need Help on Freeswitch Fail-over to avoid down time. I have installed two > Freeswitch server on two different servers. Both server have web and mysql > database. I am accessing my server using Domain name and also using domain > name to register softphone. > > Is possible, if one server will down then automatically shift everything > on other server? Right now domain pointed on server A. We are using cron to > copy backup like mysql, recording, logs etc from server A to server B. As > per my current setup, if server A will down then I need to change domain > pointing IP to server B. If this is possible to do automatically then it > will be good. > > Please advice. > > > -- > Thanks & Regard > Uday > Site:- www.shareyourknowledge.in > Mobile:- +91-9377579349 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161109/9eb60a52/attachment.html From gonzalo at parzee.com Wed Nov 9 12:24:51 2016 From: gonzalo at parzee.com (Gonzalo Gasca Meza) Date: Wed, 09 Nov 2016 03:24:51 -0600 Subject: [Freeswitch-users] Adding users programatically Message-ID: <743a84e15598477f469f042553abd742@parzee.com> Hi forum, I want to add sip users with mailbox (right now under /etc/freeswitch/directory/default as xml files) using FS API. I would like to store user information in database (using pgsql 9.5). Found these posts: http://voicebundle.com/how-to-add-sip-user-in-freeswitch https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/scripts/perl/add_user http://voicebundle.com/creating-users-extensions-in-freeswitch-creating-sip-accounts-in-freeswitch-using-add_user-script But these are scripts which manipulate XML files. Looking to store information in database and have freeswitch read this information. (I can take care of adding user, delete user, update user, handle duplicates, etc.). Similar like this post: http://telecommusings.blogspot.com/2009/10/using-modxmlcurl-in-freeswitch-how-hard_28.html Thank you From mike at jerris.com Wed Nov 9 18:29:06 2016 From: mike at jerris.com (Michael Jerris) Date: Wed, 9 Nov 2016 10:29:06 -0500 Subject: [Freeswitch-users] Adding users programatically In-Reply-To: <743a84e15598477f469f042553abd742@parzee.com> References: <743a84e15598477f469f042553abd742@parzee.com> Message-ID: <9FF42CFE-B4A1-44E5-89D8-1C5684494B88@jerris.com> use mod_xml_curl > On Nov 9, 2016, at 4:24 AM, Gonzalo Gasca Meza wrote: > > Hi forum, > > I want to add sip users with mailbox (right now under > /etc/freeswitch/directory/default as xml files) using FS API. > I would like to store user information in database (using pgsql 9.5). > > Found these posts: > > http://voicebundle.com/how-to-add-sip-user-in-freeswitch > https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/scripts/perl/add_user > http://voicebundle.com/creating-users-extensions-in-freeswitch-creating-sip-accounts-in-freeswitch-using-add_user-script > > But these are scripts which manipulate XML files. Looking to store > information in database and have freeswitch read this information. (I > can take care of adding user, delete user, update user, handle > duplicates, etc.). > > Similar like this post: > http://telecommusings.blogspot.com/2009/10/using-modxmlcurl-in-freeswitch-how-hard_28.html > > Thank you From samir.doshi at inextrix.com Wed Nov 9 20:32:04 2016 From: samir.doshi at inextrix.com (Samir Doshi) Date: Wed, 9 Nov 2016 23:02:04 +0530 Subject: [Freeswitch-users] Export CDR on failed tried calls In-Reply-To: References: Message-ID: Any feedback please? Best Regards -- Samir Doshi *iNextrix Technologie**s Pvt. Ltd*. http://www.inextrix.com *Disclaimer:* The information contained in this communication is confidential and may be legally privileged. It is intended solely for the use of the individual or entity to whom it is addressed and others authorised to receive it. If you are not the intended recipient you are hereby notified that any disclosure, copying, distribution or taking action in reliance of the contents of this information is strictly prohibited and may be unlawful. On Sun, Nov 6, 2016 at 3:43 PM, Samir Doshi wrote: > Hi Guys, > > Wondering if we can post failed tried call cdr to mod_json_cdr. I have > below dialplan generated and I want to post cdr any gateway fail to process > the call. That means if test1 fail then it should send cdr to mod_json_cdr > and then go for test2. If test2 fail then post cdr and then try test3 so > on. > > > >
> > > > > > > > > > > > > > > >
> > > Is there any variable or configuration needs to set? > > > Best Regards > -- > Samir Doshi > *iNextrix Technologie**s Pvt. Ltd*. > http://www.inextrix.com > > > > *Disclaimer:* > The information contained in this communication is confidential and may be > legally privileged. It is intended solely for the use of the individual or > entity to whom it is addressed and others authorised to receive it. If you > are not the intended recipient you are hereby notified that any disclosure, > copying, distribution or taking action in reliance of the contents of this > information is strictly prohibited and may be unlawful. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161109/9beaea15/attachment.html From udy786 at gmail.com Wed Nov 9 20:43:08 2016 From: udy786 at gmail.com (Uday kumar) Date: Wed, 9 Nov 2016 23:13:08 +0530 Subject: [Freeswitch-users] Need Help on Freeswitch Fail-over (HA) In-Reply-To: References: Message-ID: Hi, Both servers in same data center. OS CentOS. Thanks Uday. On Nov 9, 2016 8:57 PM, "Luis Daniel Lucio Quiroz" < luis.daniel.lucio at gmail.com> wrote: > Servers in same data center or different ones? > > Have you google it? > > Le 9 nov. 2016 7:33 AM, "Uday kumar" a ?crit : > >> Hi All, >> >> Need Help on Freeswitch Fail-over to avoid down time. I have installed >> two Freeswitch server on two different servers. Both server have web and >> mysql database. I am accessing my server using Domain name and also using >> domain name to register softphone. >> >> Is possible, if one server will down then automatically shift everything >> on other server? Right now domain pointed on server A. We are using cron to >> copy backup like mysql, recording, logs etc from server A to server B. As >> per my current setup, if server A will down then I need to change domain >> pointing IP to server B. If this is possible to do automatically then it >> will be good. >> >> Please advice. >> >> >> -- >> Thanks & Regard >> Uday >> Site:- www.shareyourknowledge.in >> Mobile:- +91-9377579349 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> 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/20161109/ffa8fc70/attachment.html From mike at jerris.com Wed Nov 9 20:43:20 2016 From: mike at jerris.com (Michael Jerris) Date: Wed, 9 Nov 2016 12:43:20 -0500 Subject: [Freeswitch-users] Export CDR on failed tried calls In-Reply-To: References: Message-ID: https://freeswitch.org/confluence/display/FREESWITCH/mod_json_cdr > On Nov 6, 2016, at 5:13 AM, Samir Doshi wrote: > > Hi Guys, > > Wondering if we can post failed tried call cdr to mod_json_cdr. I have below dialplan generated and I want to post cdr any gateway fail to process the call. That means if test1 fail then it should send cdr to mod_json_cdr and then go for test2. If test2 fail then post cdr and then try test3 so on. > > > >
> > > > > > > > > > > > > > > >
> > > Is there any variable or configuration needs to set? > > > Best Regards > -- > Samir Doshi > iNextrix Technologies Pvt. Ltd. > http://www.inextrix.com > > > > Disclaimer: > The information contained in this communication is confidential and may be legally privileged. It is intended solely for the use of the individual or entity to whom it is addressed and others authorised to receive it. If you are not the intended recipient you are hereby notified that any disclosure, copying, distribution or taking action in reliance of the contents of this information is strictly prohibited and may be unlawful. > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161109/caeac4da/attachment-0001.html From nbhatti at gmail.com Wed Nov 9 21:58:24 2016 From: nbhatti at gmail.com (Muhammad Naseer Bhatti) Date: Wed, 9 Nov 2016 10:58:24 -0800 Subject: [Freeswitch-users] Default connection pool size for ODBC In-Reply-To: <51DA090C-4B29-4E3B-B9F4-AE2B3DCAAB13@jerris.com> References: <51DA090C-4B29-4E3B-B9F4-AE2B3DCAAB13@jerris.com> Message-ID: As I understand from the documentation, when using ODBC it makes use of connection pooling provided by FreeSWITCH. So I understand there must be a pool where the connections are stored. I am trying to figure out why I keep on getting occasional messages like switch_odbc.c:283 The sql server is not responding for DSN andres [STATE: 08S01 CODE 2006 ERROR: [MySQL][ODBC 5.1 Driver][mysqld-5.6.33-79.0-log]MySQL server has gone away. MySQL is free at this moment and do not have any connections limitation or excessive connection limitations. -- Sent with Airmail From: Michael Jerris Reply: FreeSWITCH Users Help Date: November 9, 2016 at 6:27:31 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Default connection pool size for ODBC Freeswitch does not have a traditional connection pool with a size. What exactly are you referring to? On Nov 8, 2016, at 9:03 AM, Muhammad Naseer Bhatti wrote: Hi, what?s the default connection pool size for ODBC/Postgres is defined? _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161109/5e0e471c/attachment.html From mike at jerris.com Wed Nov 9 22:12:31 2016 From: mike at jerris.com (Michael Jerris) Date: Wed, 9 Nov 2016 14:12:31 -0500 Subject: [Freeswitch-users] Default connection pool size for ODBC In-Reply-To: References: <51DA090C-4B29-4E3B-B9F4-AE2B3DCAAB13@jerris.com> Message-ID: sounds like mysql is terminating active connections? > On Nov 9, 2016, at 1:58 PM, Muhammad Naseer Bhatti wrote: > > As I understand from the documentation, when using ODBC it makes use of connection pooling provided by FreeSWITCH. So I understand there must be a pool where the connections are stored. I am trying to figure out why I keep on getting occasional messages like switch_odbc.c:283 The sql server is not responding for DSN andres [STATE: 08S01 CODE 2006 ERROR: [MySQL][ODBC 5.1 Driver][mysqld-5.6.33-79.0-log]MySQL server has gone away. MySQL is free at this moment and do not have any connections limitation or excessive connection limitations. > > -- > > Sent with Airmail > > From: Michael Jerris > Reply: FreeSWITCH Users Help > Date: November 9, 2016 at 6:27:31 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Default connection pool size for ODBC > >> Freeswitch does not have a traditional connection pool with a size. What exactly are you referring to? >> >>> On Nov 8, 2016, at 9:03 AM, Muhammad Naseer Bhatti > wrote: >>> >>> >>> Hi, what?s the default connection pool size for ODBC/Postgres is defined? >>> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161109/7c300676/attachment.html From nbhatti at gmail.com Wed Nov 9 22:15:39 2016 From: nbhatti at gmail.com (Muhammad Naseer Bhatti) Date: Wed, 9 Nov 2016 11:15:39 -0800 Subject: [Freeswitch-users] Default connection pool size for ODBC In-Reply-To: References: <51DA090C-4B29-4E3B-B9F4-AE2B3DCAAB13@jerris.com> Message-ID: It is possible, but highly doubt since it?s s busy system. That?s why I am trying to figure out if I can change the limit of connections FreeSWITCH keeps for use or somehow see what is pool?s status and see if I can gracefully replicate this to find the root cause. -- Sent with Airmail From: Michael Jerris Reply: Michael Jerris Date: November 9, 2016 at 10:12:33 PM To: Muhammad Naseer Bhatti Cc: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Default connection pool size for ODBC sounds like mysql is terminating active connections? On Nov 9, 2016, at 1:58 PM, Muhammad Naseer Bhatti wrote: As I understand from the documentation, when using ODBC it makes use of connection pooling provided by FreeSWITCH. So I understand there must be a pool where the connections are stored. I am trying to figure out why I keep on getting occasional messages like switch_odbc.c:283 The sql server is not responding for DSN andres [STATE: 08S01 CODE 2006 ERROR: [MySQL][ODBC 5.1 Driver][mysqld-5.6.33-79.0-log]MySQL server has gone away. MySQL is free at this moment and do not have any connections limitation or excessive connection limitations. -- Sent with Airmail From: Michael Jerris Reply: FreeSWITCH Users Help Date: November 9, 2016 at 6:27:31 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Default connection pool size for ODBC Freeswitch does not have a traditional connection pool with a size. What exactly are you referring to? On Nov 8, 2016, at 9:03 AM, Muhammad Naseer Bhatti wrote: Hi, what?s the default connection pool size for ODBC/Postgres is defined? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161109/0534d303/attachment-0001.html From yehavi.bourvine at gmail.com Wed Nov 9 22:16:21 2016 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Wed, 9 Nov 2016 21:16:21 +0200 Subject: [Freeswitch-users] Default connection pool size for ODBC In-Reply-To: References: <51DA090C-4B29-4E3B-B9F4-AE2B3DCAAB13@jerris.com> Message-ID: That's reminds me a similar problem I had ages ago. As far as I remember it was idle timeout of the SQL server (MySQL in my case). Is this on a busy system, or does it happens during long idle periods? Regards, __Yehavi: 2016-11-09 20:58 GMT+02:00 Muhammad Naseer Bhatti : > As I understand from the documentation, when using ODBC it makes use of > connection pooling provided by FreeSWITCH. So I understand there must be a > pool where the connections are stored. I am trying to figure out why I keep > on getting occasional messages like switch_odbc.c:283 The sql server is not > responding for DSN andres [STATE: 08S01 CODE 2006 ERROR: [MySQL][ODBC 5.1 > Driver][mysqld-5.6.33-79.0-log]MySQL server has gone away. MySQL is free > at this moment and do not have any connections limitation or excessive > connection limitations. > > -- > > Sent with Airmail > > From: Michael Jerris > Reply: FreeSWITCH Users Help > > Date: November 9, 2016 at 6:27:31 PM > To: FreeSWITCH Users Help > > Subject: Re: [Freeswitch-users] Default connection pool size for ODBC > > Freeswitch does not have a traditional connection pool with a size. What > exactly are you referring to? > > On Nov 8, 2016, at 9:03 AM, Muhammad Naseer Bhatti > wrote: > > > Hi, what?s the default connection pool size for ODBC/Postgres is defined? > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > 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/20161109/0e1c6618/attachment.html From abaci64 at gmail.com Wed Nov 9 22:18:00 2016 From: abaci64 at gmail.com (Abaci B) Date: Wed, 9 Nov 2016 14:18:00 -0500 Subject: [Freeswitch-users] Default connection pool size for ODBC In-Reply-To: References: <51DA090C-4B29-4E3B-B9F4-AE2B3DCAAB13@jerris.com> Message-ID: are you maybe talking about this https://freeswitch.org/fisheye/browse/~br=master/freeswitch/conf/vanilla/autoload_configs/switch.conf.xml?hb=true#to43 On Wed, Nov 9, 2016 at 1:58 PM, Muhammad Naseer Bhatti wrote: > As I understand from the documentation, when using ODBC it makes use of > connection pooling provided by FreeSWITCH. So I understand there must be a > pool where the connections are stored. I am trying to figure out why I keep > on getting occasional messages like switch_odbc.c:283 The sql server is not > responding for DSN andres [STATE: 08S01 CODE 2006 ERROR: [MySQL][ODBC 5.1 > Driver][mysqld-5.6.33-79.0-log]MySQL server has gone away. MySQL is free > at this moment and do not have any connections limitation or excessive > connection limitations. > > -- > > Sent with Airmail > > From: Michael Jerris > Reply: FreeSWITCH Users Help > > Date: November 9, 2016 at 6:27:31 PM > To: FreeSWITCH Users Help > > Subject: Re: [Freeswitch-users] Default connection pool size for ODBC > > Freeswitch does not have a traditional connection pool with a size. What > exactly are you referring to? > > On Nov 8, 2016, at 9:03 AM, Muhammad Naseer Bhatti > wrote: > > > Hi, what?s the default connection pool size for ODBC/Postgres is defined? > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > 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/20161109/47740a68/attachment.html From nbhatti at gmail.com Wed Nov 9 22:27:50 2016 From: nbhatti at gmail.com (Muhammad Naseer Bhatti) Date: Wed, 09 Nov 2016 22:27:50 +0300 Subject: [Freeswitch-users] Default connection pool size for ODBC In-Reply-To: References: <51DA090C-4B29-4E3B-B9F4-AE2B3DCAAB13@jerris.com> Message-ID: I am not sure if this is max-db-handles because I never saw Max handles exceeded or Error connecting message which should be when exceeding the limits. If I can somehow figure out the current usage of the connections pool, that may help to see what?s going on. At what stage FreeSWITCH will report the server has gone away, the moment when the connection actually drops, even if it?s not used by the active script or when the script tries to use it and see?s there is nothing in the pool available? And I suppose there is no auto-reconnection here either. via Newton Mail [https://cloudmagic.com/k/d/mailapp?ct=dx&cv=9.2.6&pv=10.12.1&source=email_footer_2] On Wed, Nov 9, 2016 at 10:18 PM, Abaci B wrote: are you maybe talking about this https://freeswitch.org/fisheye/browse/~br=master/freeswitch/conf/vanilla/autoload_configs/switch.conf.xml?hb=true#to43 [https://freeswitch.org/fisheye/browse/~br=master/freeswitch/conf/vanilla/autoload_configs/switch.conf.xml?hb=true#to43] On Wed, Nov 9, 2016 at 1:58 PM, Muhammad Naseer Bhatti < nbhatti at gmail.com [nbhatti at gmail.com] > wrote: As I understand from the documentation, when using ODBC it makes use of connection pooling provided by FreeSWITCH. So I understand there must be a pool where the connections are stored. I am trying to figure out why I keep on getting occasional messages like switch_odbc.c:283 The sql server is not responding for DSN andres [STATE: 08S01 CODE 2006 ERROR: [MySQL][ODBC 5.1 Driver][mysqld-5.6.33-79.0-log]MySQL server has gone away. MySQL is free at this moment and do not have any connections limitation or excessive connection limitations. -- Sent with Airmail From: Michael Jerris [mike at jerris.com] Reply: FreeSWITCH Users Help [freeswitch-users at lists.freeswitch.org] Date: November 9, 2016 at 6:27:31 PM To: FreeSWITCH Users Help [freeswitch-users at lists.freeswitch.org] Subject: Re: [Freeswitch-users] Default connection pool size for ODBC Freeswitch does not have a traditional connection pool with a size. What exactly are you referring to? On Nov 8, 2016, at 9:03 AM, Muhammad Naseer Bhatti < nbhatti at gmail.com [nbhatti at gmail.com] > wrote: Hi, what?s the default connection pool size for ODBC/Postgres is defined? _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org [consulting at freeswitch.org] http://www.freeswitchsolutions.com [http://www.freeswitchsolutions.com] Official FreeSWITCH Sites http://www.freeswitch.org [http://www.freeswitch.org] http://confluence.freeswitch.org [http://confluence.freeswitch.org] http://www.cluecon.com [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 [http://lists.freeswitch.org/mailman/listinfo/freeswitch-users] UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users [http://lists.freeswitch.org/mailman/options/freeswitch-users] http://www.freeswitch.org [http://www.freeswitch.org] _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org [consulting at freeswitch.org] http://www.freeswitchsolutions.com [http://www.freeswitchsolutions.com] Official FreeSWITCH Sites http://www.freeswitch.org [http://www.freeswitch.org] http://confluence.freeswitch.org [http://confluence.freeswitch.org] http://www.cluecon.com [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 [http://lists.freeswitch.org/mailman/listinfo/freeswitch-users] UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users [http://lists.freeswitch.org/mailman/options/freeswitch-users] http://www.freeswitch.org [http://www.freeswitch.org] -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161109/da2afab6/attachment-0001.html From lists at kavun.ch Wed Nov 9 22:28:50 2016 From: lists at kavun.ch (Emrah) Date: Wed, 9 Nov 2016 20:28:50 +0100 Subject: [Freeswitch-users] FS-9113, still experiencing TLS crashes In-Reply-To: References: <9D820B3E-294A-49C4-9220-4233FE6994B0@kavun.ch> Message-ID: <0001A843-3487-41C8-824F-1F017015EE20@kavun.ch> It's the "reliably" part that's tricky. I'm using commercial certificates, so let me figure out how to replicate a similar environment. I'll email you the info once I have a setup, and you can circulate where needed. Thanks for helping on this > On Nov 9, 2016, at 4:25 PM, Michael Jerris wrote: > > I need a recipie to reliably reproduce this so I can dig in the code. Is there a way you can put together an environment where this can be reproduced on demand? > >> On Nov 9, 2016, at 3:39 AM, Emrah > wrote: >> >> No Sir, the response packet to the 407 Proxy Authentication Required is never received. So the session then eventually gets abandoned by FS. On the client side, and this is generalized, the packet is sent, except the TLS session breaks. >> >>> On Nov 8, 2016, at 11:41 PM, Michael Jerris > wrote: >>> >>> Can you confirm if the packet is shown in freeswitch tport_log? >>> >>>> On Nov 8, 2016, at 5:02 PM, Emrah > wrote: >>>> >>>> Hello List, >>>> Thanks to the help provided by Stanislav, I learned of issue #9113, https://freeswitch.org/jira/si/jira.issueviews:issue-html/FS-9113/FS-9113.html , which seems to be related to the issues I have been experiencing with FreeSWITCH, TLS and failed call setups. >>>> Coincidentally, or not, the fix pushed on that issue was aligned with whole months where I did not experience any TLS issues. Calls were going through fine, until all of a sudden they started failing again. This is on 2 distinct servers running a load balanced FS setup, and using Yealink phones. >>>> >>>> To sum up, here is what is going on. >>>> From the Yealink, calls with TLS work if I don't use SRTP. >>>> From the Yealink, calls crash if I use TLS and SRTP. >>>> From my laptop softphone, calls only crash sometimes if I use TLS and SRTP. >>>> >>>> How can I debug the TLS session on the FreeSWITCH side to see what happens with the TLS thread? I don't mean packet capture. >>>> >>>> I have a feeling that the packet size is too large and doesn't make it to the FS box intact after the 407 Proxy Required is received by the client. >>>> >>>> Here is the log for the Yealink: >>>> http://pastebin.com/smKP286x >>>> >>>> Your lights would be so appreciated, I'm losing my mind over 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/20161109/37a8c834/attachment.html From peter at hartmanncomputer.com Thu Nov 10 00:17:46 2016 From: peter at hartmanncomputer.com (Peter Hartmann) Date: Wed, 9 Nov 2016 16:17:46 -0500 Subject: [Freeswitch-users] bridge with esf_page_group ? Message-ID: Is it possible to bridge a sip extension with esf_page_group? (the polycom multicast function) Whichever is first wins: (esf_paging_group or bridge) I tried 224.0.1.116 and 5001 each in their own argument statement. Any ideas how to achieve this? Thanks much, Peter Hartmann Hartmann Computer Consulting http://blog.hartmanncomputer.com (212)203-8870 Public Key ID 0x1acfea8b5d88088d If I can't explain it to you in plain language, that means I don't understand it. From zsotya at jss.hu Thu Nov 10 01:42:22 2016 From: zsotya at jss.hu (Zsotya) Date: Wed, 09 Nov 2016 23:42:22 +0100 Subject: [Freeswitch-users] Lua in-band DTMF Message-ID: <20161109234222.labw5mtqaswcw844@mail.sunmonster.jss.hu> Hello All, I am trying to send audiable in-band DTMF from Lua script. If I use the session:execute("send_dtmf", 1) command I can see #1 arrives to the other party in SIP messages but I cannot hear the DTMF audio. How can I set the type of the DTMF? Can I force from lua or is it a gateway or global settings? If I need an audiable DTMF is there any better way than play dtmf sounds from wav files? Thank you! Zsotya From mike at jerris.com Thu Nov 10 01:53:49 2016 From: mike at jerris.com (Michael Jerris) Date: Wed, 9 Nov 2016 17:53:49 -0500 Subject: [Freeswitch-users] Lua in-band DTMF In-Reply-To: <20161109234222.labw5mtqaswcw844@mail.sunmonster.jss.hu> References: <20161109234222.labw5mtqaswcw844@mail.sunmonster.jss.hu> Message-ID: <647D2E03-B391-47F2-9270-A71C168B015C@jerris.com> https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools:+start_dtmf_generate > On Nov 9, 2016, at 5:42 PM, Zsotya wrote: > > Hello All, > > I am trying to send audiable in-band DTMF from Lua script. > If I use the session:execute("send_dtmf", 1) command I can see #1 > arrives to the other party in SIP messages but I cannot hear the DTMF > audio. > > How can I set the type of the DTMF? Can I force from lua or is it a > gateway or global settings? If I need an audiable DTMF is there any > better way than play dtmf sounds from wav files? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161109/fad5c1a2/attachment.html From steveayre at gmail.com Thu Nov 10 01:58:57 2016 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 9 Nov 2016 22:58:57 +0000 Subject: [Freeswitch-users] Adding users programatically In-Reply-To: <9FF42CFE-B4A1-44E5-89D8-1C5684494B88@jerris.com> References: <743a84e15598477f469f042553abd742@parzee.com> <9FF42CFE-B4A1-44E5-89D8-1C5684494B88@jerris.com> Message-ID: While mod_xml_curl would also be my preference, it's also possible within a lua script if you want to generate it within FreeSWITCH rather than from an external server. See the example script on https://freeswitch.org/confluence/display/FREESWITCH/Lua+FreeSWITCH+Dbh On 9 November 2016 at 15:29, Michael Jerris wrote: > use mod_xml_curl > > > On Nov 9, 2016, at 4:24 AM, Gonzalo Gasca Meza > wrote: > > > > Hi forum, > > > > I want to add sip users with mailbox (right now under > > /etc/freeswitch/directory/default as xml files) using FS API. > > I would like to store user information in database (using pgsql 9.5). > > > > Found these posts: > > > > http://voicebundle.com/how-to-add-sip-user-in-freeswitch > > https://freeswitch.org/stash/projects/FS/repos/freeswitch/ > browse/scripts/perl/add_user > > http://voicebundle.com/creating-users-extensions-in- > freeswitch-creating-sip-accounts-in-freeswitch-using-add_user-script > > > > But these are scripts which manipulate XML files. Looking to store > > information in database and have freeswitch read this information. (I > > can take care of adding user, delete user, update user, handle > > duplicates, etc.). > > > > Similar like this post: > > http://telecommusings.blogspot.com/2009/10/using- > modxmlcurl-in-freeswitch-how-hard_28.html > > > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161109/808812b7/attachment-0001.html From lte at lte-net.de Thu Nov 10 08:26:41 2016 From: lte at lte-net.de (Fred Schulz) Date: Thu, 10 Nov 2016 06:26:41 +0100 (CET) Subject: [Freeswitch-users] Need Help on Freeswitch Fail-over (HA) In-Reply-To: References: Message-ID: <1454940359.57256.1478755601640.JavaMail.open-xchange@app06.ox.hosteurope.de> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161110/18b9b27f/attachment.html From service at ivc.nnov.ru Thu Nov 10 11:26:39 2016 From: service at ivc.nnov.ru (Mikhail Demekhov) Date: Thu, 10 Nov 2016 11:26:39 +0300 Subject: [Freeswitch-users] Received an unsupported RTCP packet version 3 Message-ID: <58242F3F.1040106@ivc.nnov.ru> Hello! I have FreeSWITCH Version 1.6.12. After connecting partner FS-1.7 I had a lot of warning: [WARNING] switch_rtp.c:6207 Received an unsupported RTCP packet version 3 How to fix it? -- Regards, Mikhail Demekhov From lists at kavun.ch Thu Nov 10 17:01:59 2016 From: lists at kavun.ch (Emrah) Date: Thu, 10 Nov 2016 15:01:59 +0100 Subject: [Freeswitch-users] FS-9113, still experiencing TLS crashes In-Reply-To: <0001A843-3487-41C8-824F-1F017015EE20@kavun.ch> References: <9D820B3E-294A-49C4-9220-4233FE6994B0@kavun.ch> <0001A843-3487-41C8-824F-1F017015EE20@kavun.ch> Message-ID: <26C55CD4-0AC8-4D32-B49E-8FBFCF035189@kavun.ch> Michael and the list, Without doing anything on the FS side, calls are going through fine today with both TLS and SDES in use on my clients. When calls started failing, I had to disable SRTP on my clients to be able to be able to make calls again. TLS stayed on. I also disconnected a Polycom phone to switch it to a none TLS environment. Today, one of the Yealink I had left unchanged (TLS + SRTP), is working fine again. I cannot get it to break, whereas a few hours prior it would crash the TLS session consistently for every single call. Remember that FS gets the initial packets up to the ACK after the 407 Proxy Authentication Required is sent. It's the following invite that is never received, except if I disable SRTP. This made me conclude that it's a packet size / segmentation issue of my TCP packet. Are you aware of any network layer control that could be causing this? Buffering on Linux in any way? Any concept of a TCP pool that could be getting full? Anything of that nature, that was flushed or emptied by the reduced number of clients calling using TLS + SRTP? It's a long shot, but I thought I'd lay it here, see if it sparks something in your bright minds. Thanks! > On Nov 9, 2016, at 8:28 PM, Emrah wrote: > > It's the "reliably" part that's tricky. > I'm using commercial certificates, so let me figure out how to replicate a similar environment. I'll email you the info once I have a setup, and you can circulate where needed. > > Thanks for helping on this >> On Nov 9, 2016, at 4:25 PM, Michael Jerris > wrote: >> >> I need a recipie to reliably reproduce this so I can dig in the code. Is there a way you can put together an environment where this can be reproduced on demand? >> >>> On Nov 9, 2016, at 3:39 AM, Emrah > wrote: >>> >>> No Sir, the response packet to the 407 Proxy Authentication Required is never received. So the session then eventually gets abandoned by FS. On the client side, and this is generalized, the packet is sent, except the TLS session breaks. >>> >>>> On Nov 8, 2016, at 11:41 PM, Michael Jerris > wrote: >>>> >>>> Can you confirm if the packet is shown in freeswitch tport_log? >>>> >>>>> On Nov 8, 2016, at 5:02 PM, Emrah > wrote: >>>>> >>>>> Hello List, >>>>> Thanks to the help provided by Stanislav, I learned of issue #9113, https://freeswitch.org/jira/si/jira.issueviews:issue-html/FS-9113/FS-9113.html , which seems to be related to the issues I have been experiencing with FreeSWITCH, TLS and failed call setups. >>>>> Coincidentally, or not, the fix pushed on that issue was aligned with whole months where I did not experience any TLS issues. Calls were going through fine, until all of a sudden they started failing again. This is on 2 distinct servers running a load balanced FS setup, and using Yealink phones. >>>>> >>>>> To sum up, here is what is going on. >>>>> From the Yealink, calls with TLS work if I don't use SRTP. >>>>> From the Yealink, calls crash if I use TLS and SRTP. >>>>> From my laptop softphone, calls only crash sometimes if I use TLS and SRTP. >>>>> >>>>> How can I debug the TLS session on the FreeSWITCH side to see what happens with the TLS thread? I don't mean packet capture. >>>>> >>>>> I have a feeling that the packet size is too large and doesn't make it to the FS box intact after the 407 Proxy Required is received by the client. >>>>> >>>>> Here is the log for the Yealink: >>>>> http://pastebin.com/smKP286x >>>>> >>>>> Your lights would be so appreciated, I'm losing my mind over 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/20161110/51382b1e/attachment.html From loi.dangthanh at gmail.com Thu Nov 10 05:31:15 2016 From: loi.dangthanh at gmail.com (=?UTF-8?B?TOG7o2kgxJDhurduZw==?=) Date: Thu, 10 Nov 2016 09:31:15 +0700 Subject: [Freeswitch-users] FreeSwich and SIP REFER Message-ID: Hi list, I have this call scenario, it's just the simplest 2 sofia endpoints external profile, and a bridge application in dialplan Caller -> FS -> calee Then caller compose a REFER within current dialog, FS just handle that REFER, and contact the target sip uri indicated in Refer-To in REFER. Everything seems fine, since FS follow rfc3515 as a user agent, legs bridged Target transfer -> FS -> callee But I actually want my callee to receive and handle REFER message, any trick to do that? I'm thinking of creating new REFER within dialog to callee when FS receive a REFER from caller, but not sure if it's possible or how. My FS version is 1.6.12 rgds, Loi Dang Thanh Phone : 841224.735.448 Email : loi.dangthanh at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161110/1cfda4d1/attachment-0001.html From freeswitch at opencode.e4ward.com Thu Nov 10 05:50:34 2016 From: freeswitch at opencode.e4ward.com (David Matson) Date: Thu, 10 Nov 2016 02:50:34 +0000 Subject: [Freeswitch-users] License for music and sounds Message-ID: I'm maintaining some FreeSWITCH packages for Arch and have separate packages for music and sounds. What's the correct license to list for the audio tarballs available here: http://files.freeswitch.org/releases/sounds/ Is it MPL like the core code or something different? Thanks, David From freeswitch at opencode.e4ward.com Thu Nov 10 05:50:41 2016 From: freeswitch at opencode.e4ward.com (David Matson) Date: Thu, 10 Nov 2016 02:50:41 +0000 Subject: [Freeswitch-users] PGP signatures for release tarballs Message-ID: I'm maintaining some FreeSWITCH packages for Arch. If PGP signatures are available for source tarballs, the build system can use them to verify authenticity automatically. Would you be interested in producing a PGP .sig/.sign/.asc file alongside the release tarballs going forward? http://files.freeswitch.org/releases/freeswitch/ Thanks, David From devang.nathwani31589 at gmail.com Thu Nov 10 17:44:32 2016 From: devang.nathwani31589 at gmail.com (devang nathwani) Date: Thu, 10 Nov 2016 20:14:32 +0530 Subject: [Freeswitch-users] module json_cdr not creating all .json cdr when high number of calls Message-ID: Hello, My freeswitch version is 1.6.10 i have started freeswitch with nice with 19 priority, 'nice -n 19 freeswitch' i am using module json_cdr, here is my config file http://pastebin.com/Mt1AFRVv please note i am writing the logs to json file as well as sending them to nginx web server as well and both legs log are enable i am testing the calls with sipp and sending 10000 calls with 200 cps-call per second and 2000 cc-concurrent calls now after the test i am getting 5192 entires only instead of 20,000(2 legs entry for all 10,000 calls) -> in the other scenario when i send 100 calls with 10 cps and 50 cc, i am getting all 200(100*2) json entries. So what and where is the issue? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161110/44245c3e/attachment.html From modesto916 at gmail.com Thu Nov 10 19:56:14 2016 From: modesto916 at gmail.com (Antonio Modesto) Date: Thu, 10 Nov 2016 14:56:14 -0200 Subject: [Freeswitch-users] Help with FreeTDM and GSM Interface Message-ID: Hi, I'm having problems to connect calls using a digium tdm800p card. The problem only happens with the ports that are connected to a intelbras 4100 gsm interface, ports connected to the POTS are not affected. I'll try the best I can to express the situation: 1) Cold restart the gsm interface and wait for it to boot up 2) Try to place a call using it (freetdm//) 3) The call stays silent, after a while the carrier reports that a invalid number has been informed 4) At this point, all following calls always return busy. If I connect an analog phone to the gsm interface and start dialing digits, it gives a busy tone right away, before I finish dialing the number 5) Cold restart the gsm interface again 6) Try to place a call using it (freetdm//) 7) The calls stays silent, but this time I type the same number again. 8) The call connects. This problem does not happen with an analog phone or with chan_dahdi (asterisk), just with freetdm. My assumption is that when freetdm opens the channel for the first time and starting sending the dtmf digits, the gsm interface is not able to read them and sits waiting, after a while it timeouts and stops working, even for further calls. Here is my freetdm.conf: [span zt dahdi_span_1] trunk_type => FXO debugdtmf => yes analog-start-type => kewl fxo-channel => 1:1-6 zt.cont: [defaults] codec_ms => 20 wink_ms => 150 flash_ms => 750 echo_cancel_level => 64 rxgain => 0.0 txgain => 0.0 dahdi/system.conf: loadzone=br defaultzone=br fxsks=1-6 echocanceller=mg2,1-6 If anybody could help me with this I would really aprecciate. Regards. -- Atenciosamente, Ant?nio Modesto -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161110/c4f24284/attachment-0001.html From ar at cyberfonica.com Thu Nov 10 19:58:16 2016 From: ar at cyberfonica.com (Alejandro Recarey) Date: Thu, 10 Nov 2016 17:58:16 +0100 Subject: [Freeswitch-users] FS-9113, still experiencing TLS crashes In-Reply-To: <0001A843-3487-41C8-824F-1F017015EE20@kavun.ch> References: <9D820B3E-294A-49C4-9220-4233FE6994B0@kavun.ch> <0001A843-3487-41C8-824F-1F017015EE20@kavun.ch> Message-ID: <627AB283-362F-4494-B883-1C3D30D7F2F8@cyberfonica.com> You could either use a self-signed cert for a nonexistent domain (example.com?) and modify your hosts file or DNS to point to he server. I think that should give you an environment to reproduce the crash which you could share without leaking your private cert. > On 9 Nov 2016, at 20:28, Emrah wrote: > > It's the "reliably" part that's tricky. > I'm using commercial certificates, so let me figure out how to replicate a similar environment. I'll email you the info once I have a setup, and you can circulate where needed. > > Thanks for helping on this >> On Nov 9, 2016, at 4:25 PM, Michael Jerris wrote: >> >> I need a recipie to reliably reproduce this so I can dig in the code. Is there a way you can put together an environment where this can be reproduced on demand? >> >>> On Nov 9, 2016, at 3:39 AM, Emrah wrote: >>> >>> No Sir, the response packet to the 407 Proxy Authentication Required is never received. So the session then eventually gets abandoned by FS. On the client side, and this is generalized, the packet is sent, except the TLS session breaks. >>> >>>>> On Nov 8, 2016, at 11:41 PM, Michael Jerris wrote: >>>>> >>>>> Can you confirm if the packet is shown in freeswitch tport_log? >>>>> >>>>> On Nov 8, 2016, at 5:02 PM, Emrah wrote: >>>>> >>>>> Hello List, >>>>> Thanks to the help provided by Stanislav, I learned of issue #9113, https://freeswitch.org/jira/si/jira.issueviews:issue-html/FS-9113/FS-9113.html, which seems to be related to the issues I have been experiencing with FreeSWITCH, TLS and failed call setups. >>>>> Coincidentally, or not, the fix pushed on that issue was aligned with whole months where I did not experience any TLS issues. Calls were going through fine, until all of a sudden they started failing again. This is on 2 distinct servers running a load balanced FS setup, and using Yealink phones. >>>>> >>>>> To sum up, here is what is going on. >>>>> From the Yealink, calls with TLS work if I don't use SRTP. >>>>> From the Yealink, calls crash if I use TLS and SRTP. >>>>> From my laptop softphone, calls only crash sometimes if I use TLS and SRTP. >>>>> >>>>> How can I debug the TLS session on the FreeSWITCH side to see what happens with the TLS thread? I don't mean packet capture. >>>>> >>>>> I have a feeling that the packet size is too large and doesn't make it to the FS box intact after the 407 Proxy Required is received by the client. >>>>> >>>>> Here is the log for the Yealink: >>>>> http://pastebin.com/smKP286x >>>>> >>>>> Your lights would be so appreciated, I'm losing my mind over 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/20161110/d3e996e1/attachment-0001.html From senthilganesh at zohocorp.com Thu Nov 10 20:24:48 2016 From: senthilganesh at zohocorp.com (Senthil Ganesh) Date: Thu, 10 Nov 2016 22:54:48 +0530 Subject: [Freeswitch-users] Signalling from Android app Message-ID: <1584f4541c5.12a3c1bfe26782.1752127892018515516@zohocorp.com> Hi, I am developing an Audio/Video Android app. I am able to connect to the FS server using SIP protocol over UDP. I read somewhere that SIP protocol is blocked in some countries. Enterprise firewall restricts UDP packets and allow only TCP on port 443. What is the best way to do signalling from Android app? Regards,Senthil Ganesh -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161110/9ee542e8/attachment.html From manpower13.cse at gmail.com Thu Nov 10 20:53:26 2016 From: manpower13.cse at gmail.com (Murugan Pandian) Date: Thu, 10 Nov 2016 23:23:26 +0530 Subject: [Freeswitch-users] Signalling from Android app In-Reply-To: <1584f4541c5.12a3c1bfe26782.1752127892018515516@zohocorp.com> References: <1584f4541c5.12a3c1bfe26782.1752127892018515516@zohocorp.com> Message-ID: Hi, Can i know which SIP client library you are using for your application On Thu, Nov 10, 2016 at 10:54 PM, Senthil Ganesh wrote: > Hi, > > I am developing an Audio/Video Android app. I am able to connect to the FS > server using SIP protocol over UDP. > > I read somewhere that SIP protocol is blocked in some countries. > Enterprise firewall restricts UDP packets and allow only TCP on port 443. > > What is the best way to do signalling from Android app? > > > Regards, > Senthil Ganesh > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161110/d390691c/attachment.html From saumar at uol.com.br Thu Nov 10 21:00:11 2016 From: saumar at uol.com.br (Saumar Hajjar) Date: Thu, 10 Nov 2016 16:00:11 -0200 Subject: [Freeswitch-users] How to play multiple video files (not video calls) in a conference Message-ID: Hi, I'm working in a PoC and I'm considering FS. I'd like to create a conference and have several video files playing simultaneously in the canvas. I already tried: conference name play av:///var/www/vid/video.mp4 and it works great. But I need multiple files playing and apparently the above command queues a video file (or freezes if it's a rtsp stream... Later I'll confirm this and file a Jira) I also tried creating a loopback leg, joining the conference, and playing a video just to this particular member. But it fails because the member doesn't support video. Basically I'd like to do something like the examples found at the bottom of https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video. Probably I'm missing some trivial stuff... I really appreciate any advice on this. Thanks in advance, Saumar From colin.morelli at gmail.com Thu Nov 10 21:00:05 2016 From: colin.morelli at gmail.com (Colin Morelli) Date: Thu, 10 Nov 2016 18:00:05 +0000 Subject: [Freeswitch-users] Signalling from Android app In-Reply-To: References: <1584f4541c5.12a3c1bfe26782.1752127892018515516@zohocorp.com> Message-ID: If you're developing an application that you plan to distribute (as in, not just a toy app that you're playing around with), TLS over TCP should really be the only thing you even consider IMO. Using an insecure transport leaves your signaling *and* media traffic open to anyone, not to mention the difficulties of successfully handling UDP behind NATs, firewalls, and any other unknown/uncontrolled network scenario. On Thu, Nov 10, 2016 at 12:56 PM Murugan Pandian wrote: > Hi, > > Can i know which SIP client library you are using for your application > > On Thu, Nov 10, 2016 at 10:54 PM, Senthil Ganesh < > senthilganesh at zohocorp.com> wrote: > > Hi, > > I am developing an Audio/Video Android app. I am able to connect to the FS > server using SIP protocol over UDP. > > I read somewhere that SIP protocol is blocked in some countries. > Enterprise firewall restricts UDP packets and allow only TCP on port 443. > > What is the best way to do signalling from Android app? > > > Regards, > Senthil Ganesh > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > 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/20161110/0500d80d/attachment.html From lists at kavun.ch Thu Nov 10 22:24:18 2016 From: lists at kavun.ch (Emrah) Date: Thu, 10 Nov 2016 20:24:18 +0100 Subject: [Freeswitch-users] FS-9113, still experiencing TLS crashes In-Reply-To: <627AB283-362F-4494-B883-1C3D30D7F2F8@cyberfonica.com> References: <9D820B3E-294A-49C4-9220-4233FE6994B0@kavun.ch> <0001A843-3487-41C8-824F-1F017015EE20@kavun.ch> <627AB283-362F-4494-B883-1C3D30D7F2F8@cyberfonica.com> Message-ID: I agree, as long as I get to reproduce it that way. I am suspecting everything here. From the keysize to the CA to the TCP transport getting compromised to openssl not reliably transmitting certain packets to FS. Thanks for the suggestion > On Nov 10, 2016, at 5:58 PM, Alejandro Recarey wrote: > > You could either use a self-signed cert for a nonexistent domain (example.com ?) and modify your hosts file or DNS to point to he server. I think that should give you an environment to reproduce the crash which you could share without leaking your private cert. > > > On 9 Nov 2016, at 20:28, Emrah > wrote: > >> It's the "reliably" part that's tricky. >> I'm using commercial certificates, so let me figure out how to replicate a similar environment. I'll email you the info once I have a setup, and you can circulate where needed. >> >> Thanks for helping on this >>> On Nov 9, 2016, at 4:25 PM, Michael Jerris > wrote: >>> >>> I need a recipie to reliably reproduce this so I can dig in the code. Is there a way you can put together an environment where this can be reproduced on demand? >>> >>>> On Nov 9, 2016, at 3:39 AM, Emrah > wrote: >>>> >>>> No Sir, the response packet to the 407 Proxy Authentication Required is never received. So the session then eventually gets abandoned by FS. On the client side, and this is generalized, the packet is sent, except the TLS session breaks. >>>> >>>>> On Nov 8, 2016, at 11:41 PM, Michael Jerris > wrote: >>>>> >>>>> Can you confirm if the packet is shown in freeswitch tport_log? >>>>> >>>>>> On Nov 8, 2016, at 5:02 PM, Emrah > wrote: >>>>>> >>>>>> Hello List, >>>>>> Thanks to the help provided by Stanislav, I learned of issue #9113, https://freeswitch.org/jira/si/jira.issueviews:issue-html/FS-9113/FS-9113.html , which seems to be related to the issues I have been experiencing with FreeSWITCH, TLS and failed call setups. >>>>>> Coincidentally, or not, the fix pushed on that issue was aligned with whole months where I did not experience any TLS issues. Calls were going through fine, until all of a sudden they started failing again. This is on 2 distinct servers running a load balanced FS setup, and using Yealink phones. >>>>>> >>>>>> To sum up, here is what is going on. >>>>>> From the Yealink, calls with TLS work if I don't use SRTP. >>>>>> From the Yealink, calls crash if I use TLS and SRTP. >>>>>> From my laptop softphone, calls only crash sometimes if I use TLS and SRTP. >>>>>> >>>>>> How can I debug the TLS session on the FreeSWITCH side to see what happens with the TLS thread? I don't mean packet capture. >>>>>> >>>>>> I have a feeling that the packet size is too large and doesn't make it to the FS box intact after the 407 Proxy Required is received by the client. >>>>>> >>>>>> Here is the log for the Yealink: >>>>>> http://pastebin.com/smKP286x >>>>>> >>>>>> Your lights would be so appreciated, I'm losing my mind over 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 _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161110/1e2aa400/attachment-0001.html From zsotya at jss.hu Thu Nov 10 23:12:47 2016 From: zsotya at jss.hu (Zsotya) Date: Thu, 10 Nov 2016 21:12:47 +0100 Subject: [Freeswitch-users] Lua in-band DTMF In-Reply-To: <647D2E03-B391-47F2-9270-A71C168B015C@jerris.com> References: <20161109234222.labw5mtqaswcw844@mail.sunmonster.jss.hu> <647D2E03-B391-47F2-9270-A71C168B015C@jerris.com> Message-ID: <20161110211247.pjc5kp3qoo4kwkkw@mail.sunmonster.jss.hu> Hello Michael, Thank you for the helping hand but this does not work for me. This is my test code which rings my extension and after I answer there is 4 secoud silence. I still see the DTMF in the SIP messages on the soft phone debug trace but no audiable DTMF. What could be wrong here please? session=freeswitch.Session("user/1001") log.i("Leg B's UUID: " .. tostring(session:getVariable("uuid"))) if(session:ready()) then log.i("Leg ready") else log.e("Leg does not ready") end session:execute("start_dtmf_generate") session:sleep(1000) session:execute("send_dtmf", 1) session:sleep(1000) session:execute("send_dtmf", 2) session:sleep(1000) session:execute("send_dtmf", 3) session:sleep(1000) Related freeswitch log snippet: EXECUTE sofia/internal/1001 at x.y.1.2:5060 start_dtmf_generate() 2016-11-10 20:03:08.179963 [DEBUG] switch_core_media_bug.c:828 Attaching BUG to sofia/internal/1001 at x.y.1.2:5060 2016-11-10 20:03:08.239952 [DEBUG] switch_rtp.c:6694 Correct audio ip/port confirmed. 2016-11-10 20:03:08.239952 [DEBUG] switch_core_io.c:448 Setting BUG Codec PCMA:8 EXECUTE sofia/internal/1001 at x.y.1.2:5060 send_dtmf(1) 2016-11-10 20:03:09.199949 [DEBUG] switch_core_io.c:1894 sofia/internal/1001 at x.y.1.2:5060 send dtmf digit=1 ms=250 samples=2000 EXECUTE sofia/internal/1001 at 10.8.1.2:5060 send_dtmf(2) 2016-11-10 20:03:10.199951 [DEBUG] switch_core_io.c:1894 sofia/internal/1001 at x.y.1.2:5060 send dtmf digit=2 ms=250 samples=2000 EXECUTE sofia/internal/1001 at 10.8.1.2:5060 send_dtmf(3) 2016-11-10 20:03:11.199963 [DEBUG] switch_core_io.c:1894 sofia/internal/1001 at x.y.1.2:5060 send dtmf digit=3 ms=250 samples=2000 Thank you in advance! Zsotya Id?zet (Michael Jerris ): > https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools:+start_dtmf_generate > > >> On Nov 9, 2016, at 5:42 PM, Zsotya wrote: >> >> Hello All, >> >> I am trying to send audiable in-band DTMF from Lua script. >> If I use the session:execute("send_dtmf", 1) command I can see #1 >> arrives to the other party in SIP messages but I cannot hear the DTMF >> audio. >> >> How can I set the type of the DTMF? Can I force from lua or is it a >> gateway or global settings? If I need an audiable DTMF is there any >> better way than play dtmf sounds from wav files? > > From devang.nathwani31589 at gmail.com Fri Nov 11 06:36:45 2016 From: devang.nathwani31589 at gmail.com (devang nathwani) Date: Fri, 11 Nov 2016 09:06:45 +0530 Subject: [Freeswitch-users] module json_cdr not creating all .json cdr when high number of calls In-Reply-To: References: Message-ID: Any suggestion? On Nov 10, 2016 8:14 PM, "devang nathwani" wrote: > Hello, > > My freeswitch version is 1.6.10 > i have started freeswitch with nice with 19 priority, 'nice -n 19 > freeswitch' > i am using module json_cdr, here is my config file > http://pastebin.com/Mt1AFRVv > please note i am writing the logs to json file as well as sending them to > nginx web server as well and both legs log are enable > i am testing the calls with sipp and sending 10000 calls with 200 cps-call > per second and 2000 cc-concurrent calls > now after the test i am getting 5192 entires only instead of 20,000(2 legs > entry for all 10,000 calls) > > -> in the other scenario when i send 100 calls with 10 cps and 50 cc, i am > getting all 200(100*2) json entries. > > So what and where is the issue? > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161111/cddac865/attachment.html From devang.nathwani31589 at gmail.com Fri Nov 11 09:44:58 2016 From: devang.nathwani31589 at gmail.com (devang nathwani) Date: Fri, 11 Nov 2016 12:14:58 +0530 Subject: [Freeswitch-users] module json_cdr not creating all .json cdr when high number of calls In-Reply-To: References: Message-ID: anything? On Fri, Nov 11, 2016 at 9:06 AM, devang nathwani < devang.nathwani31589 at gmail.com> wrote: > Any suggestion? > > On Nov 10, 2016 8:14 PM, "devang nathwani" > wrote: > >> Hello, >> >> My freeswitch version is 1.6.10 >> i have started freeswitch with nice with 19 priority, 'nice -n 19 >> freeswitch' >> i am using module json_cdr, here is my config file >> http://pastebin.com/Mt1AFRVv >> please note i am writing the logs to json file as well as sending them to >> nginx web server as well and both legs log are enable >> i am testing the calls with sipp and sending 10000 calls with 200 >> cps-call per second and 2000 cc-concurrent calls >> now after the test i am getting 5192 entires only instead of 20,000(2 >> legs entry for all 10,000 calls) >> >> -> in the other scenario when i send 100 calls with 10 cps and 50 cc, i >> am getting all 200(100*2) json entries. >> >> So what and where is the issue? >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161111/6e9e867e/attachment.html From Alexander.Haugg at c4b.de Fri Nov 11 10:42:47 2016 From: Alexander.Haugg at c4b.de (Alexander Haugg) Date: Fri, 11 Nov 2016 07:42:47 +0000 Subject: [Freeswitch-users] String manipulation in the dialplan Message-ID: Hi, the examples on ?? are working successfully. But I need the function like ${var:0:-7}. For example: My dialplan app return a value like this ?from number?|?location? ->123456|intern. The length of the ?from number? is unknown, but the length of the location is fix (it could be intern or extern only). Now I am trying ${var:0:-7} to extract the ?123456?, but I get the complete string 123456|intern. What can I do with this function? Or is the only possibility to solve this via a regular expression ^([0-9]+)[|](intern|extern)$ -> (not tested ;-)? Thanks a lot! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161111/6fa069b5/attachment.html From Alexander.Haugg at c4b.de Fri Nov 11 10:46:42 2016 From: Alexander.Haugg at c4b.de (Alexander Haugg) Date: Fri, 11 Nov 2016 07:46:42 +0000 Subject: [Freeswitch-users] String manipulation in the dialplan In-Reply-To: References: Message-ID: <05047b9b81574299aa090b1f4143b41f@c4b.de> Hi, the examples on ? https://wiki.freeswitch.org/wiki/Manipulating_Channel_Variables ? are working successfully. But I need the function like ${var:0:-7}. For example: My dialplan app return a value like this ?from number?|?location? ->123456|intern. The length of the ?from number? is unknown, but the length of the location is fix (it could be intern or extern only). Now I am trying ${var:0:-7} to extract the ?123456?, but I get the complete string 123456|intern. What can I do with this function? Or is the only possibility to solve this via a regular expression ^([0-9]+)[|](intern|extern)$ -> (not tested ;-)? Thanks a lot! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161111/63305f5d/attachment-0001.html From mirkobrankovic at gmail.com Fri Nov 11 11:23:12 2016 From: mirkobrankovic at gmail.com (Mirko Brankovic) Date: Fri, 11 Nov 2016 09:23:12 +0100 Subject: [Freeswitch-users] Signalling from Android app In-Reply-To: <1584f4541c5.12a3c1bfe26782.1752127892018515516@zohocorp.com> References: <1584f4541c5.12a3c1bfe26782.1752127892018515516@zohocorp.com> Message-ID: Maybe you can use websockets and mod_verto ... then you can send only json messages On Thu, Nov 10, 2016 at 6:24 PM, Senthil Ganesh wrote: > Hi, > > I am developing an Audio/Video Android app. I am able to connect to the FS > server using SIP protocol over UDP. > > I read somewhere that SIP protocol is blocked in some countries. > Enterprise firewall restricts UDP packets and allow only TCP on port 443. > > What is the best way to do signalling from Android app? > > > Regards, > Senthil Ganesh > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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, Mirko -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161111/25834e18/attachment.html From gregor at infomedia.si Fri Nov 11 11:53:39 2016 From: gregor at infomedia.si (Gregor Nanger) Date: Fri, 11 Nov 2016 09:53:39 +0100 Subject: [Freeswitch-users] Callee ID on gateway bridge Message-ID: Can someone please help me with advice. I am routing call to some provider that I need to add prefix to callee number. Then I see in call log Callee ID with prefix. How can I bridge to gateway number :aa1XXXXXXXXX and see only XXXXXXX in call logs? Best regards, Gregor -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161111/0d5f2010/attachment.html From v.zakhozhai at gmail.com Fri Nov 11 11:56:17 2016 From: v.zakhozhai at gmail.com (Vladyslav Zakhozhai) Date: Fri, 11 Nov 2016 08:56:17 +0000 Subject: [Freeswitch-users] Signalling from Android app In-Reply-To: References: <1584f4541c5.12a3c1bfe26782.1752127892018515516@zohocorp.com> Message-ID: The simplest and standard way is SIP over TLS (by the way it wotks via TCP not UDP). But it does not solve your issue. I think that you may consider using some kind of tunneling built-in in your App (i.e. IPSec, SSL VPN, etc). In this case nobody knows what kind of traffic goes through this tunnel and you have not issues with NAT in SIP messaging. But this is non-default solution and it will not work out of the box in FreeSWITCH, Asterisk, etc (you need point of termination this tunnels). >From the other hand maybe websockets is an option as Mirko has mentioned. But will it work properly with Asterisk or other opensource and non-opensource or non-free PBXes? I doubt that. On Fri, Nov 11, 2016 at 10:24 AM Mirko Brankovic wrote: > Maybe you can use websockets and mod_verto ... then you can send only > json messages > > On Thu, Nov 10, 2016 at 6:24 PM, Senthil Ganesh < > senthilganesh at zohocorp.com> wrote: > > Hi, > > I am developing an Audio/Video Android app. I am able to connect to the FS > server using SIP protocol over UDP. > > I read somewhere that SIP protocol is blocked in some countries. > Enterprise firewall restricts UDP packets and allow only TCP on port 443. > > What is the best way to do signalling from Android app? > > > Regards, > Senthil Ganesh > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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, > Mirko > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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, Vladyslav Zakhozhai email: v.zakhozhai at gmail.com tel.: +380(93) 757-21-61 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161111/cef1459b/attachment.html From hawkins at hawkinsegroup.com Fri Nov 11 12:01:49 2016 From: hawkins at hawkinsegroup.com (Don Hawkins) Date: Fri, 11 Nov 2016 03:01:49 -0600 Subject: [Freeswitch-users] module json_cdr not creating all .json cdr when high number of calls In-Reply-To: References: Message-ID: The CDRs aren't perfect, i would do XML instead, I can only imagine json considering the issues i had getting XML going. My 2 cents. Sincerely, Don Hawkins Sent from my NationPCS? Nexus 6. On Nov 11, 2016 12:46 AM, "devang nathwani" wrote: > anything? > > On Fri, Nov 11, 2016 at 9:06 AM, devang nathwani < > devang.nathwani31589 at gmail.com> wrote: > >> Any suggestion? >> >> On Nov 10, 2016 8:14 PM, "devang nathwani" > m> wrote: >> >>> Hello, >>> >>> My freeswitch version is 1.6.10 >>> i have started freeswitch with nice with 19 priority, 'nice -n 19 >>> freeswitch' >>> i am using module json_cdr, here is my config file >>> http://pastebin.com/Mt1AFRVv >>> please note i am writing the logs to json file as well as sending them >>> to nginx web server as well and both legs log are enable >>> i am testing the calls with sipp and sending 10000 calls with 200 >>> cps-call per second and 2000 cc-concurrent calls >>> now after the test i am getting 5192 entires only instead of 20,000(2 >>> legs entry for all 10,000 calls) >>> >>> -> in the other scenario when i send 100 calls with 10 cps and 50 cc, i >>> am getting all 200(100*2) json entries. >>> >>> So what and where is the issue? >>> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161111/5465d24c/attachment-0001.html From v.zakhozhai at gmail.com Fri Nov 11 12:04:45 2016 From: v.zakhozhai at gmail.com (Vladyslav Zakhozhai) Date: Fri, 11 Nov 2016 09:04:45 +0000 Subject: [Freeswitch-users] module json_cdr not creating all .json cdr when high number of calls In-Reply-To: References: Message-ID: Devang very interesting issue. I'm going to use json-cdr as well. And I'm scary little bit :) Did you try to play with your configuration to exclude some "buggy" parts? My point is to try the following use cases: 1. Write json cdr only to disks 2. Write json cdr only to http service Providing configuration is not enough. We do not know the load on you disks (is it SSD, HDD, or virtual disk on SSD or HDD), load on you network adapters and response time from http service, etc. Is there any errors in freeswitch logs related to this issue? On Fri, Nov 11, 2016 at 8:45 AM devang nathwani < devang.nathwani31589 at gmail.com> wrote: > anything? > > > On Fri, Nov 11, 2016 at 9:06 AM, devang nathwani < > devang.nathwani31589 at gmail.com> wrote: > > Any suggestion? > > On Nov 10, 2016 8:14 PM, "devang nathwani" > wrote: > > Hello, > > My freeswitch version is 1.6.10 > i have started freeswitch with nice with 19 priority, 'nice -n 19 > freeswitch' > i am using module json_cdr, here is my config file > http://pastebin.com/Mt1AFRVv > please note i am writing the logs to json file as well as sending them to > nginx web server as well and both legs log are enable > i am testing the calls with sipp and sending 10000 calls with 200 cps-call > per second and 2000 cc-concurrent calls > now after the test i am getting 5192 entires only instead of 20,000(2 legs > entry for all 10,000 calls) > > -> in the other scenario when i send 100 calls with 10 cps and 50 cc, i am > getting all 200(100*2) json entries. > > So what and where is the issue? > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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, Vladyslav Zakhozhai email: v.zakhozhai at gmail.com tel.: +380(93) 757-21-61 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161111/738b2751/attachment.html From s.safarov at gmail.com Fri Nov 11 13:14:21 2016 From: s.safarov at gmail.com (Sergey Safarov) Date: Fri, 11 Nov 2016 10:14:21 +0000 Subject: [Freeswitch-users] module json_cdr not creating all .json cdr when high number of calls In-Reply-To: References: Message-ID: You can try mod_format. This module may create CDR in json format ??, 11 ????. 2016, 12:05 Vladyslav Zakhozhai : > Devang very interesting issue. I'm going to use json-cdr as well. And I'm > scary little bit :) > > Did you try to play with your configuration to exclude some "buggy" parts? > My point is to try the following use cases: > 1. Write json cdr only to disks > 2. Write json cdr only to http service > > Providing configuration is not enough. We do not know the load on you > disks (is it SSD, HDD, or virtual disk on SSD or HDD), load on you network > adapters and response time from http service, etc. > > Is there any errors in freeswitch logs related to this issue? > > On Fri, Nov 11, 2016 at 8:45 AM devang nathwani < > devang.nathwani31589 at gmail.com> wrote: > > anything? > > > On Fri, Nov 11, 2016 at 9:06 AM, devang nathwani < > devang.nathwani31589 at gmail.com> wrote: > > Any suggestion? > > On Nov 10, 2016 8:14 PM, "devang nathwani" > wrote: > > Hello, > > My freeswitch version is 1.6.10 > i have started freeswitch with nice with 19 priority, 'nice -n 19 > freeswitch' > i am using module json_cdr, here is my config file > http://pastebin.com/Mt1AFRVv > please note i am writing the logs to json file as well as sending them to > nginx web server as well and both legs log are enable > i am testing the calls with sipp and sending 10000 calls with 200 cps-call > per second and 2000 cc-concurrent calls > now after the test i am getting 5192 entires only instead of 20,000(2 legs > entry for all 10,000 calls) > > -> in the other scenario when i send 100 calls with 10 cps and 50 cc, i am > getting all 200(100*2) json entries. > > So what and where is the issue? > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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, > Vladyslav Zakhozhai > email: v.zakhozhai at gmail.com > tel.: +380(93) 757-21-61 > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161111/d6af5450/attachment.html From devang.nathwani31589 at gmail.com Fri Nov 11 13:39:41 2016 From: devang.nathwani31589 at gmail.com (devang nathwani) Date: Fri, 11 Nov 2016 16:09:41 +0530 Subject: [Freeswitch-users] module json_cdr not creating all .json cdr when high number of calls In-Reply-To: References: Message-ID: Hello , i have tried using mod_format_cdr, result not much differs my server is having hdd and its dedicated and having 24 cores i have tried two other scenarios once with only writing to disk and not sending data to web server 100 cps, 1000 cc, 5000 total top is showing all the core usage 2400 % usage, which is also critical, (how to reduce that?) i am getting 5971 json cdr entries instead of 10,000(5000*2, for 2 legs) the second case is with only sending data to web server and not writing to disk 100 cps, 1000 cc, 5000 total top is again showing the core usage of all 2400%(!) and i am getting only 704 web server request, using nginx web server On Fri, Nov 11, 2016 at 3:44 PM, Sergey Safarov wrote: > You can try mod_format. This module may create CDR in json format > > ??, 11 ????. 2016, 12:05 Vladyslav Zakhozhai : > >> Devang very interesting issue. I'm going to use json-cdr as well. And I'm >> scary little bit :) >> >> Did you try to play with your configuration to exclude some "buggy" >> parts? My point is to try the following use cases: >> 1. Write json cdr only to disks >> 2. Write json cdr only to http service >> >> Providing configuration is not enough. We do not know the load on you >> disks (is it SSD, HDD, or virtual disk on SSD or HDD), load on you network >> adapters and response time from http service, etc. >> >> Is there any errors in freeswitch logs related to this issue? >> >> On Fri, Nov 11, 2016 at 8:45 AM devang nathwani < >> devang.nathwani31589 at gmail.com> wrote: >> >> anything? >> >> >> On Fri, Nov 11, 2016 at 9:06 AM, devang nathwani < >> devang.nathwani31589 at gmail.com> wrote: >> >> Any suggestion? >> >> On Nov 10, 2016 8:14 PM, "devang nathwani" > com> wrote: >> >> Hello, >> >> My freeswitch version is 1.6.10 >> i have started freeswitch with nice with 19 priority, 'nice -n 19 >> freeswitch' >> i am using module json_cdr, here is my config file >> http://pastebin.com/Mt1AFRVv >> please note i am writing the logs to json file as well as sending them to >> nginx web server as well and both legs log are enable >> i am testing the calls with sipp and sending 10000 calls with 200 >> cps-call per second and 2000 cc-concurrent calls >> now after the test i am getting 5192 entires only instead of 20,000(2 >> legs entry for all 10,000 calls) >> >> -> in the other scenario when i send 100 calls with 10 cps and 50 cc, i >> am getting all 200(100*2) json entries. >> >> So what and where is the issue? >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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, >> Vladyslav Zakhozhai >> email: v.zakhozhai at gmail.com >> tel.: +380(93) 757-21-61 >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> 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/20161111/9596fdbc/attachment-0001.html From ovoshlook at gmail.com Fri Nov 11 14:23:42 2016 From: ovoshlook at gmail.com (Yuriy Gorlichenko) Date: Fri, 11 Nov 2016 14:23:42 +0300 Subject: [Freeswitch-users] execute_on_answer in lua Message-ID: Hi all I need to use execute_on_answer in my lua is there any possibility to call it like sessiom:executeOnAnswer or semething like this? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161111/6b3a9fa8/attachment.html From devang.nathwani31589 at gmail.com Fri Nov 11 14:34:16 2016 From: devang.nathwani31589 at gmail.com (devang nathwani) Date: Fri, 11 Nov 2016 17:04:16 +0530 Subject: [Freeswitch-users] module json_cdr not creating all .json cdr when high number of calls In-Reply-To: References: Message-ID: Gentlemen? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161111/f04e6c80/attachment.html From italo at freeswitch.org Fri Nov 11 16:00:12 2016 From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=) Date: Fri, 11 Nov 2016 13:00:12 +0000 Subject: [Freeswitch-users] module json_cdr not creating all .json cdr when high number of calls In-Reply-To: References: Message-ID: Try to use the xml_cdr and see if the same behavior is obtained. You need to know if it's a bug in json only or in the core. We can't say too much without seeing your test environment, logs, dialplan, etc. Post your findings to a JIRA with all logs and information included. You should consider emailing consulting at freeswitch.org to get assistance with this performance test Em sex, 11 de nov de 2016 ?s 08:35, devang nathwani < devang.nathwani31589 at gmail.com> escreveu: > Gentlemen? > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161111/5ddb1fea/attachment.html From fanx07 at gmail.com Fri Nov 11 16:31:49 2016 From: fanx07 at gmail.com (Anonim Stefan) Date: Fri, 11 Nov 2016 15:31:49 +0200 Subject: [Freeswitch-users] mod_vlc youtube playback Message-ID: Hi, I want to ask if sound/video playback of an youtube URL can be done with mod_vlc. Looking in freeswitch mod_vlc.c I can see "vlc_file_supported_formats[argc++] = "youtube";". I have tried without success: " EXECUTE sofia/internal/7479572798 playback(vlc:// https://youtu.be/VoeCcCQuJrM) 2016-11-11 13:23:36.772643 [DEBUG] mod_vlc.c:832 VLC attempt to open https://youtu.be/VoeCcCQuJrM read 2016-11-11 13:23:36.772643 [DEBUG] mod_vlc.c:857 VLC open https://youtu.be/VoeCcCQuJrM for reading 2016-11-11 13:23:36.772643 [NOTICE] mod_vlc.c:863 VLC Path is http https://youtu.be/VoeCcCQuJrM 2016-11-11 13:23:36.772643 [DEBUG] switch_ivr_play_say.c:1467 Codec Activated L16 at 8000hz 1 channels 20ms 2016-11-11 13:23:36.772643 [DEBUG] mod_vlc.c:242 Got a libvlc_MediaStateChanged callback. New state: 1 2016-11-11 13:23:37.472579 [DEBUG] mod_vlc.c:242 Got a libvlc_MediaStateChanged callback. New state: 3 2016-11-11 13:23:37.532634 [DEBUG] mod_vlc.c:242 Got a libvlc_MediaStateChanged callback. New state: 7 2016-11-11 13:23:37.532634 [DEBUG] mod_vlc.c:227 Got a libvlc_MediaPlayerEncounteredError callback. mediaPlayer Status: 7 2016-11-11 13:23:37.532634 [ERR] mod_vlc.c:1094 VLC error 2016-11-11 13:23:37.532634 [DEBUG] switch_ivr_play_say.c:1910 done playing file vlc://https://youtu.be/VoeCcCQuJrM 2016-11-11 13:23:37.572584 [DEBUG] mod_vlc.c:242 Got a libvlc_MediaStateChanged callback. New state: 5 2016-11-11 13:23:37.572584 [DEBUG] mod_python.c:286 Finished calling python script 2016-11-11 13:23:37.572584 [DEBUG] switch_cpp.cpp:1107 sofia/internal/7479572798 destroy/unlink session from object EXECUTE sofia/internal/7479572798 hangup() " Couldn't find any documentation/examples about youtube playback using mod_vlc. Can someone help me with some examples, please? Thank you, Stefan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161111/e9e78d54/attachment.html From david.villasmil.work at gmail.com Fri Nov 11 16:33:36 2016 From: david.villasmil.work at gmail.com (David Villasmil) Date: Fri, 11 Nov 2016 13:33:36 +0000 Subject: [Freeswitch-users] execute_on_answer in lua In-Reply-To: References: Message-ID: It's a variable you can set in lua https://wiki.freeswitch.org/wiki/Variable_execute_on_answer session:setVariable Set a variable on a session: session:setVariable("varname", "varval"); https://freeswitch.org/confluence/plugins/servlet/mobile#content/view/3965124 On Fri, Nov 11, 2016 at 12:24 PM Yuriy Gorlichenko wrote: > Hi all > I need to use execute_on_answer in my lua > is there any possibility to call it like sessiom:executeOnAnswer or > semething like 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/20161111/8e5d4b09/attachment.html From ovoshlook at gmail.com Fri Nov 11 17:26:18 2016 From: ovoshlook at gmail.com (Yuriy Gorlichenko) Date: Fri, 11 Nov 2016 17:26:18 +0300 Subject: [Freeswitch-users] execute_on_answer in lua In-Reply-To: References: Message-ID: ok thanks. I thought that it can be done through api or somethig like this but if no - will use this. Thanks one more time 2016-11-11 16:33 GMT+03:00 David Villasmil : > It's a variable you can set in lua > > https://wiki.freeswitch.org/wiki/Variable_execute_on_answer > > session:setVariable > > Set a variable on a session: > > session:setVariable("varname", "varval"); > > > > https://freeswitch.org/confluence/plugins/servlet/ > mobile#content/view/3965124 > > On Fri, Nov 11, 2016 at 12:24 PM Yuriy Gorlichenko > wrote: > >> Hi all >> I need to use execute_on_answer in my lua >> is there any possibility to call it like sessiom:executeOnAnswer or >> semething like 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/20161111/dc5183c5/attachment-0001.html From alexdruzhilov at gmail.com Fri Nov 11 19:52:00 2016 From: alexdruzhilov at gmail.com (=?UTF-8?B?0JDQu9C10LrRgdCw0L3QtNGAINCU0YDRg9C20LjQu9C+0LI=?=) Date: Fri, 11 Nov 2016 19:52:00 +0300 Subject: [Freeswitch-users] RTCP in passthrough mode Message-ID: Could somebody explain me how Freeswitch works with a RTCP packets in video passthrough mode? I ask because I found a problem when one user starts sending video to another client through Freeswitch. The problem in depth: I have two clients and whant to make WebRTC video conference between them (but use only passthrough mode because it's just a screensharing when one user sends his screen to anybody other in conference). So first user starts conference and sends his screen to Freeswitch (and he can see that Freeswitch sends him back his screen properly). When another user attaches to this conference he does not see any video, but at the same time I see in chrome://webrtc-internals that he receives exactly the same video stream that first user sends to freeswitch. I started to look deeper and found that second client sends a lot of PLI requests (about 5 per second) to Freeswitch. I tend to think that it's because second client attaches to conference in the middle of RTP stream and he didn't receive a full intra-picture and he can't start encode the video stream. Also I found that no RTCP packets was sent from Freeswitch to the first client to get this intra-picture. So returning to my first question: what if Freeswitch just ignores this RTCP packets in passthrough mode? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161111/d1e14f93/attachment.html From anthony.minessale at gmail.com Fri Nov 11 19:57:08 2016 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 11 Nov 2016 10:57:08 -0600 Subject: [Freeswitch-users] mod_vlc youtube playback In-Reply-To: References: Message-ID: It's mostly focused on broadcasting not playing youtube. I remember getting it to work at least once but its a bit fragile because it has to parse all the dash manifests etc which keep changing. I'd recommend caching it by downloading it with youtube-dl first then playing it. This will probably improve as the industry finally adopts all the streaming stuff. On Fri, Nov 11, 2016 at 7:31 AM, Anonim Stefan wrote: > Hi, > > I want to ask if sound/video playback of an youtube URL can be done with > mod_vlc. Looking in freeswitch mod_vlc.c I can see > "vlc_file_supported_formats[argc++] = "youtube";". > > I have tried without success: > > " > EXECUTE sofia/internal/7479572798 playback(vlc://https://youtu. > be/VoeCcCQuJrM) > 2016-11-11 13:23:36.772643 [DEBUG] mod_vlc.c:832 VLC attempt to open > https://youtu.be/VoeCcCQuJrM read > 2016-11-11 13:23:36.772643 [DEBUG] mod_vlc.c:857 VLC open > https://youtu.be/VoeCcCQuJrM for reading > 2016-11-11 13:23:36.772643 [NOTICE] mod_vlc.c:863 VLC Path is http > https://youtu.be/VoeCcCQuJrM > 2016-11-11 13:23:36.772643 [DEBUG] switch_ivr_play_say.c:1467 Codec > Activated L16 at 8000hz 1 channels 20ms > 2016-11-11 13:23:36.772643 [DEBUG] mod_vlc.c:242 Got a > libvlc_MediaStateChanged callback. New state: 1 > 2016-11-11 13:23:37.472579 [DEBUG] mod_vlc.c:242 Got a > libvlc_MediaStateChanged callback. New state: 3 > 2016-11-11 13:23:37.532634 [DEBUG] mod_vlc.c:242 Got a > libvlc_MediaStateChanged callback. New state: 7 > 2016-11-11 13:23:37.532634 [DEBUG] mod_vlc.c:227 Got a libvlc_MediaPlayerEncounteredError > callback. mediaPlayer Status: 7 > 2016-11-11 13:23:37.532634 [ERR] mod_vlc.c:1094 VLC error > 2016-11-11 13:23:37.532634 [DEBUG] switch_ivr_play_say.c:1910 done playing > file vlc://https://youtu.be/VoeCcCQuJrM > 2016-11-11 13:23:37.572584 [DEBUG] mod_vlc.c:242 Got a > libvlc_MediaStateChanged callback. New state: 5 > 2016-11-11 13:23:37.572584 [DEBUG] mod_python.c:286 Finished calling > python script > 2016-11-11 13:23:37.572584 [DEBUG] switch_cpp.cpp:1107 sofia/internal/ > 7479572798 destroy/unlink session from object > EXECUTE sofia/internal/7479572798 hangup() > " > > Couldn't find any documentation/examples about youtube playback using > mod_vlc. Can someone help me with some examples, please? > > Thank you, > Stefan > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161111/422a3836/attachment.html From abaci64 at gmail.com Fri Nov 11 20:52:40 2016 From: abaci64 at gmail.com (Abaci B) Date: Fri, 11 Nov 2016 12:52:40 -0500 Subject: [Freeswitch-users] module json_cdr not creating all .json cdr when high number of calls In-Reply-To: References: Message-ID: CDR should be logged for every call, if it doesn't then it's probably a bug which should be reported. On Fri, Nov 11, 2016 at 4:01 AM, Don Hawkins wrote: > The CDRs aren't perfect, i would do XML instead, I can only imagine json > considering the issues i had getting XML going. My 2 cents. > > Sincerely, > Don Hawkins > > Sent from my NationPCS? Nexus 6. > > On Nov 11, 2016 12:46 AM, "devang nathwani" m> wrote: > >> anything? >> >> On Fri, Nov 11, 2016 at 9:06 AM, devang nathwani < >> devang.nathwani31589 at gmail.com> wrote: >> >>> Any suggestion? >>> >>> On Nov 10, 2016 8:14 PM, "devang nathwani" < >>> devang.nathwani31589 at gmail.com> wrote: >>> >>>> Hello, >>>> >>>> My freeswitch version is 1.6.10 >>>> i have started freeswitch with nice with 19 priority, 'nice -n 19 >>>> freeswitch' >>>> i am using module json_cdr, here is my config file >>>> http://pastebin.com/Mt1AFRVv >>>> please note i am writing the logs to json file as well as sending them >>>> to nginx web server as well and both legs log are enable >>>> i am testing the calls with sipp and sending 10000 calls with 200 >>>> cps-call per second and 2000 cc-concurrent calls >>>> now after the test i am getting 5192 entires only instead of 20,000(2 >>>> legs entry for all 10,000 calls) >>>> >>>> -> in the other scenario when i send 100 calls with 10 cps and 50 cc, i >>>> am getting all 200(100*2) json entries. >>>> >>>> So what and where is the issue? >>>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> 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/20161111/4eec33c9/attachment.html From ssinyagin at gmail.com Sun Nov 13 00:45:37 2016 From: ssinyagin at gmail.com (Stanislav Sinyagin) Date: Sat, 12 Nov 2016 22:45:37 +0100 Subject: [Freeswitch-users] FS-9113, still experiencing TLS crashes In-Reply-To: References: <9D820B3E-294A-49C4-9220-4233FE6994B0@kavun.ch> <0001A843-3487-41C8-824F-1F017015EE20@kavun.ch> <627AB283-362F-4494-B883-1C3D30D7F2F8@cyberfonica.com> Message-ID: We actually built a test server, but weren't able to reproduce the issue. I can bring it up again if needed. On 10 Nov 2016 20:25, "Emrah" wrote: > I agree, as long as I get to reproduce it that way. I am suspecting > everything here. From the keysize to the CA to the TCP transport getting > compromised to openssl not reliably transmitting certain packets to FS. > > Thanks for the suggestion > > On Nov 10, 2016, at 5:58 PM, Alejandro Recarey wrote: > > You could either use a self-signed cert for a nonexistent domain ( > example.com?) and modify your hosts file or DNS to point to he server. I > think that should give you an environment to reproduce the crash which you > could share without leaking your private cert. > > > On 9 Nov 2016, at 20:28, Emrah wrote: > > It's the "reliably" part that's tricky. > I'm using commercial certificates, so let me figure out how to replicate a > similar environment. I'll email you the info once I have a setup, and you > can circulate where needed. > > Thanks for helping on this > > On Nov 9, 2016, at 4:25 PM, Michael Jerris wrote: > > I need a recipie to reliably reproduce this so I can dig in the code. Is > there a way you can put together an environment where this can be > reproduced on demand? > > On Nov 9, 2016, at 3:39 AM, Emrah wrote: > > No Sir, the response packet to the 407 Proxy Authentication Required is > never received. So the session then eventually gets abandoned by FS. On the > client side, and this is generalized, the packet is sent, except the TLS > session breaks. > > On Nov 8, 2016, at 11:41 PM, Michael Jerris wrote: > > Can you confirm if the packet is shown in freeswitch tport_log? > > On Nov 8, 2016, at 5:02 PM, Emrah wrote: > > Hello List, > Thanks to the help provided by Stanislav, I learned of issue #9113, > https://freeswitch.org/jira/si/jira.issueviews:issue- > html/FS-9113/FS-9113.html, which seems to be related to the issues I have > been experiencing with FreeSWITCH, TLS and failed call setups. > Coincidentally, or not, the fix pushed on that issue was aligned with > whole months where I did not experience any TLS issues. Calls were going > through fine, until all of a sudden they started failing again. This is on > 2 distinct servers running a load balanced FS setup, and using Yealink > phones. > > *To sum up, here is what is going on.* > *From the Yealink, calls with TLS work if I don't use SRTP.* > *From the Yealink, calls crash if I use TLS and SRTP.* > From my laptop softphone, calls only crash sometimes if I use TLS and SRTP. > > How can I debug the TLS session on the FreeSWITCH side to see what happens > with the TLS thread? I don't mean packet capture. > > I have a feeling that the packet size is too large and doesn't make it to > the FS box intact after the 407 Proxy Required is received by the client. > > Here is the log for the Yealink: > http://pastebin.com/smKP286x > > Your lights would be so appreciated, I'm losing my mind over 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > 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/20161112/7aca260c/attachment-0001.html From blackc2004 at gmail.com Sun Nov 13 02:29:44 2016 From: blackc2004 at gmail.com (Cj B) Date: Sat, 12 Nov 2016 15:29:44 -0800 Subject: [Freeswitch-users] Error starting profile with WSS-binding enables Message-ID: <5D77131B-5487-4745-9765-A1F9744F1CA5@gmail.com> Hi All, I have a strange issue with starting the internal profile with wss-binding enabled. If I start freeswitch, the profiles won?t start even if I try to manually start them: http://pastebin.com/bLB8XnxS Then If I go and remove the wss-binding, start the profile, then add wss-binding back and just do a rescan, it starts and the wss-binding works. This is on FreeSWITCH version: 1.6.12+git~20161111T161001Z~873c5ada12~64bit (git 873c5ad 2016-11-11 16:10:01Z 64bit) Any hints as to how to fix this? I?d really like to be able to get my profiles to start without intervention. This was not happening when I was running 1.6.8, it only started after the upgrade to 1.6.12. Thanks. Cj B -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161112/a12542a1/attachment.html From s.safarov at gmail.com Sun Nov 13 10:42:09 2016 From: s.safarov at gmail.com (Sergey Safarov) Date: Sun, 13 Nov 2016 07:42:09 +0000 Subject: [Freeswitch-users] Error starting profile with WSS-binding enables In-Reply-To: <5D77131B-5487-4745-9765-A1F9744F1CA5@gmail.com> References: <5D77131B-5487-4745-9765-A1F9744F1CA5@gmail.com> Message-ID: Maybe ssl certificate cannot be loaded. ??, 13 ????. 2016, 2:30 Cj B : > Hi All, > > I have a strange issue with starting the internal profile with wss-binding > enabled. If I start freeswitch, the profiles won?t start even if I try to > manually start them: http://pastebin.com/bLB8XnxS > > Then If I go and remove the wss-binding, start the profile, then add > wss-binding back and just do a rescan, it starts and the wss-binding works. > This is on FreeSWITCH version: 1.6.12+git~20161111T161001Z~873c5ada12~64bit > (git 873c5ad 2016-11-11 16:10:01Z 64bit) > > Any hints as to how to fix this? I?d really like to be able to get my > profiles to start without intervention. This was not happening when I was > running 1.6.8, it only started after the upgrade to 1.6.12. > > Thanks. > Cj B > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161113/866b6f6d/attachment.html From blackc2004 at gmail.com Sun Nov 13 18:56:35 2016 From: blackc2004 at gmail.com (Cj B) Date: Sun, 13 Nov 2016 07:56:35 -0800 Subject: [Freeswitch-users] Error starting profile with WSS-binding enables In-Reply-To: References: <5D77131B-5487-4745-9765-A1F9744F1CA5@gmail.com> Message-ID: <1055D98A-804E-4E47-827D-F4FDC673B56A@gmail.com> If it?s the SSL Cert, isn?t there usually a message when loglevel 9 is enabled? I don?t see any errors. Cj B > On Nov 12, 2016, at 11:42 PM, Sergey Safarov wrote: > > Maybe ssl certificate cannot be loaded. > > > ??, 13 ????. 2016, 2:30 Cj B >: > Hi All, > > I have a strange issue with starting the internal profile with wss-binding enabled. If I start freeswitch, the profiles won?t start even if I try to manually start them: http://pastebin.com/bLB8XnxS > > Then If I go and remove the wss-binding, start the profile, then add wss-binding back and just do a rescan, it starts and the wss-binding works. This is on FreeSWITCH version: 1.6.12+git~20161111T161001Z~873c5ada12~64bit (git 873c5ad 2016-11-11 16:10:01Z 64bit) > > Any hints as to how to fix this? I?d really like to be able to get my profiles to start without intervention. This was not happening when I was running 1.6.8, it only started after the upgrade to 1.6.12. > > Thanks. > Cj B > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > 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/20161113/63bd9bce/attachment.html From ovoshlook at gmail.com Sun Nov 13 20:57:51 2016 From: ovoshlook at gmail.com (Yuriy Gorlichenko) Date: Sun, 13 Nov 2016 20:57:51 +0300 Subject: [Freeswitch-users] NO any rinback signalling wjile trying to make originate with bgapi Message-ID: Hi. I using mod_conference and making outgoing call from conference for adding in existed conference All going ok (call reaches to destination) but in process of ringing i can not hear any ringback. So i hear originate leg in conference only after call was answered but i want to detect process of ringing in a call Is there any way to do it? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161113/73d2910a/attachment-0001.html From gascagonzalo at gmail.com Sun Nov 13 21:59:28 2016 From: gascagonzalo at gmail.com (Gonzalo Gasca Meza) Date: Sun, 13 Nov 2016 10:59:28 -0800 Subject: [Freeswitch-users] Error starting profile with WSS-binding enables In-Reply-To: <1055D98A-804E-4E47-827D-F4FDC673B56A@gmail.com> References: <5D77131B-5487-4745-9765-A1F9744F1CA5@gmail.com> <1055D98A-804E-4E47-827D-F4FDC673B56A@gmail.com> Message-ID: Which port are you using for WSS?, looks like FS is complaining of port already in use. MAke sure you use a different port for the one you use for UDP/TCP. Example: SIP UDP: 5060 SIP TCP: 5060 SIP TLS: 5061 SIP WS: 5062 SIP WSS: 5063 On Sun, Nov 13, 2016 at 7:56 AM, Cj B wrote: > If it?s the SSL Cert, isn?t there usually a message when loglevel 9 is > enabled? I don?t see any errors. > > Cj B > > On Nov 12, 2016, at 11:42 PM, Sergey Safarov wrote: > > Maybe ssl certificate cannot be loaded. > > ??, 13 ????. 2016, 2:30 Cj B : > >> Hi All, >> >> I have a strange issue with starting the internal profile with >> wss-binding enabled. If I start freeswitch, the profiles won?t start even >> if I try to manually start them: http://pastebin.com/bLB8XnxS >> >> Then If I go and remove the wss-binding, start the profile, then add >> wss-binding back and just do a rescan, it starts and the wss-binding works. >> This is on FreeSWITCH version: 1.6.12+git~20161111T161001Z~873c5ada12~64bit >> (git 873c5ad 2016-11-11 16:10:01Z 64bit) >> >> Any hints as to how to fix this? I?d really like to be able to get my >> profiles to start without intervention. This was not happening when I was >> running 1.6.8, it only started after the upgrade to 1.6.12. >> >> Thanks. >> Cj B >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> 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/20161113/573b0fca/attachment.html From alexdruzhilov at gmail.com Mon Nov 14 10:56:47 2016 From: alexdruzhilov at gmail.com (=?UTF-8?B?0JDQu9C10LrRgdCw0L3QtNGAINCU0YDRg9C20LjQu9C+0LI=?=) Date: Mon, 14 Nov 2016 10:56:47 +0300 Subject: [Freeswitch-users] Fwd: RTCP in passthrough mode In-Reply-To: References: Message-ID: Could somebody explain me how Freeswitch works with a RTCP packets in video passthrough mode? I ask because I found a problem when one user starts sending video to another client through Freeswitch. The problem in depth: I have two clients and whant to make WebRTC video conference between them (but use only passthrough mode because it's just a screensharing when one user sends his screen to anybody other in conference). So first user starts conference and sends his screen to Freeswitch (and he can see that Freeswitch sends him back his screen properly). When another user attaches to this conference he does not see any video, but at the same time I see in chrome://webrtc-internals that he receives exactly the same video stream that first user sends to freeswitch. I started to look deeper and found that second client sends a lot of PLI requests (about 5 per second) to Freeswitch. I tend to think that it's because second client attaches to conference in the middle of RTP stream and he didn't receive a full intra-picture and he can't start encode the video stream. Also I found that no RTCP packets was sent from Freeswitch to the first client to get this intra-picture. So returning to my first question: what if Freeswitch just ignores this RTCP packets in passthrough mode? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161114/ea192731/attachment.html From mike at jerris.com Mon Nov 14 15:56:10 2016 From: mike at jerris.com (Michael Jerris) Date: Mon, 14 Nov 2016 12:56:10 +0000 Subject: [Freeswitch-users] Fwd: RTCP in passthrough mode In-Reply-To: References: Message-ID: passthrough mode in video conference is basically useless. It's left over from before we had any real video support. It does nothing at all with rtcp. There is no plans to do anything useful with this feature other than to possibly remove it in the future. On Mon, Nov 14, 2016 at 2:58 AM ????????? ???????? wrote: > Could somebody explain me how Freeswitch works with a RTCP packets in > video passthrough mode? > > I ask because I found a problem when one user starts sending video to > another client through Freeswitch. The problem in depth: I have two clients > and whant to make WebRTC video conference between them (but use only > passthrough mode because it's just a screensharing when one user sends his > screen to anybody other in conference). So first user starts conference and > sends his screen to Freeswitch (and he can see that Freeswitch sends him > back his screen properly). When another user attaches to this conference he > does not see any video, but at the same time I see in > chrome://webrtc-internals that he receives exactly the same video stream > that first user sends to freeswitch. I started to look deeper and found > that second client sends a lot of PLI requests (about 5 per second) to > Freeswitch. I tend to think that it's because second client attaches to > conference in the middle of RTP stream and he didn't receive a full > intra-picture and he can't start encode the video stream. Also I found that > no RTCP packets was sent from Freeswitch to the first client to get this > intra-picture. So returning to my first question: what if Freeswitch just > ignores this RTCP packets in passthrough mode? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161114/a0c53786/attachment-0001.html From blackc2004 at gmail.com Mon Nov 14 18:41:58 2016 From: blackc2004 at gmail.com (Cj B) Date: Mon, 14 Nov 2016 07:41:58 -0800 Subject: [Freeswitch-users] Error starting profile with WSS-binding enables In-Reply-To: References: <5D77131B-5487-4745-9765-A1F9744F1CA5@gmail.com> <1055D98A-804E-4E47-827D-F4FDC673B56A@gmail.com> Message-ID: Hi, Here are the ports I?m using: SIP UDP/TCP: 5060 SIP TLS: 5061 SIP WS: Not sure? I don?t see this configured anywhere. SIP WSS: 7443 Cj B > On Nov 13, 2016, at 10:59 AM, Gonzalo Gasca Meza wrote: > > Which port are you using for WSS?, looks like FS is complaining of port already in use. > MAke sure you use a different port for the one you use for UDP/TCP. Example: > > SIP UDP: 5060 > SIP TCP: 5060 > SIP TLS: 5061 > SIP WS: 5062 > SIP WSS: 5063 > > > > On Sun, Nov 13, 2016 at 7:56 AM, Cj B > wrote: > If it?s the SSL Cert, isn?t there usually a message when loglevel 9 is enabled? I don?t see any errors. > > Cj B > >> On Nov 12, 2016, at 11:42 PM, Sergey Safarov > wrote: >> >> Maybe ssl certificate cannot be loaded. >> >> >> ??, 13 ????. 2016, 2:30 Cj B >: >> Hi All, >> >> I have a strange issue with starting the internal profile with wss-binding enabled. If I start freeswitch, the profiles won?t start even if I try to manually start them: http://pastebin.com/bLB8XnxS >> >> Then If I go and remove the wss-binding, start the profile, then add wss-binding back and just do a rescan, it starts and the wss-binding works. This is on FreeSWITCH version: 1.6.12+git~20161111T161001Z~873c5ada12~64bit (git 873c5ad 2016-11-11 16:10:01Z 64bit) >> >> Any hints as to how to fix this? I?d really like to be able to get my profiles to start without intervention. This was not happening when I was running 1.6.8, it only started after the upgrade to 1.6.12. >> >> Thanks. >> Cj B >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> 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/20161114/5d9b0485/attachment.html From abaci64 at gmail.com Mon Nov 14 21:11:39 2016 From: abaci64 at gmail.com (Abaci B) Date: Mon, 14 Nov 2016 13:11:39 -0500 Subject: [Freeswitch-users] Lua freeswitch.Dbh() question Message-ID: when runned a query using freeswitch.Dbh dbh:query, if there is results it a function for each returned row where you can see the data. however, if there is an error you can see the database returned error in the console on level ERR but you don't get it back in lua is someone aware of a way to get the rror back into lua? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161114/c578e30f/attachment.html From moises.silva at gmail.com Mon Nov 14 22:15:32 2016 From: moises.silva at gmail.com (Moises Silva) Date: Mon, 14 Nov 2016 14:15:32 -0500 Subject: [Freeswitch-users] Help with FreeTDM and GSM Interface In-Reply-To: References: Message-ID: On Thu, Nov 10, 2016 at 11:56 AM, Antonio Modesto wrote: > Hi, > > I'm having problems to connect calls using a digium tdm800p card. The > problem only happens with the ports that are connected to a intelbras 4100 > gsm interface, ports connected to the POTS are not affected. I'll try the > best I can to express the situation: > > 1) Cold restart the gsm interface and wait for it to boot up > 2) Try to place a call using it (freetdm//) > 3) The call stays silent, after a while the carrier reports that a invalid > number has been informed > 4) At this point, all following calls always return busy. If I connect an > analog phone to the gsm interface and start dialing digits, it gives a busy > tone right away, before I finish dialing the number > 5) Cold restart the gsm interface again > 6) Try to place a call using it (freetdm//) > 7) The calls stays silent, but this time I type the same number again. > 8) The call connects. > You need to post debug logs (use https://pastebin.freeswitch.org/ and paste the url here). You can also try the option wait-dialtone-timeout = 0 (in freetdm.conf.xml in the section), so we don't wait for dial tone, then in your dial string you can use freetdm/1/WW1234 Every upper case w (W) is a full second wait time, so the example above would wait for 2 seconds before dialing. Setting dial-timeout to 0 ensures the string is dialed, regardless of whether we receive dial tone or not. - Moy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161114/9394a90a/attachment-0001.html From anthony.minessale at gmail.com Tue Nov 15 01:11:51 2016 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 14 Nov 2016 16:11:51 -0600 Subject: [Freeswitch-users] How to play multiple video files (not video calls) in a conference In-Reply-To: References: Message-ID: Your only real option is make the calls from another box or loop them into the conference with SIP on the same box. The conference file playing mechanism is not designed for simultaneous files to play. On Thu, Nov 10, 2016 at 12:00 PM, Saumar Hajjar wrote: > Hi, > > I'm working in a PoC and I'm considering FS. > > I'd like to create a conference and have several video files playing > simultaneously in the canvas. > I already tried: conference name play av:///var/www/vid/video.mp4 and it > works great. > But I need multiple files playing and apparently the above command > queues a video file (or freezes if it's a rtsp stream... Later I'll > confirm this and file a Jira) > > I also tried creating a loopback leg, joining the conference, and > playing a video just to this particular member. But it fails because the > member doesn't support video. > > Basically I'd like to do something like the examples found at the bottom > of > https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video. > Probably I'm missing some trivial stuff... I really appreciate any > advice on this. > > Thanks in advance, > > Saumar > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161114/5e6e89a5/attachment.html From saumar at uol.com.br Tue Nov 15 02:39:51 2016 From: saumar at uol.com.br (Saumar Hajjar) Date: Mon, 14 Nov 2016 21:39:51 -0200 Subject: [Freeswitch-users] How to play multiple video files (not video calls) in a conference In-Reply-To: References: Message-ID: <83a0ac81-0466-974c-83cb-964db7675ffa@uol.com.br> Thanks Anthony, For your second suggestion, is it possible to "loop a call into the conference with SIP on the same box" using just CLI commands - or do I need a external SIP client for this? Em 14/11/2016 20:11, Anthony Minessale escreveu: > Your only real option is make the calls from another box or loop them > into the conference with SIP on the same box. > The conference file playing mechanism is not designed for simultaneous > files to play. > > > > On Thu, Nov 10, 2016 at 12:00 PM, Saumar Hajjar > wrote: > > Hi, > > I'm working in a PoC and I'm considering FS. > > I'd like to create a conference and have several video files playing > simultaneously in the canvas. > I already tried: conference name play av:///var/www/vid/video.mp4 > and it > works great. > But I need multiple files playing and apparently the above command > queues a video file (or freezes if it's a rtsp stream... Later I'll > confirm this and file a Jira) > > I also tried creating a loopback leg, joining the conference, and > playing a video just to this particular member. But it fails > because the > member doesn't support video. > > Basically I'd like to do something like the examples found at the > bottom > of > https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video > . > Probably I'm missing some trivial stuff... I really appreciate any > advice on this. > > Thanks in advance, > > Saumar > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 > > https://www.youtube.com/watch?v=9XXgW34t40s > https://www.youtube.com/watch?v=NLaDpGQuZDA > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161114/c3f446b2/attachment.html From abaci64 at gmail.com Tue Nov 15 03:08:34 2016 From: abaci64 at gmail.com (Abaci B) Date: Mon, 14 Nov 2016 19:08:34 -0500 Subject: [Freeswitch-users] How to play multiple video files (not video calls) in a conference In-Reply-To: <83a0ac81-0466-974c-83cb-964db7675ffa@uol.com.br> References: <83a0ac81-0466-974c-83cb-964db7675ffa@uol.com.br> Message-ID: you can have 2 sip profiles and use originate to create the calls going from 1 profile to the other (one can listen o local ip if it helps) On Mon, Nov 14, 2016 at 6:39 PM, Saumar Hajjar wrote: > Thanks Anthony, > > For your second suggestion, is it possible to "loop a call into the > conference with SIP on the same box" using just CLI commands - or do I need > a external SIP client for this? > > > Em 14/11/2016 20:11, Anthony Minessale escreveu: > > Your only real option is make the calls from another box or loop them into > the conference with SIP on the same box. > The conference file playing mechanism is not designed for simultaneous > files to play. > > > > On Thu, Nov 10, 2016 at 12:00 PM, Saumar Hajjar wrote: > >> Hi, >> >> I'm working in a PoC and I'm considering FS. >> >> I'd like to create a conference and have several video files playing >> simultaneously in the canvas. >> I already tried: conference name play av:///var/www/vid/video.mp4 and it >> works great. >> But I need multiple files playing and apparently the above command >> queues a video file (or freezes if it's a rtsp stream... Later I'll >> confirm this and file a Jira) >> >> I also tried creating a loopback leg, joining the conference, and >> playing a video just to this particular member. But it fails because the >> member doesn't support video. >> >> Basically I'd like to do something like the examples found at the bottom >> of >> https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video >> . >> Probably I'm missing some trivial stuff... I really appreciate any >> advice on this. >> >> Thanks in advance, >> >> Saumar >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 > > https://www.youtube.com/watch?v=9XXgW34t40s > https://www.youtube.com/watch?v=NLaDpGQuZDA > > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161114/e7664f42/attachment-0001.html From saumar at uol.com.br Tue Nov 15 07:50:54 2016 From: saumar at uol.com.br (Saumar Hajjar) Date: Tue, 15 Nov 2016 02:50:54 -0200 Subject: [Freeswitch-users] How to play multiple video files (not video calls) in a conference In-Reply-To: References: <83a0ac81-0466-974c-83cb-964db7675ffa@uol.com.br> Message-ID: <9ea8a857-2629-724b-b7d7-09855f9a20f6@uol.com.br> Thank you very much. I got it working now: sofia status Name Type Data State ================================================================================================= external profile sip:mod_sofia at 192.168.1.110:5080 RUNNING (1) internal profile sip:mod_sofia at 192.168.1.110:5060 RUNNING (0) ================================================================================================= dialplan/default.xml dialplan/public.xml CLI originate sofia/internal/3600 at 192.168.1.110:5080 &conference(3500-192.168.1.110) originate sofia/internal/3601 at 192.168.1.110:5080 &conference(3500-192.168.1.110) Em 14/11/2016 22:08, Abaci B escreveu: > you can have 2 sip profiles and use originate to create the calls > going from 1 profile to the other (one can listen o local ip if it helps) > > On Mon, Nov 14, 2016 at 6:39 PM, Saumar Hajjar > wrote: > > Thanks Anthony, > > For your second suggestion, is it possible to "loop a call into > the conference with SIP on the same box" using just CLI commands - > or do I need a external SIP client for this? > > > Em 14/11/2016 20:11, Anthony Minessale escreveu: >> Your only real option is make the calls from another box or loop >> them into the conference with SIP on the same box. >> The conference file playing mechanism is not designed for >> simultaneous files to play. >> >> >> >> On Thu, Nov 10, 2016 at 12:00 PM, Saumar Hajjar >> > wrote: >> >> Hi, >> >> I'm working in a PoC and I'm considering FS. >> >> I'd like to create a conference and have several video files >> playing >> simultaneously in the canvas. >> I already tried: conference name play >> av:///var/www/vid/video.mp4 and it >> works great. >> But I need multiple files playing and apparently the above >> command >> queues a video file (or freezes if it's a rtsp stream... >> Later I'll >> confirm this and file a Jira) >> >> I also tried creating a loopback leg, joining the conference, and >> playing a video just to this particular member. But it fails >> because the >> member doesn't support video. >> >> Basically I'd like to do something like the examples found at >> the bottom >> of >> https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video >> . >> Probably I'm missing some trivial stuff... I really >> appreciate any >> advice on this. >> >> Thanks in advance, >> >> Saumar >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 >> >> >> https://www.youtube.com/watch?v=9XXgW34t40s >> >> https://www.youtube.com/watch?v=NLaDpGQuZDA >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> 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/20161115/59a6ebf1/attachment-0001.html From findmeinwland at gmail.com Tue Nov 15 10:10:55 2016 From: findmeinwland at gmail.com (Artur Mega) Date: Tue, 15 Nov 2016 12:10:55 +0500 Subject: [Freeswitch-users] Lua freeswitch.Dbh() question In-Reply-To: References: Message-ID: Do you need to get text of an error? And to catch it in lua code? Why not just to check returned value? I dont remember, but returned value can be nul or item with type `userdata`. If you still want text of an error, i think you can try to catch it with pcall https://www.lua.org/pil/8.4.html 2016-11-14 23:11 GMT+05:00 Abaci B : > when runned a query using freeswitch.Dbh dbh:query, if there is results it > a function for each returned row where you can see the data. however, if > there is an error you can see the database returned error in the console on > level ERR but you don't get it back in lua > is someone aware of a way to get the rror back into lua? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- ????? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161115/686097b7/attachment.html From s.safarov at gmail.com Tue Nov 15 12:23:20 2016 From: s.safarov at gmail.com (Sergey Safarov) Date: Tue, 15 Nov 2016 09:23:20 +0000 Subject: [Freeswitch-users] Memory leak In-Reply-To: References: Message-ID: Hello Anthony On old FS version memory usage graph is attached. I think memory usage grows is stoped near 3Gb but last week memory usage is increased about 200Mb later i will report result on new version and with valgrind results. [image: pasted1] ??, 13 ????. 2016 ?. ? 19:12, Anthony Minessale : > That looks normal to me. > FreeSWITCH needs a minimum of 2 GB dedicated ram for prolonged use. > If you chart goes past 2 to 2.5 gigs, you may have a problem then. > > you can run valgrind but you can only run 1 call at a time testing typical > callflow. > > valgrind --tool=memcheck --log-file=vg.log --leak-check=full > --leak-resolution=high --show-reachable=yes /path/to/freeswitch -vg > > > Also when those months are over, hopefully you update cos you can keep > running the same FS for many months anyway ;) > > > > > On Tue, Sep 13, 2016 at 9:29 AM, Sergey Safarov > wrote: > > I has configured FreeSwitch process memory usage graph. > According this graph durring 24 days size of used memory ingrezed about > two times. > [image: FS-memory-chart.png] > Pastebin of FreeSwitch process memory map is placed at > https://pastebin.freeswitch.org/view/9e66572a > Are you have any suggestion how to find memory leak. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 <+1%20919-386-9900> > > https://www.youtube.com/watch?v=9XXgW34t40s > https://www.youtube.com/watch?v=NLaDpGQuZDA > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161115/f014f9c7/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: FS-memory-chart.png Type: image/png Size: 29510 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161115/f014f9c7/attachment-0002.png -------------- next part -------------- A non-text attachment was scrubbed... Name: pasted1 Type: image/png Size: 46848 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161115/f014f9c7/attachment-0003.png From evoip at ukr.net Tue Nov 15 18:47:01 2016 From: evoip at ukr.net (Ewgeny) Date: Tue, 15 Nov 2016 17:47:01 +0200 Subject: [Freeswitch-users] Problem with CALL PICKUP (intercept) in QUEUES (mod_callcenter) Message-ID: <7a9f66b1-cca1-9d01-369c-a2c6d26ceb16@ukr.net> Hi ! First about terminology: *Call Pickup*the ability to pull a ringing call to the phone you are currently on. Call Pickup = Call Intercept = Call group pickup. We're using FreeSWITCH version: 1.6.8~64bit in our complex telephony system with Kamailio and other SIP services. The problem with call pickup (intercept) when using a Queue (mod_callcenter). The group pickup scheme we are using described here: http://www.tech-invite.com/fo-sip/tinv-fo-sip-service-16.html. We do SUBSCRIBE (3 see link) to some our service - that return information about the call (call legs) in NOTIFY with XML body (5). Then we do an INVITE with REPLACES (7) that actually do the Intercept. This works on normal calls, but doesn't work with Queues. In the queue a few agents simultaneously calling, and when INVITE w Replace headers comes it didn't intercept the call. The question is: how to implement Call Group Pickup (Call Intercept) with mod_callcenter ? If it necessary i can add call sip traces for more details. Thanks in advance for any help. Regards Ewgeny. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161115/71e84c49/attachment.html From anthony.minessale at gmail.com Tue Nov 15 19:29:52 2016 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 15 Nov 2016 10:29:52 -0600 Subject: [Freeswitch-users] Memory leak In-Reply-To: References: Message-ID: Good news is 2 releases came out during that test run (and another one in a week or less) so now you can start over testing newest version again. On Tue, Nov 15, 2016 at 3:23 AM, Sergey Safarov wrote: > Hello Anthony > On old FS version memory usage graph is attached. I think memory usage > grows is stoped near 3Gb but last week memory usage is increased about 200Mb > later i will report result on new version and with valgrind results. > [image: pasted1] > > ??, 13 ????. 2016 ?. ? 19:12, Anthony Minessale < > anthony.minessale at gmail.com>: > >> That looks normal to me. >> FreeSWITCH needs a minimum of 2 GB dedicated ram for prolonged use. >> If you chart goes past 2 to 2.5 gigs, you may have a problem then. >> >> you can run valgrind but you can only run 1 call at a time testing >> typical callflow. >> >> valgrind --tool=memcheck --log-file=vg.log --leak-check=full >> --leak-resolution=high --show-reachable=yes /path/to/freeswitch -vg >> >> >> Also when those months are over, hopefully you update cos you can keep >> running the same FS for many months anyway ;) >> >> >> >> >> On Tue, Sep 13, 2016 at 9:29 AM, Sergey Safarov >> wrote: >> >> I has configured FreeSwitch process memory usage graph. >> According this graph durring 24 days size of used memory ingrezed about >> two times. >> [image: FS-memory-chart.png] >> Pastebin of FreeSwitch process memory map is placed at >> https://pastebin.freeswitch.org/view/9e66572a >> Are you have any suggestion how to find memory leak. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 <+1%20919-386-9900> >> >> https://www.youtube.com/watch?v=9XXgW34t40s >> https://www.youtube.com/watch?v=NLaDpGQuZDA >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> 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 https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161115/75c3c910/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: pasted1 Type: image/png Size: 46848 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161115/75c3c910/attachment-0002.png -------------- next part -------------- A non-text attachment was scrubbed... Name: FS-memory-chart.png Type: image/png Size: 29510 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161115/75c3c910/attachment-0003.png From abaci64 at gmail.com Tue Nov 15 19:54:02 2016 From: abaci64 at gmail.com (Abaci B) Date: Tue, 15 Nov 2016 11:54:02 -0500 Subject: [Freeswitch-users] Lua freeswitch.Dbh() question In-Reply-To: References: Message-ID: Basically what I want is the text of the error returned from the database, something like how it's handled in LuaSQL http://keplerproject.github.io/luasql/manual.html#errors pcall is for lua errors in this case there is no lua error. On Tue, Nov 15, 2016 at 2:10 AM, Artur Mega wrote: > Do you need to get text of an error? And to catch it in lua code? Why not > just to check returned value? I dont remember, but returned value can be > nul or item with type `userdata`. If you still want text of an error, i > think you can try to catch it with pcall https://www.lua.org/pil/8.4.html > > 2016-11-14 23:11 GMT+05:00 Abaci B : > >> when runned a query using freeswitch.Dbh dbh:query, if there is results >> it a function for each returned row where you can see the data. however, if >> there is an error you can see the database returned error in the console on >> level ERR but you don't get it back in lua >> is someone aware of a way to get the rror back into lua? >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> 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/20161115/4c57af97/attachment.html From brian at freeswitch.org Tue Nov 15 21:06:55 2016 From: brian at freeswitch.org (Brian West) Date: Tue, 15 Nov 2016 12:06:55 -0600 Subject: [Freeswitch-users] ClueCon Weekly - November 16th 2016 @ 1PM Eastern - Join me on the call. Message-ID: FreeSWITCHers, Tomorrow I'll be presenting on how I fight back against the card member services scams using FreeSWITCH and how effective my methods were and how I uncovered the real company behind it all. https://youtu.be/-gj5kZbUXRI Dial 888 @ https://conference.freeswitch.org/vc/ to join the conversation! -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com (50% Discount using code FreeSwitch50) http://www.freeswitchcookbook.com (50% Discount using code FreeSwitch50) https://www.gofundme.com/freeswitch_ubuntu Got Bugs? Report them here ! | Reddit: /r/freeswitch *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/20161115/a2006367/attachment.html From randomdev4 at gmail.com Tue Nov 15 21:28:10 2016 From: randomdev4 at gmail.com (Tim Smith) Date: Tue, 15 Nov 2016 18:28:10 +0000 Subject: [Freeswitch-users] sofia_glue.c:329 Invalid tls-verify-policy value: none Message-ID: I am running the latest version of freeswitch-stable from the deb repository on Debian 8 .... just like you recommend in the docs. ;-) The default config it installs seems to take issue with the following files : sip_profiles/internal.xml sip_profiles/external-ipv6.xml sip_profiles/external.xml More specifically the line Someone seems to have asked this question before http://lists.freeswitch.org/pipermail/freeswitch-users/2015-July/114881.html But nobody seems to have given an answer at the time ! From brian at freeswitch.org Tue Nov 15 21:48:52 2016 From: brian at freeswitch.org (Brian West) Date: Tue, 15 Nov 2016 12:48:52 -0600 Subject: [Freeswitch-users] sofia_glue.c:329 Invalid tls-verify-policy value: none In-Reply-To: References: Message-ID: Because when that feature was added there was no validation of the config option till later on, The defaults had NONE, I decided to leave them as they were so the end user could take action and correct it, Its harmless and doesn't effect anything. /b On Tue, Nov 15, 2016 at 12:28 PM, Tim Smith wrote: > I am running the latest version of freeswitch-stable from the deb > repository on Debian 8 .... just like you recommend in the docs. ;-) > > The default config it installs seems to take issue with the following > files : > > sip_profiles/internal.xml > sip_profiles/external-ipv6.xml > sip_profiles/external.xml > > More specifically the line > > > > > Someone seems to have asked this question before > > http://lists.freeswitch.org/pipermail/freeswitch-users/ > 2015-July/114881.html > > But nobody seems to have given an answer at the time ! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 (50% Discount using code FreeSwitch50) http://www.freeswitchcookbook.com (50% Discount using code FreeSwitch50) https://www.gofundme.com/freeswitch_ubuntu Got Bugs? Report them here ! | Reddit: /r/freeswitch *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/20161115/090b3551/attachment.html From joel at gogii.net Tue Nov 15 22:37:46 2016 From: joel at gogii.net (Joel Serrano) Date: Tue, 15 Nov 2016 11:37:46 -0800 Subject: [Freeswitch-users] freeswitch.org down? Message-ID: Getting: Service Temporarily Unavailable The server is temporarily unable to service your request due to maintenance downtime or capacity problems. Please try again later. ------------------------------ Apache/2.2.22 (Debian) Server at freeswitch.org Port 443 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161115/f23bf7fa/attachment-0001.html From brian at freeswitch.org Tue Nov 15 22:41:06 2016 From: brian at freeswitch.org (Brian West) Date: Tue, 15 Nov 2016 13:41:06 -0600 Subject: [Freeswitch-users] freeswitch.org down? In-Reply-To: References: Message-ID: Its working fine for me, Maybe Ken was doing some work on our side that triggered that. What exactly were you accessing? On Tue, Nov 15, 2016 at 1:37 PM, Joel Serrano wrote: > Getting: > > Service Temporarily Unavailable > > The server is temporarily unable to service your request due to > maintenance downtime or capacity problems. Please try again later. > ------------------------------ > Apache/2.2.22 (Debian) Server at freeswitch.org Port 443 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 (50% Discount using code FreeSwitch50) http://www.freeswitchcookbook.com (50% Discount using code FreeSwitch50) https://www.gofundme.com/freeswitch_ubuntu Got Bugs? Report them here ! | Reddit: /r/freeswitch *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/20161115/3faf8d8c/attachment.html From joel at gogii.net Tue Nov 15 22:44:25 2016 From: joel at gogii.net (Joel Serrano) Date: Tue, 15 Nov 2016 11:44:25 -0800 Subject: [Freeswitch-users] freeswitch.org down? In-Reply-To: References: Message-ID: Tried with https://freeswitch.org/category/releases/ and failed so I tried with https://freeswitch.org ... failed too so I posted. Just tried again and now it is working again. Thanks an sorry for the noise :) Joel. On Tue, Nov 15, 2016 at 11:41 AM, Brian West wrote: > Its working fine for me, Maybe Ken was doing some work on our side that > triggered that. What exactly were you accessing? > > On Tue, Nov 15, 2016 at 1:37 PM, Joel Serrano wrote: > >> Getting: >> >> Service Temporarily Unavailable >> >> The server is temporarily unable to service your request due to >> maintenance downtime or capacity problems. Please try again later. >> ------------------------------ >> Apache/2.2.22 (Debian) Server at freeswitch.org Port 443 >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 (50% Discount using code FreeSwitch50) > http://www.freeswitchcookbook.com (50% Discount using code FreeSwitch50) > https://www.gofundme.com/freeswitch_ubuntu > > Got Bugs? Report them here ! | Reddit: > /r/freeswitch > > *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/20161115/30237340/attachment.html From grcamauer at gmail.com Tue Nov 15 23:50:14 2016 From: grcamauer at gmail.com (Guillermo Ruiz Camauer) Date: Tue, 15 Nov 2016 17:50:14 -0300 Subject: [Freeswitch-users] sofia_glue.c:329 Invalid tls-verify-policy value: none In-Reply-To: References: Message-ID: If you don't use TLS, just choose one of the other options, like "all" and the error will go away. Guillermo On Tue, Nov 15, 2016 at 3:48 PM, Brian West wrote: > Because when that feature was added there was no validation of the config > option till later on, The defaults had NONE, I decided to leave them as > they were so the end user could take action and correct it, Its harmless > and doesn't effect anything. > > /b > > > On Tue, Nov 15, 2016 at 12:28 PM, Tim Smith wrote: > >> I am running the latest version of freeswitch-stable from the deb >> repository on Debian 8 .... just like you recommend in the docs. ;-) >> >> The default config it installs seems to take issue with the following >> files : >> >> sip_profiles/internal.xml >> sip_profiles/external-ipv6.xml >> sip_profiles/external.xml >> >> More specifically the line >> >> >> >> >> Someone seems to have asked this question before >> >> http://lists.freeswitch.org/pipermail/freeswitch-users/2015- >> July/114881.html >> >> But nobody seems to have given an answer at the time ! >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 (50% Discount using code FreeSwitch50) > http://www.freeswitchcookbook.com (50% Discount using code FreeSwitch50) > https://www.gofundme.com/freeswitch_ubuntu > > Got Bugs? Report them here ! | Reddit: > /r/freeswitch > > *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 > -- Guillermo Ruiz Camauer -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161115/2abf160c/attachment-0001.html From randomdev4 at gmail.com Wed Nov 16 00:43:43 2016 From: randomdev4 at gmail.com (Tim Smith) Date: Tue, 15 Nov 2016 21:43:43 +0000 Subject: [Freeswitch-users] sofia_glue.c:329 Invalid tls-verify-policy value: none In-Reply-To: References: Message-ID: May I humbly suggest that if you're going to leave it as an unhandled (but safe) exception that you at least tone down the severity level so it doesn't show up in big bold ERROR red letters in the console ! ;-) On 15 November 2016 at 18:48, Brian West wrote: > Because when that feature was added there was no validation of the config > option till later on, The defaults had NONE, I decided to leave them as > they were so the end user could take action and correct it, Its harmless > and doesn't effect anything. > > /b > > > On Tue, Nov 15, 2016 at 12:28 PM, Tim Smith wrote: > >> I am running the latest version of freeswitch-stable from the deb >> repository on Debian 8 .... just like you recommend in the docs. ;-) >> >> The default config it installs seems to take issue with the following >> files : >> >> sip_profiles/internal.xml >> sip_profiles/external-ipv6.xml >> sip_profiles/external.xml >> >> More specifically the line >> >> >> >> >> Someone seems to have asked this question before >> >> http://lists.freeswitch.org/pipermail/freeswitch-users/2015- >> July/114881.html >> >> But nobody seems to have given an answer at the time ! >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 (50% Discount using code FreeSwitch50) > http://www.freeswitchcookbook.com (50% Discount using code FreeSwitch50) > https://www.gofundme.com/freeswitch_ubuntu > > Got Bugs? Report them here ! | Reddit: > /r/freeswitch > > *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/20161115/6be55b8f/attachment.html From randomdev4 at gmail.com Wed Nov 16 00:44:30 2016 From: randomdev4 at gmail.com (Tim Smith) Date: Tue, 15 Nov 2016 21:44:30 +0000 Subject: [Freeswitch-users] sofia_glue.c:329 Invalid tls-verify-policy value: none In-Reply-To: References: Message-ID: That's the problem. ;-) The erroneous error message was interfering with my diagnostics. I was trying to figure out why TLS isn't working and the error message was throwing me off track ! On 15 November 2016 at 20:50, Guillermo Ruiz Camauer wrote: > If you don't use TLS, just choose one of the other options, like "all" and > the error will go away. > > Guillermo > > On Tue, Nov 15, 2016 at 3:48 PM, Brian West wrote: > >> Because when that feature was added there was no validation of the config >> option till later on, The defaults had NONE, I decided to leave them as >> they were so the end user could take action and correct it, Its harmless >> and doesn't effect anything. >> >> /b >> >> >> On Tue, Nov 15, 2016 at 12:28 PM, Tim Smith wrote: >> >>> I am running the latest version of freeswitch-stable from the deb >>> repository on Debian 8 .... just like you recommend in the docs. ;-) >>> >>> The default config it installs seems to take issue with the following >>> files : >>> >>> sip_profiles/internal.xml >>> sip_profiles/external-ipv6.xml >>> sip_profiles/external.xml >>> >>> More specifically the line >>> >>> >>> >>> >>> Someone seems to have asked this question before >>> >>> http://lists.freeswitch.org/pipermail/freeswitch-users/2015- >>> July/114881.html >>> >>> But nobody seems to have given an answer at the time ! >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 (50% Discount using code FreeSwitch50) >> http://www.freeswitchcookbook.com (50% Discount using code FreeSwitch50) >> https://www.gofundme.com/freeswitch_ubuntu >> >> Got Bugs? Report them here ! | Reddit: >> /r/freeswitch >> >> *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 >> > > > > -- > 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/20161115/035a33f0/attachment.html From brian at freeswitch.org Wed Nov 16 00:47:58 2016 From: brian at freeswitch.org (Brian West) Date: Tue, 15 Nov 2016 15:47:58 -0600 Subject: [Freeswitch-users] sofia_glue.c:329 Invalid tls-verify-policy value: none In-Reply-To: References: Message-ID: The new 1.8 configs have this corrected. /b On Tue, Nov 15, 2016 at 3:43 PM, Tim Smith wrote: > May I humbly suggest that if you're going to leave it as an unhandled (but > safe) exception that you at least tone down the severity level so it > doesn't show up in big bold ERROR red letters in the console ! ;-) > > On 15 November 2016 at 18:48, Brian West wrote: > >> Because when that feature was added there was no validation of the config >> option till later on, The defaults had NONE, I decided to leave them as >> they were so the end user could take action and correct it, Its harmless >> and doesn't effect anything. >> >> /b >> >> >> On Tue, Nov 15, 2016 at 12:28 PM, Tim Smith wrote: >> >>> I am running the latest version of freeswitch-stable from the deb >>> repository on Debian 8 .... just like you recommend in the docs. ;-) >>> >>> The default config it installs seems to take issue with the following >>> files : >>> >>> sip_profiles/internal.xml >>> sip_profiles/external-ipv6.xml >>> sip_profiles/external.xml >>> >>> More specifically the line >>> >>> >>> >>> >>> Someone seems to have asked this question before >>> >>> http://lists.freeswitch.org/pipermail/freeswitch-users/2015- >>> July/114881.html >>> >>> But nobody seems to have given an answer at the time ! >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 (50% Discount using code FreeSwitch50) >> http://www.freeswitchcookbook.com (50% Discount using code FreeSwitch50) >> https://www.gofundme.com/freeswitch_ubuntu >> >> Got Bugs? Report them here ! | Reddit: >> /r/freeswitch >> >> *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 (50% Discount using code FreeSwitch50) http://www.freeswitchcookbook.com (50% Discount using code FreeSwitch50) https://www.gofundme.com/freeswitch_ubuntu Got Bugs? Report them here ! | Reddit: /r/freeswitch *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/20161115/0bb41ff9/attachment-0001.html From randomdev4 at gmail.com Wed Nov 16 01:53:47 2016 From: randomdev4 at gmail.com (Tim Smith) Date: Tue, 15 Nov 2016 22:53:47 +0000 Subject: [Freeswitch-users] Can't get SRTP working with Freeswitch Message-ID: Debian GNU/Linux 8 (jessie) Linux vxf 3.16.0-4-amd64 #1 SMP Debian 3.16.36-1+deb8u2 (2016-10-19) x86_64 GNU/Linux FreeSWITCH Version 1.6.12-20-b91a0a6~64bit (-20-b91a0a6 64bit) I'm trying to get a simple freeswitch setup working with a softphone. The problem I am having is when I dial the FS local talking clock (9170), I can hear the time when I just have TLS enabled, but when I have TLS transport and call-encryption (SRTP) enabled, I can't hear the time any more ? This is the fs_cli log when using just TLS : http://pastebin.com/yUuS6rWX This is the fs_cli log with TLS and SRTP: http://pastebin.com/ydi6Rkyx From brian at freeswitch.org Wed Nov 16 02:27:53 2016 From: brian at freeswitch.org (Brian West) Date: Tue, 15 Nov 2016 17:27:53 -0600 Subject: [Freeswitch-users] Can't get SRTP working with Freeswitch In-Reply-To: References: Message-ID: Nothing really stands out there, Have you restarted Bria before testing? On Tue, Nov 15, 2016 at 4:53 PM, Tim Smith wrote: > Debian GNU/Linux 8 (jessie) > Linux vxf 3.16.0-4-amd64 #1 SMP Debian 3.16.36-1+deb8u2 (2016-10-19) > x86_64 GNU/Linux > FreeSWITCH Version 1.6.12-20-b91a0a6~64bit (-20-b91a0a6 64bit) > > I'm trying to get a simple freeswitch setup working with a softphone. > > The problem I am having is when I dial the FS local talking clock > (9170), I can hear the time when I just have TLS enabled, but when I > have TLS transport and call-encryption (SRTP) enabled, I can't hear > the time any more ? > > This is the fs_cli log when using just TLS : > http://pastebin.com/yUuS6rWX > > This is the fs_cli log with TLS and SRTP: > http://pastebin.com/ydi6Rkyx > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 (50% Discount using code FreeSwitch50) http://www.freeswitchcookbook.com (50% Discount using code FreeSwitch50) https://www.gofundme.com/freeswitch_ubuntu Got Bugs? Report them here ! | Reddit: /r/freeswitch *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/20161115/8c144bf3/attachment.html From randomdev4 at gmail.com Wed Nov 16 11:06:16 2016 From: randomdev4 at gmail.com (Tim Smith) Date: Wed, 16 Nov 2016 08:06:16 +0000 Subject: [Freeswitch-users] Can't get SRTP working with Freeswitch In-Reply-To: References: Message-ID: Yeah, its really odd. Bria is all up to date too at v4.6. On 15 November 2016 at 23:27, Brian West wrote: > Nothing really stands out there, Have you restarted Bria before testing? > > On Tue, Nov 15, 2016 at 4:53 PM, Tim Smith wrote: > >> Debian GNU/Linux 8 (jessie) >> Linux vxf 3.16.0-4-amd64 #1 SMP Debian 3.16.36-1+deb8u2 (2016-10-19) >> x86_64 GNU/Linux >> FreeSWITCH Version 1.6.12-20-b91a0a6~64bit (-20-b91a0a6 64bit) >> >> I'm trying to get a simple freeswitch setup working with a softphone. >> >> The problem I am having is when I dial the FS local talking clock >> (9170), I can hear the time when I just have TLS enabled, but when I >> have TLS transport and call-encryption (SRTP) enabled, I can't hear >> the time any more ? >> >> This is the fs_cli log when using just TLS : >> http://pastebin.com/yUuS6rWX >> >> This is the fs_cli log with TLS and SRTP: >> http://pastebin.com/ydi6Rkyx >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 (50% Discount using code FreeSwitch50) > http://www.freeswitchcookbook.com (50% Discount using code FreeSwitch50) > https://www.gofundme.com/freeswitch_ubuntu > > Got Bugs? Report them here ! | Reddit: > /r/freeswitch > > *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/20161116/9a994af3/attachment.html From silviu.cpp at gmail.com Wed Nov 16 12:32:04 2016 From: silviu.cpp at gmail.com (Caragea Silviu) Date: Wed, 16 Nov 2016 11:32:04 +0200 Subject: [Freeswitch-users] problems with playback between v1.6.9 and last 1.6 version Message-ID: Hello, I have an app that's playing several files once the call is answered: On v1.6.9 or older works perfect. I tried on last 1.6.x master form today and only first file I can hear being played. All other are not played and in logs I can see very fast: switch_ivr_play_say.c:1910 done playing file /usr/local/woow-vm/sounds/en/1.wav On the v1.6.9 this message is triggered at the end of the file. I will try to find the commit that broken this but help is appreciated. Silviu -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161116/6d0e27ee/attachment.html From silviu.cpp at gmail.com Wed Nov 16 13:22:37 2016 From: silviu.cpp at gmail.com (Caragea Silviu) Date: Wed, 16 Nov 2016 12:22:37 +0200 Subject: [Freeswitch-users] problems with playback between v1.6.9 and last 1.6 version In-Reply-To: References: Message-ID: I found the bug. Seems this commit is breaking my IVR feature: 7c80f6c717f8b2538c843d6743521cfe6471ccd8 (FS-9705 #resolve [Files using prebuffer do not play properly when seeking back to the beginning once the file is done playing]) Silviu On Wed, Nov 16, 2016 at 11:32 AM, Caragea Silviu wrote: > Hello, > > I have an app that's playing several files once the call is answered: > > > > > > > > On v1.6.9 or older works perfect. I tried on last 1.6.x master form today > > and only first file I can hear being played. All other are not played and in logs > > I can see very fast: > > switch_ivr_play_say.c:1910 done playing file /usr/local/woow-vm/sounds/en/1.wav > > On the v1.6.9 this message is triggered at the end of the file. > > I will try to find the commit that broken this but help is appreciated. > > Silviu > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161116/bdf866be/attachment-0001.html From vma at 440hz.fr Wed Nov 16 14:33:14 2016 From: vma at 440hz.fr (Vallimamod Abdullah) Date: Wed, 16 Nov 2016 12:33:14 +0100 Subject: [Freeswitch-users] problems with playback between v1.6.9 and last 1.6 version In-Reply-To: References: Message-ID: Hi, You can also play all the files with one call to callback app with the the file_string url (https://wiki.freeswitch.org/wiki/File_string) like this: Even if it does not solve the core issue, it his may help as a temporary workaround (if it works.) Best Regards, Vallimamod . > On 16 Nov 2016, at 10:32, Caragea Silviu wrote: > > Hello, > > I have an app that's playing several files once the call is answered: > > > > > > > > On v1.6.9 or older works perfect. I tried on last 1.6.x master form today > and only first file I can hear being played. All other are not played and in logs > I can see very fast: > > switch_ivr_play_say.c:1910 done playing file /usr/local/woow-vm/sounds/en/1.wav > > On the v1.6.9 this message is triggered at the end of the file. > > I will try to find the commit that broken this but help is appreciated. > > Silviu > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 evoip at ukr.net Wed Nov 16 14:56:32 2016 From: evoip at ukr.net (Ewgeny) Date: Wed, 16 Nov 2016 13:56:32 +0200 Subject: [Freeswitch-users] Problem with CALL PICKUP (intercept) in QUEUES (mod_callcenter) In-Reply-To: <7a9f66b1-cca1-9d01-369c-a2c6d26ceb16@ukr.net> References: <7a9f66b1-cca1-9d01-369c-a2c6d26ceb16@ukr.net> Message-ID: <2a6bef5e-9238-dc62-fc43-59063e37d4fb@ukr.net> Some more information about the issue with group pickup. Why intercept doesn't work with Queues ? 15.11.2016 17:47, Ewgeny ?????: > > Hi ! > > First about terminology: *Call Pickup*the ability to pull a ringing > call to the phone you are currently on. > > Call Pickup = Call Intercept = Call group pickup. > > > We're using FreeSWITCH version: 1.6.8~64bit in our complex telephony > system with Kamailio and other SIP services. > > The problem with call pickup (intercept) when using a Queue > (mod_callcenter). > > The group pickup scheme we are using described here: > http://www.tech-invite.com/fo-sip/tinv-fo-sip-service-16.html. > > We do SUBSCRIBE (3 see link) to some our service - that return > information about the call (call legs) in NOTIFY with XML body (5). > > Then we do an INVITE with REPLACES (7) that actually do the Intercept. > > This works on normal calls, but doesn't work with Queues. > > In the queue a few agents simultaneously calling, and when INVITE w > Replace headers comes it didn't intercept the call. > > The question is: how to implement Call Group Pickup (Call Intercept) > with mod_callcenter ? > > If it necessary i can add call sip traces for more details. > > Thanks in advance for any help. > > > Regards Ewgeny. > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161116/91eb0662/attachment-0001.html -------------- next part -------------- Call flow with normal Group Pickup Here: - UA1 call to UA2; - UA2 - ringing 180; - UA3 (Intercepter) - send INVITE with replaces to Freeswitch - to intercept UA2; - dialog established with UA1 & UA3; - UA2 - was CANCELED; - Result: successful Gr. Pickup; UA1 Freeswitch UA2 ???????????????????? ???????????????????? ???????????????????? ? INVITE (SDP) ? ? ? ??????????????????????????> ? ? ? 100 Trying ? ? ? ? ? ? 100 trying -- your call is ? ? ? | ? ? ? ? ? ? ? 200 OK (SDP) ? ? ? CANCEL ? ? ? ??????????????????????????> ? ? ? 200 canceling ? ? ? ? ? ? ? ? Call flow with Group Pickup in Queues (mod_callcenter) Here: - UA1 call to Queue some number, the Queue has 2 agents. The Queue is based on mod_callcenter; - Queue Agent 1 & Queue Agent 2 - are ringing; - UA3 (Intercepter) - send INVITE with replaces to Freeswitch - to intercept Queue Agent 1; - for both queue agents are sent CANCEL. To UA3 (Intercepter) sent "502 Bad Gateway"; - In Freeswitch console logs: "Reason: Q.850;cause=27;text="DESTINATION_OUT_OF_ORDER"; - Result: Gr. Pickup failed; UA1 Freeswitch (Queue mod_callcenter) Queue Agent 1 Queue Agent 2 ???????????????????? ???????????????????? ???????????????????? ???????????????????? ? INVITE (SDP) ? ? ? ? ??????????????????????????> ? ? ? ? 100 Trying ? ? ? ? ? ? ? ? ? ? ? 180 Ringing ? 180/183 Session Progress ? ? ? ? INVITE (SDP) ? ? ? | ? ? CANCEL ? ? ? ????????????????????????????????????????????????????> ? ? ? 200 canceling ? ? ? ????????????????????????????????????????????????????> ? ? ? ? ? ? ? ? ? ? ? ? CANCEL ? ? ? ??????????????????????????> ? ? ? 200 canceling ? ? ? ? ? 502 Bad Gateway ? ? ???????????????????????????????????????????????????????????????????????????????????> ? ? ACK ? ? ???????????????????????????????????????????????????????????????????????????????????> ? ? ? ? ? ? From evoip at ukr.net Wed Nov 16 15:02:25 2016 From: evoip at ukr.net (Ewgeny) Date: Wed, 16 Nov 2016 14:02:25 +0200 Subject: [Freeswitch-users] Problem with CALL PICKUP (intercept) in QUEUES (mod_callcenter) In-Reply-To: <2a6bef5e-9238-dc62-fc43-59063e37d4fb@ukr.net> References: <7a9f66b1-cca1-9d01-369c-a2c6d26ceb16@ukr.net> <2a6bef5e-9238-dc62-fc43-59063e37d4fb@ukr.net> Message-ID: P.S. see attached file with call-flows. Thanks in advance for any help. Regards Ewgeny. 16.11.2016 13:56, Ewgeny ?????: > > Some more information about the issue with group pickup. > > Why intercept doesn't work with Queues ? > > > 15.11.2016 17:47, Ewgeny ?????: >> >> Hi ! >> >> First about terminology: *Call Pickup*the ability to pull a ringing >> call to the phone you are currently on. >> >> Call Pickup = Call Intercept = Call group pickup. >> >> >> We're using FreeSWITCH version: 1.6.8~64bit in our complex telephony >> system with Kamailio and other SIP services. >> >> The problem with call pickup (intercept) when using a Queue >> (mod_callcenter). >> >> The group pickup scheme we are using described here: >> http://www.tech-invite.com/fo-sip/tinv-fo-sip-service-16.html. >> >> We do SUBSCRIBE (3 see link) to some our service - that return >> information about the call (call legs) in NOTIFY with XML body (5). >> >> Then we do an INVITE with REPLACES (7) that actually do the Intercept. >> >> This works on normal calls, but doesn't work with Queues. >> >> In the queue a few agents simultaneously calling, and when INVITE w >> Replace headers comes it didn't intercept the call. >> >> The question is: how to implement Call Group Pickup (Call Intercept) >> with mod_callcenter ? >> >> If it necessary i can add call sip traces for more details. >> >> Thanks in advance for any help. >> >> >> Regards Ewgeny. >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> 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/20161116/5200275b/attachment.html From mike at jerris.com Wed Nov 16 18:26:32 2016 From: mike at jerris.com (Michael Jerris) Date: Wed, 16 Nov 2016 10:26:32 -0500 Subject: [Freeswitch-users] problems with playback between v1.6.9 and last 1.6 version In-Reply-To: References: Message-ID: please file a bug at https://freeswitch.org/jira > On Nov 16, 2016, at 5:22 AM, Caragea Silviu wrote: > > I found the bug. Seems this commit is breaking my IVR feature: 7c80f6c717f8b2538c843d6743521cfe6471ccd8 (FS-9705 #resolve [Files using prebuffer do not play properly when seeking back to the beginning once the file is done playing]) > > Silviu > > On Wed, Nov 16, 2016 at 11:32 AM, Caragea Silviu > wrote: > Hello, > > I have an app that's playing several files once the call is answered: > > > > > > > > On v1.6.9 or older works perfect. I tried on last 1.6.x master form today > and only first file I can hear being played. All other are not played and in logs > I can see very fast: > > switch_ivr_play_say.c:1910 done playing file /usr/local/woow-vm/sounds/en/1.wav > > On the v1.6.9 this message is triggered at the end of the file. > > I will try to find the commit that broken this but help is appreciated. > > Silviu > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161116/45c4c8b6/attachment.html From brian at freeswitch.org Wed Nov 16 18:35:22 2016 From: brian at freeswitch.org (Brian West) Date: Wed, 16 Nov 2016 09:35:22 -0600 Subject: [Freeswitch-users] problems with playback between v1.6.9 and last 1.6 version In-Reply-To: References: Message-ID: Can you provide the FULL extension? On Wed, Nov 16, 2016 at 3:32 AM, Caragea Silviu wrote: > Hello, > > I have an app that's playing several files once the call is answered: > > > > > > > > On v1.6.9 or older works perfect. I tried on last 1.6.x master form today > > and only first file I can hear being played. All other are not played and in logs > > I can see very fast: > > switch_ivr_play_say.c:1910 done playing file /usr/local/woow-vm/sounds/en/1.wav > > On the v1.6.9 this message is triggered at the end of the file. > > I will try to find the commit that broken this but help is appreciated. > > Silviu > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 (50% Discount using code FreeSwitch50) http://www.freeswitchcookbook.com (50% Discount using code FreeSwitch50) https://www.gofundme.com/freeswitch_ubuntu Got Bugs? Report them here ! | Reddit: /r/freeswitch *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/20161116/f7ce830a/attachment-0001.html From brian at freeswitch.org Wed Nov 16 18:37:00 2016 From: brian at freeswitch.org (Brian West) Date: Wed, 16 Nov 2016 09:37:00 -0600 Subject: [Freeswitch-users] problems with playback between v1.6.9 and last 1.6 version In-Reply-To: References: Message-ID: I can not replicate this with this extension: On Wed, Nov 16, 2016 at 9:35 AM, Brian West wrote: > Can you provide the FULL extension? > > On Wed, Nov 16, 2016 at 3:32 AM, Caragea Silviu > wrote: > >> Hello, >> >> I have an app that's playing several files once the call is answered: >> >> >> >> >> >> >> >> On v1.6.9 or older works perfect. I tried on last 1.6.x master form today >> >> and only first file I can hear being played. All other are not played and in logs >> >> I can see very fast: >> >> switch_ivr_play_say.c:1910 done playing file /usr/local/woow-vm/sounds/en/1.wav >> >> On the v1.6.9 this message is triggered at the end of the file. >> >> I will try to find the commit that broken this but help is appreciated. >> >> Silviu >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 (50% Discount using code FreeSwitch50) > http://www.freeswitchcookbook.com (50% Discount using code FreeSwitch50) > https://www.gofundme.com/freeswitch_ubuntu > > Got Bugs? Report them here ! | Reddit: > /r/freeswitch > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com (50% Discount using code FreeSwitch50) http://www.freeswitchcookbook.com (50% Discount using code FreeSwitch50) https://www.gofundme.com/freeswitch_ubuntu Got Bugs? Report them here ! | Reddit: /r/freeswitch *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/20161116/a82400cf/attachment.html From jm at mayfirst.org Wed Nov 16 18:54:57 2016 From: jm at mayfirst.org (Jamie McClelland) Date: Wed, 16 Nov 2016 10:54:57 -0500 Subject: [Freeswitch-users] permissions for tmute Message-ID: <20161116155457.hrvaf74bdzvk5qpk@mayfirst.org> Hi all - Thanks to the devs for the excellent verto code! After a year in use, I'm planning to improve my verto web app by allowing administrators to mute or kick *other* participants from the call. I'm following these instructions: https://evoluxbr.github.io/verto-docs/tut/sending-conference-commands.html Thankfully, when I tried it as a regular logged in user that has the proper permissions to join the conference, I received a permission denied message. Now I'd like to login as a user with the proper permissions to mute another conference participant, but I'm not sure how to set those permissions. The regular conference participant has the following: What would I add/change for a user that has permission to mute or kick another conference participant? thanks! jamie -- May First/People Link Growing networks to build a just world http://www.mayfirst.org https://support.mayfirst.org OpenPGP Key: http://current.workingdirectory.net/pages/identity/ xmpp: jamie at mayfirst.org -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 801 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161116/569e4f2e/attachment.bin From brian at freeswitch.org Wed Nov 16 19:33:28 2016 From: brian at freeswitch.org (Brian West) Date: Wed, 16 Nov 2016 10:33:28 -0600 Subject: [Freeswitch-users] permissions for tmute In-Reply-To: <20161116155457.hrvaf74bdzvk5qpk@mayfirst.org> References: <20161116155457.hrvaf74bdzvk5qpk@mayfirst.org> Message-ID: You need the moderator flag, see the vanilla config. /b On Wed, Nov 16, 2016 at 9:54 AM, Jamie McClelland wrote: > Hi all - Thanks to the devs for the excellent verto code! > > After a year in use, I'm planning to improve my verto web app by > allowing administrators to mute or kick *other* participants from the > call. > > I'm following these instructions: > > https://evoluxbr.github.io/verto-docs/tut/sending-conference-commands.html > > Thankfully, when I tried it as a regular logged in user that has the proper > permissions to join the conference, I received a permission denied message. > > Now I'd like to login as a user with the proper permissions to mute another > conference participant, but I'm not sure how to set those permissions. > > The regular conference participant has the following: > > > > > > > What would I add/change for a user that has permission to mute or kick > another > conference participant? > > thanks! > > jamie > > -- > May First/People Link > Growing networks to build a just world > http://www.mayfirst.org > https://support.mayfirst.org > > OpenPGP Key: http://current.workingdirectory.net/pages/identity/ > xmpp: jamie at mayfirst.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 (50% Discount using code FreeSwitch50) http://www.freeswitchcookbook.com (50% Discount using code FreeSwitch50) https://www.gofundme.com/freeswitch_ubuntu Got Bugs? Report them here ! | Reddit: /r/freeswitch *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/20161116/0e5e1f6d/attachment-0001.html From mike at jerris.com Wed Nov 16 19:37:44 2016 From: mike at jerris.com (Michael Jerris) Date: Wed, 16 Nov 2016 11:37:44 -0500 Subject: [Freeswitch-users] permissions for tmute In-Reply-To: <20161116155457.hrvaf74bdzvk5qpk@mayfirst.org> References: <20161116155457.hrvaf74bdzvk5qpk@mayfirst.org> Message-ID: <4FF85CF6-4E65-460D-B927-EC60849DC13B@jerris.com> https://freeswitch.org/confluence/display/FREESWITCH/mod_conference member flag moderator? > On Nov 16, 2016, at 10:54 AM, Jamie McClelland wrote: > > Hi all - Thanks to the devs for the excellent verto code! > > After a year in use, I'm planning to improve my verto web app by > allowing administrators to mute or kick *other* participants from the > call. > > I'm following these instructions: > > https://evoluxbr.github.io/verto-docs/tut/sending-conference-commands.html > > Thankfully, when I tried it as a regular logged in user that has the proper > permissions to join the conference, I received a permission denied message. > > Now I'd like to login as a user with the proper permissions to mute another > conference participant, but I'm not sure how to set those permissions. > > The regular conference participant has the following: > > > > > > > What would I add/change for a user that has permission to mute or kick another > conference participant? > > thanks! > > jamie -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161116/c72b3747/attachment.html From silviu.cpp at gmail.com Wed Nov 16 19:41:39 2016 From: silviu.cpp at gmail.com (Caragea Silviu) Date: Wed, 16 Nov 2016 18:41:39 +0200 Subject: [Freeswitch-users] problems with playback between v1.6.9 and last 1.6 version In-Reply-To: References: Message-ID: Can you replicate with the sound files from here: https://dl.dropboxusercontent.com/u/30548263/en.7z Silviu On Wed, Nov 16, 2016 at 5:37 PM, Brian West wrote: > I can not replicate this with this extension: > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > On Wed, Nov 16, 2016 at 9:35 AM, Brian West wrote: > >> Can you provide the FULL extension? >> >> On Wed, Nov 16, 2016 at 3:32 AM, Caragea Silviu >> wrote: >> >>> Hello, >>> >>> I have an app that's playing several files once the call is answered: >>> >>> >>> >>> >>> >>> >>> >>> On v1.6.9 or older works perfect. I tried on last 1.6.x master form today >>> >>> and only first file I can hear being played. All other are not played and in logs >>> >>> I can see very fast: >>> >>> switch_ivr_play_say.c:1910 done playing file /usr/local/woow-vm/sounds/en/1.wav >>> >>> On the v1.6.9 this message is triggered at the end of the file. >>> >>> I will try to find the commit that broken this but help is appreciated. >>> >>> Silviu >>> >>> >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 (50% Discount using code FreeSwitch50) >> http://www.freeswitchcookbook.com (50% Discount using code FreeSwitch50) >> https://www.gofundme.com/freeswitch_ubuntu >> >> Got Bugs? Report them here ! | Reddit: >> /r/freeswitch >> >> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >> > > > > -- > > *Brian West* > brian at freeswitch.org > > > *Twitter: @FreeSWITCH , @briankwest* > http://www.freeswitchbook.com (50% Discount using code FreeSwitch50) > http://www.freeswitchcookbook.com (50% Discount using code FreeSwitch50) > https://www.gofundme.com/freeswitch_ubuntu > > Got Bugs? Report them here ! | Reddit: > /r/freeswitch > > *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/20161116/0c81314c/attachment-0001.html From anthony.minessale at gmail.com Wed Nov 16 19:50:30 2016 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 16 Nov 2016 10:50:30 -0600 Subject: [Freeswitch-users] problems with playback between v1.6.9 and last 1.6 version In-Reply-To: References: Message-ID: 1) There was another commit to address that so you should update again. 2) *Please* do not report bugs on this mailing list, bugs are supposed to be reported at https://freeswitch.org/jira 3) Do not use the mailing list to ask if you think something is a bug or not, if there is any question its a bug or a feature request or anything to do with commits or changing code. Address the issue on JIRA. On Wed, Nov 16, 2016 at 10:41 AM, Caragea Silviu wrote: > Can you replicate with the sound files from here: > > https://dl.dropboxusercontent.com/u/30548263/en.7z > > Silviu > > On Wed, Nov 16, 2016 at 5:37 PM, Brian West wrote: > >> I can not replicate this with this extension: >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> On Wed, Nov 16, 2016 at 9:35 AM, Brian West wrote: >> >>> Can you provide the FULL extension? >>> >>> On Wed, Nov 16, 2016 at 3:32 AM, Caragea Silviu >>> wrote: >>> >>>> Hello, >>>> >>>> I have an app that's playing several files once the call is answered: >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> On v1.6.9 or older works perfect. I tried on last 1.6.x master form today >>>> >>>> and only first file I can hear being played. All other are not played and in logs >>>> >>>> I can see very fast: >>>> >>>> switch_ivr_play_say.c:1910 done playing file /usr/local/woow-vm/sounds/en/1.wav >>>> >>>> On the v1.6.9 this message is triggered at the end of the file. >>>> >>>> I will try to find the commit that broken this but help is appreciated. >>>> >>>> Silviu >>>> >>>> >>>> >>>> ____________________________________________________________ >>>> _____________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>> switch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> >>> -- >>> >>> *Brian West* >>> brian at freeswitch.org >>> >>> >>> *Twitter: @FreeSWITCH , @briankwest* >>> http://www.freeswitchbook.com (50% Discount using code FreeSwitch50) >>> http://www.freeswitchcookbook.com (50% Discount using code FreeSwitch50) >>> https://www.gofundme.com/freeswitch_ubuntu >>> >>> Got Bugs? Report them here ! | Reddit: >>> /r/freeswitch >>> >>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >>> >> >> >> >> -- >> >> *Brian West* >> brian at freeswitch.org >> >> >> *Twitter: @FreeSWITCH , @briankwest* >> http://www.freeswitchbook.com (50% Discount using code FreeSwitch50) >> http://www.freeswitchcookbook.com (50% Discount using code FreeSwitch50) >> https://www.gofundme.com/freeswitch_ubuntu >> >> Got Bugs? Report them here ! | Reddit: >> /r/freeswitch >> >> *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 https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161116/76de6a88/attachment-0001.html From randomdev4 at gmail.com Wed Nov 16 20:16:29 2016 From: randomdev4 at gmail.com (Tim Smith) Date: Wed, 16 Nov 2016 17:16:29 +0000 Subject: [Freeswitch-users] Problems with diaplan var settings Message-ID: Debian GNU/Linux 8 (jessie) Linux vxf 3.16.0-4-amd64 #1 SMP Debian 3.16.36-1+deb8u2 (2016-10-19) x86_64 GNU/Linux FreeSWITCH Version 1.6.12-20-b91a0a6~64bit (-20-b91a0a6 64bit) Can anyone explain to my why this works : And yet this doesn't (nothing happens, no From rewrite, no privacy settings, nothing ... I have tried swapping "set" for "export" but that makes no difference either) : From samir.doshi at inextrix.com Wed Nov 16 20:37:27 2016 From: samir.doshi at inextrix.com (Samir Doshi) Date: Wed, 16 Nov 2016 23:07:27 +0530 Subject: [Freeswitch-users] Issue with cc_warning_tone in mod_callcenter module In-Reply-To: References: Message-ID: Reported https://freeswitch.org/jira/browse/FS-9723 Best Regards -- Samir Doshi *iNextrix Technologie**s Pvt. Ltd*. http://www.inextrix.com *Disclaimer:* The information contained in this communication is confidential and may be legally privileged. It is intended solely for the use of the individual or entity to whom it is addressed and others authorised to receive it. If you are not the intended recipient you are hereby notified that any disclosure, copying, distribution or taking action in reliance of the contents of this information is strictly prohibited and may be unlawful. On Tue, Nov 8, 2016 at 5:45 PM, ?talo Rossi wrote: > Fila a JIRA with your configs and debug logs > > On Tue, Nov 8, 2016 at 3:49 AM, Samir Doshi > wrote: > >> Hi, >> >> I tried with cc_outbound_announce variable but getting same result. Not >> getting alert tone. >> Any other things should I look? >> >> >> Best Regards >> -- >> Samir Doshi >> *iNextrix Technologie**s Pvt. Ltd*. >> http://www.inextrix.com >> >> >> >> *Disclaimer:* >> The information contained in this communication is confidential and may >> be legally privileged. It is intended solely for the use of the individual >> or entity to whom it is addressed and others authorised to receive it. If >> you are not the intended recipient you are hereby notified that any >> disclosure, copying, distribution or taking action in reliance of the >> contents of this information is strictly prohibited and may be unlawful. >> >> On Sun, Nov 6, 2016 at 3:30 PM, Samir Doshi >> wrote: >> >>> Thanks. I will give a try and get back to you with result. >>> >>> >>> Best Regards >>> -- >>> Samir Doshi >>> *iNextrix Technologie**s Pvt. Ltd*. >>> http://www.inextrix.com >>> >>> >>> >>> *Disclaimer:* >>> The information contained in this communication is confidential and may >>> be legally privileged. It is intended solely for the use of the individual >>> or entity to whom it is addressed and others authorised to receive it. If >>> you are not the intended recipient you are hereby notified that any >>> disclosure, copying, distribution or taking action in reliance of the >>> contents of this information is strictly prohibited and may be unlawful. >>> >>> On Sat, Nov 5, 2016 at 1:31 AM, ?talo Rossi >>> wrote: >>> >>>> mod_callcenter.c:1776 Agent 8003 answered "Outbound Call" >>>> from queue agent8003 at default (Recorded) >>>> >>>> From this line you're using callback agents and the variable should be >>>> cc_outbound_announce instead of cc_warning_tone, this one is specific for >>>> uuid-standby agents >>>> >>>> On Fri, Nov 4, 2016 at 3:36 PM, Samir Doshi >>>> wrote: >>>> >>>>> Hi, >>>>> >>>>> We are doing impementation using mod_callcenter module by taking >>>>> reference from documentation (https://freeswitch.org/conflu >>>>> ence/display/FREESWITCH/mod_callcenter). So far everything works well >>>>> except cc_warning_tone playback. We are never getting tone when some >>>>> connect to agent. >>>>> >>>>> Debug Logs : >>>>> >>>>> FS cli Log : https://pastebin.freeswitch.org/view/f9be26b9 >>>>> Log with sip debug : https://pastebin.freeswitch.org/view/aab8ea85 >>>>> Log with sip debug + loglevel 9 : https://pastebin.freeswitch.or >>>>> g/view/ff793521 >>>>> >>>>> Any hint? >>>>> >>>>> >>>>> Best Regards >>>>> -- >>>>> Samir Doshi >>>>> *iNextrix Technologie**s Pvt. Ltd*. >>>>> http://www.inextrix.com >>>>> >>>>> >>>>> >>>>> *Disclaimer:* >>>>> The information contained in this communication is confidential and >>>>> may be legally privileged. It is intended solely for the use of the >>>>> individual or entity to whom it is addressed and others authorised to >>>>> receive it. If you are not the intended recipient you are hereby notified >>>>> that any disclosure, copying, distribution or taking action in reliance of >>>>> the contents of this information is strictly prohibited and may be unlawful. >>>>> >>>>> ____________________________________________________________ >>>>> _____________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>>> switch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> >>>> -- >>>> ?talo Rossi >>>> italo at freeswitch.org >>>> >>>> ____________________________________________________________ >>>> _____________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>> switch-users >>>> 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 > italo at freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161116/204b983b/attachment-0001.html From samir.doshi at inextrix.com Wed Nov 16 20:38:26 2016 From: samir.doshi at inextrix.com (Samir Doshi) Date: Wed, 16 Nov 2016 23:08:26 +0530 Subject: [Freeswitch-users] Export CDR on failed tried calls In-Reply-To: References: Message-ID: Thanks. Its exporting cdr for failed trial. Best Regards -- Samir Doshi *iNextrix Technologie**s Pvt. Ltd*. http://www.inextrix.com *Disclaimer:* The information contained in this communication is confidential and may be legally privileged. It is intended solely for the use of the individual or entity to whom it is addressed and others authorised to receive it. If you are not the intended recipient you are hereby notified that any disclosure, copying, distribution or taking action in reliance of the contents of this information is strictly prohibited and may be unlawful. On Wed, Nov 9, 2016 at 11:13 PM, Michael Jerris wrote: > https://freeswitch.org/confluence/display/FREESWITCH/mod_json_cdr > > > > > > On Nov 6, 2016, at 5:13 AM, Samir Doshi wrote: > > Hi Guys, > > Wondering if we can post failed tried call cdr to mod_json_cdr. I have > below dialplan generated and I want to post cdr any gateway fail to process > the call. That means if test1 fail then it should send cdr to mod_json_cdr > and then go for test2. If test2 fail then post cdr and then try test3 so > on. > > > >
> > > > > > > > > > > > > > > >
> > > Is there any variable or configuration needs to set? > > > Best Regards > -- > Samir Doshi > *iNextrix Technologie**s Pvt. Ltd*. > http://www.inextrix.com > > > > *Disclaimer:* > The information contained in this communication is confidential and may be > legally privileged. It is intended solely for the use of the individual or > entity to whom it is addressed and others authorised to receive it. If you > are not the intended recipient you are hereby notified that any disclosure, > copying, distribution or taking action in reliance of the contents of this > information is strictly prohibited and may be unlawful. > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > 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/20161116/9a4e6e5c/attachment.html From jude19love at gmail.com Wed Nov 16 20:50:29 2016 From: jude19love at gmail.com (Jude Mukundane) Date: Wed, 16 Nov 2016 17:50:29 +0000 Subject: [Freeswitch-users] No DTMF in conference Message-ID: Hello All, I have two servers - one running 1.5.15 and another running 1.6.12. In both servers if I issue the originate command on fs_cli "originate user/1000 &conference(h)", I do see DTMF digits on the console for 1.5 but not 1.6. Am using zoiper on the same phone for both servers and the DTMF settings (duration etc) seem to be the same on both servers. I can get DTMF for one to one calls, but not conferences. I googled... Has any one run into this and know how to solve it? Jude -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161116/1d37f3fc/attachment.html From blackc2004 at gmail.com Wed Nov 16 22:03:43 2016 From: blackc2004 at gmail.com (Cj B) Date: Wed, 16 Nov 2016 11:03:43 -0800 Subject: [Freeswitch-users] Verto conference and Yealink T49 video In-Reply-To: <09FB5EEC-246A-4C04-B40E-0319C004CA18@gmail.com> References: <09FB5EEC-246A-4C04-B40E-0319C004CA18@gmail.com> Message-ID: <38F623D6-AED6-429C-A9BC-4E6F891EB813@gmail.com> Finally found my issue here. I hadn?t started mod_av! It works now with video from web->t49! Cj B > On Nov 2, 2016, at 10:39 AM, Cj B wrote: > > Thanks Brian for the quick reply. I added that to my dialplan and am still not getting video from the T49. Here?s a log of the call: http://pastebin.com/zk9TPYfp > > Extension 1007 is the Yealink T49. Any other ideas? > > Thanks for the help! > > Cj B > >> On Nov 2, 2016, at 9:12 AM, Brian West > wrote: >> >> I'm going to guess your canvas size is larger than the phone can support, You'll probably want to set this variable before you answer video_mirror_input=true >> >> /b >> >> On Wed, Nov 2, 2016 at 11:01 AM, Cj B > wrote: >> Hi all, >> >> I have verto working and am able to do video conferences with multiple people. But I was wondering what it takes to get a yealink t49 or other video capable phone to be able to join the conference and share video? I haven?t been able to find anything in documentation or the email list. >> >> I can do T49->T49 video right now, but when they join the conference there?s no video. >> >> Thanks! >> Cj B >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 (50% Discount using code FreeSwitch50) >> http://www.freeswitchcookbook.com (50% Discount using code FreeSwitch50) >> https://www.gofundme.com/freeswitch_ubuntu >> Got Bugs? Report them here ! | Reddit: /r/freeswitch >> 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/20161116/0e95407d/attachment-0001.html From italo at freeswitch.org Wed Nov 16 23:56:41 2016 From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=) Date: Wed, 16 Nov 2016 17:56:41 -0300 Subject: [Freeswitch-users] Problem with CALL PICKUP (intercept) in QUEUES (mod_callcenter) In-Reply-To: References: <7a9f66b1-cca1-9d01-369c-a2c6d26ceb16@ukr.net> <2a6bef5e-9238-dc62-fc43-59063e37d4fb@ukr.net> Message-ID: Bug reports go to JIRA. Don't forget to put debug logs (/log 7 siptrace etc) and your configurations. On Wed, Nov 16, 2016 at 9:02 AM, Ewgeny wrote: > P.S. > > see attached file with call-flows. > > > > > Thanks in advance for any help. > Regards Ewgeny. > > > > 16.11.2016 13:56, Ewgeny ?????: > > Some more information about the issue with group pickup. > > Why intercept doesn't work with Queues ? > > > 15.11.2016 17:47, Ewgeny ?????: > > Hi ! > > First about terminology: *Call Pickup* the ability to pull a ringing call > to the phone you are currently on. > > Call Pickup = Call Intercept = Call group pickup. > > > We're using FreeSWITCH version: 1.6.8~64bit in our complex telephony > system with Kamailio and other SIP services. > > The problem with call pickup (intercept) when using a Queue > (mod_callcenter). > > The group pickup scheme we are using described here: > http://www.tech-invite.com/fo-sip/tinv-fo-sip-service-16.html. > > We do SUBSCRIBE (3 see link) to some our service - that return information > about the call (call legs) in NOTIFY with XML body (5). > > Then we do an INVITE with REPLACES (7) that actually do the Intercept. > > This works on normal calls, but doesn't work with Queues. > > In the queue a few agents simultaneously calling, and when INVITE w > Replace headers comes it didn't intercept the call. > > The question is: how to implement Call Group Pickup (Call Intercept) with > mod_callcenter ? > > If it necessary i can add call sip traces for more details. > > Thanks in advance for any help. > > > Regards Ewgeny. > > > > > _________________________________________________________________________ > 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.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 > -- ?talo Rossi italo at freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161116/ddb443c8/attachment.html From joel at gogii.net Thu Nov 17 01:02:07 2016 From: joel at gogii.net (Joel Serrano) Date: Wed, 16 Nov 2016 14:02:07 -0800 Subject: [Freeswitch-users] Freeswitch with shared MySQL over ODBC Message-ID: Hi, I'm preparing 2 servers with shared MySQL database. I'm not sure if the doc are outdated... are the following 3 settings required in switch.conf.xml: Or is it enough only with "core-db-dsn" ? Should any of them go in the profile xml rather than the switch.conf.xml? What is the difference between "core-db-dsn" and "odbc-dsn" parameters? (in case "odbc-dsn" is still required) The idea is to run "sofia recover" on second node on the event of a failure on the first node, and to continue with call that are in progress. Thanks. Joel. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161116/f49fc106/attachment.html From jm at mayfirst.org Wed Nov 16 22:01:48 2016 From: jm at mayfirst.org (Jamie McClelland) Date: Wed, 16 Nov 2016 14:01:48 -0500 Subject: [Freeswitch-users] permissions for tmute In-Reply-To: <4FF85CF6-4E65-460D-B927-EC60849DC13B@jerris.com> References: <20161116155457.hrvaf74bdzvk5qpk@mayfirst.org> <4FF85CF6-4E65-460D-B927-EC60849DC13B@jerris.com> Message-ID: <20161116190148.rup63nwasctdhlud@mayfirst.org> Thanks Brian and Michael for pointing me to the moderator flag. However, I'm not having luck with it. In my web app, everyone logs into via verto using the same username. I've configured freeswitch to assign one of them the moderator flag (I can confirm using fs_cli that the given conference member has the flag set). However, when that member tries to mute another member via verto I'm still getting permission denied. Is it because all members are logging in as the same user (as defined in the directory)? Or, if this really should be working I can test so more to make sure I'm not making any other mistakes. I'm having a hard time figuring out what permissions belong to a logged in verto user and which permissions belong to a given member of a given conference. thanks, jamie On Wed Nov 16, Michael Jerris wrote: > https://freeswitch.org/confluence/display/FREESWITCH/mod_conference > > member flag moderator? > > > On Nov 16, 2016, at 10:54 AM, Jamie McClelland wrote: > > > > Hi all - Thanks to the devs for the excellent verto code! > > > > After a year in use, I'm planning to improve my verto web app by > > allowing administrators to mute or kick *other* participants from the > > call. > > > > I'm following these instructions: > > > > https://evoluxbr.github.io/verto-docs/tut/sending-conference-commands.html > > > > Thankfully, when I tried it as a regular logged in user that has the proper > > permissions to join the conference, I received a permission denied message. > > > > Now I'd like to login as a user with the proper permissions to mute another > > conference participant, but I'm not sure how to set those permissions. > > > > The regular conference participant has the following: > > > > > > > > > > > > > > What would I add/change for a user that has permission to mute or kick another > > conference participant? > > > > thanks! > > > > jamie > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- May First/People Link Growing networks to build a just world http://www.mayfirst.org https://support.mayfirst.org OpenPGP Key: http://current.workingdirectory.net/pages/identity/ xmpp: jamie at mayfirst.org -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 801 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161116/5bff80f1/attachment.bin From italo at freeswitch.org Thu Nov 17 01:31:17 2016 From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=) Date: Wed, 16 Nov 2016 19:31:17 -0300 Subject: [Freeswitch-users] permissions for tmute In-Reply-To: <20161116190148.rup63nwasctdhlud@mayfirst.org> References: <20161116155457.hrvaf74bdzvk5qpk@mayfirst.org> <4FF85CF6-4E65-460D-B927-EC60849DC13B@jerris.com> <20161116190148.rup63nwasctdhlud@mayfirst.org> Message-ID: Change your dialplan to: On Wed, Nov 16, 2016 at 4:01 PM, Jamie McClelland wrote: > Thanks Brian and Michael for pointing me to the moderator flag. > > However, I'm not having luck with it. > > In my web app, everyone logs into via verto using the same username. > > I've configured freeswitch to assign one of them the moderator flag (I > can confirm using fs_cli that the given conference member has the flag > set). > > However, when that member tries to mute another member via verto I'm > still getting permission denied. > > Is it because all members are logging in as the same user (as defined in > the directory)? > > Or, if this really should be working I can test so more to make sure I'm > not making any other mistakes. > > I'm having a hard time figuring out what permissions belong to a logged > in verto user and which permissions belong to a given member of a given > conference. > > thanks, > jamie > > > On Wed Nov 16, Michael Jerris wrote: > > https://freeswitch.org/confluence/display/FREESWITCH/mod_conference < > https://freeswitch.org/confluence/display/FREESWITCH/mod_conference> > > > > member flag moderator? > > > > > On Nov 16, 2016, at 10:54 AM, Jamie McClelland > wrote: > > > > > > Hi all - Thanks to the devs for the excellent verto code! > > > > > > After a year in use, I'm planning to improve my verto web app by > > > allowing administrators to mute or kick *other* participants from the > > > call. > > > > > > I'm following these instructions: > > > > > > https://evoluxbr.github.io/verto-docs/tut/sending- > conference-commands.html > > > > > > Thankfully, when I tried it as a regular logged in user that has the > proper > > > permissions to join the conference, I received a permission denied > message. > > > > > > Now I'd like to login as a user with the proper permissions to mute > another > > > conference participant, but I'm not sure how to set those permissions. > > > > > > The regular conference participant has the following: > > > > > > > > > > > > > > > > > > > > > What would I add/change for a user that has permission to mute or kick > another > > > conference participant? > > > > > > thanks! > > > > > > jamie > > > > > ____________________________________________________________ > _____________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://confluence.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > -- > May First/People Link > Growing networks to build a just world > http://www.mayfirst.org > https://support.mayfirst.org > > OpenPGP Key: http://current.workingdirectory.net/pages/identity/ > xmpp: jamie at mayfirst.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 italo at freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161116/e1587dfd/attachment-0001.html From mike at jerris.com Thu Nov 17 01:32:19 2016 From: mike at jerris.com (Michael Jerris) Date: Wed, 16 Nov 2016 17:32:19 -0500 Subject: [Freeswitch-users] permissions for tmute In-Reply-To: <20161116190148.rup63nwasctdhlud@mayfirst.org> References: <20161116155457.hrvaf74bdzvk5qpk@mayfirst.org> <4FF85CF6-4E65-460D-B927-EC60849DC13B@jerris.com> <20161116190148.rup63nwasctdhlud@mayfirst.org> Message-ID: <7169CB34-45E4-4E48-962F-A0C489707A74@jerris.com> all you should need to do that i can think of: > On Nov 16, 2016, at 2:01 PM, Jamie McClelland wrote: > > Thanks Brian and Michael for pointing me to the moderator flag. > > However, I'm not having luck with it. > > In my web app, everyone logs into via verto using the same username. > > I've configured freeswitch to assign one of them the moderator flag (I > can confirm using fs_cli that the given conference member has the flag > set). > > However, when that member tries to mute another member via verto I'm > still getting permission denied. > > Is it because all members are logging in as the same user (as defined in > the directory)? > > Or, if this really should be working I can test so more to make sure I'm > not making any other mistakes. > > I'm having a hard time figuring out what permissions belong to a logged > in verto user and which permissions belong to a given member of a given > conference. > > thanks, > jamie > > > On Wed Nov 16, Michael Jerris wrote: >> https://freeswitch.org/confluence/display/FREESWITCH/mod_conference > >> >> member flag moderator? >> >>> On Nov 16, 2016, at 10:54 AM, Jamie McClelland wrote: >>> >>> Hi all - Thanks to the devs for the excellent verto code! >>> >>> After a year in use, I'm planning to improve my verto web app by >>> allowing administrators to mute or kick *other* participants from the >>> call. >>> >>> I'm following these instructions: >>> >>> https://evoluxbr.github.io/verto-docs/tut/sending-conference-commands.html >>> >>> Thankfully, when I tried it as a regular logged in user that has the proper >>> permissions to join the conference, I received a permission denied message. >>> >>> Now I'd like to login as a user with the proper permissions to mute another >>> conference participant, but I'm not sure how to set those permissions. >>> >>> The regular conference participant has the following: >>> >>> >>> >>> >>> >>> >>> What would I add/change for a user that has permission to mute or kick another >>> conference participant? >>> >>> thanks! >>> >>> jamie >> > >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > -- > May First/People Link > Growing networks to build a just world > http://www.mayfirst.org > https://support.mayfirst.org > > OpenPGP Key: http://current.workingdirectory.net/pages/identity/ > xmpp: jamie at mayfirst.org > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161116/f8421068/attachment-0001.html From 568691 at gmail.com Thu Nov 17 02:07:39 2016 From: 568691 at gmail.com (Alexandru Covalschi) Date: Thu, 17 Nov 2016 01:07:39 +0200 Subject: [Freeswitch-users] FAX issues/questions Message-ID: Hello list, In my set up Freeswitch acts as fax originator, so an image is being processed and sent by FS. I do not receive any fax, so all my questions/issues are related to sending it. All fax is over IP. First a question - how to understand if Freeswitch is sending in T.38? Why do I always see T.30 FLOW in log even if in SDP is mentioned T.38? I do see some "T.38 rxfax set to 0/txfax set to N", I understand that rx is receive and tx is transmit, but why any other log strings shows T.30 FLOW? And the issue - sometimes when I initiate fax I get ----------------- Codec RAW Signed Linear (16 bit) encoder error mod_g729.c:102 This codec is only usable in passthrough mode! ----------------- I can't understand why it's happening as it occurs from time to time, but it always happens before Freeswitch makes re-invite to initiate a T.38 session. I suppose that's because I have no license for g729, but the supplier side negotiates it before FS has chance to re-invite. But why does it work with same SDP in other times? Does it have sense to exclude g729 for the initial codec negotiation, or g729 is essential to keep T.38 working? What shall I do in case of Asia where everything is on g729,? Maybe it has sense to purchase a license - but why then it works unstable right now, I mean, why does it even work if it requires mod_g729_com? Or it doesn't? I'm really confused on all that. Thanks in advance! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161117/53368264/attachment.html From brian at freeswitch.org Thu Nov 17 02:13:46 2016 From: brian at freeswitch.org (Brian West) Date: Wed, 16 Nov 2016 17:13:46 -0600 Subject: [Freeswitch-users] FAX issues/questions In-Reply-To: References: Message-ID: On Wed, Nov 16, 2016 at 5:07 PM, Alexandru Covalschi <568691 at gmail.com> wrote: > Hello list, > > In my set up Freeswitch acts as fax originator, so an image is being > processed and sent by FS. I do not receive any fax, so all my > questions/issues are related to sending it. All fax is over IP. > > First a question - how to understand if Freeswitch is sending in T.38? Why > do I always see T.30 FLOW in log even if in SDP is mentioned T.38? I do see > some "T.38 rxfax set to 0/txfax set to N", I understand that rx is receive > and tx is transmit, but why any other log strings shows T.30 FLOW? > Because its still T.30, you're just encapsulating it . > > And the issue - sometimes when I initiate fax I get > ----------------- > Codec RAW Signed Linear (16 bit) encoder error > mod_g729.c:102 This codec is only usable in passthrough mode! > ----------------- > I can't understand why it's happening as it occurs from time to time, but > it always happens before Freeswitch makes re-invite to initiate a T.38 > session. > I suppose that's because I have no license for g729, but the supplier side > negotiates it before FS has chance to re-invite. But why does it work with > same SDP in other times? > Does it have sense to exclude g729 for the initial codec negotiation, or > g729 is essential to keep T.38 working? What shall I do in case of Asia > where everything is on g729,? Maybe it has sense to purchase a license - > but why then it works unstable right now, I mean, why does it even work if > it requires mod_g729_com? Or it doesn't? I'm really confused on all that. > You'll need the codec to negotiate and process the audio before the t.38 re-invite, I do not believe there are any other ways to do this. > > > Thanks 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-users mailing list > FreeSWITCH-users at 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 (50% Discount using code FreeSwitch50) http://www.freeswitchcookbook.com (50% Discount using code FreeSwitch50) https://www.gofundme.com/freeswitch_ubuntu Got Bugs? Report them here ! | Reddit: /r/freeswitch *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/20161116/6e73eab6/attachment.html From 568691 at gmail.com Thu Nov 17 13:40:35 2016 From: 568691 at gmail.com (Alexandru Covalschi) Date: Thu, 17 Nov 2016 12:40:35 +0200 Subject: [Freeswitch-users] FAX issues/questions Message-ID: First of all thanks for answer! It's ok to buy several licenses, but why then it works at all right now? To the same destination 8/10 are successful, 2/10 have that issue with codec. Also, can you point me to any material I can read to understand the T.38 encapsulation? Is that a usual practice or it's just the way FS does that? I'm very "weak" at faxing and maybe I'm asking silly questions, but I really need to understand how things work... Cheers, Alex From lists at kavun.ch Thu Nov 17 15:40:44 2016 From: lists at kavun.ch (Emrah) Date: Thu, 17 Nov 2016 13:40:44 +0100 Subject: [Freeswitch-users] FS-9113, still experiencing TLS crashes In-Reply-To: References: <9D820B3E-294A-49C4-9220-4233FE6994B0@kavun.ch> <0001A843-3487-41C8-824F-1F017015EE20@kavun.ch> <627AB283-362F-4494-B883-1C3D30D7F2F8@cyberfonica.com> Message-ID: <05CC95CF-14D9-478E-A39A-244316D28547@kavun.ch> Stan, we were able to reproduce the issue. You even commented on the jira about it. I went MIA after using the fix provided by Ethan so we never really got to the bottom of it. Could you please share the snapshot of that machine with me? We should take that environment as the base and try the same environment updated to the latest unstable also. Best, Emrah > On Nov 12, 2016, at 10:45 PM, Stanislav Sinyagin wrote: > > We actually built a test server, but weren't able to reproduce the issue. > I can bring it up again if needed. > > > On 10 Nov 2016 20:25, "Emrah" > wrote: > I agree, as long as I get to reproduce it that way. I am suspecting everything here. From the keysize to the CA to the TCP transport getting compromised to openssl not reliably transmitting certain packets to FS. > > Thanks for the suggestion >> On Nov 10, 2016, at 5:58 PM, Alejandro Recarey > wrote: >> >> You could either use a self-signed cert for a nonexistent domain (example.com ?) and modify your hosts file or DNS to point to he server. I think that should give you an environment to reproduce the crash which you could share without leaking your private cert. >> >> >> On 9 Nov 2016, at 20:28, Emrah > wrote: >> >>> It's the "reliably" part that's tricky. >>> I'm using commercial certificates, so let me figure out how to replicate a similar environment. I'll email you the info once I have a setup, and you can circulate where needed. >>> >>> Thanks for helping on this >>>> On Nov 9, 2016, at 4:25 PM, Michael Jerris > wrote: >>>> >>>> I need a recipie to reliably reproduce this so I can dig in the code. Is there a way you can put together an environment where this can be reproduced on demand? >>>> >>>>> On Nov 9, 2016, at 3:39 AM, Emrah > wrote: >>>>> >>>>> No Sir, the response packet to the 407 Proxy Authentication Required is never received. So the session then eventually gets abandoned by FS. On the client side, and this is generalized, the packet is sent, except the TLS session breaks. >>>>> >>>>>> On Nov 8, 2016, at 11:41 PM, Michael Jerris > wrote: >>>>>> >>>>>> Can you confirm if the packet is shown in freeswitch tport_log? >>>>>> >>>>>>> On Nov 8, 2016, at 5:02 PM, Emrah > wrote: >>>>>>> >>>>>>> Hello List, >>>>>>> Thanks to the help provided by Stanislav, I learned of issue #9113, https://freeswitch.org/jira/si/jira.issueviews:issue-html/FS-9113/FS-9113.html , which seems to be related to the issues I have been experiencing with FreeSWITCH, TLS and failed call setups. >>>>>>> Coincidentally, or not, the fix pushed on that issue was aligned with whole months where I did not experience any TLS issues. Calls were going through fine, until all of a sudden they started failing again. This is on 2 distinct servers running a load balanced FS setup, and using Yealink phones. >>>>>>> >>>>>>> To sum up, here is what is going on. >>>>>>> From the Yealink, calls with TLS work if I don't use SRTP. >>>>>>> From the Yealink, calls crash if I use TLS and SRTP. >>>>>>> From my laptop softphone, calls only crash sometimes if I use TLS and SRTP. >>>>>>> >>>>>>> How can I debug the TLS session on the FreeSWITCH side to see what happens with the TLS thread? I don't mean packet capture. >>>>>>> >>>>>>> I have a feeling that the packet size is too large and doesn't make it to the FS box intact after the 407 Proxy Required is received by the client. >>>>>>> >>>>>>> Here is the log for the Yealink: >>>>>>> http://pastebin.com/smKP286x >>>>>>> >>>>>>> Your lights would be so appreciated, I'm losing my mind over 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 _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> 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/20161117/92252c57/attachment-0001.html From evoip at ukr.net Thu Nov 17 16:40:59 2016 From: evoip at ukr.net (Ewgeny) Date: Thu, 17 Nov 2016 15:40:59 +0200 Subject: [Freeswitch-users] Problem with CALL PICKUP (intercept) in QUEUES (mod_callcenter) In-Reply-To: References: <7a9f66b1-cca1-9d01-369c-a2c6d26ceb16@ukr.net> <2a6bef5e-9238-dc62-fc43-59063e37d4fb@ukr.net> Message-ID: <3aa89065-7ecf-4ee3-2cf0-00f30beea831@ukr.net> the Bug report created - https://freeswitch.org/jira/browse/FS-9752 Regards Ewgeny. 16.11.2016 22:56, ?talo Rossi ?????: > Bug reports go to JIRA. > > Don't forget to put debug logs (/log 7 siptrace etc) and your > configurations. > > On Wed, Nov 16, 2016 at 9:02 AM, Ewgeny > wrote: > > P.S. > > see attached file with call-flows. > > > > > Thanks in advance for any help. > > Regards Ewgeny. > > > > 16.11.2016 13:56, Ewgeny ?????: >> >> Some more information about the issue with group pickup. >> >> Why intercept doesn't work with Queues ? >> >> >> 15.11.2016 17:47, Ewgeny ?????: >>> >>> Hi ! >>> >>> First about terminology: *Call Pickup*the ability to pull a >>> ringing call to the phone you are currently on. >>> >>> Call Pickup = Call Intercept = Call group pickup. >>> >>> >>> We're using FreeSWITCH version: 1.6.8~64bit in our complex >>> telephony system with Kamailio and other SIP services. >>> >>> The problem with call pickup (intercept) when using a Queue >>> (mod_callcenter). >>> >>> The group pickup scheme we are using described here: >>> http://www.tech-invite.com/fo-sip/tinv-fo-sip-service-16.html >>> . >>> >>> We do SUBSCRIBE (3 see link) to some our service - that return >>> information about the call (call legs) in NOTIFY with XML body (5). >>> >>> Then we do an INVITE with REPLACES (7) that actually do the >>> Intercept. >>> >>> This works on normal calls, but doesn't work with Queues. >>> >>> In the queue a few agents simultaneously calling, and when >>> INVITE w Replace headers comes it didn't intercept the call. >>> >>> The question is: how to implement Call Group Pickup (Call >>> Intercept) with mod_callcenter ? >>> >>> If it necessary i can add call sip traces for more details. >>> >>> Thanks in advance for any help. >>> >>> >>> Regards Ewgeny. >>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >>> 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 > > -- > ?talo Rossi > italo at freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161117/bac80d62/attachment.html From brian at freeswitch.org Thu Nov 17 17:19:53 2016 From: brian at freeswitch.org (Brian West) Date: Thu, 17 Nov 2016 08:19:53 -0600 Subject: [Freeswitch-users] FAX issues/questions In-Reply-To: References: Message-ID: http://soft-switch.org/t38/index.html http://soft-switch.org/foip.html On Thu, Nov 17, 2016 at 4:40 AM, Alexandru Covalschi <568691 at gmail.com> wrote: > First of all thanks for answer! > > It's ok to buy several licenses, but why then it works at all right now? > To the same destination 8/10 are successful, 2/10 have that issue with > codec. > Also, can you point me to any material I can read to understand the T.38 > encapsulation? Is that a usual practice or it's just the way FS does that? > I'm very "weak" at faxing and maybe I'm asking silly questions, but I > really need to understand how things work... > > Cheers, > Alex > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 (50% Discount using code FreeSwitch50) http://www.freeswitchcookbook.com (50% Discount using code FreeSwitch50) https://www.gofundme.com/freeswitch_ubuntu Got Bugs? Report them here ! | Reddit: /r/freeswitch *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/20161117/4a8225ea/attachment.html From steveayre at gmail.com Thu Nov 17 20:00:12 2016 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 17 Nov 2016 17:00:12 +0000 Subject: [Freeswitch-users] Lua freeswitch.Dbh() question In-Reply-To: References: Message-ID: Currently the answer is no. It is possible with a patch though. If you look at src/mod/languages/mod_lua/freeswitch_lua.cpp you'll see it calls switch_cache_db_execute_sql_callback/switch_cache_db_execute_sql with NULL as the err parameter. If it was given that would return NULL (no error) or a the error string, the same one you see logged. That could be copied to the object and exposed with a last_error() function. Steve On 15 November 2016 at 16:54, Abaci B wrote: > Basically what I want is the text of the error returned from the database, > something like how it's handled in LuaSQL http://keplerproject.github. > io/luasql/manual.html#errors pcall is for lua errors in this case there > is no lua error. > > On Tue, Nov 15, 2016 at 2:10 AM, Artur Mega > wrote: > >> Do you need to get text of an error? And to catch it in lua code? Why not >> just to check returned value? I dont remember, but returned value can be >> nul or item with type `userdata`. If you still want text of an error, i >> think you can try to catch it with pcall https://www.lua.org/pil/8.4.html >> >> 2016-11-14 23:11 GMT+05:00 Abaci B : >> >>> when runned a query using freeswitch.Dbh dbh:query, if there is results >>> it a function for each returned row where you can see the data. however, if >>> there is an error you can see the database returned error in the console on >>> level ERR but you don't get it back in lua >>> is someone aware of a way to get the rror back into lua? >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> 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/20161117/a76a53c5/attachment-0001.html From rtreleaven at bunnykick.ca Fri Nov 18 01:29:25 2016 From: rtreleaven at bunnykick.ca (Russell Treleaven) Date: Thu, 17 Nov 2016 17:29:25 -0500 Subject: [Freeswitch-users] Skype In-Reply-To: References: <9F01883B-FEAF-437B-8AFA-0C559863E067@magicmail.mooo.com> Message-ID: How did this land? On Aug 4, 2016 1:35 PM, "Giovanni Maruzzelli" wrote: About skypopen: I suspect they (M$) will retract support soon for the protocol/API used by mod_skypopen. Don't seriously invest in it until situation will clarify, probably in Autumn... 6 concurrent call limit can be overridden in many ways, but I repeat, I would not counseil to seriously invest in it until Autumn. I'm mod_skypopen author, btw. If someone has more info, please step up and let us know. -giovanni On Thu, Aug 4, 2016 at 5:53 PM, Rick Jarvis wrote: > Skype are telling me that it is no longer possible to receive calls to > Skype names / addresses (as appose to Skype numbers) via SIP with Skype > Connect. Does anyone know if this is the case? Their support is awful, but > this would seem a bit of a backward step, and I don?t know whether to trust > what I?m being told. > > The alternative is Skypopen, of course. But I notice that it has a maximum > of 6 concurrent calls. Is this for resource reasons? I?m thinking of > building a server to handle all Skype to SIP calls for our customers, so > would be interested to know if it?s possible to bypass the limit with a > beefy spec?? > > Thanks > R > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161117/8fa49476/attachment.html From murat.knecht at googlemail.com Fri Nov 18 09:32:51 2016 From: murat.knecht at googlemail.com (Murat) Date: Fri, 18 Nov 2016 14:32:51 +0800 Subject: [Freeswitch-users] Only one bug of this type allowed! Message-ID: Hey, occasionally, we're getting the following error log from FS. [ERR] switch_core_media_bug.c:432 Only one bug of this type allowed! I'd like to understand what's happening. I've googled and searched around a but can't find anything useful aside from the code [1]. The code seems to indicate that multi-threading problems are recognized, I assume from a faulty plugin? Our version is 1.5.8b. Since it happens very rarely, I'm not sure what the implications are. Any pointers, thoughts, ideas? Thanks, murat [1] https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/src/switch_core_media_bug.c From anthony.minessale at gmail.com Fri Nov 18 10:58:20 2016 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 18 Nov 2016 01:58:20 -0600 Subject: [Freeswitch-users] Only one bug of this type allowed! In-Reply-To: References: Message-ID: Some media bugs (used for recording calls, eavesdropping etc) can only have one instance in use per channel. Recordings can be multiple, you can uuid_record 3 different files at the same time. Whatever one you are using is limited to one use and you found a path to cause it to be attempted to be used more than once so you get that error. Another error is you should not be using 1.5 anymore, its a temporary development release feeding 1.4 which is also EOL. Have a look at 1.6...... On Fri, Nov 18, 2016 at 12:32 AM, Murat wrote: > Hey, > > occasionally, we're getting the following error log from FS. > > [ERR] switch_core_media_bug.c:432 Only one bug of this type allowed! > > I'd like to understand what's happening. I've googled and searched > around a but can't find anything useful aside from the code [1]. The > code seems to indicate that multi-threading problems are recognized, I > assume from a faulty plugin? Our version is 1.5.8b. > > Since it happens very rarely, I'm not sure what the implications are. > Any pointers, thoughts, ideas? > > Thanks, > murat > > [1] https://freeswitch.org/stash/projects/FS/repos/freeswitch/ > browse/src/switch_core_media_bug.c > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161118/686cc46b/attachment.html From aubalde at presenceco.com Fri Nov 18 11:06:28 2016 From: aubalde at presenceco.com (=?UTF-8?Q?Agust=C3=AD_Ubalde?=) Date: Fri, 18 Nov 2016 09:06:28 +0100 Subject: [Freeswitch-users] Allow only one register Message-ID: Hi all, I'm using FreeSWITCH with WebRTC and I'm very interested on allow only one user per user. Is it possible? Thanks, PRESENCE TECHNOLOGY An ENGHOUSE INTERACTIVE Company Agust? Ubalde Bellot Chief Developer C/ Comte Urgell 240 3?-A Barcelona 08036 aubalde at presenceco.com Ph: +34 93 10 10 322 Enghouse is listed on the Toronto Stock Exchange, stock symbol: ENGH -- *Presence Technology - DisclaimerThis message, its content and any file attached thereto is for the intended recipient only and is confidential and /or privileged. If you have received this e-mail in error or had access to it, you should note that the information in it is private and any use thereof is unauthorized. In such an event please notify us by e-mail or by telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by whatsoever means and any transmission or dissemination thereof to other persons is prohibited. It should be deleted immediately from your system. Presence Technology reserves the right to take legal action against any persons unlawfully gaining access to the content of any external message it has emitted.* *For additional information, please visit our website **www.presenceco.com * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161118/59daeb20/attachment-0001.html From mirkobrankovic at gmail.com Fri Nov 18 12:29:28 2016 From: mirkobrankovic at gmail.com (Mirko Brankovic) Date: Fri, 18 Nov 2016 10:29:28 +0100 Subject: [Freeswitch-users] Allow only one register In-Reply-To: References: Message-ID: what do you mean by "user per user."? one web user per directory user? On Fri, Nov 18, 2016 at 9:06 AM, Agust? Ubalde wrote: > Hi all, > > I'm using FreeSWITCH with WebRTC and I'm very interested on allow only one > user per user. Is it possible? > > > Thanks, > > PRESENCE TECHNOLOGY > > An ENGHOUSE INTERACTIVE Company > > Agust? Ubalde Bellot > > Chief Developer > > C/ Comte Urgell 240 3?-A > > Barcelona 08036 > > aubalde at presenceco.com > > Ph: +34 93 10 10 322 > > Enghouse is listed on the Toronto Stock Exchange, stock symbol: ENGH > > > > *Presence Technology - DisclaimerThis message, its content and any file > attached thereto is for the intended recipient only and is confidential and > /or privileged. If you have received this e-mail in error or had access to > it, you should note that the information in it is private and any use > thereof is unauthorized. In such an event please notify us by e-mail or by > telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by > whatsoever means and any transmission or dissemination thereof to other > persons is prohibited. It should be deleted immediately from your system. > Presence Technology reserves the right to take legal action against any > persons unlawfully gaining access to the content of any external message it > has emitted.* > > *For additional information, please visit our website **www.presenceco.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 > -- Regards, Mirko -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161118/294f6e7a/attachment.html From aubalde at presenceco.com Fri Nov 18 12:36:36 2016 From: aubalde at presenceco.com (=?UTF-8?Q?Agust=C3=AD_Ubalde?=) Date: Fri, 18 Nov 2016 10:36:36 +0100 Subject: [Freeswitch-users] Allow only one register In-Reply-To: References: Message-ID: Hi Mirko, I'm talking about the possibility to block two registrations for the same extension from 2 different browsers (client WebRTC). Thanks, PRESENCE TECHNOLOGY An ENGHOUSE INTERACTIVE Company Agust? Ubalde Bellot Chief Developer C/ Comte Urgell 240 3?-A Barcelona 08036 aubalde at presenceco.com Ph: +34 93 10 10 322 Enghouse is listed on the Toronto Stock Exchange, stock symbol: ENGH 2016-11-18 10:29 GMT+01:00 Mirko Brankovic : > what do you mean by "user per user."? one web user per directory user? > > On Fri, Nov 18, 2016 at 9:06 AM, Agust? Ubalde > wrote: > >> Hi all, >> >> I'm using FreeSWITCH with WebRTC and I'm very interested on allow only >> one user per user. Is it possible? >> >> >> Thanks, >> >> PRESENCE TECHNOLOGY >> >> An ENGHOUSE INTERACTIVE Company >> >> Agust? Ubalde Bellot >> >> Chief Developer >> >> C/ Comte Urgell 240 3?-A >> >> Barcelona 08036 >> >> aubalde at presenceco.com >> >> Ph: +34 93 10 10 322 >> >> Enghouse is listed on the Toronto Stock Exchange, stock symbol: ENGH >> >> >> >> *Presence Technology - DisclaimerThis message, its content and any file >> attached thereto is for the intended recipient only and is confidential and >> /or privileged. If you have received this e-mail in error or had access to >> it, you should note that the information in it is private and any use >> thereof is unauthorized. In such an event please notify us by e-mail or by >> telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by >> whatsoever means and any transmission or dissemination thereof to other >> persons is prohibited. It should be deleted immediately from your system. >> Presence Technology reserves the right to take legal action against any >> persons unlawfully gaining access to the content of any external message it >> has emitted.* >> >> *For additional information, please visit our website **www.presenceco.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 >> > > > > -- > Regards, > Mirko > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Presence Technology - DisclaimerThis message, its content and any file attached thereto is for the intended recipient only and is confidential and /or privileged. If you have received this e-mail in error or had access to it, you should note that the information in it is private and any use thereof is unauthorized. In such an event please notify us by e-mail or by telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by whatsoever means and any transmission or dissemination thereof to other persons is prohibited. It should be deleted immediately from your system. Presence Technology reserves the right to take legal action against any persons unlawfully gaining access to the content of any external message it has emitted.* *For additional information, please visit our website **www.presenceco.com * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161118/770c7b54/attachment-0001.html From randomdev4 at gmail.com Fri Nov 18 13:08:38 2016 From: randomdev4 at gmail.com (Tim Smith) Date: Fri, 18 Nov 2016 10:08:38 +0000 Subject: [Freeswitch-users] Freeswitch routing inbound calls over SIP instead of TLS/SRTP Message-ID: Debian GNU/Linux 8 (jessie) Linux my 3.16.0-4-amd64 #1 SMP Debian 3.16.36-1+deb8u2 (2016-10-19) x86_64 GNU/Linux FreeSWITCH Version 1.6.12-20-b91a0a6~64bit (-20-b91a0a6 64bit) I have a Vtech handset with TLS/SRTP enabled registered with Freeswitch as below: Call-ID: a0000a0a000aa000 User: 2001 at my.example.com Contact: "my" Agent: Vtech Vesa VSP736A 2.0.3.2-0 Status: Registered(TLS)(unknown) EXP(2016-11-18 10:56:57) EXPSECS(3646) Ping-Status: Reachable Ping-Time: 0.00 Host: my IP: 198.51.100.81 Port: 58348 Auth-User: 2001 Auth-Realm: my.example.com MWI-Account: 2001 at my.example.com sofia_contact is happy : freeswitch at my>sofia_contact internal/2001 sofia/internal/sip:2001 at 198.51.100.81:58348 I have an inbound dial plan configured as follows: The problem is Freeswitch is sending invites over SIP/RTP and not TLS/SRTP and so the calls never get through : INVITE sip:2001 at 198.51.100.81:58348 SIP/2.0 Via: SIP/2.0/UDP 203.0.113.4;rport;branch=z9hG4bKvHjgXXpFF77XK From: "Anonymous" ;tag=rKmXQjZN8SFXp To: m=audio 32190 RTP/AVP 8 98 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:98 G726-32/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 From mirkobrankovic at gmail.com Fri Nov 18 13:31:22 2016 From: mirkobrankovic at gmail.com (Mirko Brankovic) Date: Fri, 18 Nov 2016 11:31:22 +0100 Subject: [Freeswitch-users] Allow only one register In-Reply-To: References: Message-ID: I'm not sure you can or you should limit that on FS side, since you can have multiple devices registering on same account. It is probably better to add a logic on client side as of what user will be used, either by logging in or incremental usernames,.... On Fri, Nov 18, 2016 at 10:36 AM, Agust? Ubalde wrote: > Hi Mirko, > > I'm talking about the possibility to block two registrations for the same > extension from 2 different browsers (client WebRTC). > > > Thanks, > > PRESENCE TECHNOLOGY > > An ENGHOUSE INTERACTIVE Company > > Agust? Ubalde Bellot > > Chief Developer > > C/ Comte Urgell 240 3?-A > > Barcelona 08036 > > aubalde at presenceco.com > > Ph: +34 93 10 10 322 > > Enghouse is listed on the Toronto Stock Exchange, stock symbol: ENGH > > > 2016-11-18 10:29 GMT+01:00 Mirko Brankovic : > >> what do you mean by "user per user."? one web user per directory user? >> >> On Fri, Nov 18, 2016 at 9:06 AM, Agust? Ubalde >> wrote: >> >>> Hi all, >>> >>> I'm using FreeSWITCH with WebRTC and I'm very interested on allow only >>> one user per user. Is it possible? >>> >>> >>> Thanks, >>> >>> PRESENCE TECHNOLOGY >>> >>> An ENGHOUSE INTERACTIVE Company >>> >>> Agust? Ubalde Bellot >>> >>> Chief Developer >>> >>> C/ Comte Urgell 240 3?-A >>> >>> Barcelona 08036 >>> >>> aubalde at presenceco.com >>> >>> Ph: +34 93 10 10 322 >>> >>> Enghouse is listed on the Toronto Stock Exchange, stock symbol: ENGH >>> >>> >>> >>> *Presence Technology - DisclaimerThis message, its content and any file >>> attached thereto is for the intended recipient only and is confidential and >>> /or privileged. If you have received this e-mail in error or had access to >>> it, you should note that the information in it is private and any use >>> thereof is unauthorized. In such an event please notify us by e-mail or by >>> telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by >>> whatsoever means and any transmission or dissemination thereof to other >>> persons is prohibited. It should be deleted immediately from your system. >>> Presence Technology reserves the right to take legal action against any >>> persons unlawfully gaining access to the content of any external message it >>> has emitted.* >>> >>> *For additional information, please visit our website **www.presenceco.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 >>> >> >> >> >> -- >> Regards, >> Mirko >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > *Presence Technology - DisclaimerThis message, its content and any file > attached thereto is for the intended recipient only and is confidential and > /or privileged. If you have received this e-mail in error or had access to > it, you should note that the information in it is private and any use > thereof is unauthorized. In such an event please notify us by e-mail or by > telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by > whatsoever means and any transmission or dissemination thereof to other > persons is prohibited. It should be deleted immediately from your system. > Presence Technology reserves the right to take legal action against any > persons unlawfully gaining access to the content of any external message it > has emitted.* > > *For additional information, please visit our website **www.presenceco.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 > -- Regards, Mirko -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161118/dec0214e/attachment-0001.html From gregor at infomedia.si Fri Nov 18 13:38:40 2016 From: gregor at infomedia.si (Gregor Nanger) Date: Fri, 18 Nov 2016 11:38:40 +0100 Subject: [Freeswitch-users] Allow only one register In-Reply-To: References: Message-ID: ?I am solving this problem with ESL, monitoring verto registration? and saving in my database. Then in XML curl I respond to registration request according to user state in my database. Or you can skip first step and use show verto json to get all registered verto users in json format and parse it. Each browser registration has it's own port, even if it is registered on same ip. 2016-11-18 10:36 GMT+01:00 Agust? Ubalde : > Hi Mirko, > > I'm talking about the possibility to block two registrations for the same > extension from 2 different browsers (client WebRTC). > > > Thanks, > > PRESENCE TECHNOLOGY > > An ENGHOUSE INTERACTIVE Company > > Agust? Ubalde Bellot > > Chief Developer > > C/ Comte Urgell 240 3?-A > > Barcelona 08036 > > aubalde at presenceco.com > > Ph: +34 93 10 10 322 > > Enghouse is listed on the Toronto Stock Exchange, stock symbol: ENGH > > > 2016-11-18 10:29 GMT+01:00 Mirko Brankovic : > >> what do you mean by "user per user."? one web user per directory user? >> >> On Fri, Nov 18, 2016 at 9:06 AM, Agust? Ubalde >> wrote: >> >>> Hi all, >>> >>> I'm using FreeSWITCH with WebRTC and I'm very interested on allow only >>> one user per user. Is it possible? >>> >>> >>> Thanks, >>> >>> PRESENCE TECHNOLOGY >>> >>> An ENGHOUSE INTERACTIVE Company >>> >>> Agust? Ubalde Bellot >>> >>> Chief Developer >>> >>> C/ Comte Urgell 240 3?-A >>> >>> Barcelona 08036 >>> >>> aubalde at presenceco.com >>> >>> Ph: +34 93 10 10 322 >>> >>> Enghouse is listed on the Toronto Stock Exchange, stock symbol: ENGH >>> >>> >>> >>> *Presence Technology - DisclaimerThis message, its content and any file >>> attached thereto is for the intended recipient only and is confidential and >>> /or privileged. If you have received this e-mail in error or had access to >>> it, you should note that the information in it is private and any use >>> thereof is unauthorized. In such an event please notify us by e-mail or by >>> telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by >>> whatsoever means and any transmission or dissemination thereof to other >>> persons is prohibited. It should be deleted immediately from your system. >>> Presence Technology reserves the right to take legal action against any >>> persons unlawfully gaining access to the content of any external message it >>> has emitted.* >>> >>> *For additional information, please visit our website **www.presenceco.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 >>> >> >> >> >> -- >> Regards, >> Mirko >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > *Presence Technology - DisclaimerThis message, its content and any file > attached thereto is for the intended recipient only and is confidential and > /or privileged. If you have received this e-mail in error or had access to > it, you should note that the information in it is private and any use > thereof is unauthorized. In such an event please notify us by e-mail or by > telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by > whatsoever means and any transmission or dissemination thereof to other > persons is prohibited. It should be deleted immediately from your system. > Presence Technology reserves the right to take legal action against any > persons unlawfully gaining access to the content of any external message it > has emitted.* > > *For additional information, please visit our website **www.presenceco.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 > -- Gregor Nanger *CTO* t./f.: 00386 (0) 7 6000 308/309 ? m:. 00386 (0)41 756485 ? Infomedia d.o.o. ? Jerebova 3, Novo mesto, Slovenia ? www.infomedia.si -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161118/7892d33a/attachment.html From david.villasmil.work at gmail.com Fri Nov 18 14:44:10 2016 From: david.villasmil.work at gmail.com (David Villasmil) Date: Fri, 18 Nov 2016 12:44:10 +0100 Subject: [Freeswitch-users] Editing confluence Message-ID: Hello guys, I'm trying to edit https://freeswitch.org/confluence/display/FREESWITCH/mod_sofia to document the new "sofia filter", but i don't seem to have permissions? Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161118/84a6f233/attachment-0001.html From devang.nathwani31589 at gmail.com Fri Nov 18 15:07:40 2016 From: devang.nathwani31589 at gmail.com (devang nathwani) Date: Fri, 18 Nov 2016 17:37:40 +0530 Subject: [Freeswitch-users] freeswitch sending '481 Call Does Not Exist' in return of BYE Message-ID: Hello, Here are the sip traces of two legs leg1 UAC -> proxy -> media leg2 media -> proxy -> provider leg1 http://pastebin.com/G0jnF75t leg2 http://pastebin.com/299dZyN4 from leg1, when freeswitch(media) is sending back 480 temporarily unavailable after 183 session progress, UAC is sending ack and bye, now in return freeswitch is sending '481 Call Does Not Exist' my question is why? why freeswitch is sending 481 in return of ack and bye? here, 11.23.16.12 is UAC 11.23.16.13 is Opensips(Proxy) 11.23.16.16 is Freeswitch(Media) Thanks, Devang Nathwani -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161118/3752eb5c/attachment.html From mirkobrankovic at gmail.com Fri Nov 18 15:47:29 2016 From: mirkobrankovic at gmail.com (Mirko Brankovic) Date: Fri, 18 Nov 2016 13:47:29 +0100 Subject: [Freeswitch-users] freeswitch sending '481 Call Does Not Exist' in return of BYE In-Reply-To: References: Message-ID: HI, I'm not sure if .17 is Freeswitch and who is 12 and wh 13 but I see that on first 480 no answer To:.... tag is changed: 1. 2016/11/18 11:30:42.698223 11.23.16.13:5060 -> 11.23.16.12:5060 2. SIP/2.0 480 NO_ANSWER 3. Via: SIP/2.0/UDP 11.23.16.12:5060;branch=z9hG4bKlj0b84100gnjom3o1320.1 4. From: ;tag=09002861124617 5. To: ;tag=f949601897706458a5166612fe67c373-1439 6. Call-ID: 03cS323551118140bcGhEfCmJej at RBM2S1.MSS.MTN.CO.ZA 7. CSeq: 986549121 INVITE 8. Content-Length: 0 WHich I think you should avoid, changing tags. I think this might be the problem. On Fri, Nov 18, 2016 at 1:07 PM, devang nathwani < devang.nathwani31589 at gmail.com> wrote: > Hello, > > Here are the sip traces of two legs > leg1 UAC -> proxy -> media > leg2 media -> proxy -> provider > leg1 > http://pastebin.com/G0jnF75t > leg2 > http://pastebin.com/299dZyN4 > > from leg1, > when freeswitch(media) is sending back 480 temporarily unavailable after > 183 session progress, UAC is sending ack and bye, now in return freeswitch > is sending '481 Call Does Not Exist' > > my question is why? why freeswitch is sending 481 in return of ack and bye? > > here, > 11.23.16.12 is UAC > 11.23.16.13 is Opensips(Proxy) > 11.23.16.16 is Freeswitch(Media) > > Thanks, > Devang Nathwani > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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, Mirko -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161118/24a4887a/attachment.html From mike at jerris.com Fri Nov 18 17:46:41 2016 From: mike at jerris.com (Michael Jerris) Date: Fri, 18 Nov 2016 09:46:41 -0500 Subject: [Freeswitch-users] Allow only one register In-Reply-To: References: Message-ID: <010A6C24-069D-4DA5-86E7-672BD39156E1@jerris.com> We can?t block registrations (sip) or connections (verto) but you could use our limit functionality to block concurrent calls. With a little bit of cleverness, and dynamic directory, you could detect active current reg/connections and serve up user different if already registered. You can on sip disable multi registration, but the new one will just replace the old one. > On Nov 18, 2016, at 3:06 AM, Agust? Ubalde wrote: > > Hi all, > > I'm using FreeSWITCH with WebRTC and I'm very interested on allow only one user per user. Is it possible? > From S.Boomstra at telecats.nl Fri Nov 18 15:09:12 2016 From: S.Boomstra at telecats.nl (Sjoerd Boomstra) Date: Fri, 18 Nov 2016 12:09:12 +0000 Subject: [Freeswitch-users] bypass_media after receiving invite with replaces header Message-ID: Hello, I've got a problem with FreeSwitch version 1.4.26. Our setup is like this: We have the phones registered at a SipXecs cluster. Freeswitch is used as a gateway (sbc) between SipXecs and another SIP device (Audiocodes Mediant). We aim to bypass the media at the FreeSwitch. At first this works as expected, the media does not go through the FreeSwitch. However when a phone issues a transfer, the refer is handled by SipXecs. The invite with replaces header is sent to Freeswitch (via SipX). FreeSwitch handles this SIP message correctly, but it puts itself in the RTP-stream. The original channel has the variables bypass_media and bypass_media_after_bridge set to true. It seems that these variables are not checked when the invite with replaces header is handled. Anyone an idea how to fix this? Best regards, Sjoerd. ----- Sjoerd Boomstra | Senior Business Consultant | Telecats bv | KvK Enschede 06069106 | Tel: 053 488 99 07 | Fax: 053 488 99 10 | E?mail: s.boomstra at telecats.nl | -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161118/707e1c45/attachment-0001.html From mike at jerris.com Fri Nov 18 17:51:45 2016 From: mike at jerris.com (Michael Jerris) Date: Fri, 18 Nov 2016 09:51:45 -0500 Subject: [Freeswitch-users] bypass_media after receiving invite with replaces header In-Reply-To: References: Message-ID: <77974EB6-48DE-4430-B95F-2BDC776D3C97@jerris.com> 1.4 is long since EOL. Please try latest release version and confirm this is still an issue. > On Nov 18, 2016, at 7:09 AM, Sjoerd Boomstra wrote: > > Hello, > > I've got a problem with FreeSwitch version 1.4.26. Our setup is like this: > > We have the phones registered at a SipXecs cluster. Freeswitch is used as a gateway (sbc) between SipXecs and another SIP device (Audiocodes Mediant). > We aim to bypass the media at the FreeSwitch. > At first this works as expected, the media does not go through the FreeSwitch. > > However when a phone issues a transfer, the refer is handled by SipXecs. The invite with replaces header is sent to Freeswitch (via SipX). FreeSwitch handles this SIP message correctly, but it puts itself in the RTP-stream. > The original channel has the variables bypass_media and bypass_media_after_bridge set to true. > > It seems that these variables are not checked when the invite with replaces header is handled. > > Anyone an idea how to fix this? > > Best regards, > Sjoerd. > > ----- > Sjoerd Boomstra | Senior Business Consultant | Telecats bv | KvK Enschede 06069106 | Tel: 053 488 99 07 | Fax: 053 488 99 10 | E?mail: s.boomstra at telecats.nl | > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161118/61c8eae1/attachment.html From alexdruzhilov at gmail.com Fri Nov 18 18:45:36 2016 From: alexdruzhilov at gmail.com (=?UTF-8?B?0JDQu9C10LrRgdCw0L3QtNGAINCU0YDRg9C20LjQu9C+0LI=?=) Date: Fri, 18 Nov 2016 18:45:36 +0300 Subject: [Freeswitch-users] Video conference with screensharing. Good quality without high CPU usage. Message-ID: Hello everybody! My case is: make video conference (with layout 1x1 only) and with possibility to share screen or video from camera. The main problem for me is a high CPU usage for good quality video (for example screen sharing 1920x1280. I use latest version of Freeswitch and mod_sofia. My config in mod_conference is: In case of "mux" with this configuration if somebody tries to send his video from camera (800x600) then everybody will see him bordered by black canvas. But if I will set video-canvas-size = 800x600, then if somebody tries to share his screen in resolution 1920x1280 then everybody will see only 800x600 and quality will be not really good. The same situation with the video-fps parameter. My questions are: 1) If I have only 1x1 layout, so what video-mode should I use for best quality and lowest CPU usage? transcode or mux? 2) How to make screen sharing video stream and camera video stream to have different resolution/bitrate/fps? 3) Should I use mod_verto for better quality/performance? 4) What kind of optimizations could you recommend me for better quality/performance? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161118/8919ba92/attachment.html From ignaciodurli at cinchcast.com Fri Nov 18 21:12:16 2016 From: ignaciodurli at cinchcast.com (Ignacio Durli) Date: Fri, 18 Nov 2016 18:12:16 +0000 Subject: [Freeswitch-users] FS not receiving incoming calls Message-ID: I upgraded to FS 1.6.12 which I installed in a new Debian 8 instance ran in VirtualBox. After compiling and running all the config scripts I'm not able to establish a SIP call. After some troubleshooting I found that calls are being answered on my eth1 interface (10.0.x.x) but not on my eth0 interface (192.168.99.x). The IP is configured in vars.xml properly and sofia status is showing: freeswitch at freeswitch> sofia status Name Type Data State ================================================================================================= external-ipv6 profile sip:mod_sofia@[::1]:5080 RUNNING (0) external-ipv6 profile sip:mod_sofia@[::1]:5081 RUNNING (0) (TLS) external profile sip:mod_sofia at 192.168.99.103:5080 RUNNING (0) external profile sip:mod_sofia at 192.168.99.103:5081 RUNNING (0) (TLS) 192.168.99.103 alias internal ALIASED internal-ipv6 profile sip:mod_sofia@[::1]:5060 RUNNING (0) internal-ipv6 profile sip:mod_sofia@[::1]:5061 RUNNING (0) (TLS) internal profile sip:mod_sofia at 192.168.99.103:5060 RUNNING (0) internal profile sip:mod_sofia at 192.168.99.103:5061 RUNNING (0) (TLS) ================================================================================================= 4 profiles 1 alias Which looks fine. Also the ifconfig of the new Debian instance seems OK, since it's the same that I have on another VM where an older FS version is running. root at osboxes:/usr/local/freeswitch/bin# ifconfig eth0 Link encap:Ethernet HWaddr 08:00:27:3a:f0:c6 inet addr:192.168.99.103 Bcast:192.168.99.255 Mask:255.255.255.0 inet6 addr: fe80::a00:27ff:fe3a:f0c6/64 Scope:Link UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1 RX packets:2482 errors:0 dropped:0 overruns:0 frame:0 TX packets:1034 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:1000 RX bytes:353837 (345.5 KiB) TX bytes:227253 (221.9 KiB) eth1 Link encap:Ethernet HWaddr 08:00:27:f7:f1:24 inet addr:10.0.3.15 Bcast:10.0.3.255 Mask:255.255.255.0 inet6 addr: fe80::a00:27ff:fef7:f124/64 Scope:Link UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1 RX packets:4298 errors:0 dropped:0 overruns:0 frame:0 TX packets:7037 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:1000 RX bytes:437241 (426.9 KiB) TX bytes:829683 (810.2 KiB) lo Link encap:Local Loopback inet addr:127.0.0.1 Mask:255.0.0.0 inet6 addr: ::1/128 Scope:Host UP LOOPBACK RUNNING MTU:65536 Metric:1 RX packets:49100 errors:0 dropped:0 overruns:0 frame:0 TX packets:49100 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:0 RX bytes:8431576 (8.0 MiB) TX bytes:8431576 (8.0 MiB) I honestly don't know what's going on. Acls are fine, and I don't see any error in the FS log. I traced the network traffic and when trying to call to 192.168.99.103:5060 I get an ICM (Destination unreachable) which doesn't make sense according to what sofia status is showing. Any hint on this? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161118/2a67a9bf/attachment-0001.html From joel at gogii.net Fri Nov 18 21:36:56 2016 From: joel at gogii.net (Joel Serrano) Date: Fri, 18 Nov 2016 10:36:56 -0800 Subject: [Freeswitch-users] FS not receiving incoming calls In-Reply-To: References: Message-ID: Output of: # iptables -L -n # netstat -putan | grep -i freeswitch Please :) On Fri, Nov 18, 2016 at 10:12 AM, Ignacio Durli wrote: > I upgraded to FS 1.6.12 which I installed in a new Debian 8 instance ran > in VirtualBox. After compiling and running all the config scripts I?m not > able to establish a SIP call. After some troubleshooting I found that calls > are being answered on my eth1 interface (10.0.x.x) but not on my eth0 > interface (192.168.99.x). > > > > The IP is configured in vars.xml properly and sofia status is showing: > > > > freeswitch at freeswitch> sofia status > > Name Type > Data State > > ============================================================ > ===================================== > > external-ipv6 profile sip:mod_sofia@ > [::1]:5080 RUNNING (0) > > external-ipv6 profile sip:mod_sofia@ > [::1]:5081 RUNNING (0) (TLS) > > external profile > sip:mod_sofia at 192.168.99.103:5080 RUNNING (0) > > external profile > sip:mod_sofia at 192.168.99.103:5081 RUNNING (0) (TLS) > > 192.168.99.103 alias > internal ALIASED > > internal-ipv6 profile sip:mod_sofia@ > [::1]:5060 RUNNING (0) > > internal-ipv6 profile sip:mod_sofia@ > [::1]:5061 RUNNING (0) (TLS) > > internal profile > sip:mod_sofia at 192.168.99.103:5060 RUNNING (0) > > internal profile > sip:mod_sofia at 192.168.99.103:5061 RUNNING (0) (TLS) > > ============================================================ > ===================================== > > 4 profiles 1 alias > > > > > > Which looks fine. Also the ifconfig of the new Debian instance seems OK, > since it?s the same that I have on another VM where an older FS version is > running. > > > > root at osboxes:/usr/local/freeswitch/bin# ifconfig > > eth0 Link encap:Ethernet HWaddr 08:00:27:3a:f0:c6 > > inet addr:192.168.99.103 Bcast:192.168.99.255 > Mask:255.255.255.0 > > inet6 addr: fe80::a00:27ff:fe3a:f0c6/64 Scope:Link > > UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1 > > RX packets:2482 errors:0 dropped:0 overruns:0 frame:0 > > TX packets:1034 errors:0 dropped:0 overruns:0 carrier:0 > > collisions:0 txqueuelen:1000 > > RX bytes:353837 (345.5 KiB) TX bytes:227253 (221.9 KiB) > > > > eth1 Link encap:Ethernet HWaddr 08:00:27:f7:f1:24 > > inet addr:10.0.3.15 Bcast:10.0.3.255 Mask:255.255.255.0 > > inet6 addr: fe80::a00:27ff:fef7:f124/64 Scope:Link > > UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1 > > RX packets:4298 errors:0 dropped:0 overruns:0 frame:0 > > TX packets:7037 errors:0 dropped:0 overruns:0 carrier:0 > > collisions:0 txqueuelen:1000 > > RX bytes:437241 (426.9 KiB) TX bytes:829683 (810.2 KiB) > > > > lo Link encap:Local Loopback > > inet addr:127.0.0.1 Mask:255.0.0.0 > > inet6 addr: ::1/128 Scope:Host > > UP LOOPBACK RUNNING MTU:65536 Metric:1 > > RX packets:49100 errors:0 dropped:0 overruns:0 frame:0 > > TX packets:49100 errors:0 dropped:0 overruns:0 carrier:0 > > collisions:0 txqueuelen:0 > > RX bytes:8431576 (8.0 MiB) TX bytes:8431576 (8.0 MiB) > > > > > > I honestly don?t know what?s going on. Acls are fine, and I don?t see any > error in the FS log. I traced the network traffic and when trying to call > to 192.168.99.103:5060 I get an ICM (Destination unreachable) which > doesn?t make sense according to what sofia status is showing. Any hint 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/20161118/54771523/attachment.html From s.safarov at gmail.com Sat Nov 19 04:54:47 2016 From: s.safarov at gmail.com (Sergey Safarov) Date: Sat, 19 Nov 2016 01:54:47 +0000 Subject: [Freeswitch-users] bypass_media after receiving invite with replaces header In-Reply-To: <77974EB6-48DE-4430-B95F-2BDC776D3C97@jerris.com> References: <77974EB6-48DE-4430-B95F-2BDC776D3C97@jerris.com> Message-ID: Alternatively, configure the SIP profile to use proxy media by default: ??, 18 ????. 2016 ?. ? 17:52, Michael Jerris : > 1.4 is long since EOL. Please try latest release version and confirm this > is still an issue. > > On Nov 18, 2016, at 7:09 AM, Sjoerd Boomstra > wrote: > > Hello, > > I've got a problem with FreeSwitch version 1.4.26. Our setup is like this: > > We have the phones registered at a SipXecs cluster. Freeswitch is used as > a gateway (sbc) between SipXecs and another SIP device (Audiocodes > Mediant). > We aim to bypass the media at the FreeSwitch. > At first this works as expected, the media does not go through the > FreeSwitch. > > However when a phone issues a transfer, the refer is handled by SipXecs. > The invite with replaces header is sent to Freeswitch (via SipX). > FreeSwitch handles this SIP message correctly, but it puts itself in the > RTP-stream. > The original channel has the variables bypass_media and > bypass_media_after_bridge set to true. > > It seems that these variables are not checked when the invite with > replaces header is handled. > > Anyone an idea how to fix this? > > Best regards, > Sjoerd. > > ----- > Sjoerd Boomstra | Senior Business Consultant | Telecats bv | > KvK Enschede 06069106 | Tel: 053 488 99 07 | Fax: 053 488 99 10 | > E?mail: s.boomstra at telecats.nl | > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161119/51807702/attachment-0001.html From mike at jerris.com Sat Nov 19 19:24:17 2016 From: mike at jerris.com (Michael Jerris) Date: Sat, 19 Nov 2016 16:24:17 +0000 Subject: [Freeswitch-users] bypass_media after receiving invite with replaces header In-Reply-To: References: <77974EB6-48DE-4430-B95F-2BDC776D3C97@jerris.com> Message-ID: Never use proxy media unless there is a specific very good reason to use it. It was originally built for codecs we didn't have proper support for and now has little to no reason to ever use it now that we have video codec and fax support. On Fri, Nov 18, 2016 at 9:00 PM Sergey Safarov wrote: > Alternatively, configure the SIP profile to use proxy media by default: > > > > > > > ??, 18 ????. 2016 ?. ? 17:52, Michael Jerris : > > 1.4 is long since EOL. Please try latest release version and confirm this > is still an issue. > > On Nov 18, 2016, at 7:09 AM, Sjoerd Boomstra > wrote: > > Hello, > > I've got a problem with FreeSwitch version 1.4.26. Our setup is like this: > > We have the phones registered at a SipXecs cluster. Freeswitch is used as > a gateway (sbc) between SipXecs and another SIP device (Audiocodes > Mediant). > We aim to bypass the media at the FreeSwitch. > At first this works as expected, the media does not go through the > FreeSwitch. > > However when a phone issues a transfer, the refer is handled by SipXecs. > The invite with replaces header is sent to Freeswitch (via SipX). > FreeSwitch handles this SIP message correctly, but it puts itself in the > RTP-stream. > The original channel has the variables bypass_media and > bypass_media_after_bridge set to true. > > It seems that these variables are not checked when the invite with > replaces header is handled. > > Anyone an idea how to fix this? > > Best regards, > Sjoerd. > > ----- > Sjoerd Boomstra | Senior Business Consultant | Telecats bv | > KvK Enschede 06069106 | Tel: 053 488 99 07 | Fax: 053 488 99 10 | > E?mail: s.boomstra at telecats.nl | > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > 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/20161119/d1fa874d/attachment.html From saumar at uol.com.br Sun Nov 20 00:21:54 2016 From: saumar at uol.com.br (Saumar Hajjar) Date: Sat, 19 Nov 2016 19:21:54 -0200 Subject: [Freeswitch-users] Sofia endpoint that only sends video In-Reply-To: References: <83a0ac81-0466-974c-83cb-964db7675ffa@uol.com.br> Message-ID: <8eade16d-9d6e-220d-0f0b-b55c9206eef7@uol.com.br> Hi I'm originating calls from (mod_sofia's) internal to external profile and joining them into a conference. The purpose is to play multiple video streams in a conference and consume the resulting canvas by a single VERTO RTC caller. My setup works but there's a LOT useless RTP traffic (lo) coming from the conference to the video playback channels. Is there any way to disable the audio/video conference feedback to those play-only video channels? Perhaps an existing channel variable that I've missed, or some way to tweak the SDP and put a sendonly for audio and video? My setup is (on top of the vanilla config): sofia status Name Type Data State ================================================================================================= external profile sip:mod_sofia at 192.168.1.110:5080 RUNNING (1) internal profile sip:mod_sofia at 192.168.1.110:5060 RUNNING (0) ================================================================================================= dialplan/default.xml dialplan/public.xml CLI (add 2 video streams to an existing conference) originate sofia/internal/3600 at 192.168.1.110:5080 &conference(3500-192.168.1.110) originate sofia/internal/3601 at 192.168.1.110:5080 &conference(3500-192.168.1.110) Thanks Em 14/11/2016 22:08, Abaci B escreveu: > you can have 2 sip profiles and use originate to create the calls > going from 1 profile to the other (one can listen o local ip if it helps) > > On Mon, Nov 14, 2016 at 6:39 PM, Saumar Hajjar > wrote: > > Thanks Anthony, > > For your second suggestion, is it possible to "loop a call into > the conference with SIP on the same box" using just CLI commands - > or do I need a external SIP client for this? > > > Em 14/11/2016 20:11, Anthony Minessale escreveu: >> Your only real option is make the calls from another box or loop >> them into the conference with SIP on the same box. >> The conference file playing mechanism is not designed for >> simultaneous files to play. >> >> >> >> On Thu, Nov 10, 2016 at 12:00 PM, Saumar Hajjar >> > wrote: >> >> Hi, >> >> I'm working in a PoC and I'm considering FS. >> >> I'd like to create a conference and have several video files >> playing >> simultaneously in the canvas. >> I already tried: conference name play >> av:///var/www/vid/video.mp4 and it >> works great. >> But I need multiple files playing and apparently the above >> command >> queues a video file (or freezes if it's a rtsp stream... >> Later I'll >> confirm this and file a Jira) >> >> I also tried creating a loopback leg, joining the conference, and >> playing a video just to this particular member. But it fails >> because the >> member doesn't support video. >> >> Basically I'd like to do something like the examples found at >> the bottom >> of >> https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video >> . >> Probably I'm missing some trivial stuff... I really >> appreciate any >> advice on this. >> >> Thanks in advance, >> >> Saumar >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 >> >> >> https://www.youtube.com/watch?v=9XXgW34t40s >> >> https://www.youtube.com/watch?v=NLaDpGQuZDA >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> 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/20161119/c1642d8c/attachment-0001.html From steveayre at gmail.com Sun Nov 20 03:33:56 2016 From: steveayre at gmail.com (Steven Ayre) Date: Sun, 20 Nov 2016 00:33:56 +0000 Subject: [Freeswitch-users] Freeswitch routing inbound calls over SIP instead of TLS/SRTP In-Reply-To: References: Message-ID: Looks like a bug to me. Your first snippet shows the contact stored in the database uses the 'sips:' scheme, but sofia_contact is returning 'sip:' In the code it looks like sofia_contact fetches the contact using select_from_profile which invokes contact_callback. In contact_callback it's hardcoded to use sip: plus the result of sofia_glue_strip_proto. That looks to me like it can never return a sips URI even though it's stored in the database. I'd file a jira. Steve On 18 November 2016 at 10:08, Tim Smith wrote: > Debian GNU/Linux 8 (jessie) > Linux my 3.16.0-4-amd64 #1 SMP Debian 3.16.36-1+deb8u2 (2016-10-19) > x86_64 GNU/Linux > FreeSWITCH Version 1.6.12-20-b91a0a6~64bit (-20-b91a0a6 64bit) > > I have a Vtech handset with TLS/SRTP enabled registered with > Freeswitch as below: > > > Call-ID: a0000a0a000aa000 > User: 2001 at my.example.com > Contact: "my" > Agent: Vtech Vesa VSP736A 2.0.3.2-0 > Status: Registered(TLS)(unknown) EXP(2016-11-18 10:56:57) > EXPSECS(3646) > Ping-Status: Reachable > Ping-Time: 0.00 > Host: my > IP: 198.51.100.81 > Port: 58348 > Auth-User: 2001 > Auth-Realm: my.example.com > MWI-Account: 2001 at my.example.com > > > sofia_contact is happy : > > freeswitch at my>sofia_contact internal/2001 > sofia/internal/sip:2001 at 198.51.100.81:58348 > > I have an inbound dial plan configured as follows: > > > > > > > > > > > The problem is Freeswitch is sending invites over SIP/RTP and not > TLS/SRTP and so the calls never get through : > > INVITE sip:2001 at 198.51.100.81:58348 SIP/2.0 > Via: SIP/2.0/UDP 203.0.113.4;rport;branch=z9hG4bKvHjgXXpFF77XK > From: "Anonymous" ;tag=rKmXQjZN8SFXp > To: > m=audio 32190 RTP/AVP 8 98 9 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:98 G726-32/8000 > a=rtpmap:9 G722/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161120/e518c270/attachment.html From lists at kavun.ch Sun Nov 20 14:23:23 2016 From: lists at kavun.ch (Emrah) Date: Sun, 20 Nov 2016 12:23:23 +0100 Subject: [Freeswitch-users] Freeswitch routing inbound calls over SIP instead of TLS/SRTP In-Reply-To: References: Message-ID: <931AA1AF-4E99-46CF-A6F0-CE4AE3AE1C93@kavun.ch> Actually, your sofia_contact is not happy.. I'm not seeing the transport param for tls in there. How many profiles are you running? Are you sure sofia_contact isn't giving you the value of another endpoint registered with UDP? Also, I went through your message fast, but I don't think you're securing your rtp... > On Nov 18, 2016, at 11:08 AM, Tim Smith wrote: > > Debian GNU/Linux 8 (jessie) > Linux my 3.16.0-4-amd64 #1 SMP Debian 3.16.36-1+deb8u2 (2016-10-19) > x86_64 GNU/Linux > FreeSWITCH Version 1.6.12-20-b91a0a6~64bit (-20-b91a0a6 64bit) > > I have a Vtech handset with TLS/SRTP enabled registered with > Freeswitch as below: > > > Call-ID: a0000a0a000aa000 > User: 2001 at my.example.com > Contact: "my" > Agent: Vtech Vesa VSP736A 2.0.3.2-0 > Status: Registered(TLS)(unknown) EXP(2016-11-18 10:56:57) EXPSECS(3646) > Ping-Status: Reachable > Ping-Time: 0.00 > Host: my > IP: 198.51.100.81 > Port: 58348 > Auth-User: 2001 > Auth-Realm: my.example.com > MWI-Account: 2001 at my.example.com > > > sofia_contact is happy : > > freeswitch at my>sofia_contact internal/2001 > sofia/internal/sip:2001 at 198.51.100.81:58348 > > I have an inbound dial plan configured as follows: > > > > > > > > > > > The problem is Freeswitch is sending invites over SIP/RTP and not > TLS/SRTP and so the calls never get through : > > INVITE sip:2001 at 198.51.100.81:58348 SIP/2.0 > Via: SIP/2.0/UDP 203.0.113.4;rport;branch=z9hG4bKvHjgXXpFF77XK > From: "Anonymous" ;tag=rKmXQjZN8SFXp > To: > m=audio 32190 RTP/AVP 8 98 9 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:98 G726-32/8000 > a=rtpmap:9 G722/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 italorossib at gmail.com Sun Nov 20 18:16:00 2016 From: italorossib at gmail.com (=?UTF-8?B?w410YWxvIFJvc3Np?=) Date: Sun, 20 Nov 2016 15:16:00 +0000 Subject: [Freeswitch-users] Editing confluence In-Reply-To: References: Message-ID: Hey David, You're now in the editors group. Please let me know if you need assistance. Em sex, 18 de nov de 2016 ?s 09:45, David Villasmil < david.villasmil.work at gmail.com> escreveu: > Hello guys, > > I'm trying to edit > https://freeswitch.org/confluence/display/FREESWITCH/mod_sofia to > document the new "sofia filter", but i don't seem to have permissions? > > > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > phone: +34669448337 > ? > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161120/b8bcb15e/attachment.html From anthony.minessale at gmail.com Sun Nov 20 18:45:36 2016 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 20 Nov 2016 15:45:36 +0000 Subject: [Freeswitch-users] Editing confluence In-Reply-To: References: Message-ID: Thank you! On Sun, Nov 20, 2016 at 9:17 AM ?talo Rossi wrote: > Hey David, > > You're now in the editors group. Please let me know if you need assistance. > > Em sex, 18 de nov de 2016 ?s 09:45, David Villasmil < > david.villasmil.work at gmail.com> escreveu: > > Hello guys, > > I'm trying to edit > https://freeswitch.org/confluence/display/FREESWITCH/mod_sofia to > document the new "sofia filter", but i don't seem to have permissions? > > > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > phone: +34669448337 > ? > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > 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/20161120/56ac9254/attachment-0001.html From S.Boomstra at telecats.nl Mon Nov 21 13:15:55 2016 From: S.Boomstra at telecats.nl (Sjoerd Boomstra) Date: Mon, 21 Nov 2016 10:15:55 +0000 Subject: [Freeswitch-users] bypass_media after receiving invite with replaces header In-Reply-To: <77974EB6-48DE-4430-B95F-2BDC776D3C97@jerris.com> References: <77974EB6-48DE-4430-B95F-2BDC776D3C97@jerris.com> Message-ID: We will try this issue with the latest version. We have a lot of things attached to FS, so this will take a while (a few weeks). Best regards, Sjoerd. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris Sent: vrijdag 18 november 2016 15:52 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] bypass_media after receiving invite with replaces header 1.4 is long since EOL. Please try latest release version and confirm this is still an issue. On Nov 18, 2016, at 7:09 AM, Sjoerd Boomstra > wrote: Hello, I've got a problem with FreeSwitch version 1.4.26. Our setup is like this: We have the phones registered at a SipXecs cluster. Freeswitch is used as a gateway (sbc) between SipXecs and another SIP device (Audiocodes Mediant). We aim to bypass the media at the FreeSwitch. At first this works as expected, the media does not go through the FreeSwitch. However when a phone issues a transfer, the refer is handled by SipXecs. The invite with replaces header is sent to Freeswitch (via SipX). FreeSwitch handles this SIP message correctly, but it puts itself in the RTP-stream. The original channel has the variables bypass_media and bypass_media_after_bridge set to true. It seems that these variables are not checked when the invite with replaces header is handled. Anyone an idea how to fix this? Best regards, Sjoerd. ----- Sjoerd Boomstra | Senior Business Consultant | Telecats bv | KvK Enschede 06069106 | Tel: 053 488 99 07 | Fax: 053 488 99 10 | E?mail: s.boomstra at telecats.nl | -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161121/7e951430/attachment.html From catogonzalez at gmail.com Mon Nov 21 16:35:20 2016 From: catogonzalez at gmail.com (Cato Gonzalez) Date: Mon, 21 Nov 2016 08:35:20 -0500 Subject: [Freeswitch-users] Bridge an incoming call to an external SIP address Message-ID: Hi everyone, I am trying to get FS to route inbound calls to an outside sip provider but I am getting the call hung up on the FS side just upon the first or second ring on the B-leg. The diaplan action being executed is: I have a softphone registered to this provider sip.linphone.org and it works well when calling XXX from some other IP phone. FS reports this in the logs: [DEBUG] switch_ivr_originate.c:1274 Raw Codec Activation Success L16 at 48000hz 1 channel 20ms [DEBUG] switch_core_codec.c:221 sofia/internal/ anonymous at webrtc.sip.mydomain.co Push codec L16:100 [DEBUG] switch_ivr_originate.c:1343 Play Ringback Tone [%(2000,4000,440,480)] [DEBUG] sofia.c:6760 Channel sofia/internal/anonymous at webrtc.sip.mydomain.co entering state [terminated][500] [NOTICE] sofia.c:7779 Hangup sofia/internal/anonymous at webrtc.sip.mydomain.co [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] [DEBUG] switch_core_codec.c:246 sofia/internal/ anonymous at webrtc.sip.mydomain.co Restore previous codec opus:116. [NOTICE] switch_ivr_originate.c:3523 Hangup sofia/external/ XXX at sip.linphone.org [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL] Result is the same if I try to call any other sip destination. I appreciate any input anyone may give: have been working on this for a week already. Thanks, Cato -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161121/5737a464/attachment.html From ignacio.durli at gmail.com Sat Nov 19 00:13:41 2016 From: ignacio.durli at gmail.com (Ignacio Durli) Date: Fri, 18 Nov 2016 16:13:41 -0500 Subject: [Freeswitch-users] (no subject) Message-ID: Thanks for your reply Joel. Here's the output, I see that the FS process is binding 10.X interface with port 5060, but I don't know why that doesn't match what sofia status is showing: root at osboxes:~# iptables -L -n Chain INPUT (policy ACCEPT) target prot opt source destination Chain FORWARD (policy ACCEPT) target prot opt source destination Chain OUTPUT (policy ACCEPT) target prot opt source destination root at osboxes:~# netstat -putan | grep -i freeswitch tcp 0 0 10.0.3.15:5080 0.0.0.0:* LISTEN 3519/freeswitch tcp 0 0 10.0.3.15:5081 0.0.0.0:* LISTEN 3519/freeswitch tcp 0 0 10.0.3.15:5060 0.0.0.0:* LISTEN 3519/freeswitch tcp 0 0 10.0.3.15:5061 0.0.0.0:* LISTEN 3519/freeswitch tcp 0 0 0.0.0.0:8080 0.0.0.0:* LISTEN 3519/freeswitch tcp 0 0 0.0.0.0:8081 0.0.0.0:* LISTEN 3519/freeswitch tcp 0 0 0.0.0.0:8082 0.0.0.0:* LISTEN 3519/freeswitch tcp 0 0 10.0.3.15:7443 0.0.0.0:* LISTEN 3519/freeswitch tcp 0 0 192.168.99.103:59607 192.168.99.103:3306 ESTABLISHED 3519/freeswitch tcp 0 0 192.168.99.103:59606 192.168.99.103:3306 ESTABLISHED 3519/freeswitch tcp 0 0 192.168.99.103:59611 192.168.99.103:3306 ESTABLISHED 3519/freeswitch tcp6 0 0 ::1:5080 :::* LISTEN 3519/freeswitch tcp6 0 0 ::1:5081 :::* LISTEN 3519/freeswitch tcp6 0 0 ::1:5060 :::* LISTEN 3519/freeswitch tcp6 0 0 ::1:5061 :::* LISTEN 3519/freeswitch tcp6 0 0 ::1:8081 :::* LISTEN 3519/freeswitch tcp6 0 0 ::1:8082 :::* LISTEN 3519/freeswitch tcp6 0 0 :::8021 :::* LISTEN 3519/freeswitch udp 0 0 10.0.3.15:5060 0.0.0.0:* 3519/freeswitch udp 0 0 10.0.3.15:5080 0.0.0.0:* 3519/freeswitch udp 0 0 0.0.0.0:1337 0.0.0.0:* 3519/freeswitch udp6 0 0 ::1:5060 :::* 3519/freeswitch udp6 0 0 ::1:5080 :::* 3519/freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161118/fb445860/attachment-0001.html From mario_fs at mgtech.com Mon Nov 21 19:39:38 2016 From: mario_fs at mgtech.com (Mario G) Date: Mon, 21 Nov 2016 08:39:38 -0800 Subject: [Freeswitch-users] Forward fails with deflect and REFER to null Message-ID: <627DB024-A0FE-4EBC-93F3-8774BF3D463F@mgtech.com> What are the bold deflect and refer to lines below telling me? This worked until last year, not sure if it?s related to 1.6, I can?t nail down a start date. A ring group rings. When one phone?s FWD button is pressed prior to a call or during a call the call is forwarded to VM extension 2921, and the other phones stopped ringing. This no longer works, the set phone does not ring, the call is not transferred to VM and the other phones keep ringing. I couldn?t find anything on the wiki (it only described using codes to turn FWD on/off), etc. to explain why this stopped working. Any help to get the phone forwarding working again is greatly appreciated. Thanks! Mario G YEALINK T48G DSS button is set to forward to 2921, pressing again turns it off. All other DSS keys such as call park, etc work fine. This log shows the only lines that contain 2921 in bold: 2016-11-19 16:27:26.673609 [INFO] mod_dialplan_xml.c:637 Processing Mario G <15051234567>->2921 in context public Dialplan: sofia/internal/15051234567 at 11.123.1.10 parsing [public->unloop] continue=false Dialplan: sofia/internal/15051234567 at 11.123.1.10 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/internal/15051234567 at 11.123.1.10 Regex (PASS) [unloop] ${sip_looped_call}(true) =~ /^true$/ break=on-false Dialplan: sofia/internal/15051234567 at 11.123.1.10 Action deflect(${destination_number}) 2016-11-19 16:27:26.673609 [DEBUG] switch_core_state_machine.c:286 (sofia/internal/15051234567 at 11.123.1.10) State Change CS_ROUTING -> CS_EXECUTE 2016-11-19 16:27:26.673609 [DEBUG] switch_core_state_machine.c:643 (sofia/internal/15051234567 at 11.123.1.10) State ROUTING going to sleep 2016-11-19 16:27:26.673609 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/15051234567 at 11.123.1.10) Running State Change CS_EXECUTE 2016-11-19 16:27:26.673609 [DEBUG] switch_core_state_machine.c:650 (sofia/internal/15051234567 at 11.123.1.10) State EXECUTE 2016-11-19 16:27:26.673609 [DEBUG] mod_sofia.c:198 sofia/internal/15051234567 at 11.123.1.10 SOFIA EXECUTE 2016-11-19 16:27:26.673609 [DEBUG] switch_core_state_machine.c:328 sofia/internal/15051234567 at 11.123.1.10 Standard EXECUTE EXECUTE sofia/internal/15051234567 at 11.123.1.10 deflect(2921) 2016-11-19 16:27:26.673609 [DEBUG] sofia.c:8349 Process REFER to [(null)@2921] -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161121/1fc10f7f/attachment.html From randomdev4 at gmail.com Mon Nov 21 19:39:23 2016 From: randomdev4 at gmail.com (Tim Smith) Date: Mon, 21 Nov 2016 16:39:23 +0000 Subject: [Freeswitch-users] Freeswitch routing inbound calls over SIP instead of TLS/SRTP In-Reply-To: References: Message-ID: Hi Steve, Sorry for the delay ack'ing your mail ... yeah, guess I should maybe look into filing a JIRA. On 20 November 2016 at 00:33, Steven Ayre wrote: > Looks like a bug to me. Your first snippet shows the contact stored in the > database uses the 'sips:' scheme, but sofia_contact is returning 'sip:' > > In the code it looks like sofia_contact fetches the contact using > select_from_profile which invokes contact_callback. In contact_callback it's > hardcoded to use sip: plus the result of sofia_glue_strip_proto. That looks > to me like it can never return a sips URI even though it's stored in the > database. > > I'd file a jira. > > Steve > > On 18 November 2016 at 10:08, Tim Smith wrote: >> >> Debian GNU/Linux 8 (jessie) >> Linux my 3.16.0-4-amd64 #1 SMP Debian 3.16.36-1+deb8u2 (2016-10-19) >> x86_64 GNU/Linux >> FreeSWITCH Version 1.6.12-20-b91a0a6~64bit (-20-b91a0a6 64bit) >> >> I have a Vtech handset with TLS/SRTP enabled registered with >> Freeswitch as below: >> >> >> Call-ID: a0000a0a000aa000 >> User: 2001 at my.example.com >> Contact: "my" >> Agent: Vtech Vesa VSP736A 2.0.3.2-0 >> Status: Registered(TLS)(unknown) EXP(2016-11-18 10:56:57) >> EXPSECS(3646) >> Ping-Status: Reachable >> Ping-Time: 0.00 >> Host: my >> IP: 198.51.100.81 >> Port: 58348 >> Auth-User: 2001 >> Auth-Realm: my.example.com >> MWI-Account: 2001 at my.example.com >> >> >> sofia_contact is happy : >> >> freeswitch at my>sofia_contact internal/2001 >> sofia/internal/sip:2001 at 198.51.100.81:58348 >> >> I have an inbound dial plan configured as follows: >> >> >> >> >> >> > data="${sofia_contact(internal/2001)}"/> >> >> >> >> >> The problem is Freeswitch is sending invites over SIP/RTP and not >> TLS/SRTP and so the calls never get through : >> >> INVITE sip:2001 at 198.51.100.81:58348 SIP/2.0 >> Via: SIP/2.0/UDP 203.0.113.4;rport;branch=z9hG4bKvHjgXXpFF77XK >> From: "Anonymous" ;tag=rKmXQjZN8SFXp >> To: >> m=audio 32190 RTP/AVP 8 98 9 101 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:98 G726-32/8000 >> a=rtpmap:9 G722/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=ptime:20 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> 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 randomdev4 at gmail.com Mon Nov 21 19:40:20 2016 From: randomdev4 at gmail.com (Tim Smith) Date: Mon, 21 Nov 2016 16:40:20 +0000 Subject: [Freeswitch-users] Freeswitch routing inbound calls over SIP instead of TLS/SRTP In-Reply-To: <931AA1AF-4E99-46CF-A6F0-CE4AE3AE1C93@kavun.ch> References: <931AA1AF-4E99-46CF-A6F0-CE4AE3AE1C93@kavun.ch> Message-ID: Hi Emrah, Its a test box. So one internal, one external, one endpoint .... no scope for confusion. ;-) On 20 November 2016 at 11:23, Emrah wrote: > Actually, your sofia_contact is not happy.. I'm not seeing the transport param for tls in there. > How many profiles are you running? > Are you sure sofia_contact isn't giving you the value of another endpoint registered with UDP? > Also, I went through your message fast, but I don't think you're securing your rtp... > >> On Nov 18, 2016, at 11:08 AM, Tim Smith wrote: >> >> Debian GNU/Linux 8 (jessie) >> Linux my 3.16.0-4-amd64 #1 SMP Debian 3.16.36-1+deb8u2 (2016-10-19) >> x86_64 GNU/Linux >> FreeSWITCH Version 1.6.12-20-b91a0a6~64bit (-20-b91a0a6 64bit) >> >> I have a Vtech handset with TLS/SRTP enabled registered with >> Freeswitch as below: >> >> >> Call-ID: a0000a0a000aa000 >> User: 2001 at my.example.com >> Contact: "my" >> Agent: Vtech Vesa VSP736A 2.0.3.2-0 >> Status: Registered(TLS)(unknown) EXP(2016-11-18 10:56:57) EXPSECS(3646) >> Ping-Status: Reachable >> Ping-Time: 0.00 >> Host: my >> IP: 198.51.100.81 >> Port: 58348 >> Auth-User: 2001 >> Auth-Realm: my.example.com >> MWI-Account: 2001 at my.example.com >> >> >> sofia_contact is happy : >> >> freeswitch at my>sofia_contact internal/2001 >> sofia/internal/sip:2001 at 198.51.100.81:58348 >> >> I have an inbound dial plan configured as follows: >> >> >> >> >> >> >> >> >> >> >> The problem is Freeswitch is sending invites over SIP/RTP and not >> TLS/SRTP and so the calls never get through : >> >> INVITE sip:2001 at 198.51.100.81:58348 SIP/2.0 >> Via: SIP/2.0/UDP 203.0.113.4;rport;branch=z9hG4bKvHjgXXpFF77XK >> From: "Anonymous" ;tag=rKmXQjZN8SFXp >> To: >> m=audio 32190 RTP/AVP 8 98 9 101 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:98 G726-32/8000 >> a=rtpmap:9 G722/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=ptime:20 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> 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 rajil.s at gmail.com Mon Nov 21 19:50:48 2016 From: rajil.s at gmail.com (Rajil Saraswat) Date: Mon, 21 Nov 2016 10:50:48 -0600 Subject: [Freeswitch-users] Letsencrypt and TLS In-Reply-To: References: Message-ID: Hello, I have been using self generated certificates ( https://wiki.freeswitch.org/wiki/SIP_TLS) until now. Is it possible to use Letsencrypt generated certificates for TLS? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161121/e80c4975/attachment.html From josefu at gmail.com Mon Nov 21 20:05:20 2016 From: josefu at gmail.com (=?UTF-8?Q?Jose_Fco=2E_Irles_Dur=C3=A1?=) Date: Mon, 21 Nov 2016 18:05:20 +0100 Subject: [Freeswitch-users] Members in fifo at startup Message-ID: Hi, I'm testing the fifo module with the default configuration (configuration packaged in debs) and I am not able to make it work. The default configuration (/etc/freeswitch/autoload_configs/fifo.conf.xml): {fifo_member_wait=nowait}user/1001@$${domain} {fifo_member_wait=nowait}user/1002@$${domain} I can see two members in the "cool_fifo" (1001 and 1002) configured, but when I start freeswitch, If I execute the command "fifo list" I don't see thats members (the fifo has no members) Only when I execute the command "fifo reparse" after freeswitch is started I can see the "cool_fifo" with the members. Where is the problem? Should I execute "fifo reparse" once freeswitch is started? Thanks in advance -- Jose Fco. Irles Dur? From mike at jerris.com Mon Nov 21 20:09:58 2016 From: mike at jerris.com (Michael Jerris) Date: Mon, 21 Nov 2016 12:09:58 -0500 Subject: [Freeswitch-users] Letsencrypt and TLS In-Reply-To: References: Message-ID: <94FB5E2B-E2EB-42D5-8B1F-10D74ECC4317@jerris.com> there are some instructions here: https://freeswitch.org/confluence/display/FREESWITCH/Debian+8+Jessie#Debian8Jessie-Scriptinstallfreeswitchdemowithverto_communicator > On Nov 21, 2016, at 11:50 AM, Rajil Saraswat wrote: > > Hello, > > I have been using self generated certificates (https://wiki.freeswitch.org/wiki/SIP_TLS ) until now. > Is it possible to use Letsencrypt generated certificates for TLS? > > Thanks > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161121/c2837f7c/attachment.html From jaradmorgan at gmail.com Mon Nov 21 22:19:23 2016 From: jaradmorgan at gmail.com (Jarad Morgan) Date: Mon, 21 Nov 2016 19:19:23 +0000 Subject: [Freeswitch-users] one way audio coming off hold Message-ID: Hello, we are running into some issues with one way audio coming off hold with messaging between Avaya's. Running FS 1.6.10. Is there a FS config setting to leave out the media type attribute (sendonly,sendrecv) in the 200 OK sent by FS when doing late media negotiation between a Avaya device? Thanks!! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161121/68cf0b4c/attachment.html From mike at jerris.com Mon Nov 21 22:37:21 2016 From: mike at jerris.com (Michael Jerris) Date: Mon, 21 Nov 2016 14:37:21 -0500 Subject: [Freeswitch-users] one way audio coming off hold In-Reply-To: References: Message-ID: <324F9FE2-65A2-4A3F-9285-D0FF52B67093@jerris.com> if there is an sdp and it does not specify sendonly, recvonly, or inactive, it means that stream is sendrecv. This is normal operation of sdp, many things won?t actually include the sendrecv as size of sdp is a bigger and bigger issue these days. What do you mean about a config setting to leave it out? > On Nov 21, 2016, at 2:19 PM, Jarad Morgan wrote: > > Hello, we are running into some issues with one way audio coming off hold with messaging between Avaya's. Running FS 1.6.10. > > Is there a FS config setting to leave out the media type attribute (sendonly,sendrecv) in the 200 OK sent by FS when doing late media negotiation between a Avaya device? > > Thanks!! From jaradmorgan at gmail.com Mon Nov 21 22:59:51 2016 From: jaradmorgan at gmail.com (Jarad Morgan) Date: Mon, 21 Nov 2016 19:59:51 +0000 Subject: [Freeswitch-users] one way audio coming off hold In-Reply-To: <324F9FE2-65A2-4A3F-9285-D0FF52B67093@jerris.com> References: <324F9FE2-65A2-4A3F-9285-D0FF52B67093@jerris.com> Message-ID: So long story short here is the issue were running into. Sorry for the crappy messaging diagram. //when going on hold. Avaya INVITE to FS no SDP FS 100 trying no SDP to Avaya FS 200 OK w SDP to Avaya Avaya ACK w SDP(a=sendonly) to FS //were now on hold. //going off hold where we run into the one way audio. Avaya INVITE to FS no SDP FS 100 trying no SDP to Avaya FS 200 OK w SDP (a=sendonly) to Avaya Avaya ACK w SDP (a=recvonly) to FS We get a recvonly from Avaya so were in listen mode instead of sendrecv. This is a new issue since we upgraded from and older version to 1.6. The previous version of FS we didnt see the a=sendonly in the 200 OK when going off hold so i thought that may be configurable in the newer version to leave those media attributes out of the SDP? Thanks!! On Mon, Nov 21, 2016 at 2:38 PM Michael Jerris wrote: > if there is an sdp and it does not specify sendonly, recvonly, or > inactive, it means that stream is sendrecv. This is normal operation of > sdp, many things won?t actually include the sendrecv as size of sdp is a > bigger and bigger issue these days. What do you mean about a config > setting to leave it out? > > > On Nov 21, 2016, at 2:19 PM, Jarad Morgan wrote: > > > > Hello, we are running into some issues with one way audio coming off > hold with messaging between Avaya's. Running FS 1.6.10. > > > > Is there a FS config setting to leave out the media type attribute > (sendonly,sendrecv) in the 200 OK sent by FS when doing late media > negotiation between a Avaya device? > > > > 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/20161121/9ddf3fde/attachment.html From mike at jerris.com Mon Nov 21 23:05:48 2016 From: mike at jerris.com (Michael Jerris) Date: Mon, 21 Nov 2016 15:05:48 -0500 Subject: [Freeswitch-users] one way audio coming off hold In-Reply-To: References: <324F9FE2-65A2-4A3F-9285-D0FF52B67093@jerris.com> Message-ID: > On Nov 21, 2016, at 2:59 PM, Jarad Morgan wrote: > > So long story short here is the issue were running into. Sorry for the crappy messaging diagram. > > //when going on hold. > Avaya INVITE to FS no SDP > FS 100 trying no SDP to Avaya > FS 200 OK w SDP to Avaya > Avaya ACK w SDP(a=sendonly) to FS > > //were now on hold. > > //going off hold where we run into the one way audio. > Avaya INVITE to FS no SDP > FS 100 trying no SDP to Avaya > FS 200 OK w SDP (a=sendonly) to Avaya > Avaya ACK w SDP (a=recvonly) to FS This is not a valid way to go off hold. The avaya would need to somehow indicate that it is changing the stream to sendrecv, they don?t in what you describe here. > > We get a recvonly from Avaya so were in listen mode instead of sendrecv. > > This is a new issue since we upgraded from and older version to 1.6. The previous version of FS we didnt see the a=sendonly in the 200 OK when going off hold so i thought that may be configurable in the newer version to leave those media attributes out of the SDP? Not sure what you mean. We didn?t ?see? it? > > Thanks!! > > On Mon, Nov 21, 2016 at 2:38 PM Michael Jerris > wrote: > if there is an sdp and it does not specify sendonly, recvonly, or inactive, it means that stream is sendrecv. This is normal operation of sdp, many things won?t actually include the sendrecv as size of sdp is a bigger and bigger issue these days. What do you mean about a config setting to leave it out? > > > On Nov 21, 2016, at 2:19 PM, Jarad Morgan > wrote: > > > > Hello, we are running into some issues with one way audio coming off hold with messaging between Avaya's. Running FS 1.6.10. > > > > Is there a FS config setting to leave out the media type attribute (sendonly,sendrecv) in the 200 OK sent by FS when doing late media negotiation between a Avaya device? > > > > 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 _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161121/a20afe99/attachment-0001.html From joel at gogii.net Mon Nov 21 23:08:48 2016 From: joel at gogii.net (Joel Serrano) Date: Mon, 21 Nov 2016 12:08:48 -0800 Subject: [Freeswitch-users] FS not receiving incoming calls In-Reply-To: References: Message-ID: Hi Ignacio, Please remember to always reply to the thread. Can you check the sip-ip and/or ext-sip-ip of your active profiles? Looks like the issue could be there. On Fri, Nov 18, 2016 at 1:13 PM, Ignacio Durli wrote: > Thanks for your reply Joel. Here's the output, I see that the FS process > is binding 10.X interface with port 5060, but I don't know why that doesn't > match what sofia status is showing: > > root at osboxes:~# iptables -L -n > Chain INPUT (policy ACCEPT) > target prot opt source destination > > Chain FORWARD (policy ACCEPT) > target prot opt source destination > > Chain OUTPUT (policy ACCEPT) > target prot opt source destination > root at osboxes:~# netstat -putan | grep -i freeswitch > tcp 0 0 10.0.3.15:5080 0.0.0.0:* > LISTEN 3519/freeswitch > tcp 0 0 10.0.3.15:5081 0.0.0.0:* > LISTEN 3519/freeswitch > tcp 0 0 10.0.3.15:5060 0.0.0.0:* > LISTEN 3519/freeswitch > tcp 0 0 10.0.3.15:5061 0.0.0.0:* > LISTEN 3519/freeswitch > tcp 0 0 0.0.0.0:8080 0.0.0.0:* > LISTEN 3519/freeswitch > tcp 0 0 0.0.0.0:8081 0.0.0.0:* > LISTEN 3519/freeswitch > tcp 0 0 0.0.0.0:8082 0.0.0.0:* > LISTEN 3519/freeswitch > tcp 0 0 10.0.3.15:7443 0.0.0.0:* > LISTEN 3519/freeswitch > tcp 0 0 192.168.99.103:59607 192.168.99.103:3306 > ESTABLISHED 3519/freeswitch > tcp 0 0 192.168.99.103:59606 192.168.99.103:3306 > ESTABLISHED 3519/freeswitch > tcp 0 0 192.168.99.103:59611 192.168.99.103:3306 > ESTABLISHED 3519/freeswitch > tcp6 0 0 ::1:5080 :::* LISTEN > 3519/freeswitch > tcp6 0 0 ::1:5081 :::* LISTEN > 3519/freeswitch > tcp6 0 0 ::1:5060 :::* LISTEN > 3519/freeswitch > tcp6 0 0 ::1:5061 :::* LISTEN > 3519/freeswitch > tcp6 0 0 ::1:8081 :::* LISTEN > 3519/freeswitch > tcp6 0 0 ::1:8082 :::* LISTEN > 3519/freeswitch > tcp6 0 0 :::8021 :::* LISTEN > 3519/freeswitch > udp 0 0 10.0.3.15:5060 0.0.0.0:* > 3519/freeswitch > udp 0 0 10.0.3.15:5080 0.0.0.0:* > 3519/freeswitch > udp 0 0 0.0.0.0:1337 0.0.0.0:* > 3519/freeswitch > udp6 0 0 ::1:5060 :::* > 3519/freeswitch > udp6 0 0 ::1:5080 :::* > 3519/freeswitch > > On Fri, Nov 18, 2016 at 10:36 AM, Joel Serrano wrote: > Output of: > > # iptables -L -n > # netstat -putan | grep -i freeswitch > > Please :) > > On Fri, Nov 18, 2016 at 10:12 AM, Ignacio Durli < > ignaciodurli at cinchcast.com> wrote: > >> I upgraded to FS 1.6.12 which I installed in a new Debian 8 instance ran >> in VirtualBox. After compiling and running all the config scripts I?m not >> able to establish a SIP call. After some troubleshooting I found that calls >> are being answered on my eth1 interface (10.0.x.x) but not on my eth0 >> interface (192.168.99.x). >> >> >> >> The IP is configured in vars.xml properly and sofia status is showing: >> >> >> >> freeswitch at freeswitch> sofia status >> >> Name Type >> Data State >> >> ============================================================ >> ===================================== >> >> external-ipv6 profile sip:mod_sofia@ >> [::1]:5080 RUNNING (0) >> >> external-ipv6 profile sip:mod_sofia@ >> [::1]:5081 RUNNING (0) (TLS) >> >> external profile >> sip:mod_sofia at 192.168.99.103:5080 RUNNING (0) >> >> external profile >> sip:mod_sofia at 192.168.99.103:5081 RUNNING (0) (TLS) >> >> 192.168.99.103 alias >> internal ALIASED >> >> internal-ipv6 profile >> sip:mod_sofia@[::1]:5060 RUNNING (0) >> >> internal-ipv6 profile >> sip:mod_sofia@[::1]:5061 RUNNING (0) (TLS) >> >> internal profile >> sip:mod_sofia at 192.168.99.103:5060 RUNNING (0) >> >> internal profile >> sip:mod_sofia at 192.168.99.103:5061 RUNNING (0) (TLS) >> >> ============================================================ >> ===================================== >> >> 4 profiles 1 alias >> >> >> >> >> >> Which looks fine. Also the ifconfig of the new Debian instance seems OK, >> since it?s the same that I have on another VM where an older FS version is >> running. >> >> >> >> root at osboxes:/usr/local/freeswitch/bin# ifconfig >> >> eth0 Link encap:Ethernet HWaddr 08:00:27:3a:f0:c6 >> >> inet addr:192.168.99.103 Bcast:192.168.99.255 >> Mask:255.255.255.0 >> >> inet6 addr: fe80::a00:27ff:fe3a:f0c6/64 Scope:Link >> >> UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1 >> >> RX packets:2482 errors:0 dropped:0 overruns:0 frame:0 >> >> TX packets:1034 errors:0 dropped:0 overruns:0 carrier:0 >> >> collisions:0 txqueuelen:1000 >> >> RX bytes:353837 (345.5 KiB) TX bytes:227253 (221.9 KiB) >> >> >> >> eth1 Link encap:Ethernet HWaddr 08:00:27:f7:f1:24 >> >> inet addr:10.0.3.15 Bcast:10.0.3.255 Mask:255.255.255.0 >> >> inet6 addr: fe80::a00:27ff:fef7:f124/64 Scope:Link >> >> UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1 >> >> RX packets:4298 errors:0 dropped:0 overruns:0 frame:0 >> >> TX packets:7037 errors:0 dropped:0 overruns:0 carrier:0 >> >> collisions:0 txqueuelen:1000 >> >> RX bytes:437241 (426.9 KiB) TX bytes:829683 (810.2 KiB) >> >> >> >> lo Link encap:Local Loopback >> >> inet addr:127.0.0.1 Mask:255.0.0.0 >> >> inet6 addr: ::1/128 Scope:Host >> >> UP LOOPBACK RUNNING MTU:65536 Metric:1 >> >> RX packets:49100 errors:0 dropped:0 overruns:0 frame:0 >> >> TX packets:49100 errors:0 dropped:0 overruns:0 carrier:0 >> >> collisions:0 txqueuelen:0 >> >> RX bytes:8431576 (8.0 MiB) TX bytes:8431576 (8.0 MiB) >> >> >> >> >> >> I honestly don?t know what?s going on. Acls are fine, and I don?t see any >> error in the FS log. I traced the network traffic and when trying to call >> to 192.168.99.103:5060 I get an ICM (Destination unreachable) which >> doesn?t make sense according to what sofia status is showing. Any hint 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/20161121/b4e541f0/attachment-0001.html From joel at gogii.net Mon Nov 21 23:10:31 2016 From: joel at gogii.net (Joel Serrano) Date: Mon, 21 Nov 2016 12:10:31 -0800 Subject: [Freeswitch-users] Letsencrypt and TLS In-Reply-To: <94FB5E2B-E2EB-42D5-8B1F-10D74ECC4317@jerris.com> References: <94FB5E2B-E2EB-42D5-8B1F-10D74ECC4317@jerris.com> Message-ID: You can also try startssl.com They have longer expirations than letsencrypt. On Mon, Nov 21, 2016 at 9:09 AM, Michael Jerris wrote: > there are some instructions here: > > https://freeswitch.org/confluence/display/FREESWITCH/ > Debian+8+Jessie#Debian8Jessie-Scriptinstallfreeswitchdemowit > hverto_communicator > > On Nov 21, 2016, at 11:50 AM, Rajil Saraswat wrote: > > Hello, > > I have been using self generated certificates ( > https://wiki.freeswitch.org/wiki/SIP_TLS) until now. > Is it possible to use Letsencrypt generated certificates for TLS? > > 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/20161121/ab56107f/attachment.html From edwardludd at gmail.com Mon Nov 21 23:15:46 2016 From: edwardludd at gmail.com (Ned Ludd) Date: Mon, 21 Nov 2016 15:15:46 -0500 Subject: [Freeswitch-users] Help calling phrase macros from Lua Message-ID: So I'm trying to develop a simple voicemail IVR. In "sounds.xml" there's a nice macro for "voicemail_message_count", but I can't seem to figure out how to get the count to feed it. Can you help? I know I can use this to get the count in JSON format: count = api:executeString("vm_fsdb_msg_count default default " .. domain .. " " .. extension .. " inbox"); But how do I get a count that I can send to that phrase macro? Confused. (And excuse my ignorance). ----- Ned Ludd -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161121/e68d60e6/attachment.html From mike at jerris.com Mon Nov 21 23:22:51 2016 From: mike at jerris.com (Michael Jerris) Date: Mon, 21 Nov 2016 15:22:51 -0500 Subject: [Freeswitch-users] Help calling phrase macros from Lua In-Reply-To: References: Message-ID: <9CDBBBF7-F18D-441F-88B0-61D22F6BC426@jerris.com> I don?t know a ton about this one as it was contributed but this was designed to let you do what you are looking to do: https://freeswitch.org/confluence/display/FREESWITCH/mod_voicemail_ivr > On Nov 21, 2016, at 3:15 PM, Ned Ludd wrote: > > So I'm trying to develop a simple voicemail IVR. > > In "sounds.xml" there's a nice macro for "voicemail_message_count", but I can't seem to figure out how to get the count to feed it. Can you help? > > I know I can use this to get the count in JSON format: > > count = api:executeString("vm_fsdb_msg_count default default " .. domain .. " " .. extension .. " inbox"); > > But how do I get a count that I can send to that phrase macro? > > Confused. (And excuse my ignorance). -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161121/3ae7d858/attachment.html From abaci64 at gmail.com Mon Nov 21 23:28:11 2016 From: abaci64 at gmail.com (Abaci B) Date: Mon, 21 Nov 2016 15:28:11 -0500 Subject: [Freeswitch-users] Help calling phrase macros from Lua In-Reply-To: References: Message-ID: https://freeswitch.org/confluence/display/FREESWITCH/mod_voicemail#mod_voicemail-vm_boxcount On Mon, Nov 21, 2016 at 3:15 PM, Ned Ludd wrote: > So I'm trying to develop a simple voicemail IVR. > > In "sounds.xml" there's a nice macro for "voicemail_message_count", but I > can't seem to figure out how to get the count to feed it. Can you help? > > I know I can use this to get the count in JSON format: > > count = api:executeString("vm_fsdb_msg_count default default " .. > domain .. " " .. extension .. " inbox"); > > But how do I get a count that I can send to that phrase macro? > > Confused. (And excuse my ignorance). > > ----- > Ned Ludd > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161121/8b27db25/attachment.html From jaradmorgan at gmail.com Mon Nov 21 23:33:08 2016 From: jaradmorgan at gmail.com (Jarad Morgan) Date: Mon, 21 Nov 2016 20:33:08 +0000 Subject: [Freeswitch-users] one way audio coming off hold In-Reply-To: References: <324F9FE2-65A2-4A3F-9285-D0FF52B67093@jerris.com> Message-ID: Sorry what i meant was on the older FS version the sendonly attribute was not present in the 200 OK message back to Avaya. For example SIP flow on older version. Avaya INVITE to FS no SDP FS 100 trying no SDP to Avaya FS 200 OK w SDP to Avaya <-- no sendonly media attribute specified here. Avaya ACK w SDP to FS <-- no media attribute so default sendrecv Call is taken off hold.. Older version and 1.6 are running the same configs. I can bug the Avaya guys to fix their stuff but was just hoping to fix in FS if it was easily possible. On Mon, Nov 21, 2016 at 3:06 PM Michael Jerris wrote: > On Nov 21, 2016, at 2:59 PM, Jarad Morgan wrote: > > So long story short here is the issue were running into. Sorry for the > crappy messaging diagram. > > //when going on hold. > Avaya INVITE to FS no SDP > FS 100 trying no SDP to Avaya > FS 200 OK w SDP to Avaya > Avaya ACK w SDP(a=sendonly) to FS > > //were now on hold. > > //going off hold where we run into the one way audio. > Avaya INVITE to FS no SDP > FS 100 trying no SDP to Avaya > FS 200 OK w SDP (a=sendonly) to Avaya > Avaya ACK w SDP (a=recvonly) to FS > > > This is not a valid way to go off hold. The avaya would need to somehow > indicate that it is changing the stream to sendrecv, they don?t in what you > describe here. > > > We get a recvonly from Avaya so were in listen mode instead of sendrecv. > > This is a new issue since we upgraded from and older version to 1.6. The > previous version of FS we didnt see the a=sendonly in the 200 OK when going > off hold so i thought that may be configurable in the newer version to > leave those media attributes out of the SDP? > > > Not sure what you mean. We didn?t ?see? it? > > > Thanks!! > > On Mon, Nov 21, 2016 at 2:38 PM Michael Jerris wrote: > > if there is an sdp and it does not specify sendonly, recvonly, or > inactive, it means that stream is sendrecv. This is normal operation of > sdp, many things won?t actually include the sendrecv as size of sdp is a > bigger and bigger issue these days. What do you mean about a config > setting to leave it out? > > > On Nov 21, 2016, at 2:19 PM, Jarad Morgan wrote: > > > > Hello, we are running into some issues with one way audio coming off > hold with messaging between Avaya's. Running FS 1.6.10. > > > > Is there a FS config setting to leave out the media type attribute > (sendonly,sendrecv) in the 200 OK sent by FS when doing late media > negotiation between a Avaya device? > > > > 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > 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/20161121/4640739b/attachment-0001.html From gled at remote-shell.net Tue Nov 22 00:32:37 2016 From: gled at remote-shell.net (=?UTF-8?Q?Tristan_Mah=c3=a9?=) Date: Mon, 21 Nov 2016 13:32:37 -0800 Subject: [Freeswitch-users] Letsencrypt and TLS In-Reply-To: References: <94FB5E2B-E2EB-42D5-8B1F-10D74ECC4317@jerris.com> Message-ID: <13249e30-0ac2-c5ed-cc5a-b9e683892ac9@remote-shell.net> Not to step on any toes, but startssl certificates are to be untrusted by the major browsers very soon, if not already IIRC. ( some shady behaviour from the parent company, google can give more info if you want ). On 11/21/2016 12:10 PM, Joel Serrano wrote: > You can also try startssl.com > > They have longer expirations than letsencrypt. > > > > On Mon, Nov 21, 2016 at 9:09 AM, Michael Jerris > wrote: > > there are some instructions here: > > https://freeswitch.org/confluence/display/FREESWITCH/Debian+8+Jessie#Debian8Jessie-Scriptinstallfreeswitchdemowithverto_communicator > > >> On Nov 21, 2016, at 11:50 AM, Rajil Saraswat > > wrote: >> >> Hello, >> >> I have been using self generated certificates >> (https://wiki.freeswitch.org/wiki/SIP_TLS >> ) until now. >> Is it possible to use Letsencrypt generated certificates for TLS? >> >> 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 > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161121/b32fa456/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 516 bytes Desc: OpenPGP digital signature Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161121/b32fa456/attachment.bin From mike at jerris.com Tue Nov 22 00:59:53 2016 From: mike at jerris.com (Michael Jerris) Date: Mon, 21 Nov 2016 16:59:53 -0500 Subject: [Freeswitch-users] one way audio coming off hold In-Reply-To: References: <324F9FE2-65A2-4A3F-9285-D0FF52B67093@jerris.com> Message-ID: If we were doing that, that was a bug on our part. Nothing told us to take the call off hold. > On Nov 21, 2016, at 3:33 PM, Jarad Morgan wrote: > > Sorry what i meant was on the older FS version the sendonly attribute was not present in the 200 OK message back to Avaya. For example SIP flow on older version. > > Avaya INVITE to FS no SDP > FS 100 trying no SDP to Avaya > FS 200 OK w SDP to Avaya <-- no sendonly media attribute specified here. > Avaya ACK w SDP to FS <-- no media attribute so default sendrecv > > Call is taken off hold.. > > Older version and 1.6 are running the same configs. > > I can bug the Avaya guys to fix their stuff but was just hoping to fix in FS if it was easily possible. > > > > On Mon, Nov 21, 2016 at 3:06 PM Michael Jerris > wrote: >> On Nov 21, 2016, at 2:59 PM, Jarad Morgan > wrote: >> >> So long story short here is the issue were running into. Sorry for the crappy messaging diagram. >> >> //when going on hold. >> Avaya INVITE to FS no SDP >> FS 100 trying no SDP to Avaya >> FS 200 OK w SDP to Avaya >> Avaya ACK w SDP(a=sendonly) to FS >> >> //were now on hold. >> >> //going off hold where we run into the one way audio. >> Avaya INVITE to FS no SDP >> FS 100 trying no SDP to Avaya >> FS 200 OK w SDP (a=sendonly) to Avaya >> Avaya ACK w SDP (a=recvonly) to FS > > This is not a valid way to go off hold. The avaya would need to somehow indicate that it is changing the stream to sendrecv, they don?t in what you describe here. > >> >> We get a recvonly from Avaya so were in listen mode instead of sendrecv. >> >> This is a new issue since we upgraded from and older version to 1.6. The previous version of FS we didnt see the a=sendonly in the 200 OK when going off hold so i thought that may be configurable in the newer version to leave those media attributes out of the SDP? > > Not sure what you mean. We didn?t ?see? it? > >> >> Thanks!! >> >> On Mon, Nov 21, 2016 at 2:38 PM Michael Jerris > wrote: >> if there is an sdp and it does not specify sendonly, recvonly, or inactive, it means that stream is sendrecv. This is normal operation of sdp, many things won?t actually include the sendrecv as size of sdp is a bigger and bigger issue these days. What do you mean about a config setting to leave it out? >> >> > On Nov 21, 2016, at 2:19 PM, Jarad Morgan > wrote: >> > >> > Hello, we are running into some issues with one way audio coming off hold with messaging between Avaya's. Running FS 1.6.10. >> > >> > Is there a FS config setting to leave out the media type attribute (sendonly,sendrecv) in the 200 OK sent by FS when doing late media negotiation between a Avaya device? >> > >> > Thanks!! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161121/d91dfa18/attachment.html From jungleboogie0 at gmail.com Tue Nov 22 03:50:06 2016 From: jungleboogie0 at gmail.com (jungle Boogie) Date: Mon, 21 Nov 2016 16:50:06 -0800 Subject: [Freeswitch-users] Letsencrypt and TLS In-Reply-To: <94FB5E2B-E2EB-42D5-8B1F-10D74ECC4317@jerris.com> References: <94FB5E2B-E2EB-42D5-8B1F-10D74ECC4317@jerris.com> Message-ID: On 21 November 2016 at 09:09, Michael Jerris wrote: > there are some instructions here: > > https://freeswitch.org/confluence/display/FREESWITCH/Debian+8+Jessie#Debian8Jessie-Scriptinstallfreeswitchdemowithverto_communicator Are these the instructions used to build the usb freeswitch image? I don't remember what Brian called it and I don't see an obvious blog entry about it, either: https://freeswitch.org/blog/ There's lots of text that looks like it should be links on the blog. -- ------- inum: 883510009027723 sip: jungleboogie at sip2sip.info From krice at freeswitch.org Tue Nov 22 04:09:12 2016 From: krice at freeswitch.org (Ken Rice) Date: Mon, 21 Nov 2016 19:09:12 -0600 Subject: [Freeswitch-users] Letsencrypt and TLS In-Reply-To: References: <94FB5E2B-E2EB-42D5-8B1F-10D74ECC4317@jerris.com> Message-ID: <045a01d2445d$0a16cf40$1e446dc0$@freeswitch.org> No these scripts aren't used to build the USB image. It uses self signed certs as to avoid the DNS requirements and such of lets encrypt -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of jungle Boogie Sent: Monday, November 21, 2016 6:50 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Letsencrypt and TLS On 21 November 2016 at 09:09, Michael Jerris wrote: > there are some instructions here: > > https://freeswitch.org/confluence/display/FREESWITCH/Debian+8+Jessie#D > ebian8Jessie-Scriptinstallfreeswitchdemowithverto_communicator Are these the instructions used to build the usb freeswitch image? I don't remember what Brian called it and I don't see an obvious blog entry about it, either: https://freeswitch.org/blog/ There's lots of text that looks like it should be links on the blog. -- ------- inum: 883510009027723 sip: jungleboogie at sip2sip.info _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From brian at freeswitch.org Tue Nov 22 04:53:40 2016 From: brian at freeswitch.org (Brian West) Date: Tue, 22 Nov 2016 01:53:40 +0000 Subject: [Freeswitch-users] Letsencrypt and TLS In-Reply-To: References: <94FB5E2B-E2EB-42D5-8B1F-10D74ECC4317@jerris.com> Message-ID: Untrusted by many due to shady practices, I gave up on StartSSL! On Mon, Nov 21, 2016 at 3:11 PM Joel Serrano wrote: > You can also try startssl.com > > They have longer expirations than letsencrypt. > > > > On Mon, Nov 21, 2016 at 9:09 AM, Michael Jerris wrote: > > there are some instructions here: > > > https://freeswitch.org/confluence/display/FREESWITCH/Debian+8+Jessie#Debian8Jessie-Scriptinstallfreeswitchdemowithverto_communicator > > On Nov 21, 2016, at 11:50 AM, Rajil Saraswat wrote: > > Hello, > > I have been using self generated certificates ( > https://wiki.freeswitch.org/wiki/SIP_TLS) until now. > Is it possible to use Letsencrypt generated certificates for TLS? > > 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161122/b713c459/attachment.html From krice at freeswitch.org Tue Nov 22 05:14:44 2016 From: krice at freeswitch.org (Ken Rice) Date: Mon, 21 Nov 2016 20:14:44 -0600 Subject: [Freeswitch-users] Letsencrypt and TLS In-Reply-To: References: <94FB5E2B-E2EB-42D5-8B1F-10D74ECC4317@jerris.com> Message-ID: Not to mention who cares about the short expire on lets encrypt. Thats why they give you a client to automate it all and you can just cron the auto renew monthly Sent from my iPhone > On Nov 21, 2016, at 19:53, Brian West wrote: > > Untrusted by many due to shady practices, I gave up on StartSSL! > >> On Mon, Nov 21, 2016 at 3:11 PM Joel Serrano wrote: >> You can also try startssl.com >> >> They have longer expirations than letsencrypt. >> >> >> >> On Mon, Nov 21, 2016 at 9:09 AM, Michael Jerris wrote: >> there are some instructions here: >> >> https://freeswitch.org/confluence/display/FREESWITCH/Debian+8+Jessie#Debian8Jessie-Scriptinstallfreeswitchdemowithverto_communicator >> >>> On Nov 21, 2016, at 11:50 AM, Rajil Saraswat wrote: >>> >>> Hello, >>> >>> I have been using self generated certificates (https://wiki.freeswitch.org/wiki/SIP_TLS) until now. >>> Is it possible to use Letsencrypt generated certificates for TLS? >>> >>> 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 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> 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/20161121/38bf27a2/attachment-0001.html From rajil.s at gmail.com Tue Nov 22 07:06:41 2016 From: rajil.s at gmail.com (Rajil Saraswat) Date: Mon, 21 Nov 2016 22:06:41 -0600 Subject: [Freeswitch-users] Letsencrypt and TLS In-Reply-To: <94FB5E2B-E2EB-42D5-8B1F-10D74ECC4317@jerris.com> References: <94FB5E2B-E2EB-42D5-8B1F-10D74ECC4317@jerris.com> Message-ID: On 11/21/2016 11:09 AM, Michael Jerris wrote: > there are some instructions here: > > https://freeswitch.org/confluence/display/FREESWITCH/Debian+8+Jessie#Debian8Jessie-Scriptinstallfreeswitchdemowithverto_communicator > >From these instructions, the certificate is wss.pem | cat| |/etc/letsencrypt/live/||$DOMAIN||/fullchain||.pem ||/etc/letsencrypt/live/||$DOMAIN||/privkey||.pem > ||/etc/freeswitch/tls/wss||.pem| The TLS page suggests i need CA/ folder with certificates and conf/ssl/agent.pem. How do i generate these from Letsencrypt? | | From trever at middleearth.sapphiresunday.org Tue Nov 22 07:29:19 2016 From: trever at middleearth.sapphiresunday.org (Trever L. Adams) Date: Mon, 21 Nov 2016 21:29:19 -0700 Subject: [Freeswitch-users] Letsencrypt and TLS In-Reply-To: References: <94FB5E2B-E2EB-42D5-8B1F-10D74ECC4317@jerris.com> Message-ID: <3dd01ac4-36f0-0ca6-1e2f-310f8713a875@middleearth.sapphiresunday.org> On 11/21/2016 01:10 PM, Joel Serrano wrote: > You can also try startssl.com > > They have longer expirations than letsencrypt. Several web browsers and other organizations are stopping their trust in the CA database for these certificates. This is based on them being acquired by another entity and the very real reasons to not trust that entity. Trever -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161121/a8507c0b/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 872 bytes Desc: OpenPGP digital signature Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161121/a8507c0b/attachment.bin From dominique.jeannerod at interact-iv.com Tue Nov 22 13:23:13 2016 From: dominique.jeannerod at interact-iv.com (Dominique Jeannerod) Date: Tue, 22 Nov 2016 11:23:13 +0100 Subject: [Freeswitch-users] mod_fifo and early media Message-ID: Hello, i'm working on a project where I need to manage a call waiting queue, in early media mode, to have a free waiting time, and start billing only after the call is really answered. The general call flow is : 1- incoming call to a service number 2- early media answer (pre-answer) to manage voice messages 3- the call is put in a waiting queue, and the caller ears a message (MOH), still in early media 4- the queue is associated to a unique destination number, and then one an only call at a time is picked-up from the queue and bridged to the destination. 5- When the call is established (200 OK), the incoming call is also answered (200 OK), which starts the billing. Question : this looks like a basic call queue handling, and mod_fifo could be a perfect fit for that ... except that mod_fifo answers the call with a 200 OK Is it possible to manage early media with mod_fifo ? Does someone have a best practice, experience, or advice to share on this matter ? Thanks with anticipation D. Jeannerod -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161122/ed2f026b/attachment.html From v.zakhozhai at gmail.com Tue Nov 22 13:33:02 2016 From: v.zakhozhai at gmail.com (Vladyslav Zakhozhai) Date: Tue, 22 Nov 2016 12:33:02 +0200 Subject: [Freeswitch-users] FreeSWITCH Registrar TLS offload Message-ID: Hi, I'm trying to understand what is the best or suitable approach to the following use case. Let me simplify thing a little bit. Suppose we have one FreeSWITCH registrar behind SIP proxy (kamailio). I'd like to offload SSL/TLS encryption/decryption to SIP proxy: REGISTER: Request: UAC == SIP/TLS ==> Kamailio == UDP ==> FreeSWITCH:50 Reply: UAC <== SIP/TLS == Kamailio <== UDP == FreeSWITCH INVITE: UAC1 == SIP/TLS ==> Kamailio == UDP == > FreeSWITCH == UDP ==> Kamailio == SIP/TLS ==> UAC2 (FreeSWITCH uses kamailio as outbound proxy with fs_path tag appended in dialplan). The main problem is in Contact header which contains transport=tls and we can see it in FreeSWITCH console: User: user at domain.com Contact: "" Status: Registered(TLS)(unknown) EXP(2016-11-22 10:16:59) EXPSECS(108) IP: SIP_PROXY_IP Port: 5060 When FreeSWITCH sends INVITE to UAC2 (during call) it tries to establish TLS session to UAC2. It fails because there is no TLS-enabled sofia profiles in the config of FreeSWITCH. I have only one solution in my mind: rewrite transport tag in Contact header on SIP proxy (transport=udp to FreeSWITCH, and transport=tls to UAC). I'd like to know it this solution ok or there is more elegant solutions. I've tried appending tag transport=udp in FreeSWITCH's dialplan but no success. Thank you in advance. -- ? ?????????, ????????? ??????? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161122/99e73abf/attachment.html From 568691 at gmail.com Tue Nov 22 14:03:55 2016 From: 568691 at gmail.com (Alexandru Covalschi) Date: Tue, 22 Nov 2016 13:03:55 +0200 Subject: [Freeswitch-users] FreeSWITCH Registrar TLS offload In-Reply-To: References: Message-ID: Do you have set_contact_alias or add_contact_alias in Kamailio? Anyways you're doing something wrong as AFAIK Kamailio translates contact header to udp automatically. You should try to post on sr-users list. 2016-11-22 12:33 GMT+02:00 Vladyslav Zakhozhai : > Hi, > > I'm trying to understand what is the best or suitable approach to the > following use case. Let me simplify thing a little bit. > > Suppose we have one FreeSWITCH registrar behind SIP proxy (kamailio). I'd > like to offload SSL/TLS encryption/decryption to SIP proxy: > > REGISTER: > > Request: UAC == SIP/TLS ==> Kamailio == UDP ==> FreeSWITCH:50 > Reply: UAC <== SIP/TLS == Kamailio <== UDP == FreeSWITCH > > INVITE: > UAC1 == SIP/TLS ==> Kamailio == UDP == > FreeSWITCH == UDP ==> Kamailio == > SIP/TLS ==> UAC2 > > (FreeSWITCH uses kamailio as outbound proxy with fs_path tag appended in > dialplan). > > The main problem is in Contact header which contains transport=tls and we > can see it in FreeSWITCH console: > > User: user at domain.com > Contact: "" > Status: Registered(TLS)(unknown) EXP(2016-11-22 10:16:59) EXPSECS(108) > IP: SIP_PROXY_IP > Port: 5060 > > When FreeSWITCH sends INVITE to UAC2 (during call) it tries to establish > TLS session to UAC2. It fails because there is no TLS-enabled sofia > profiles in the config of FreeSWITCH. > > I have only one solution in my mind: rewrite transport tag in Contact > header on SIP proxy (transport=udp to FreeSWITCH, and transport=tls to UAC). > > I'd like to know it this solution ok or there is more elegant solutions. > > I've tried appending tag transport=udp in FreeSWITCH's dialplan but no > success. > > 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-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Alexandru Covalschi VoIP engineer and system administrator tel: +37367398493 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161122/80f6ac9f/attachment-0001.html From devang.nathwani31589 at gmail.com Tue Nov 22 16:09:34 2016 From: devang.nathwani31589 at gmail.com (devang nathwani) Date: Tue, 22 Nov 2016 18:39:34 +0530 Subject: [Freeswitch-users] freeswitch sending '481 Call Does Not Exist' in return of BYE In-Reply-To: References: Message-ID: Hello, Now the tags are not changing any where in the call, here is the sip trace for the call http://pastebin.com/rYqAeYyi Yet freeswitch is sending '481 Call Does Not Exist' Where could be issue? On Fri, Nov 18, 2016 at 6:17 PM, Mirko Brankovic wrote: > HI, > I'm not sure if .17 is Freeswitch and who is 12 and wh 13 but I see that > on first 480 no answer To:.... tag is changed: > > 1. 2016/11/18 11:30:42.698223 11.23.16.13:5060 -> 11.23.16.12:5060 > 2. SIP/2.0 480 NO_ANSWER > 3. Via: SIP/2.0/UDP 11.23.16.12:5060;branch= > z9hG4bKlj0b84100gnjom3o1320.1 > 4. From: ;tag= > 09002861124617 > 5. To: ;tag= > f949601897706458a5166612fe67c373-1439 > 6. Call-ID: 03cS323551118140bcGhEfCmJej at RBM2S1.MSS.MTN.CO.ZA > 7. CSeq: 986549121 INVITE > 8. Content-Length: 0 > > WHich I think you should avoid, changing tags. I think this might be the > problem. > > On Fri, Nov 18, 2016 at 1:07 PM, devang nathwani < > devang.nathwani31589 at gmail.com> wrote: > >> Hello, >> >> Here are the sip traces of two legs >> leg1 UAC -> proxy -> media >> leg2 media -> proxy -> provider >> leg1 >> http://pastebin.com/G0jnF75t >> leg2 >> http://pastebin.com/299dZyN4 >> >> from leg1, >> when freeswitch(media) is sending back 480 temporarily unavailable after >> 183 session progress, UAC is sending ack and bye, now in return freeswitch >> is sending '481 Call Does Not Exist' >> >> my question is why? why freeswitch is sending 481 in return of ack and >> bye? >> >> here, >> 11.23.16.12 is UAC >> 11.23.16.13 is Opensips(Proxy) >> 11.23.16.16 is Freeswitch(Media) >> >> Thanks, >> Devang Nathwani >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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, > Mirko > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161122/b8264be2/attachment.html From vijaya_penugonda at yahoo.com Tue Nov 22 07:36:44 2016 From: vijaya_penugonda at yahoo.com (vijaya alavalpati) Date: Tue, 22 Nov 2016 04:36:44 +0000 (UTC) Subject: [Freeswitch-users] aravindan@teledna.com References: <1229608476.1801452.1479789404194.ref@mail.yahoo.com> Message-ID: <1229608476.1801452.1479789404194@mail.yahoo.com> Hi , ? I have configured freeswitch server for making one-one call.I am making test call from IMSDroid(https://github.com/DoubangoTelecom/imsdroid)?android client to SIPML5 client( i.e https://www.doubango.org/sipml5/call.htm#). When both are in the same wifi call is working. But when my android phone ison 3g and SIPML5 client is on wifi, call is established but voice is notaudible. Can you please help us what is the issue and how to fix. My Freeswitchserver is natted to a public IP. ? Following is the invite going from my android client.1004 user is SIPLM5 client and 1003 user is Android client. ? ?? INVITE sip:1004 at freeswitch.nowconfer.comSIP/2.0?? Via: SIP/2.0/TCP10.44.15.65:50583;branch=z9hG4bK-600328989;rport?? From: ;tag=1909344009?? To: ?? Contact: ;+g.oma.sip-im;language="en,fr";+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"?? Call-ID: bfaff8ce-8a80-cbdc-c106-7713da878fcb?? CSeq: 571345640 INVITE?? Content-Type: application/sdp?? Content-Length: 905?? Max-Forwards: 70?? Accept-Contact:*;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"?? Allow: INVITE, ACK, CANCEL, BYE, MESSAGE,OPTIONS, NOTIFY, PRACK, UPDATE, REFER?? Privacy: none?? User-Agent: IM-client/OMA1.0android-ngn-stack/v2.586.1328 (doubango r1328 - MotoG3)?? P-Preferred-Identity: ?? Supported: 100rel ? ?? v=0?? o=doubango 1983 678902 IN IP4 10.44.15.65?? s=-?? c=IN IP4 10.44.15.65?? t=0 0?? a=tcap:1 RTP/AVPF?? m=audio 45812 RTP/AVP 98 97 8 0 113 112 101 1119?? a=ptime:20?? a=minptime:1?? a=maxptime:255?? a=silenceSupp:off - - - -?? a=rtpmap:98 SPEEX/16000/1?? a=rtpmap:97 SPEEX/8000/1?? a=rtpmap:8 PCMA/8000/1?? a=rtpmap:0 PCMU/8000/1?? a=rtpmap:113 AMR/8000/1?? a=imageattr:113 octet-align=1?? a=fmtp:113 octet-align=1?? a=rtpmap:112 AMR/8000/1?? a=imageattr:112 octet-align=0?? a=fmtp:112 octet-align=0?? a=rtpmap:101 telephone-event/8000/1?? a=fmtp:101 0-16?? a=rtpmap:111 opus/48000/2?? a=fmtp:111 maxplaybackrate=16000;sprop-maxcapturerate=16000; stereo=0; sprop-stereo=0; useinbandfec=0; usedtx=0?? a=rtpmap:9 G722/8000/1?? a=pcfg:1 t=1?? a=sendrecv?? a=rtcp-mux?? a=ssrc:4194994927cname:a797995ff45befd0f562f852791dc543?? a=ssrc:4194994927mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2?? a=ssrc:4194994927 label:doubango at audio ? RegardsVijaya ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161122/a519725a/attachment-0001.html From loi.dangthanh at gmail.com Tue Nov 22 13:27:08 2016 From: loi.dangthanh at gmail.com (=?UTF-8?B?TOG7o2kgxJDhurduZw==?=) Date: Tue, 22 Nov 2016 17:27:08 +0700 Subject: [Freeswitch-users] absolute_codec_string not working Message-ID: Hi List, I got some trouble with using `absolute_codec_string` param. My call scenario is pretty simple: caller <--> FS <--> callee. My caller compose `m=audio 7078 RTP/AVP 8 0 101` in its INVITE, and I'm doing `` in the dialplan. But FS still use `m=audio 22952 RTP/AVP 8 0 101` in the INVITE to the callee. not sure what I'm missing, helps would be appreciated. Note that when I'm using `originate` application in fs_cli, things are good. `originate {absolute_codec_string=PCMU}sofia/gateway/caller/100 &bridge({absolute_codec_string=PCMA}sofia/gateway/callee/100`. I have FS with proper behavior in transcoding, caller has `m=audio 31184 RTP/AVP 0 101` received, and callee has `m=audio 21922 RTP/AVP 8 101` received. rgds, Loi Dang Thanh Phone : 84.1224.735.448 Email : loi.dangthanh at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161122/818b4a1a/attachment.html From mike at jerris.com Tue Nov 22 17:57:57 2016 From: mike at jerris.com (Michael Jerris) Date: Tue, 22 Nov 2016 09:57:57 -0500 Subject: [Freeswitch-users] absolute_codec_string not working In-Reply-To: References: Message-ID: <40268DB6-3518-4C8E-A022-D67FB585E2D9@jerris.com> using proxy_media is my best guess but can?t tell with this little info. > On Nov 22, 2016, at 5:27 AM, L?i ??ng wrote: > > > Hi List, I got some trouble with using `absolute_codec_string` param. > My call scenario is pretty simple: caller <--> FS <--> callee. > My caller compose `m=audio 7078 RTP/AVP 8 0 101` in its INVITE, and I'm doing `` in the dialplan. > But FS still use `m=audio 22952 RTP/AVP 8 0 101` in the INVITE to the callee. > not sure what I'm missing, helps would be appreciated. > > Note that when I'm using `originate` application in fs_cli, things are good. > `originate {absolute_codec_string=PCMU}sofia/gateway/caller/100 &bridge({absolute_codec_string=PCMA}sofia/gateway/callee/100`. > I have FS with proper behavior in transcoding, caller has `m=audio 31184 RTP/AVP 0 101` received, and callee has `m=audio 21922 RTP/AVP 8 101` received. > > rgds, > Loi Dang Thanh > Phone : 84.1224.735.448 > Email : loi.dangthanh 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/20161122/9574e4e8/attachment.html From jaradmorgan at gmail.com Tue Nov 22 18:13:48 2016 From: jaradmorgan at gmail.com (Jarad Morgan) Date: Tue, 22 Nov 2016 15:13:48 +0000 Subject: [Freeswitch-users] one way audio coming off hold In-Reply-To: References: <324F9FE2-65A2-4A3F-9285-D0FF52B67093@jerris.com> Message-ID: Ok thanks for the help Michael. Much appreciated! On Mon, Nov 21, 2016 at 5:00 PM Michael Jerris wrote: > If we were doing that, that was a bug on our part. Nothing told us to > take the call off hold. > > On Nov 21, 2016, at 3:33 PM, Jarad Morgan wrote: > > Sorry what i meant was on the older FS version the sendonly attribute was > not present in the 200 OK message back to Avaya. For example SIP flow on > older version. > > Avaya INVITE to FS no SDP > FS 100 trying no SDP to Avaya > FS 200 OK w SDP to Avaya <-- no sendonly media attribute specified here. > Avaya ACK w SDP to FS <-- no media attribute so default sendrecv > > Call is taken off hold.. > > Older version and 1.6 are running the same configs. > > I can bug the Avaya guys to fix their stuff but was just hoping to fix in > FS if it was easily possible. > > > > On Mon, Nov 21, 2016 at 3:06 PM Michael Jerris wrote: > > On Nov 21, 2016, at 2:59 PM, Jarad Morgan wrote: > > So long story short here is the issue were running into. Sorry for the > crappy messaging diagram. > > //when going on hold. > Avaya INVITE to FS no SDP > FS 100 trying no SDP to Avaya > FS 200 OK w SDP to Avaya > Avaya ACK w SDP(a=sendonly) to FS > > //were now on hold. > > //going off hold where we run into the one way audio. > Avaya INVITE to FS no SDP > FS 100 trying no SDP to Avaya > FS 200 OK w SDP (a=sendonly) to Avaya > Avaya ACK w SDP (a=recvonly) to FS > > > This is not a valid way to go off hold. The avaya would need to somehow > indicate that it is changing the stream to sendrecv, they don?t in what you > describe here. > > > We get a recvonly from Avaya so were in listen mode instead of sendrecv. > > This is a new issue since we upgraded from and older version to 1.6. The > previous version of FS we didnt see the a=sendonly in the 200 OK when going > off hold so i thought that may be configurable in the newer version to > leave those media attributes out of the SDP? > > > Not sure what you mean. We didn?t ?see? it? > > > Thanks!! > > On Mon, Nov 21, 2016 at 2:38 PM Michael Jerris wrote: > > if there is an sdp and it does not specify sendonly, recvonly, or > inactive, it means that stream is sendrecv. This is normal operation of > sdp, many things won?t actually include the sendrecv as size of sdp is a > bigger and bigger issue these days. What do you mean about a config > setting to leave it out? > > > On Nov 21, 2016, at 2:19 PM, Jarad Morgan wrote: > > > > Hello, we are running into some issues with one way audio coming off > hold with messaging between Avaya's. Running FS 1.6.10. > > > > Is there a FS config setting to leave out the media type attribute > (sendonly,sendrecv) in the 200 OK sent by FS when doing late media > negotiation between a Avaya device? > > > > 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/20161122/3ad2e886/attachment.html From mirkobrankovic at gmail.com Tue Nov 22 18:39:21 2016 From: mirkobrankovic at gmail.com (Mirko Brankovic) Date: Tue, 22 Nov 2016 16:39:21 +0100 Subject: [Freeswitch-users] freeswitch sending '481 Call Does Not Exist' in return of BYE In-Reply-To: References: Message-ID: I think that your UAC is not adding Contact header to BYE message. So try Adding : > Contact: from the initial invite. Also make sure that all VIA headers are there and unchanged Hope it helps now :) Mirko On Tue, Nov 22, 2016 at 2:09 PM, devang nathwani < devang.nathwani31589 at gmail.com> wrote: > Hello, > > Now the tags are not changing any where in the call, here is the sip trace > for the call > http://pastebin.com/rYqAeYyi > > Yet freeswitch is sending '481 Call Does Not Exist' > Where could be issue? > > On Fri, Nov 18, 2016 at 6:17 PM, Mirko Brankovic > wrote: > >> HI, >> I'm not sure if .17 is Freeswitch and who is 12 and wh 13 but I see that >> on first 480 no answer To:.... tag is changed: >> >> 1. 2016/11/18 11:30:42.698223 11.23.16.13:5060 -> 11.23.16.12:5060 >> 2. SIP/2.0 480 NO_ANSWER >> 3. Via: SIP/2.0/UDP 11.23.16.12:5060;branch=z9hG4b >> Klj0b84100gnjom3o1320.1 >> 4. From: ;tag=09002861 >> 124617 >> 5. To: ;tag=f94960189 >> 7706458a5166612fe67c373-1439 >> 6. Call-ID: 03cS323551118140bcGhEfCmJej at RBM2S1.MSS.MTN.CO.ZA >> 7. CSeq: 986549121 INVITE >> 8. Content-Length: 0 >> >> WHich I think you should avoid, changing tags. I think this might be the >> problem. >> >> On Fri, Nov 18, 2016 at 1:07 PM, devang nathwani < >> devang.nathwani31589 at gmail.com> wrote: >> >>> Hello, >>> >>> Here are the sip traces of two legs >>> leg1 UAC -> proxy -> media >>> leg2 media -> proxy -> provider >>> leg1 >>> http://pastebin.com/G0jnF75t >>> leg2 >>> http://pastebin.com/299dZyN4 >>> >>> from leg1, >>> when freeswitch(media) is sending back 480 temporarily unavailable after >>> 183 session progress, UAC is sending ack and bye, now in return freeswitch >>> is sending '481 Call Does Not Exist' >>> >>> my question is why? why freeswitch is sending 481 in return of ack and >>> bye? >>> >>> here, >>> 11.23.16.12 is UAC >>> 11.23.16.13 is Opensips(Proxy) >>> 11.23.16.16 is Freeswitch(Media) >>> >>> Thanks, >>> Devang Nathwani >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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, >> Mirko >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> 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, Mirko -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161122/235e8626/attachment-0001.html From anthony.minessale at gmail.com Tue Nov 22 19:39:25 2016 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 22 Nov 2016 10:39:25 -0600 Subject: [Freeswitch-users] absolute_codec_string not working In-Reply-To: <40268DB6-3518-4C8E-A022-D67FB585E2D9@jerris.com> References: <40268DB6-3518-4C8E-A022-D67FB585E2D9@jerris.com> Message-ID: escape the comma list with single quotes On Tue, Nov 22, 2016 at 8:57 AM, Michael Jerris wrote: > using proxy_media is my best guess but can?t tell with this little info. > > On Nov 22, 2016, at 5:27 AM, L?i ??ng wrote: > > > Hi List, I got some trouble with using `absolute_codec_string` param. > My call scenario is pretty simple: caller <--> FS <--> callee. > My caller compose `m=audio 7078 RTP/AVP 8 0 101` in its INVITE, and I'm > doing `` in the dialplan. > But FS still use `m=audio 22952 RTP/AVP 8 0 101` in the INVITE to the > callee. > not sure what I'm missing, helps would be appreciated. > > Note that when I'm using `originate` application in fs_cli, things are > good. > `originate {absolute_codec_string=PCMU}sofia/gateway/caller/100 > &bridge({absolute_codec_string=PCMA}sofia/gateway/callee/100`. > I have FS with proper behavior in transcoding, caller has `m=audio 31184 > RTP/AVP 0 101` received, and callee has `m=audio 21922 RTP/AVP 8 101` > received. > > rgds, > Loi Dang Thanh > Phone : 84.1224.735.448 > Email : loi.dangthanh 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 > -- 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 https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161122/a01b0f7a/attachment.html From lists at kavun.ch Tue Nov 22 20:49:16 2016 From: lists at kavun.ch (Emrah) Date: Tue, 22 Nov 2016 18:49:16 +0100 Subject: [Freeswitch-users] Freeswitch routing inbound calls over SIP instead of TLS/SRTP In-Reply-To: References: Message-ID: FreeSWITCH doesn't support sips, although I think attempts were made by bkw to get it supported. See if your client can get registered using SIP over TLS as opposed to using SIPS. Are you using SRV entries? > On Nov 21, 2016, at 5:39 PM, Tim Smith wrote: > > Hi Steve, > > Sorry for the delay ack'ing your mail ... yeah, guess I should maybe > look into filing a JIRA. > > On 20 November 2016 at 00:33, Steven Ayre wrote: >> Looks like a bug to me. Your first snippet shows the contact stored in the >> database uses the 'sips:' scheme, but sofia_contact is returning 'sip:' >> >> In the code it looks like sofia_contact fetches the contact using >> select_from_profile which invokes contact_callback. In contact_callback it's >> hardcoded to use sip: plus the result of sofia_glue_strip_proto. That looks >> to me like it can never return a sips URI even though it's stored in the >> database. >> >> I'd file a jira. >> >> Steve >> >> On 18 November 2016 at 10:08, Tim Smith wrote: >>> >>> Debian GNU/Linux 8 (jessie) >>> Linux my 3.16.0-4-amd64 #1 SMP Debian 3.16.36-1+deb8u2 (2016-10-19) >>> x86_64 GNU/Linux >>> FreeSWITCH Version 1.6.12-20-b91a0a6~64bit (-20-b91a0a6 64bit) >>> >>> I have a Vtech handset with TLS/SRTP enabled registered with >>> Freeswitch as below: >>> >>> >>> Call-ID: a0000a0a000aa000 >>> User: 2001 at my.example.com >>> Contact: "my" >>> Agent: Vtech Vesa VSP736A 2.0.3.2-0 >>> Status: Registered(TLS)(unknown) EXP(2016-11-18 10:56:57) >>> EXPSECS(3646) >>> Ping-Status: Reachable >>> Ping-Time: 0.00 >>> Host: my >>> IP: 198.51.100.81 >>> Port: 58348 >>> Auth-User: 2001 >>> Auth-Realm: my.example.com >>> MWI-Account: 2001 at my.example.com >>> >>> >>> sofia_contact is happy : >>> >>> freeswitch at my>sofia_contact internal/2001 >>> sofia/internal/sip:2001 at 198.51.100.81:58348 >>> >>> I have an inbound dial plan configured as follows: >>> >>> >>> >>> >>> >>> >> data="${sofia_contact(internal/2001)}"/> >>> >>> >>> >>> >>> The problem is Freeswitch is sending invites over SIP/RTP and not >>> TLS/SRTP and so the calls never get through : >>> >>> INVITE sip:2001 at 198.51.100.81:58348 SIP/2.0 >>> Via: SIP/2.0/UDP 203.0.113.4;rport;branch=z9hG4bKvHjgXXpFF77XK >>> From: "Anonymous" ;tag=rKmXQjZN8SFXp >>> To: >>> m=audio 32190 RTP/AVP 8 98 9 101 >>> a=rtpmap:8 PCMA/8000 >>> a=rtpmap:98 G726-32/8000 >>> a=rtpmap:9 G722/8000 >>> a=rtpmap:101 telephone-event/8000 >>> a=fmtp:101 0-16 >>> a=ptime:20 >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> 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 s.safarov at gmail.com Tue Nov 22 20:52:23 2016 From: s.safarov at gmail.com (Sergey Safarov) Date: Tue, 22 Nov 2016 17:52:23 +0000 Subject: [Freeswitch-users] mod_fifo and early media In-Reply-To: References: Message-ID: You can try patch mod_fifo source code and comment "switch_channel_answer(channel);" strings. Then you can manage signal is passed to a-leg from dialplan. ??, 22 ????. 2016 ?. ? 13:24, Dominique Jeannerod < dominique.jeannerod at interact-iv.com>: > Hello, > > i'm working on a project where I need to manage a call waiting queue, in > early media mode, to have a free waiting time, and start billing only after > the call is really answered. > > The general call flow is : > 1- incoming call to a service number > 2- early media answer (pre-answer) to manage voice messages > 3- the call is put in a waiting queue, and the caller ears a message > (MOH), still in early media > 4- the queue is associated to a unique destination number, and then one an > only call at a time is picked-up from the queue and bridged to the > destination. > 5- When the call is established (200 OK), the incoming call is also > answered (200 OK), which starts the billing. > > > Question : this looks like a basic call queue handling, and mod_fifo could > be a perfect fit for that ... except that mod_fifo answers the call with a > 200 OK > > Is it possible to manage early media with mod_fifo ? > > Does someone have a best practice, experience, or advice to share on this > matter ? > > > Thanks with anticipation > > D. Jeannerod > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161122/0fe39da9/attachment.html From jose.lopes at itcenter.com.pt Tue Nov 22 21:40:26 2016 From: jose.lopes at itcenter.com.pt (=?UTF-8?Q?Jos=C3=A9_Lopes?=) Date: Tue, 22 Nov 2016 18:40:26 +0000 Subject: [Freeswitch-users] When activate MoH two times from different legs, the call is silent or terminated Message-ID: Hello, I have a FreeSwitch with vanilla configuration connected to an SIP Provider. I configured the proxy-hold=true configuration on Sofia Internal profile so when i activate the Hold on internal sip regs, it will be forwarded to SIP Provider. The SIP Provider provides MoH when requested. I have a problem that results on the call on silent or terminated by SIP Provider when I do the next actions: 1- I receive a call from the SIP Provider 2- the caller activate MoH and deactivate MoH 3- the callee activate the MoH 4- Now the call is silent (with other SIP Provider the call is dropped by SIP Provider) I notice that on step 3, the FreeSwitch send a Re-INVITE without SDP, so I think this causes the call to be silent, or sometimes the SIP Provider drop the call. I replicate this issue creating a sipp scenario and verified this issue. SIPP scenario to replicate this issue https://pastebin.freeswitch.org/view/afbae914 Freeswitch logs: https://pastebin.freeswitch.org/view/7f36d5ba Command to use sipp scenario sipp -i 172.21.0.1 172.21.0.11:5080 -s 1001 -sf scenario.xml -m 1 -l 1 -r 1 -rp 1000 #172.21.0.1 -> local ip #172.21.0.11 -> Freeswitch IP Best regards, Jose Lopes -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161122/88145182/attachment-0001.html From colin.morelli at gmail.com Wed Nov 23 00:05:13 2016 From: colin.morelli at gmail.com (Colin Morelli) Date: Tue, 22 Nov 2016 21:05:13 +0000 Subject: [Freeswitch-users] uuid_getvar / uuid_setvar Safety Message-ID: Hey all, This should hopefully be a very simple question. If I have a variable in one channel that I want to copy to another (these are unrelated, un-bridged channels) is it safe to just do, for example uuid_setvar some_var ${uuid_getvar other-call-id some_var} (from the dialplan)? Is there a better alternative? Without being an expert in how the variable expansion works, I want to make sure this would work for all possible values of the var, and I won't be caught off-guard with buffer size issues, separator issues, etc. Thanks in advance, Colin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161122/b49e7f5a/attachment.html From steveayre at gmail.com Wed Nov 23 02:14:43 2016 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 22 Nov 2016 23:14:43 +0000 Subject: [Freeswitch-users] Bridge an incoming call to an external SIP address In-Reply-To: References: Message-ID: Are you sure 'internal' is the sofia profile you want to be using to reach an external address? On 21 November 2016 at 13:35, Cato Gonzalez wrote: > Hi everyone, > > I am trying to get FS to route inbound calls to an outside sip provider > but I am getting the call hung up on the FS side just upon the first or > second ring on the B-leg. The diaplan action being executed is: > > > > I have a softphone registered to this provider sip.linphone.org and it > works well when calling XXX from some other IP phone. FS reports this in > the logs: > > [DEBUG] switch_ivr_originate.c:1274 Raw Codec Activation Success > L16 at 48000hz 1 channel 20ms > [DEBUG] switch_core_codec.c:221 sofia/internal/anonymous@ > webrtc.sip.mydomain.co Push codec L16:100 > [DEBUG] switch_ivr_originate.c:1343 Play Ringback Tone > [%(2000,4000,440,480)] > [DEBUG] sofia.c:6760 Channel sofia/internal/anonymous@ > webrtc.sip.mydomain.co entering state [terminated][500] > [NOTICE] sofia.c:7779 Hangup sofia/internal/anonymous@ > webrtc.sip.mydomain.co [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] > [DEBUG] switch_core_codec.c:246 sofia/internal/anonymous@ > webrtc.sip.mydomain.co Restore previous codec opus:116. > [NOTICE] switch_ivr_originate.c:3523 Hangup sofia/external/XXX at sip. > linphone.org [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL] > > Result is the same if I try to call any other sip destination. I > appreciate any input anyone may give: have been working on this for a week > already. > > Thanks, > > Cato > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161122/66043b5c/attachment.html From anthony.minessale at gmail.com Wed Nov 23 02:16:48 2016 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 22 Nov 2016 17:16:48 -0600 Subject: [Freeswitch-users] uuid_getvar / uuid_setvar Safety In-Reply-To: References: Message-ID: yes On Tue, Nov 22, 2016 at 3:05 PM, Colin Morelli wrote: > Hey all, > > This should hopefully be a very simple question. If I have a variable in > one channel that I want to copy to another (these are unrelated, un-bridged > channels) is it safe to just do, for example uuid_setvar some_var > ${uuid_getvar other-call-id some_var} (from the dialplan)? Is there a > better alternative? > > Without being an expert in how the variable expansion works, I want to > make sure this would work for all possible values of the var, and I won't > be caught off-guard with buffer size issues, separator issues, etc. > > Thanks in advance, > Colin > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161122/4b1aa573/attachment.html From loi.dangthanh at gmail.com Wed Nov 23 06:07:24 2016 From: loi.dangthanh at gmail.com (=?UTF-8?B?TOG7o2kgxJDhurduZw==?=) Date: Wed, 23 Nov 2016 10:07:24 +0700 Subject: [Freeswitch-users] absolute_codec_string not working In-Reply-To: <40268DB6-3518-4C8E-A022-D67FB585E2D9@jerris.com> References: <40268DB6-3518-4C8E-A022-D67FB585E2D9@jerris.com> Message-ID: Hi @Michael, you were right, I'm intentionally using media_proxy for FS, since I want to reduce CPU usage on FS machine. In this case, I just want to limit the codecs used for each endpoint, and codec negotiation will be handled by them. e.g: caller use PCMA, PCMU, GSM by its own in INVITE, I want to limit the callee to only use PCMA,GSM. Look like `absolute_codec_string` is not what I'm looking for right? Any way out? Loi Dang Thanh Phone : 01224.735.448 Email : loi.dangthanh at gmail.com On Tue, Nov 22, 2016 at 9:57 PM, Michael Jerris wrote: > using proxy_media is my best guess but can?t tell with this little info. > > On Nov 22, 2016, at 5:27 AM, L?i ??ng wrote: > > > Hi List, I got some trouble with using `absolute_codec_string` param. > My call scenario is pretty simple: caller <--> FS <--> callee. > My caller compose `m=audio 7078 RTP/AVP 8 0 101` in its INVITE, and I'm > doing `` in the dialplan. > But FS still use `m=audio 22952 RTP/AVP 8 0 101` in the INVITE to the > callee. > not sure what I'm missing, helps would be appreciated. > > Note that when I'm using `originate` application in fs_cli, things are > good. > `originate {absolute_codec_string=PCMU}sofia/gateway/caller/100 > &bridge({absolute_codec_string=PCMA}sofia/gateway/callee/100`. > I have FS with proper behavior in transcoding, caller has `m=audio 31184 > RTP/AVP 0 101` received, and callee has `m=audio 21922 RTP/AVP 8 101` > received. > > rgds, > Loi Dang Thanh > Phone : 84.1224.735.448 > Email : loi.dangthanh 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/20161123/5f43d404/attachment-0001.html From krice at freeswitch.org Wed Nov 23 06:28:08 2016 From: krice at freeswitch.org (Ken Rice) Date: Tue, 22 Nov 2016 21:28:08 -0600 Subject: [Freeswitch-users] absolute_codec_string not working In-Reply-To: References: <40268DB6-3518-4C8E-A022-D67FB585E2D9@jerris.com> Message-ID: <09ed01d24539$9e24ccf0$da6e66d0$@freeswitch.org> If you are limiting the calls to specific codecs and avoiding transcoding, proxy media doesn?t really reduce the overhead anymore? that changed a few years ago but the notion its better still hangs on today From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of L?i ??ng Sent: Tuesday, November 22, 2016 9:07 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] absolute_codec_string not working Hi @Michael, you were right, I'm intentionally using media_proxy for FS, since I want to reduce CPU usage on FS machine. In this case, I just want to limit the codecs used for each endpoint, and codec negotiation will be handled by them. e.g: caller use PCMA, PCMU, GSM by its own in INVITE, I want to limit the callee to only use PCMA,GSM. Look like `absolute_codec_string` is not what I'm looking for right? Any way out? Loi Dang Thanh Phone : 01224.735.448 Email : loi.dangthanh at gmail.com On Tue, Nov 22, 2016 at 9:57 PM, Michael Jerris > wrote: using proxy_media is my best guess but can?t tell with this little info. On Nov 22, 2016, at 5:27 AM, L?i ??ng > wrote: Hi List, I got some trouble with using `absolute_codec_string` param. My call scenario is pretty simple: caller <--> FS <--> callee. My caller compose `m=audio 7078 RTP/AVP 8 0 101` in its INVITE, and I'm doing `` in the dialplan. But FS still use `m=audio 22952 RTP/AVP 8 0 101` in the INVITE to the callee. not sure what I'm missing, helps would be appreciated. Note that when I'm using `originate` application in fs_cli, things are good. `originate {absolute_codec_string=PCMU}sofia/gateway/caller/100 &bridge({absolute_codec_string=PCMA}sofia/gateway/callee/100`. I have FS with proper behavior in transcoding, caller has `m=audio 31184 RTP/AVP 0 101` received, and callee has `m=audio 21922 RTP/AVP 8 101` received. rgds, Loi Dang Thanh Phone : 84.1224.735.448 Email : loi.dangthanh 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/20161122/81ed9cf5/attachment.html From devang.nathwani31589 at gmail.com Wed Nov 23 11:20:34 2016 From: devang.nathwani31589 at gmail.com (devang nathwani) Date: Wed, 23 Nov 2016 13:50:34 +0530 Subject: [Freeswitch-users] freeswitch sending '481 Call Does Not Exist' in return of BYE In-Reply-To: References: Message-ID: Yes this is helping but for the 'Contact' part in BYE header below is the 'BYE' sip block from the working call, and it also not containing 'Contact:' ========= 2016/11/23 08:30:05.595033 UAC_IP:5060 -> PROXY_IP:5060 BYE sip:2648784 at 41.218.72.6:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP UAC_IP:5060;branch=z9hG4bK76fdsg00e0oh925na7l1cd0000g10.1 Call-ID: tlgfv877nn82btngv2vlbtagn9a799cn at SoftX3000 From: ;tag=vf77f89n-CC-23 To: ;tag=yvXByKme39FKF CSeq: 3 BYE Reason: Q.850;cause=16;text="normal call clearing" Max-Forwards: 69 Content-Length: 0 Route: ========= To compare above block with not working one, below is the block which is getting '481 Call Does Not Exist' from media ========= 2016/11/21 14:21:36.931816 UAC_IP:5060 -> PROXY_IP:5060 BYE sip:2648362 at MEDIA_IP:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP UAC_IP:5060;branch=z9hG4bKnabcr510eo49dgoabt50.1 From: ;tag=07001094154893 To: ;tag=aDa8SH7vpgg5F Max-Forwards: 69 Call-ID: 00bT41746142115DbcGhEfCoOni at JHM2S3.MSS.MTN.CO.ZA CSeq: 448960898 BYE Reason: Q.850;cause=16 Content-Length: 0 Route: ========= However, i find values of 'CSeq' and 'Reason' fields bit suspicious. On Tue, Nov 22, 2016 at 9:09 PM, Mirko Brankovic wrote: > I think that your UAC is not adding Contact header to BYE message. > So try Adding : > >> Contact: > > from the initial invite. > > Also make sure that all VIA headers are there and unchanged > > Hope it helps now :) > > Mirko > > On Tue, Nov 22, 2016 at 2:09 PM, devang nathwani < > devang.nathwani31589 at gmail.com> wrote: > >> Hello, >> >> Now the tags are not changing any where in the call, here is the sip >> trace for the call >> http://pastebin.com/rYqAeYyi >> >> Yet freeswitch is sending '481 Call Does Not Exist' >> Where could be issue? >> >> On Fri, Nov 18, 2016 at 6:17 PM, Mirko Brankovic < >> mirkobrankovic at gmail.com> wrote: >> >>> HI, >>> I'm not sure if .17 is Freeswitch and who is 12 and wh 13 but I see that >>> on first 480 no answer To:.... tag is changed: >>> >>> 1. 2016/11/18 11:30:42.698223 11.23.16.13:5060 -> 11.23.16.12:5060 >>> 2. SIP/2.0 480 NO_ANSWER >>> 3. Via: SIP/2.0/UDP 11.23.16.12:5060;branch=z9hG4b >>> Klj0b84100gnjom3o1320.1 >>> 4. From: ;tag=09002861 >>> 124617 >>> 5. To: ;tag=f94960189 >>> 7706458a5166612fe67c373-1439 >>> 6. Call-ID: 03cS323551118140bcGhEfCmJej at RBM2S1.MSS.MTN.CO.ZA >>> 7. CSeq: 986549121 INVITE >>> 8. Content-Length: 0 >>> >>> WHich I think you should avoid, changing tags. I think this might be the >>> problem. >>> >>> On Fri, Nov 18, 2016 at 1:07 PM, devang nathwani < >>> devang.nathwani31589 at gmail.com> wrote: >>> >>>> Hello, >>>> >>>> Here are the sip traces of two legs >>>> leg1 UAC -> proxy -> media >>>> leg2 media -> proxy -> provider >>>> leg1 >>>> http://pastebin.com/G0jnF75t >>>> leg2 >>>> http://pastebin.com/299dZyN4 >>>> >>>> from leg1, >>>> when freeswitch(media) is sending back 480 temporarily unavailable >>>> after 183 session progress, UAC is sending ack and bye, now in return >>>> freeswitch is sending '481 Call Does Not Exist' >>>> >>>> my question is why? why freeswitch is sending 481 in return of ack and >>>> bye? >>>> >>>> here, >>>> 11.23.16.12 is UAC >>>> 11.23.16.13 is Opensips(Proxy) >>>> 11.23.16.16 is Freeswitch(Media) >>>> >>>> Thanks, >>>> Devang Nathwani >>>> >>>> ____________________________________________________________ >>>> _____________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>> switch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> >>> -- >>> Regards, >>> Mirko >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> 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, > Mirko > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161123/539480f4/attachment-0001.html From randomdev4 at gmail.com Wed Nov 23 11:20:58 2016 From: randomdev4 at gmail.com (Tim Smith) Date: Wed, 23 Nov 2016 08:20:58 +0000 Subject: [Freeswitch-users] Freeswitch routing inbound calls over SIP instead of TLS/SRTP In-Reply-To: References: Message-ID: Unfortunatley I don't think the vtech handsets offer many options in that respect. All the ports in the vtech config are set to 5060. Transport options are only UDP,TCP,TLS And SRTP is a boolean option. On 22 November 2016 at 17:49, Emrah wrote: > FreeSWITCH doesn't support sips, although I think attempts were made by bkw to get it supported. See if your client can get registered using SIP over TLS as opposed to using SIPS. > Are you using SRV entries? > >> On Nov 21, 2016, at 5:39 PM, Tim Smith wrote: >> >> Hi Steve, >> >> Sorry for the delay ack'ing your mail ... yeah, guess I should maybe >> look into filing a JIRA. >> >> On 20 November 2016 at 00:33, Steven Ayre wrote: >>> Looks like a bug to me. Your first snippet shows the contact stored in the >>> database uses the 'sips:' scheme, but sofia_contact is returning 'sip:' >>> >>> In the code it looks like sofia_contact fetches the contact using >>> select_from_profile which invokes contact_callback. In contact_callback it's >>> hardcoded to use sip: plus the result of sofia_glue_strip_proto. That looks >>> to me like it can never return a sips URI even though it's stored in the >>> database. >>> >>> I'd file a jira. >>> >>> Steve >>> >>> On 18 November 2016 at 10:08, Tim Smith wrote: >>>> >>>> Debian GNU/Linux 8 (jessie) >>>> Linux my 3.16.0-4-amd64 #1 SMP Debian 3.16.36-1+deb8u2 (2016-10-19) >>>> x86_64 GNU/Linux >>>> FreeSWITCH Version 1.6.12-20-b91a0a6~64bit (-20-b91a0a6 64bit) >>>> >>>> I have a Vtech handset with TLS/SRTP enabled registered with >>>> Freeswitch as below: >>>> >>>> >>>> Call-ID: a0000a0a000aa000 >>>> User: 2001 at my.example.com >>>> Contact: "my" >>>> Agent: Vtech Vesa VSP736A 2.0.3.2-0 >>>> Status: Registered(TLS)(unknown) EXP(2016-11-18 10:56:57) >>>> EXPSECS(3646) >>>> Ping-Status: Reachable >>>> Ping-Time: 0.00 >>>> Host: my >>>> IP: 198.51.100.81 >>>> Port: 58348 >>>> Auth-User: 2001 >>>> Auth-Realm: my.example.com >>>> MWI-Account: 2001 at my.example.com >>>> >>>> >>>> sofia_contact is happy : >>>> >>>> freeswitch at my>sofia_contact internal/2001 >>>> sofia/internal/sip:2001 at 198.51.100.81:58348 >>>> >>>> I have an inbound dial plan configured as follows: >>>> >>>> >>>> >>>> >>>> >>>> >>> data="${sofia_contact(internal/2001)}"/> >>>> >>>> >>>> >>>> >>>> The problem is Freeswitch is sending invites over SIP/RTP and not >>>> TLS/SRTP and so the calls never get through : >>>> >>>> INVITE sip:2001 at 198.51.100.81:58348 SIP/2.0 >>>> Via: SIP/2.0/UDP 203.0.113.4;rport;branch=z9hG4bKvHjgXXpFF77XK >>>> From: "Anonymous" ;tag=rKmXQjZN8SFXp >>>> To: >>>> m=audio 32190 RTP/AVP 8 98 9 101 >>>> a=rtpmap:8 PCMA/8000 >>>> a=rtpmap:98 G726-32/8000 >>>> a=rtpmap:9 G722/8000 >>>> a=rtpmap:101 telephone-event/8000 >>>> a=fmtp:101 0-16 >>>> a=ptime:20 >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> 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 mirkobrankovic at gmail.com Wed Nov 23 12:34:08 2016 From: mirkobrankovic at gmail.com (Mirko Brankovic) Date: Wed, 23 Nov 2016 10:34:08 +0100 Subject: [Freeswitch-users] freeswitch sending '481 Call Does Not Exist' in return of BYE In-Reply-To: References: Message-ID: reason shouldn't make any difference. CSec should 'increase' or just be a new one from the invite. I see that VIA branch have changed from invite and further messages and the bye one. I think that branch should stay the same as from the Invite. mirko On Wed, Nov 23, 2016 at 9:20 AM, devang nathwani < devang.nathwani31589 at gmail.com> wrote: > Yes this is helping > but for the 'Contact' part in BYE header > > below is the 'BYE' sip block from the working call, and it also not > containing 'Contact:' > ========= > 2016/11/23 08:30:05.595033 UAC_IP:5060 -> PROXY_IP:5060 > BYE sip:2648784 at 41.218.72.6:5060;transport=udp SIP/2.0 > Via: SIP/2.0/UDP UAC_IP:5060;branch=z9hG4bK76fdsg00e0oh925na7l1cd0000g10.1 > Call-ID: tlgfv877nn82btngv2vlbtagn9a799cn at SoftX3000 > From: ;tag=vf77f89n-CC-23 > To: ;tag=yvXByKme39FKF > CSeq: 3 BYE > Reason: Q.850;cause=16;text="normal call clearing" > Max-Forwards: 69 > Content-Length: 0 > Route: > ========= > > To compare above block with not working one, below is the block which is > getting '481 Call Does Not Exist' from media > ========= > 2016/11/21 14:21:36.931816 UAC_IP:5060 -> PROXY_IP:5060 > BYE sip:2648362 at MEDIA_IP:5060;transport=udp SIP/2.0 > Via: SIP/2.0/UDP UAC_IP:5060;branch=z9hG4bKnabcr510eo49dgoabt50.1 > From: ;tag=07001094154893 > To: ;tag=aDa8SH7vpgg5F > Max-Forwards: 69 > Call-ID: 00bT41746142115DbcGhEfCoOni at JHM2S3.MSS.MTN.CO.ZA > CSeq: 448960898 BYE > Reason: Q.850;cause=16 > Content-Length: 0 > Route: > ========= > > However, i find values of 'CSeq' and 'Reason' fields bit suspicious. > > On Tue, Nov 22, 2016 at 9:09 PM, Mirko Brankovic > wrote: > >> I think that your UAC is not adding Contact header to BYE message. >> So try Adding : >> >>> Contact: >> >> from the initial invite. >> >> Also make sure that all VIA headers are there and unchanged >> >> Hope it helps now :) >> >> Mirko >> >> On Tue, Nov 22, 2016 at 2:09 PM, devang nathwani < >> devang.nathwani31589 at gmail.com> wrote: >> >>> Hello, >>> >>> Now the tags are not changing any where in the call, here is the sip >>> trace for the call >>> http://pastebin.com/rYqAeYyi >>> >>> Yet freeswitch is sending '481 Call Does Not Exist' >>> Where could be issue? >>> >>> On Fri, Nov 18, 2016 at 6:17 PM, Mirko Brankovic < >>> mirkobrankovic at gmail.com> wrote: >>> >>>> HI, >>>> I'm not sure if .17 is Freeswitch and who is 12 and wh 13 but I see >>>> that on first 480 no answer To:.... tag is changed: >>>> >>>> 1. 2016/11/18 11:30:42.698223 11.23.16.13:5060 -> 11.23.16.12:5060 >>>> 2. SIP/2.0 480 NO_ANSWER >>>> 3. Via: SIP/2.0/UDP 11.23.16.12:5060;branch=z9hG4b >>>> Klj0b84100gnjom3o1320.1 >>>> 4. From: ;tag= >>>> 09002861124617 >>>> 5. To: ;tag=f94960189 >>>> 7706458a5166612fe67c373-1439 >>>> 6. Call-ID: 03cS323551118140bcGhEfCmJej at RBM2S1.MSS.MTN.CO.ZA >>>> 7. CSeq: 986549121 INVITE >>>> 8. Content-Length: 0 >>>> >>>> WHich I think you should avoid, changing tags. I think this might be >>>> the problem. >>>> >>>> On Fri, Nov 18, 2016 at 1:07 PM, devang nathwani < >>>> devang.nathwani31589 at gmail.com> wrote: >>>> >>>>> Hello, >>>>> >>>>> Here are the sip traces of two legs >>>>> leg1 UAC -> proxy -> media >>>>> leg2 media -> proxy -> provider >>>>> leg1 >>>>> http://pastebin.com/G0jnF75t >>>>> leg2 >>>>> http://pastebin.com/299dZyN4 >>>>> >>>>> from leg1, >>>>> when freeswitch(media) is sending back 480 temporarily unavailable >>>>> after 183 session progress, UAC is sending ack and bye, now in return >>>>> freeswitch is sending '481 Call Does Not Exist' >>>>> >>>>> my question is why? why freeswitch is sending 481 in return of ack and >>>>> bye? >>>>> >>>>> here, >>>>> 11.23.16.12 is UAC >>>>> 11.23.16.13 is Opensips(Proxy) >>>>> 11.23.16.16 is Freeswitch(Media) >>>>> >>>>> Thanks, >>>>> Devang Nathwani >>>>> >>>>> ____________________________________________________________ >>>>> _____________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>>> switch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> >>>> -- >>>> Regards, >>>> Mirko >>>> >>>> ____________________________________________________________ >>>> _____________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>> switch-users >>>> 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, >> Mirko >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> 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, Mirko -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161123/e7aadd3a/attachment-0001.html From v.zakhozhai at gmail.com Wed Nov 23 12:36:51 2016 From: v.zakhozhai at gmail.com (Vladyslav Zakhozhai) Date: Wed, 23 Nov 2016 11:36:51 +0200 Subject: [Freeswitch-users] FreeSWITCH Registrar TLS offload In-Reply-To: References: Message-ID: Alexandru, thank you for the answer. I think you've given me right direction to investigate. As you've mentioned this is really kamailio issue/question. So I'm moving to sr-users list. 2016-11-22 13:03 GMT+02:00 Alexandru Covalschi <568691 at gmail.com>: > Do you have set_contact_alias or add_contact_alias in Kamailio? Anyways > you're doing something wrong as AFAIK Kamailio translates contact header to > udp automatically. You should try to post on sr-users list. > > 2016-11-22 12:33 GMT+02:00 Vladyslav Zakhozhai : > >> Hi, >> >> I'm trying to understand what is the best or suitable approach to the >> following use case. Let me simplify thing a little bit. >> >> Suppose we have one FreeSWITCH registrar behind SIP proxy (kamailio). I'd >> like to offload SSL/TLS encryption/decryption to SIP proxy: >> >> REGISTER: >> >> Request: UAC == SIP/TLS ==> Kamailio == UDP ==> FreeSWITCH:50 >> Reply: UAC <== SIP/TLS == Kamailio <== UDP == FreeSWITCH >> >> INVITE: >> UAC1 == SIP/TLS ==> Kamailio == UDP == > FreeSWITCH == UDP ==> Kamailio >> == SIP/TLS ==> UAC2 >> >> (FreeSWITCH uses kamailio as outbound proxy with fs_path tag appended in >> dialplan). >> >> The main problem is in Contact header which contains transport=tls and we >> can see it in FreeSWITCH console: >> >> User: user at domain.com >> Contact: "" >> Status: Registered(TLS)(unknown) EXP(2016-11-22 10:16:59) >> EXPSECS(108) >> IP: SIP_PROXY_IP >> Port: 5060 >> >> When FreeSWITCH sends INVITE to UAC2 (during call) it tries to establish >> TLS session to UAC2. It fails because there is no TLS-enabled sofia >> profiles in the config of FreeSWITCH. >> >> I have only one solution in my mind: rewrite transport tag in Contact >> header on SIP proxy (transport=udp to FreeSWITCH, and transport=tls to UAC). >> >> I'd like to know it this solution ok or there is more elegant solutions. >> >> I've tried appending tag transport=udp in FreeSWITCH's dialplan but no >> success. >> >> 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-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Alexandru Covalschi > VoIP engineer and system administrator > tel: +37367398493 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- ? ?????????, ????????? ??????? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161123/5fc69ac9/attachment.html From loi.dangthanh at gmail.com Wed Nov 23 13:37:14 2016 From: loi.dangthanh at gmail.com (=?UTF-8?B?TOG7o2kgxJDhurduZw==?=) Date: Wed, 23 Nov 2016 17:37:14 +0700 Subject: [Freeswitch-users] absolute_codec_string not working In-Reply-To: <09ed01d24539$9e24ccf0$da6e66d0$@freeswitch.org> References: <40268DB6-3518-4C8E-A022-D67FB585E2D9@jerris.com> <09ed01d24539$9e24ccf0$da6e66d0$@freeswitch.org> Message-ID: In FS document of media proxy mode: > FreeSWITCH has no control or even understanding of other SDP parameters. Look like I have to find another way, like writing a custom module listening on specific event. Any suggest? Thanks to all of you. rgds, Loi Dang Thanh Phone : 841224.735.448 Email : loi.dangthanh at gmail.com On Wed, Nov 23, 2016 at 10:28 AM, Ken Rice wrote: > If you are limiting the calls to specific codecs and avoiding transcoding, > proxy media doesn?t really reduce the overhead anymore? that changed a few > years ago but the notion its better still hangs on today > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *L?i ??ng > *Sent:* Tuesday, November 22, 2016 9:07 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] absolute_codec_string not working > > > > Hi @Michael, you were right, I'm intentionally using media_proxy for FS, > since I want to reduce CPU usage on FS machine. > > In this case, I just want to limit the codecs used for each endpoint, and > codec negotiation will be handled by them. > > e.g: caller use PCMA, PCMU, GSM by its own in INVITE, I want to limit the > callee to only use PCMA,GSM. > > Look like `absolute_codec_string` is not what I'm looking for right? Any > way out? > > > Loi Dang Thanh > > Phone : 01224.735.448 > > Email : loi.dangthanh at gmail.com > > > > On Tue, Nov 22, 2016 at 9:57 PM, Michael Jerris wrote: > > using proxy_media is my best guess but can?t tell with this little info. > > > > On Nov 22, 2016, at 5:27 AM, L?i ??ng wrote: > > > > > > Hi List, I got some trouble with using `absolute_codec_string` param. > > My call scenario is pretty simple: caller <--> FS <--> callee. > > My caller compose `m=audio 7078 RTP/AVP 8 0 101` in its INVITE, and I'm > doing `` in the dialplan. > > But FS still use `m=audio 22952 RTP/AVP 8 0 101` in the INVITE to the > callee. > > not sure what I'm missing, helps would be appreciated. > > Note that when I'm using `originate` application in fs_cli, things are > good. > > `originate {absolute_codec_string=PCMU}sofia/gateway/caller/100 > &bridge({absolute_codec_string=PCMA}sofia/gateway/callee/100`. > > I have FS with proper behavior in transcoding, caller has `m=audio 31184 > RTP/AVP 0 101` received, and callee has `m=audio 21922 RTP/AVP 8 101` > received. > > rgds, > > Loi Dang Thanh > > Phone : 84.1224.735.448 > > Email : loi.dangthanh 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161123/9d3fdaca/attachment-0001.html From catogonzalez at gmail.com Wed Nov 23 15:48:30 2016 From: catogonzalez at gmail.com (Cato Gonzalez) Date: Wed, 23 Nov 2016 07:48:30 -0500 Subject: [Freeswitch-users] Bridge an incoming call to an external SIP address In-Reply-To: References: Message-ID: Hi Steven, thanks for your reply. Yes, you are right, I had it all backwards :( I solved this by bridging to a gateway but *also* had to place this before bridging the call: Thanks again for replying. On Tue, Nov 22, 2016 at 6:14 PM, Steven Ayre wrote: > Are you sure 'internal' is the sofia profile you want to be using to reach > an external address? > > On 21 November 2016 at 13:35, Cato Gonzalez > wrote: > >> Hi everyone, >> >> I am trying to get FS to route inbound calls to an outside sip provider >> but I am getting the call hung up on the FS side just upon the first or >> second ring on the B-leg. The diaplan action being executed is: >> >> >> >> I have a softphone registered to this provider sip.linphone.org and it >> works well when calling XXX from some other IP phone. FS reports this in >> the logs: >> >> [DEBUG] switch_ivr_originate.c:1274 Raw Codec Activation Success >> L16 at 48000hz 1 channel 20ms >> [DEBUG] switch_core_codec.c:221 sofia/internal/anonymous at webrt >> c.sip.mydomain.co Push codec L16:100 >> [DEBUG] switch_ivr_originate.c:1343 Play Ringback Tone >> [%(2000,4000,440,480)] >> [DEBUG] sofia.c:6760 Channel sofia/internal/anonymous at webrt >> c.sip.mydomain.co entering state [terminated][500] >> [NOTICE] sofia.c:7779 Hangup sofia/internal/anonymous at webrt >> c.sip.mydomain.co [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] >> [DEBUG] switch_core_codec.c:246 sofia/internal/anonymous at webrt >> c.sip.mydomain.co Restore previous codec opus:116. >> [NOTICE] switch_ivr_originate.c:3523 Hangup sofia/external/ >> XXX at sip.linphone.org [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL] >> >> Result is the same if I try to call any other sip destination. I >> appreciate any input anyone may give: have been working on this for a week >> already. >> >> Thanks, >> >> Cato >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> 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/20161123/a07a5cd0/attachment.html From fanx07 at gmail.com Wed Nov 23 16:55:13 2016 From: fanx07 at gmail.com (Anonim Stefan) Date: Wed, 23 Nov 2016 15:55:13 +0200 Subject: [Freeswitch-users] mod_vlc playback hangup issue Message-ID: Hi, I'm trying to playback a .wav using mod_vlc (.wav obtained via maryTTS). Dialplan extension looks like this: The .wav gets played correctly with [1] Freeswitch log output. The problem is that after the file ends, the channel never hangup(i.e. show channels won't decrease), even after I end call. Actually, I won't receive 200OK for my BYE, as seen in the attached picture. I am using FreeSWITCH version: 1.6.12-20-b91a0a6~64bit (-20-b91a0a6 64bit). Is this a bug ? Thank you, Stefan [1] http://pastebin.com/Q364uHvp -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161123/ce83e7b0/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: FS.png Type: image/png Size: 45018 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161123/ce83e7b0/attachment-0001.png From fanx07 at gmail.com Wed Nov 23 17:07:48 2016 From: fanx07 at gmail.com (Anonim Stefan) Date: Wed, 23 Nov 2016 16:07:48 +0200 Subject: [Freeswitch-users] mod_vlc playback hangup issue In-Reply-To: References: Message-ID: Small typo in the above playback; should have been: If I end the call before .wav finishes, all is good: 200 OK received and channel deleted. So this is happening only if mod_vlc playback finishes playing the .wav file. Thank you, Stefan On Wed, Nov 23, 2016 at 3:55 PM, Anonim Stefan wrote: > Hi, > > I'm trying to playback a .wav using mod_vlc (.wav obtained via maryTTS). > Dialplan extension looks like this: > > > > > > > > > > > > > The .wav gets played correctly with [1] Freeswitch log output. > > The problem is that after the file ends, the channel never hangup(i.e. > show channels won't decrease), even after I end call. Actually, I won't > receive 200OK for my BYE, as seen in the attached picture. I am using > FreeSWITCH version: 1.6.12-20-b91a0a6~64bit (-20-b91a0a6 64bit). > > Is this a bug ? > > Thank you, > Stefan > > [1] http://pastebin.com/Q364uHvp > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161123/c22ee10b/attachment.html From kiranravuri04 at gmail.com Wed Nov 23 09:29:24 2016 From: kiranravuri04 at gmail.com (Kiran Ravuri) Date: Wed, 23 Nov 2016 11:59:24 +0530 Subject: [Freeswitch-users] Unable to make audio-only conference Message-ID: Hi all, I am unable to make audio-only conference call. I am using version 1.6. Even after I choose "wideband" profile, video is negotiated in SDP. Please let me what i should do. BRs, Kiran -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161123/ad6edf91/attachment.html From aphoticwalker at gmail.com Wed Nov 23 09:58:58 2016 From: aphoticwalker at gmail.com (Charles Yu) Date: Wed, 23 Nov 2016 14:58:58 +0800 Subject: [Freeswitch-users] How to enable IPv6? Message-ID: Hi, We want to use Freeswitch at IPv6 environment. How to do it? I saw there's a global variable "local_ip_v6". I think I need to change the default global domain variable ( at vars.xml ) as "local_ip_v6" and other module configs that use global variable "local_ip_v4". Are these right? Thanks in advance, Charles -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161123/0e3b264f/attachment.html From mike at jerris.com Wed Nov 23 20:33:01 2016 From: mike at jerris.com (Michael Jerris) Date: Wed, 23 Nov 2016 12:33:01 -0500 Subject: [Freeswitch-users] absolute_codec_string not working In-Reply-To: References: <40268DB6-3518-4C8E-A022-D67FB585E2D9@jerris.com> <09ed01d24539$9e24ccf0$da6e66d0$@freeswitch.org> Message-ID: I?m not totally sure what you are trying to accomplish but proxy_media is completely unnecessary and undesired for what you are doing. It should ONLY be used in the case where you are trying to pass codecs we don?t know about at all. Take a look at the codec negotiation page on freeswitch.org/confluence and I think you will find your answers. I don?t think you need a custom mod looking for events with what you have described so far. > On Nov 23, 2016, at 5:37 AM, L?i ??ng wrote: > > In FS document of media proxy mode: > > FreeSWITCH has no control or even understanding of other SDP parameters. > Look like I have to find another way, like writing a custom module listening on specific event. > Any suggest? > > Thanks to all of you. > rgds, > > Loi Dang Thanh > Phone : 841224.735.448 > Email : loi.dangthanh at gmail.com > > On Wed, Nov 23, 2016 at 10:28 AM, Ken Rice > wrote: > If you are limiting the calls to specific codecs and avoiding transcoding, proxy media doesn?t really reduce the overhead anymore? that changed a few years ago but the notion its better still hangs on today > > > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org ] On Behalf Of L?i ??ng > Sent: Tuesday, November 22, 2016 9:07 PM > To: FreeSWITCH Users Help > > Subject: Re: [Freeswitch-users] absolute_codec_string not working > > > > Hi @Michael, you were right, I'm intentionally using media_proxy for FS, since I want to reduce CPU usage on FS machine. > > In this case, I just want to limit the codecs used for each endpoint, and codec negotiation will be handled by them. > > e.g: caller use PCMA, PCMU, GSM by its own in INVITE, I want to limit the callee to only use PCMA,GSM. > > Look like `absolute_codec_string` is not what I'm looking for right? Any way out? > > > > Loi Dang Thanh > > Phone : 01224.735.448 > > Email : loi.dangthanh at gmail.com > > > On Tue, Nov 22, 2016 at 9:57 PM, Michael Jerris > wrote: > > using proxy_media is my best guess but can?t tell with this little info. > > > > On Nov 22, 2016, at 5:27 AM, L?i ??ng > wrote: > > > > > > Hi List, I got some trouble with using `absolute_codec_string` param. > > My call scenario is pretty simple: caller <--> FS <--> callee. > > My caller compose `m=audio 7078 RTP/AVP 8 0 101` in its INVITE, and I'm doing `` in the dialplan. > > But FS still use `m=audio 22952 RTP/AVP 8 0 101` in the INVITE to the callee. > > not sure what I'm missing, helps would be appreciated. > > Note that when I'm using `originate` application in fs_cli, things are good. > > `originate {absolute_codec_string=PCMU}sofia/gateway/caller/100 &bridge({absolute_codec_string=PCMA}sofia/gateway/callee/100`. > > I have FS with proper behavior in transcoding, caller has `m=audio 31184 RTP/AVP 0 101` received, and callee has `m=audio 21922 RTP/AVP 8 101` received. > > rgds, > > Loi Dang Thanh > > Phone : 84.1224.735.448 > > Email : loi.dangthanh 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > 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/20161123/4e0de866/attachment-0001.html From mike at jerris.com Wed Nov 23 20:36:25 2016 From: mike at jerris.com (Michael Jerris) Date: Wed, 23 Nov 2016 12:36:25 -0500 Subject: [Freeswitch-users] Unable to make audio-only conference In-Reply-To: References: Message-ID: <6873240E-2CD4-4CB2-955E-790CAAE9C183@jerris.com> This would likely be based on codec preferences. Check the codec prefs you have configured. > On Nov 23, 2016, at 1:29 AM, Kiran Ravuri wrote: > > Hi all, > > I am unable to make audio-only conference call. > I am using version 1.6. > Even after I choose "wideband" profile, video is negotiated in SDP. > > Please let me what i should do. > > BRs, > Kiran From mario_fs at mgtech.com Wed Nov 23 23:04:21 2016 From: mario_fs at mgtech.com (Mario G) Date: Wed, 23 Nov 2016 12:04:21 -0800 Subject: [Freeswitch-users] How to enable IPv6? In-Reply-To: References: Message-ID: <7F7A31EA-1E59-40E8-BA37-8A8DF5D8CD0F@mgtech.com> I was able to get it working with only adding the IP address to local_ip_v6 and making sure the internal-ipv6 profile was active. Then setting the Bria clients (I was a Bria IPV6 beta tester) and my Yealink phones to IPV6. However, I had to back off of IPV6 in FS due to some issues and am wondering how may people are using IPV6 in FS. Before you commit see the kinds of things I ran into: FS-9428 F S-9428 and FS-9430 . Mario G > On Nov 22, 2016, at 10:58 PM, Charles Yu wrote: > > Hi, > > We want to use Freeswitch at IPv6 environment. How to do it? > I saw there's a global variable "local_ip_v6". I think I need to change the default global domain variable ( at vars.xml ) as "local_ip_v6" and other module configs that use global variable "local_ip_v4". Are these right? > > Thanks in advance, > Charles > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161123/ca51f3f3/attachment.html From loi.dangthanh at gmail.com Thu Nov 24 07:35:57 2016 From: loi.dangthanh at gmail.com (=?UTF-8?B?TOG7o2kgxJDhurduZw==?=) Date: Thu, 24 Nov 2016 11:35:57 +0700 Subject: [Freeswitch-users] absolute_codec_string not working In-Reply-To: References: <40268DB6-3518-4C8E-A022-D67FB585E2D9@jerris.com> <09ed01d24539$9e24ccf0$da6e66d0$@freeswitch.org> Message-ID: actually, my FS needs to be in proxy_media mode, since it always deal with codecs it doesn't know about, g729. real case: my caller(asterisk) always compose INVITE with `G729,PCMA,PCMU,GSM` to FS, some of my callee only accept G729, while others accept G729,PCMA, and so on ... I want to limit codecs choice for each callee accordingly, instead of fully pass `G729,PCMA,PCMU,GSM` to every callee, so that they don't know my full supported codec. rgds Loi Dang Thanh Phone : 84.1224.735.448 Email : loi.dangthanh at gmail.com On Thu, Nov 24, 2016 at 12:33 AM, Michael Jerris wrote: > I?m not totally sure what you are trying to accomplish but proxy_media is > completely unnecessary and undesired for what you are doing. It should > ONLY be used in the case where you are trying to pass codecs we don?t know > about at all. Take a look at the codec negotiation page on > freeswitch.org/confluence and I think you will find your answers. I > don?t think you need a custom mod looking for events with what you have > described so far. > > On Nov 23, 2016, at 5:37 AM, L?i ??ng wrote: > > In FS document of media proxy mode: > > FreeSWITCH has no control or even understanding of other SDP parameters. > Look like I have to find another way, like writing a custom module > listening on specific event. > Any suggest? > > Thanks to all of you. > rgds, > > Loi Dang Thanh > Phone : 841224.735.448 > Email : loi.dangthanh at gmail.com > > On Wed, Nov 23, 2016 at 10:28 AM, Ken Rice wrote: > >> If you are limiting the calls to specific codecs and avoiding >> transcoding, proxy media doesn?t really reduce the overhead anymore? that >> changed a few years ago but the notion its better still hangs on today >> >> >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *L?i ??ng >> *Sent:* Tuesday, November 22, 2016 9:07 PM >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] absolute_codec_string not working >> >> >> >> Hi @Michael, you were right, I'm intentionally using media_proxy for FS, >> since I want to reduce CPU usage on FS machine. >> >> In this case, I just want to limit the codecs used for each endpoint, and >> codec negotiation will be handled by them. >> >> e.g: caller use PCMA, PCMU, GSM by its own in INVITE, I want to limit the >> callee to only use PCMA,GSM. >> >> Look like `absolute_codec_string` is not what I'm looking for right? Any >> way out? >> >> >> Loi Dang Thanh >> >> Phone : 01224.735.448 >> >> Email : loi.dangthanh at gmail.com >> >> >> >> On Tue, Nov 22, 2016 at 9:57 PM, Michael Jerris wrote: >> >> using proxy_media is my best guess but can?t tell with this little info. >> >> >> >> On Nov 22, 2016, at 5:27 AM, L?i ??ng wrote: >> >> >> >> >> >> Hi List, I got some trouble with using `absolute_codec_string` param. >> >> My call scenario is pretty simple: caller <--> FS <--> callee. >> >> My caller compose `m=audio 7078 RTP/AVP 8 0 101` in its INVITE, and I'm >> doing `` in the dialplan. >> >> But FS still use `m=audio 22952 RTP/AVP 8 0 101` in the INVITE to the >> callee. >> >> not sure what I'm missing, helps would be appreciated. >> >> Note that when I'm using `originate` application in fs_cli, things are >> good. >> >> `originate {absolute_codec_string=PCMU}sofia/gateway/caller/100 >> &bridge({absolute_codec_string=PCMA}sofia/gateway/callee/100`. >> >> I have FS with proper behavior in transcoding, caller has `m=audio 31184 >> RTP/AVP 0 101` received, and callee has `m=audio 21922 RTP/AVP 8 101` >> received. >> >> rgds, >> >> Loi Dang Thanh >> >> Phone : 84.1224.735.448 >> >> Email : loi.dangthanh 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 >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> 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/20161124/ea898f41/attachment-0001.html From udy786 at gmail.com Thu Nov 24 08:02:48 2016 From: udy786 at gmail.com (Uday kumar) Date: Thu, 24 Nov 2016 10:32:48 +0530 Subject: [Freeswitch-users] SMPP Incoming Message Message-ID: Hi, I have installed freeswitch 1.6.12 and SMPP on this. Configured SMPP account and able to send outgoing message from softphone but when some replying to me then message coming on freeswitch but its coming on my softphone. I followed https://quentustech.com/smpp-support-in-freeswitch.html created by by William King. Log showing same as per William included but in my case its never reach to my extension. Local extension to extension chat working but only when message coming on my DID then its not going routed extension. *My chat plan:- * Please guide me. -- Thanks & Regard Uday Site:- www.shareyourknowledge.in Mobile:- +91-9377579349 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161124/2a4c000b/attachment.html From jose.lopes at itcenter.com.pt Thu Nov 24 12:21:04 2016 From: jose.lopes at itcenter.com.pt (=?UTF-8?Q?Jos=C3=A9_Lopes?=) Date: Thu, 24 Nov 2016 09:21:04 +0000 Subject: [Freeswitch-users] Distinctive ringtone on mod_verto like SIP Header Alert-Info Message-ID: Hello, On mod_verto there is any way to forward the SIP Alert-Info Header (distinctive ringtone) to verto client for a new call? Thanks in advance. Best regards, Jose Lopes -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161124/514b59b3/attachment.html From david.villasmil.work at gmail.com Thu Nov 24 13:30:24 2016 From: david.villasmil.work at gmail.com (David Villasmil) Date: Thu, 24 Nov 2016 10:30:24 +0000 Subject: [Freeswitch-users] FreeSWITCH Registrar TLS offload In-Reply-To: References: Message-ID: Hello, Please come back with the solution when you have it. It should be interesting for people using kamailio/freeswitch. Regards, David On Wed, Nov 23, 2016 at 10:37 AM Vladyslav Zakhozhai wrote: > Alexandru, thank you for the answer. I think you've given me right > direction to investigate. > > As you've mentioned this is really kamailio issue/question. So I'm moving > to sr-users list. > > > 2016-11-22 13:03 GMT+02:00 Alexandru Covalschi <568691 at gmail.com>: > > Do you have set_contact_alias or add_contact_alias in Kamailio? Anyways > you're doing something wrong as AFAIK Kamailio translates contact header to > udp automatically. You should try to post on sr-users list. > > 2016-11-22 12:33 GMT+02:00 Vladyslav Zakhozhai : > > Hi, > > I'm trying to understand what is the best or suitable approach to the > following use case. Let me simplify thing a little bit. > > Suppose we have one FreeSWITCH registrar behind SIP proxy (kamailio). I'd > like to offload SSL/TLS encryption/decryption to SIP proxy: > > REGISTER: > > Request: UAC == SIP/TLS ==> Kamailio == UDP ==> FreeSWITCH:50 > Reply: UAC <== SIP/TLS == Kamailio <== UDP == FreeSWITCH > > INVITE: > UAC1 == SIP/TLS ==> Kamailio == UDP == > FreeSWITCH == UDP ==> Kamailio == > SIP/TLS ==> UAC2 > > (FreeSWITCH uses kamailio as outbound proxy with fs_path tag appended in > dialplan). > > The main problem is in Contact header which contains transport=tls and we > can see it in FreeSWITCH console: > > User: user at domain.com > Contact: "" > Status: Registered(TLS)(unknown) EXP(2016-11-22 10:16:59) EXPSECS(108) > IP: SIP_PROXY_IP > Port: 5060 > > When FreeSWITCH sends INVITE to UAC2 (during call) it tries to establish > TLS session to UAC2. It fails because there is no TLS-enabled sofia > profiles in the config of FreeSWITCH. > > I have only one solution in my mind: rewrite transport tag in Contact > header on SIP proxy (transport=udp to FreeSWITCH, and transport=tls to UAC). > > I'd like to know it this solution ok or there is more elegant solutions. > > I've tried appending tag transport=udp in FreeSWITCH's dialplan but no > success. > > 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-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Alexandru Covalschi > VoIP engineer and system administrator > tel: +37367398493 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > 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/20161124/760b9799/attachment-0001.html From athompson at successos.com Thu Nov 24 19:06:40 2016 From: athompson at successos.com (Adrian Thompson) Date: Thu, 24 Nov 2016 10:06:40 -0600 Subject: [Freeswitch-users] Best Way To Chat / Presence / File Xfer Message-ID: <003501d2466c$c11a6330$434f2990$@successos.com> Hello, Been a user and fan of Freeswitch for two years now - thank you for the great software! I've done days of research and would like to reach out to see if anyone else has some ideas here: We are looking into giving all of our techs a softphone for their mobile device so they can communicate back to base and to each other while out in the field. Our desired features are as follows: - Video (for troubleshooting equipment with other remote techs) - Chat (for quick updates) - Group Chat (for group reminders) - Page (reach out to a busy individual with an important message) - Group Page (reach out to a busy group of workers with an important message) - File Transfer (send each other marketing material and tech manuals on the fly) - Presence (so employees can let everyone else know if they are available) Video - Check, Working Chat - Check, SIMPLE, Working Group Chat - Not an option with SIMPLE?, Not working Page - Check, Working Group Page - Check, Working File Transfer - Not an option with SIMPLE?, Not working Presence - I have BLF working great, but what about extended presence such as 'busy' or 'I'm a leprechaun' Do I have to start using XMPP for these things? It could complicate the setup three-fold if I have to go that route and I would rather not.. But if I have to, is there documents supporting a configuration that includes Freeswitch for voice/video and XMPP for everything else? Also, does anyone have a suggestion for a mobile softphone with video, chat, file xfer and can be provisioned server side? (I've tried the Grandstream Wave, but it cannot use an HTTPS server with authentication for some reason so I'm looking for more options) Thanks in advance for advice! Adrian -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161124/4ddd3a9a/attachment.html From aphoticwalker at gmail.com Fri Nov 25 07:28:54 2016 From: aphoticwalker at gmail.com (Charles Yu) Date: Fri, 25 Nov 2016 12:28:54 +0800 Subject: [Freeswitch-users] How to enable IPv6? In-Reply-To: <7F7A31EA-1E59-40E8-BA37-8A8DF5D8CD0F@mgtech.com> References: <7F7A31EA-1E59-40E8-BA37-8A8DF5D8CD0F@mgtech.com> Message-ID: Hi, Thanks for your reply. We'll confirm the network environment and try this.. Charles 2016-11-24 4:04 GMT+08:00 Mario G : > I was able to get it working with only adding the IP address to > local_ip_v6 and making sure the internal-ipv6 profile was active. Then > setting the Bria clients (I was a Bria IPV6 beta tester) and my Yealink > phones to IPV6. However, I had to back off of IPV6 in FS due to some issues > and am wondering how may people are using IPV6 in FS. Before you commit see > the kinds of things I ran into: > F > S-9428 and FS-9430 > . > Mario G > > On Nov 22, 2016, at 10:58 PM, Charles Yu wrote: > > Hi, > > We want to use Freeswitch at IPv6 environment. How to do it? > I saw there's a global variable "local_ip_v6". I think I need to change > the default global domain variable ( at vars.xml ) as "local_ip_v6" and > other module configs that use global variable "local_ip_v4". Are these > right? > > Thanks in advance, > Charles > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > 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/20161125/d3a9da44/attachment.html From v.zakhozhai at gmail.com Fri Nov 25 14:15:35 2016 From: v.zakhozhai at gmail.com (Vladyslav Zakhozhai) Date: Fri, 25 Nov 2016 13:15:35 +0200 Subject: [Freeswitch-users] FreeSWITCH Registrar TLS offload In-Reply-To: References: Message-ID: David, yes of course I'll be back with solution here :) But I'm not sure when exactly. 2016-11-24 12:30 GMT+02:00 David Villasmil : > Hello, > > Please come back with the solution when you have it. It should be > interesting for people using kamailio/freeswitch. > > Regards, > > David > > On Wed, Nov 23, 2016 at 10:37 AM Vladyslav Zakhozhai < > v.zakhozhai at gmail.com> wrote: > >> Alexandru, thank you for the answer. I think you've given me right >> direction to investigate. >> >> As you've mentioned this is really kamailio issue/question. So I'm moving >> to sr-users list. >> >> >> 2016-11-22 13:03 GMT+02:00 Alexandru Covalschi <568691 at gmail.com>: >> >> Do you have set_contact_alias or add_contact_alias in Kamailio? Anyways >> you're doing something wrong as AFAIK Kamailio translates contact header to >> udp automatically. You should try to post on sr-users list. >> >> 2016-11-22 12:33 GMT+02:00 Vladyslav Zakhozhai : >> >> Hi, >> >> I'm trying to understand what is the best or suitable approach to the >> following use case. Let me simplify thing a little bit. >> >> Suppose we have one FreeSWITCH registrar behind SIP proxy (kamailio). I'd >> like to offload SSL/TLS encryption/decryption to SIP proxy: >> >> REGISTER: >> >> Request: UAC == SIP/TLS ==> Kamailio == UDP ==> FreeSWITCH:50 >> Reply: UAC <== SIP/TLS == Kamailio <== UDP == FreeSWITCH >> >> INVITE: >> UAC1 == SIP/TLS ==> Kamailio == UDP == > FreeSWITCH == UDP ==> Kamailio >> == SIP/TLS ==> UAC2 >> >> (FreeSWITCH uses kamailio as outbound proxy with fs_path tag appended in >> dialplan). >> >> The main problem is in Contact header which contains transport=tls and we >> can see it in FreeSWITCH console: >> >> User: user at domain.com >> Contact: "" >> Status: Registered(TLS)(unknown) EXP(2016-11-22 10:16:59) >> EXPSECS(108) >> IP: SIP_PROXY_IP >> Port: 5060 >> >> When FreeSWITCH sends INVITE to UAC2 (during call) it tries to establish >> TLS session to UAC2. It fails because there is no TLS-enabled sofia >> profiles in the config of FreeSWITCH. >> >> I have only one solution in my mind: rewrite transport tag in Contact >> header on SIP proxy (transport=udp to FreeSWITCH, and transport=tls to UAC). >> >> I'd like to know it this solution ok or there is more elegant solutions. >> >> I've tried appending tag transport=udp in FreeSWITCH's dialplan but no >> success. >> >> 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-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> -- >> Alexandru Covalschi >> VoIP engineer and system administrator >> tel: +37367398493 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> 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/20161125/b6ee2852/attachment-0001.html From shaun.stokes at itec-support.co.uk Fri Nov 25 17:54:50 2016 From: shaun.stokes at itec-support.co.uk (Shaun Stokes) Date: Fri, 25 Nov 2016 14:54:50 +0000 Subject: [Freeswitch-users] Best Way To Chat / Presence / File Xfer In-Reply-To: <003501d2466c$c11a6330$434f2990$@successos.com> References: <003501d2466c$c11a6330$434f2990$@successos.com> Message-ID: <6FD2F8B5BB72834E9939AEDF9FB802A901E86132D4@mbx-01.sysconfig.co.uk> This is something we've also been looking into. I believe the FreeSWITCH Verto module can support these features over WebRTC and it looks very promising but you would need to develop the client app or web page to facilitate this, you're best bet would be to chat with FreeSWITCH Solutions. We have some fairly tight deadlines so we're planning to re-visit Verto at a later date, for the time being we intend to use the Jitsi softphone client (javascript based) with a combination of SIP (FreeSWITCH) and XMPP you might be able to package this into a phone app. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Adrian Thompson Sent: 24 November 2016 16:07 To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Best Way To Chat / Presence / File Xfer Hello, Been a user and fan of Freeswitch for two years now - thank you for the great software! I've done days of research and would like to reach out to see if anyone else has some ideas here: We are looking into giving all of our techs a softphone for their mobile device so they can communicate back to base and to each other while out in the field. Our desired features are as follows: - Video (for troubleshooting equipment with other remote techs) - Chat (for quick updates) - Group Chat (for group reminders) - Page (reach out to a busy individual with an important message) - Group Page (reach out to a busy group of workers with an important message) - File Transfer (send each other marketing material and tech manuals on the fly) - Presence (so employees can let everyone else know if they are available) Video - Check, Working Chat - Check, SIMPLE, Working Group Chat - Not an option with SIMPLE?, Not working Page - Check, Working Group Page - Check, Working File Transfer - Not an option with SIMPLE?, Not working Presence - I have BLF working great, but what about extended presence such as 'busy' or 'I'm a leprechaun' Do I have to start using XMPP for these things? It could complicate the setup three-fold if I have to go that route and I would rather not.... But if I have to, is there documents supporting a configuration that includes Freeswitch for voice/video and XMPP for everything else? Also, does anyone have a suggestion for a mobile softphone with video, chat, file xfer and can be provisioned server side? (I've tried the Grandstream Wave, but it cannot use an HTTPS server with authentication for some reason so I'm looking for more options) Thanks in advance for advice! Adrian ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ [http://www.itec-support.co.uk/wp-content/uploads/2016/07/email_logo.jpg] Shaun Stokes - Infrastructure Analyst T : 01453 700713 E : shaun.stokes at itec-support.co.uk W : www.itec-support.co.uk Registered Address :- ITEC Support, Suite 2 Prospect House, Bath Road, Stroud, Gloucestershire GL5 3QF Company No. 06908001 CONFIDENTIALITY NOTICE This communication and the information it contains are intended for the person or organisation to which it is addressed. Its contents are confidential and may be protected in law. Unauthorised use, copying or disclosure of any of it may be unlawful. If you are not the intended recipient, please contact us immediately. The contents of any attachments in this e-mail may contain software viruses, which could damage your own computer system. While ITEC Support has taken every reasonable precaution to minimise this risk, we cannot accept liability for any damage which you sustain as a result of software viruses. You should carry out your own virus checking procedure before opening any attachment. ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161125/ff5da2cd/attachment.html From bilaln018 at gmail.com Fri Nov 25 18:03:12 2016 From: bilaln018 at gmail.com (Bilal Abbasi) Date: Fri, 25 Nov 2016 20:03:12 +0500 Subject: [Freeswitch-users] [Bridge two channels on talk] Message-ID: Hi Users, I want to bridge two channels on talk event,scenario is Call comes into FS and put on park. Another call is originated and on talk event i want to bridge that with the parked leg uuid. I will use uuid_brige to bridge two legs, But how can i execute bridge on talk event? i am looking onto some thing "execute on talk" that i can pass with originate command(ideally). P.S: i found on talk event but dont know how to use that, any help is highly appreciated. Regards Abbasi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161125/83af4e38/attachment.html From v.zakhozhai at gmail.com Fri Nov 25 18:49:27 2016 From: v.zakhozhai at gmail.com (Vladyslav Zakhozhai) Date: Fri, 25 Nov 2016 17:49:27 +0200 Subject: [Freeswitch-users] Best Way To Chat / Presence / File Xfer In-Reply-To: <6FD2F8B5BB72834E9939AEDF9FB802A901E86132D4@mbx-01.sysconfig.co.uk> References: <003501d2466c$c11a6330$434f2990$@successos.com> <6FD2F8B5BB72834E9939AEDF9FB802A901E86132D4@mbx-01.sysconfig.co.uk> Message-ID: Many things depend on client capabilities. I.e. jitsi may act as SIP cleint and XMPP client. Linphone can act as SIP client (including SIP SIMPLE), etc. In the first example we can setup separate XMPP server and configure client app. But there is some draft - https://tools.ietf.org/html/draft-ietf-mmusic-file-transfer-mech-11 (sorry for maybe obsoleted doc, I beleive there is more fresh standard). Kamailio supports MSRP for example. And I think that it can be solution of this issue. >From the other side Kamailio can act as XMPP gateway also. In this use case there is SIP proxy and XMPP server still separate. But one entry point for client app. Sorry for mess of thoughts. I'm investigating this topic too. 2016-11-25 16:54 GMT+02:00 Shaun Stokes : > This is something we?ve also been looking into. > > > > I believe the FreeSWITCH Verto module can support these features over > WebRTC and it looks very promising but you would need to develop the client > app or web page to facilitate this, you?re best bet would be to chat with > FreeSWITCH Solutions. > > > > We have some fairly tight deadlines so we?re planning to re-visit Verto at > a later date, for the time being we intend to use the Jitsi softphone > client (javascript based) with a combination of SIP (FreeSWITCH) and XMPP > you might be able to package this into a phone app. > > > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Adrian > Thompson > *Sent:* 24 November 2016 16:07 > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* [Freeswitch-users] Best Way To Chat / Presence / File Xfer > > > > > > Hello, > > > > Been a user and fan of Freeswitch for two years now ? thank you for the > great software! > > I?ve done days of research and would like to reach out to see if anyone > else has some ideas here: > > We are looking into giving all of our techs a softphone for their mobile > device so they can communicate back to base and to each other while out in > the field. Our desired features are as follows: > > - Video (for troubleshooting equipment with other remote techs) > > - Chat (for quick updates) > > - Group Chat (for group reminders) > > - Page (reach out to a busy individual with an important message) > > - Group Page (reach out to a busy group of workers with an > important message) > > - File Transfer (send each other marketing material and tech > manuals on the fly) > > - Presence (so employees can let everyone else know if they are > available) > > > > Video ? Check, Working > > Chat ? Check, SIMPLE, Working > > Group Chat ? Not an option with SIMPLE?, Not working > > Page ? Check, Working > > Group Page ? Check, Working > > File Transfer ? Not an option with SIMPLE?, Not working > > Presence ? I have BLF working great, but what about extended presence such > as ?busy? or ?I?m a leprechaun? > > > > Do I have to start using XMPP for these things? It could complicate the > setup three-fold if I have to go that route and I would rather not?. But > if I have to, is there documents supporting a configuration that includes > Freeswitch for voice/video and XMPP for everything else? > > > > Also, does anyone have a suggestion for a mobile softphone with video, > chat, file xfer and can be provisioned server side? (I?ve tried the > Grandstream Wave, but it cannot use an HTTPS server with authentication for > some reason so I?m looking for more options) > > > > Thanks in advance for advice! > > Adrian > > > ______________________________________________________________________ > This message has been checked for all known viruses by MessageLabs Virus > Scanning Service. > ______________________________________________________________________ > Shaun Stokes - Infrastructure Analyst > T : 01453 700713 > E : shaun.stokes at itec-support.co.uk > W : www.itec-support.co.uk > Registered Address :- ITEC Support, Suite 2 Prospect House, Bath Road, > Stroud, Gloucestershire GL5 3QF > Company No. 06908001 > > CONFIDENTIALITY NOTICE > This communication and the information it contains are intended for the > person or organisation to which it is addressed. Its contents are > confidential and may be protected in law. Unauthorised use, copying or > disclosure of any of it may be unlawful. If you are not the intended > recipient, please contact us immediately. > The contents of any attachments in this e-mail may contain software > viruses, which could damage your own computer system. While ITEC Support > has taken every reasonable precaution to minimise this risk, we cannot > accept liability for any damage which you sustain as a result of software > viruses. You should carry out your own virus checking procedure before > opening any attachment. > > ______________________________________________________________________ > This message has been checked for all known viruses by MessageLabs Virus > Scanning Service. > ______________________________________________________________________ > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- ? ?????????, ????????? ??????? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161125/c0b9f239/attachment-0001.html From dominique.jeannerod at interact-iv.com Fri Nov 25 19:52:31 2016 From: dominique.jeannerod at interact-iv.com (Dominique Jeannerod) Date: Fri, 25 Nov 2016 17:52:31 +0100 Subject: [Freeswitch-users] mod_fifo and early media Message-ID: Hi, thanks for your answer. I'm not a developper, but will look at that with someone in my team. What I'm currently planning to do (if possible), and submit as a patch for mod_fifo : Add a parameter to make mod_fifo manage two modes : 1. Normal mode, where the A-Leg gets answered (200 OK). 2. "Early media" mode, where the A-Leg is pre-answered, and kept in early media mode, until the B-Leg answers. The A-leg is then (automatically) answered when the B-Leg answers. Does it make sense ? Dominique Jeannerod Interact-iv.com Infrastructure Manager 2016-11-22 18:52 GMT+01:00 Sergey Safarov : > You can try patch mod_fifo source code and comment > "switch_channel_answer(channel);" strings. > Then you can manage signal is passed to a-leg from dialplan. > > ??, 22 ????. 2016 ?. ? 13:24, Dominique Jeannerod > : >> >> Hello, >> >> i'm working on a project where I need to manage a call waiting queue, in >> early media mode, to have a free waiting time, and start billing only after >> the call is really answered. >> >> The general call flow is : >> 1- incoming call to a service number >> 2- early media answer (pre-answer) to manage voice messages >> 3- the call is put in a waiting queue, and the caller ears a message >> (MOH), still in early media >> 4- the queue is associated to a unique destination number, and then one an >> only call at a time is picked-up from the queue and bridged to the >> destination. >> 5- When the call is established (200 OK), the incoming call is also >> answered (200 OK), which starts the billing. >> >> >> Question : this looks like a basic call queue handling, and mod_fifo could >> be a perfect fit for that ... except that mod_fifo answers the call with a >> 200 OK >> >> Is it possible to manage early media with mod_fifo ? >> >> Does someone have a best practice, experience, or advice to share on this >> matter ? >> >> >> Thanks with anticipation >> >> D. Jeannerod >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> 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 s.safarov at gmail.com Fri Nov 25 20:47:36 2016 From: s.safarov at gmail.com (Sergey Safarov) Date: Fri, 25 Nov 2016 17:47:36 +0000 Subject: [Freeswitch-users] mod_fifo and early media In-Reply-To: References: Message-ID: Both cases may be implemented. Also you must know - many provider restrict time while channel state is early media. In my case some operator's disconnect calls after 60 of early media, other after 120 sec. Are is this may take place in you case? ??, 25 ????. 2016, 19:53 Dominique Jeannerod < dominique.jeannerod at interact-iv.com>: > Hi, > > thanks for your answer. > I'm not a developper, but will look at that with someone in my team. > > What I'm currently planning to do (if possible), and submit as a patch > for mod_fifo : > Add a parameter to make mod_fifo manage two modes : > 1. Normal mode, where the A-Leg gets answered (200 OK). > 2. "Early media" mode, where the A-Leg is pre-answered, and kept in > early media mode, until the B-Leg answers. The A-leg is then > (automatically) answered when the B-Leg answers. > > Does it make sense ? > > > Dominique Jeannerod > Interact-iv.com > Infrastructure Manager > > > > 2016-11-22 18:52 GMT+01:00 Sergey Safarov : > > You can try patch mod_fifo source code and comment > > "switch_channel_answer(channel);" strings. > > Then you can manage signal is passed to a-leg from dialplan. > > > > ??, 22 ????. 2016 ?. ? 13:24, Dominique Jeannerod > > : > >> > >> Hello, > >> > >> i'm working on a project where I need to manage a call waiting queue, in > >> early media mode, to have a free waiting time, and start billing only > after > >> the call is really answered. > >> > >> The general call flow is : > >> 1- incoming call to a service number > >> 2- early media answer (pre-answer) to manage voice messages > >> 3- the call is put in a waiting queue, and the caller ears a message > >> (MOH), still in early media > >> 4- the queue is associated to a unique destination number, and then one > an > >> only call at a time is picked-up from the queue and bridged to the > >> destination. > >> 5- When the call is established (200 OK), the incoming call is also > >> answered (200 OK), which starts the billing. > >> > >> > >> Question : this looks like a basic call queue handling, and mod_fifo > could > >> be a perfect fit for that ... except that mod_fifo answers the call > with a > >> 200 OK > >> > >> Is it possible to manage early media with mod_fifo ? > >> > >> Does someone have a best practice, experience, or advice to share on > this > >> matter ? > >> > >> > >> Thanks with anticipation > >> > >> D. Jeannerod > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://confluence.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> 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/20161125/45a4fc41/attachment.html From athompson at successos.com Fri Nov 25 20:53:31 2016 From: athompson at successos.com (Adrian Thompson) Date: Fri, 25 Nov 2016 11:53:31 -0600 Subject: [Freeswitch-users] Best Way To Chat / Presence / File Xfer In-Reply-To: References: <003501d2466c$c11a6330$434f2990$@successos.com> <6FD2F8B5BB72834E9939AEDF9FB802A901E86132D4@mbx-01.sysconfig.co.uk> Message-ID: <003401d24744$d60621c0$82126540$@successos.com> Thanks for the advice ? I ended up installing openfire and jitsi (desktop) and Bria (Mobile). Seems to be the most painless in the short term. Both clients accept SIP and XMPP with video, chat and presence. The only drawback is that Bria doesn?t seem to support the video bridging or XMPP file transfers. In the long run, I am going to play with verto and see what I come up with. Cheers Thanks! From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Vladyslav Zakhozhai Sent: November 25, 2016 9:49 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Best Way To Chat / Presence / File Xfer Many things depend on client capabilities. I.e. jitsi may act as SIP cleint and XMPP client. Linphone can act as SIP client (including SIP SIMPLE), etc. In the first example we can setup separate XMPP server and configure client app. But there is some draft - https://tools.ietf.org/html/draft-ietf-mmusic-file-transfer-mech-11 (sorry for maybe obsoleted doc, I beleive there is more fresh standard). Kamailio supports MSRP for example. And I think that it can be solution of this issue. >From the other side Kamailio can act as XMPP gateway also. In this use case there is SIP proxy and XMPP server still separate. But one entry point for client app. Sorry for mess of thoughts. I'm investigating this topic too. 2016-11-25 16:54 GMT+02:00 Shaun Stokes : This is something we?ve also been looking into. I believe the FreeSWITCH Verto module can support these features over WebRTC and it looks very promising but you would need to develop the client app or web page to facilitate this, you?re best bet would be to chat with FreeSWITCH Solutions. We have some fairly tight deadlines so we?re planning to re-visit Verto at a later date, for the time being we intend to use the Jitsi softphone client (javascript based) with a combination of SIP (FreeSWITCH) and XMPP you might be able to package this into a phone app. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Adrian Thompson Sent: 24 November 2016 16:07 To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Best Way To Chat / Presence / File Xfer Hello, Been a user and fan of Freeswitch for two years now ? thank you for the great software! I?ve done days of research and would like to reach out to see if anyone else has some ideas here: We are looking into giving all of our techs a softphone for their mobile device so they can communicate back to base and to each other while out in the field. Our desired features are as follows: - Video (for troubleshooting equipment with other remote techs) - Chat (for quick updates) - Group Chat (for group reminders) - Page (reach out to a busy individual with an important message) - Group Page (reach out to a busy group of workers with an important message) - File Transfer (send each other marketing material and tech manuals on the fly) - Presence (so employees can let everyone else know if they are available) Video ? Check, Working Chat ? Check, SIMPLE, Working Group Chat ? Not an option with SIMPLE?, Not working Page ? Check, Working Group Page ? Check, Working File Transfer ? Not an option with SIMPLE?, Not working Presence ? I have BLF working great, but what about extended presence such as ?busy? or ?I?m a leprechaun? Do I have to start using XMPP for these things? It could complicate the setup three-fold if I have to go that route and I would rather not?. But if I have to, is there documents supporting a configuration that includes Freeswitch for voice/video and XMPP for everything else? Also, does anyone have a suggestion for a mobile softphone with video, chat, file xfer and can be provisioned server side? (I?ve tried the Grandstream Wave, but it cannot use an HTTPS server with authentication for some reason so I?m looking for more options) Thanks in advance for advice! Adrian ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ Shaun Stokes - Infrastructure Analyst T : 01453 700713 E : shaun.stokes at itec-support.co.uk W : www.itec-support.co.uk Registered Address :- ITEC Support, Suite 2 Prospect House, Bath Road, Stroud, Gloucestershire GL5 3QF Company No. 06908001 CONFIDENTIALITY NOTICE This communication and the information it contains are intended for the person or organisation to which it is addressed. Its contents are confidential and may be protected in law. Unauthorised use, copying or disclosure of any of it may be unlawful. If you are not the intended recipient, please contact us immediately. The contents of any attachments in this e-mail may contain software viruses, which could damage your own computer system. While ITEC Support has taken every reasonable precaution to minimise this risk, we cannot accept liability for any damage which you sustain as a result of software viruses. You should carry out your own virus checking procedure before opening any attachment. ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- ? ?????????, ????????? ??????? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161125/5bebe7ae/attachment-0001.html From dominique.jeannerod at interact-iv.com Fri Nov 25 20:55:24 2016 From: dominique.jeannerod at interact-iv.com (Dominique Jeannerod) Date: Fri, 25 Nov 2016 18:55:24 +0100 Subject: [Freeswitch-users] mod_fifo and early media In-Reply-To: References: Message-ID: Yes, we are building a service to manage a early media phase for call price announcement, which is a legal requirement in Germany, and has to be free. The billing starts when the call is really answered. This "answer waiting" phase is limited to 120s, and I will manage a timer to kill the call if necessary, after 120s. From bilaln018 at gmail.com Sun Nov 27 11:08:44 2016 From: bilaln018 at gmail.com (Bilal Abbasi) Date: Sun, 27 Nov 2016 13:08:44 +0500 Subject: [Freeswitch-users] [Unable to get VAD event TALK] Message-ID: Hi users, I am currently playing with freeswitch to detect TALK event, i have the following set at profile level, Plus i am exporting fire_talk_events=true while originate as well. Using python pyswitch self.eventsocket.apiOriginate("{execute_on_answer='transfer 05123456789 XML default',originate_timeout='120',origination_caller_id_name='123456789',origination_caller_id_number='123456789'}sofia/external/"+NUMBER+"@X.X.X.X",cidname="godson", cidnum="123" ,channelvars={'rtp_enable_vad_in':'True','rtp_enable_vad_out':'True','origination_uuid':uid,'DTMF':UUID,'fire_talk_events':'true','fire_not_talk_events':'true'}) I am getting this on fs_cli 2016-11-27 09:49:45.160380 [DEBUG] switch_rtp.c:7962 Activate VAD codec G722 20ms 2016-11-27 09:49:45.160380 [DEBUG] switch_core_media.c:6754 AUDIO RTP Engage VAD for sofia/external/05123456789 at X.X.X.X ( in out ) I am getting all other registered events(CHANNEL_ANSWER, CHANNEL_DESTROY etc), but i am not able to get TALK event. Can someone please help me out to get the talk event. Regards Abbasi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161127/c9070cbf/attachment.html From nbhatti at gmail.com Sun Nov 27 13:21:48 2016 From: nbhatti at gmail.com (Muhammad Naseer Bhatti) Date: Sun, 27 Nov 2016 02:21:48 -0800 Subject: [Freeswitch-users] 2 CS_DESTROY events without trying the second one in the bridge Message-ID: I have a simple Dialplan with two gateways to try, and I have tried to bridge the same with 2 lines of execute with hangup_after_bridge set to true. > > > > > > > > > > > > > > > > > > > > > > > > > > > Please guide me. > > -- > Thanks & Regard > Uday > Site:- www.shareyourknowledge.in > Mobile:- +91-9377579349 > -- Thanks & Regard Uday Site:- www.shareyourknowledge.in Mobile:- +91-9377579349 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161128/f3d7182c/attachment-0001.html From s.safarov at gmail.com Mon Nov 28 09:52:22 2016 From: s.safarov at gmail.com (Sergey Safarov) Date: Mon, 28 Nov 2016 06:52:22 +0000 Subject: [Freeswitch-users] How to drop a voice mail to user who doesn't exist In-Reply-To: References: Message-ID: Think is required dynamic directory and dialling generation. Try mod_curl ??, 28 ????. 2016, 9:28 Naveen Tamanam : > Hi, > > How to create a dialplan to facilitate to drop a voicemail to the user > and domain who/which doesn't exist in freeswitch. > > > > -- > Thanks & Regards, > Naveen Tamanam > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161128/1bb9916d/attachment.html From bilaln018 at gmail.com Mon Nov 28 10:41:35 2016 From: bilaln018 at gmail.com (Bilal Abbasi) Date: Mon, 28 Nov 2016 12:41:35 +0500 Subject: [Freeswitch-users] [Unable to get VAD event TALK] In-Reply-To: References: Message-ID: Just for users info i have activated following event. self.eventsocket.registerEvent("TALK", True, self.channelAnswer) But no luck. Regards Abbasi On Sun, Nov 27, 2016 at 1:08 PM, Bilal Abbasi wrote: > Hi users, > > I am currently playing with freeswitch to detect TALK event, i have the > following set at profile level, > > Plus i am exporting fire_talk_events=true while originate as well. > Using python pyswitch > self.eventsocket.apiOriginate("{execute_on_answer='transfer 05123456789 > XML default',originate_timeout='120',origination_caller_id_ > name='123456789',origination_caller_id_number='123456789'} > sofia/external/"+NUMBER+"@X.X.X.X",cidname="godson", cidnum="123" > ,channelvars={'rtp_enable_vad_in':'True','rtp_enable_vad_ > out':'True','origination_uuid':uid,'DTMF':UUID,'fire_talk_ > events':'true','fire_not_talk_events':'true'}) > > I am getting this on fs_cli > 2016-11-27 09:49:45.160380 [DEBUG] switch_rtp.c:7962 Activate VAD codec > G722 20ms > 2016-11-27 09:49:45.160380 [DEBUG] switch_core_media.c:6754 AUDIO RTP > Engage VAD for sofia/external/05123456789 at X.X.X.X ( in out ) > > I am getting all other registered events(CHANNEL_ANSWER, CHANNEL_DESTROY > etc), but i am not able to get TALK event. > > Can someone please help me out to get the talk event. > > Regards > Abbasi > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161128/6232efc2/attachment.html From mike at jerris.com Mon Nov 28 19:09:01 2016 From: mike at jerris.com (Michael Jerris) Date: Mon, 28 Nov 2016 11:09:01 -0500 Subject: [Freeswitch-users] absolute_codec_string not working In-Reply-To: References: <40268DB6-3518-4C8E-A022-D67FB585E2D9@jerris.com> <09ed01d24539$9e24ccf0$da6e66d0$@freeswitch.org> Message-ID: <9EEB3B0B-F454-458B-88E6-38611227751A@jerris.com> FreeSWITCH knows about g729 just fine. you shouldn?t be using proxy. Check out the page i referenced below for information on how to accomplish it, Proxy media mode is not the way you want to use for sure. > On Nov 23, 2016, at 11:35 PM, L?i ??ng wrote: > > actually, my FS needs to be in proxy_media mode, since it always deal with codecs it doesn't know about, g729. > real case: my caller(asterisk) always compose INVITE with `G729,PCMA,PCMU,GSM` to FS, some of my callee only accept G729, while others accept G729,PCMA, and so on ... > I want to limit codecs choice for each callee accordingly, instead of fully pass `G729,PCMA,PCMU,GSM` to every callee, so that they don't know my full supported codec. > > rgds > > Loi Dang Thanh > Phone : 84.1224.735.448 > Email : loi.dangthanh at gmail.com > > On Thu, Nov 24, 2016 at 12:33 AM, Michael Jerris > wrote: > I?m not totally sure what you are trying to accomplish but proxy_media is completely unnecessary and undesired for what you are doing. It should ONLY be used in the case where you are trying to pass codecs we don?t know about at all. Take a look at the codec negotiation page on freeswitch.org/confluence and I think you will find your answers. I don?t think you need a custom mod looking for events with what you have described so far. > >> On Nov 23, 2016, at 5:37 AM, L?i ??ng > wrote: >> >> In FS document of media proxy mode: >> > FreeSWITCH has no control or even understanding of other SDP parameters. >> Look like I have to find another way, like writing a custom module listening on specific event. >> Any suggest? >> >> Thanks to all of you. >> rgds, >> >> Loi Dang Thanh >> Phone : 841224.735.448 >> Email : loi.dangthanh at gmail.com >> >> On Wed, Nov 23, 2016 at 10:28 AM, Ken Rice > wrote: >> If you are limiting the calls to specific codecs and avoiding transcoding, proxy media doesn?t really reduce the overhead anymore? that changed a few years ago but the notion its better still hangs on today >> >> >> >> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org ] On Behalf Of L?i ??ng >> Sent: Tuesday, November 22, 2016 9:07 PM >> To: FreeSWITCH Users Help > >> Subject: Re: [Freeswitch-users] absolute_codec_string not working >> >> >> >> Hi @Michael, you were right, I'm intentionally using media_proxy for FS, since I want to reduce CPU usage on FS machine. >> >> In this case, I just want to limit the codecs used for each endpoint, and codec negotiation will be handled by them. >> >> e.g: caller use PCMA, PCMU, GSM by its own in INVITE, I want to limit the callee to only use PCMA,GSM. >> >> Look like `absolute_codec_string` is not what I'm looking for right? Any way out? >> >> >> >> Loi Dang Thanh >> >> Phone : 01224.735.448 >> >> Email : loi.dangthanh at gmail.com >> >> >> On Tue, Nov 22, 2016 at 9:57 PM, Michael Jerris > wrote: >> >> using proxy_media is my best guess but can?t tell with this little info. >> >> >> >> On Nov 22, 2016, at 5:27 AM, L?i ??ng > wrote: >> >> >> >> >> >> Hi List, I got some trouble with using `absolute_codec_string` param. >> >> My call scenario is pretty simple: caller <--> FS <--> callee. >> >> My caller compose `m=audio 7078 RTP/AVP 8 0 101` in its INVITE, and I'm doing `` in the dialplan. >> >> But FS still use `m=audio 22952 RTP/AVP 8 0 101` in the INVITE to the callee. >> >> not sure what I'm missing, helps would be appreciated. >> >> Note that when I'm using `originate` application in fs_cli, things are good. >> >> `originate {absolute_codec_string=PCMU}sofia/gateway/caller/100 &bridge({absolute_codec_string=PCMA}sofia/gateway/callee/100`. >> >> I have FS with proper behavior in transcoding, caller has `m=audio 31184 RTP/AVP 0 101` received, and callee has `m=audio 21922 RTP/AVP 8 101` received. >> >> rgds, >> >> Loi Dang Thanh >> >> Phone : 84.1224.735.448 >> >> Email : loi.dangthanh at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161128/406b61db/attachment-0001.html From tezcancirakoglu at gmail.com Sat Nov 26 15:15:31 2016 From: tezcancirakoglu at gmail.com (=?UTF-8?B?VGV6Y2FuIMOHSVJBS0/EnkxV?=) Date: Sat, 26 Nov 2016 15:15:31 +0300 Subject: [Freeswitch-users] Management of freeswitch with event socket library Message-ID: Hi there, I am currently trying to develop a management console for freeswitch which is supposed to manage couple of freeswitch instances on different servers. So i am using event socket library to manage them remotely. My question is, can i edit configuration xml files (like dialplan, acl, directory etc) with inbound or outbound socket interface? Also is there a detailed documentation about commands that i can execute over sockets? Thanks for further help Tezcan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161126/16d9db47/attachment.html From mail at paulzillmann.de Thu Nov 24 21:02:38 2016 From: mail at paulzillmann.de (Paul Zillmann) Date: Thu, 24 Nov 2016 19:02:38 +0100 Subject: [Freeswitch-users] all-reg-options-ping don't work Message-ID: Hello there, I've set the options all-reg-options-ping and registration-thread-frequency on multiple gateway profile, but they don't seem to work. siptrace on those profiles don't show any activity. I've put them in the settings part of the profiles. What do I miss? How do I enable options ping on gateways (for example every 20 seconds)? Thanks, From am at voipit.pt Mon Nov 28 16:58:09 2016 From: am at voipit.pt (=?UTF-8?Q?Andr=C3=A9_Maricato?=) Date: Mon, 28 Nov 2016 13:58:09 +0000 Subject: [Freeswitch-users] Freeswitch and STUN behind NAT. Message-ID: I?m trying to configure Freeswitch behind NAT but on following the instructions in https://wiki.freeswitch.org/wiki/NAT_Traversal#FreeSWITCH_ behind_NAT Freeswitch starts to listen for SIP on the random a port obtained from the STUN process, instead of the configured 5080 port for the external profile. Is this expected behaviour? Shouldn't the STUN port be ignored for the SIP profile? Can it be configured in any other way? I would appreciate any input. Already tried it with a static configuration and it works, but I wanted to have a system behind a dynamic IP link which only seems plausible with STUN enabled. Thanks. -- Andr? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161128/2b4723ef/attachment.html From c.blau at foodpanda.com Thu Nov 24 18:26:33 2016 From: c.blau at foodpanda.com (Christoph Blau) Date: Thu, 24 Nov 2016 16:26:33 +0100 Subject: [Freeswitch-users] mod_python InputCallback() Message-ID: Hi There I want to catch some DTMF and if possibly events from mod_callcenter using a Python script. For now DTMF would be a good start to get me going. Following the examples on the old and new Wiki brought no luck since it is just not executing my callback function I hope it is okay that I dumped dialplan, script and CLI output into on gist https://gist.github.com/cblaupanda/587a07b746f0e6427ceaf013576f29af Am I missing something? Could someone please point me in a direction that I get this working? Many Thanks, Christoph Blau -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161124/8c489824/attachment-0001.html From claire.zxy at gmail.com Sun Nov 27 21:08:56 2016 From: claire.zxy at gmail.com (xiyu zhao) Date: Sun, 27 Nov 2016 13:08:56 -0500 Subject: [Freeswitch-users] SIP TLS failed with FSClient 1.2.3.5 Message-ID: <000001d248d9$523258a0$f69709e0$@gmail.com> Hi All, Please help me when you get a chance. I've follow the instruction link below to configure TLS in my freeswitch server, but it failed with my FSClient 1.2.3.5. I copied cafile.pem from my freeswitch to my windows desktop and gived the right directory under "TLS Certificate Directory" shown as below screenshot (also attached). https://freeswitch.org/confluence/display/FREESWITCH/SIP+TLS But I still cannot log in with tls, console log output, and configuration files are below. Kindly take a look and let me know if additional info is needed. P.S. I used IP instead of domain name to create the certificate, is it a problem? E.g: I used ./gentls_cert setup -cn pbx.freeswitch.org -alt DNS: 52.35.22.204 -org 52.35.22.204. Console output: tport.c:2749 tport_wakeup_pri() tport_wakeup_pri(0x7fcee8050770): events IN tport.c:862 tport_alloc_secondary() tport_alloc_secondary(0x7fcee8050770): new secondary tport 0x7fcee8252ea0 tport_type_tcp.c:203 tport_tcp_init_secondary() tport_tcp_init_secondary(0x7fcee8252ea0): Setting TCP_KEEPIDLE to 30 tport_type_tcp.c:209 tport_tcp_init_secondary() tport_tcp_init_secondary(0x7fcee8252ea0): Setting TCP_KEEPINTVL to 30 tport_type_tls.c:610 tport_tls_accept() tport_tls_accept(0x7fcee8252ea0): new connection from tls/50.187.205.251:56612/sips tport_tls.c:955 tls_connect() tls_connect(0x7fcee8252ea0): events NEGOTIATING tport_tls.c:1044 tls_connect() tls_connect(0x7fcee8252ea0): TLS setup failed (error:00000001:lib(0):func(0):reason(1)) tport.c:2090 tport_close() tport_close(0x7fcee8252ea0): tls/50.187.205.251:56612/sips tport.c:2263 tport_set_secondary_timer() tport(0x7fcee8252ea0): set timer at 0 ms because zap freeswitch at ip-172-31-28-201> sofia status Name Type Data State ============================================================================ ===================== external-ipv6 profile sip:mod_sofia@[::1]:5080 RUNNING (0) 172.31.28.201 alias internal ALIASED external profile sip:mod_sofia at 52.35.22.204:5080 RUNNING (0) external::example.com gateway sip:joeuser at example.com NOREG internal-ipv6 profile sip:mod_sofia@[::1]:5060 RUNNING (0) internal-ipv6 profile sip:mod_sofia@[::1]:5061 RUNNING (0) (TLS) internal profile sip:mod_sofia at 52.35.22.204:5060 RUNNING (0) internal profile sip:mod_sofia at 52.35.22.204:5061 RUNNING (0) (TLS) ============================================================================ ===================== 4 profiles 1 alias Under vars.xml: Under internel.xml: Thanks, Clarie -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161127/f3d8cad2/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/png Size: 49134 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161127/f3d8cad2/attachment-0002.png -------------- next part -------------- A non-text attachment was scrubbed... Name: Capture.PNG Type: image/png Size: 32537 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161127/f3d8cad2/attachment-0003.png From claire.zxy at gmail.com Sun Nov 27 21:14:19 2016 From: claire.zxy at gmail.com (xiyu zhao) Date: Sun, 27 Nov 2016 13:14:19 -0500 Subject: [Freeswitch-users] SIP TLS failed with FSClient 1.2.3.5 In-Reply-To: <000001d248d9$523258a0$f69709e0$@gmail.com> References: <000001d248d9$523258a0$f69709e0$@gmail.com> Message-ID: <000901d248da$12fa7540$38ef5fc0$@gmail.com> Hi All, Sorry for the bad example, it's actually E.g: I used ./gentls_cert setup -cn 52.35.22.204 -alt DNS: 52.35.22.204 -org 52.35.22.204. Below is the view of one cert: root at ip-172-31-28-201:/usr/local/freeswitch/conf/ssl# openssl x509 -noout -inform pem -text -in /usr/local/freeswitch/conf/ssl/agent.pem Certificate: Data: Version: 3 (0x2) Serial Number: be:37:19:a3:98:6e:82:19 Signature Algorithm: sha1WithRSAEncryption Issuer: CN=52.35.22.204, O=52.35.22.204 Validity Not Before: Nov 12 21:20:24 2016 GMT Not After : Nov 11 21:20:24 2022 GMT Subject: CN=52.35.22.204, O=52.35.22.204 Subject Public Key Info: Public Key Algorithm: rsaEncryption Public-Key: (2048 bit) Modulus: 00:bd:01:6a:df:ae:35:f2:82:1f:ca:af:cf:7b:97: 2f:ec:a5:2d:ec:7c:3d:0a:c3:fb:e2:17:d3:78:b6: dc:c6:60:b6:14:eb:6e:5e:96:c2:ef:bf:d8:9f:a7: 19:a1:36:a5:82:37:5b:8b:0a:5d:95:00:9c:11:f0: 90:77:e6:34:f1:36:b3:c9:62:8e:82:28:d3:41:fd: 0a:3e:67:32:57:c2:52:71:8a:9b:99:4c:e0:4b:e4: 15:e0:53:0c:46:d0:98:1a:05:8e:79:f4:c6:d4:0b: b8:16:ea:24:80:1c:67:67:12:16:c4:29:f1:d5:81: ab:4b:b6:a4:b7:f7:a7:ad:11:34:ef:9c:70:dc:a9: 4a:da:9f:dd:14:71:7e:7d:b1:91:ab:f6:fb:f3:fd: a0:9f:56:ab:89:eb:91:fd:1e:74:d6:55:a0:bb:6e: 1d:94:1d:08:c7:26:2d:85:45:46:b4:44:84:e5:ed: 68:83:e6:25:2b:fd:82:d5:7c:67:ce:32:d9:15:d1: de:00:85:62:d7:f7:ad:a8:c2:17:a1:55:c3:64:08: a3:9e:d8:6d:55:f7:4d:a9:4f:73:75:31:74:3c:21: 3b:1e:27:6b:fb:3c:40:49:80:55:0c:dd:90:fe:4c: da:8c:a4:10:d8:bf:1b:12:15:56:81:0a:15:64:04: cc:d3 Exponent: 65537 (0x10001) X509v3 extensions: Netscape Comment: FS Server Cert X509v3 Basic Constraints: CA:FALSE X509v3 Subject Key Identifier: 74:5E:4B:09:21:37:50:1F:BB:F1:A8:D5:1D:6D:D7:36:D9:D5:EE:AD X509v3 Authority Key Identifier: keyid:0B:51:AF:BF:BF:8F:2A:94:8A:18:B6:70:4F:9A:0B:FA:EB:4B:49:FC DirName:/CN=52.35.22.204/O=52.35.22.204 serial:F5:5B:BD:AA:25:4E:16:0B X509v3 Subject Alternative Name: DNS:52.35.22.204 Netscape Cert Type: SSL Server X509v3 Extended Key Usage: TLS Web Server Authentication Signature Algorithm: sha1WithRSAEncryption e7:35:1e:9a:70:6c:1c:61:2f:c8:50:8f:5d:a8:7d:73:cc:a4: c0:7a:54:02:65:91:49:82:0b:86:7f:45:44:91:b2:14:32:c3: d6:50:5c:41:28:f3:80:ca:ea:2b:c3:2c:d7:d8:09:90:11:8b: fe:4e:8d:35:4f:ca:ec:cb:6b:05:ee:63:e3:17:17:4f:be:bb: f7:85:f4:4a:3a:34:b6:4f:c1:5c:d7:07:7e:f5:d5:a5:ae:40: 3c:25:2a:70:24:6d:0e:3c:e4:e1:64:43:7a:6e:10:ad:a2:9e: 38:d5:e3:91:de:4f:e5:60:27:44:58:7c:2a:42:2a:f2:6f:19: 60:d5:01:48:01:39:1a:18:30:3a:f5:e7:d8:fd:c6:00:22:a4: f7:4b:44:c9:c7:4d:02:2a:d3:d4:1b:f2:e6:35:63:7b:c9:0d: 69:2c:38:7f:04:e1:5e:9a:0c:13:21:50:d5:78:3b:22:f4:11: f4:09:73:e8:58:c5:c4:ba:33:28:88:cc:28:c7:7b:1b:73:11: 06:15:ad:29:1a:25:47:0c:91:be:6d:20:7d:88:6e:6a:a1:53: a6:95:84:cc:d3:bc:10:18:e5:43:fa:5c:96:c3:7b:ce:98:c0: d3:dc:81:8c:ea:85:83:69:39:63:2e:fa:a1:03:0e:69:5e:be: c4:52:8c:25 Any ideas? Thanks Claire From: xiyu zhao [mailto:claire.zxy at gmail.com] Sent: Sunday, November 27, 2016 1:09 PM To: freeswitch-users at lists.freeswitch.org Subject: SIP TLS failed with FSClient 1.2.3.5 Hi All, Please help me when you get a chance. I've follow the instruction link below to configure TLS in my freeswitch server, but it failed with my FSClient 1.2.3.5. I copied cafile.pem from my freeswitch to my windows desktop and gived the right directory under "TLS Certificate Directory" shown as below screenshot (also attached). https://freeswitch.org/confluence/display/FREESWITCH/SIP+TLS But I still cannot log in with tls, console log output, and configuration files are below. Kindly take a look and let me know if additional info is needed. P.S. I used IP instead of domain name to create the certificate, is it a problem? E.g: I used ./gentls_cert setup -cn pbx.freeswitch.org -alt DNS: 52.35.22.204 -org 52.35.22.204. Console output: tport.c:2749 tport_wakeup_pri() tport_wakeup_pri(0x7fcee8050770): events IN tport.c:862 tport_alloc_secondary() tport_alloc_secondary(0x7fcee8050770): new secondary tport 0x7fcee8252ea0 tport_type_tcp.c:203 tport_tcp_init_secondary() tport_tcp_init_secondary(0x7fcee8252ea0): Setting TCP_KEEPIDLE to 30 tport_type_tcp.c:209 tport_tcp_init_secondary() tport_tcp_init_secondary(0x7fcee8252ea0): Setting TCP_KEEPINTVL to 30 tport_type_tls.c:610 tport_tls_accept() tport_tls_accept(0x7fcee8252ea0): new connection from tls/50.187.205.251:56612/sips tport_tls.c:955 tls_connect() tls_connect(0x7fcee8252ea0): events NEGOTIATING tport_tls.c:1044 tls_connect() tls_connect(0x7fcee8252ea0): TLS setup failed (error:00000001:lib(0):func(0):reason(1)) tport.c:2090 tport_close() tport_close(0x7fcee8252ea0): tls/50.187.205.251:56612/sips tport.c:2263 tport_set_secondary_timer() tport(0x7fcee8252ea0): set timer at 0 ms because zap freeswitch at ip-172-31-28-201> sofia status Name Type Data State ============================================================================ ===================== external-ipv6 profile sip:mod_sofia@[::1]:5080 RUNNING (0) 172.31.28.201 alias internal ALIASED external profile sip:mod_sofia at 52.35.22.204:5080 RUNNING (0) external::example.com gateway sip:joeuser at example.com NOREG internal-ipv6 profile sip:mod_sofia@[::1]:5060 RUNNING (0) internal-ipv6 profile sip:mod_sofia@[::1]:5061 RUNNING (0) (TLS) internal profile sip:mod_sofia at 52.35.22.204:5060 RUNNING (0) internal profile sip:mod_sofia at 52.35.22.204:5061 RUNNING (0) (TLS) ============================================================================ ===================== 4 profiles 1 alias Under vars.xml: Under internel.xml: Thanks, Clarie -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161127/c9c71fd6/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/png Size: 49134 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161127/c9c71fd6/attachment-0001.png From luis.daniel.lucio at gmail.com Mon Nov 28 18:46:07 2016 From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz) Date: Mon, 28 Nov 2016 10:46:07 -0500 Subject: [Freeswitch-users] homer archtecture In-Reply-To: References: Message-ID: I know maybe this is not the best place to ask, but does anyone have a link where to read about homer archtecture.I want to understand relationship among elements before doing a blind install Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161128/b930f1fa/attachment.html From mike at jerris.com Mon Nov 28 19:20:20 2016 From: mike at jerris.com (Michael Jerris) Date: Mon, 28 Nov 2016 11:20:20 -0500 Subject: [Freeswitch-users] Management of freeswitch with event socket library In-Reply-To: References: Message-ID: <5B73FCFF-7AA4-4337-A871-CBCAB8034613@jerris.com> We provide no api interface to edit config files. You could do some of this by way of serving up config dynamically using mod_xml_curl and editing the underlying store for that information that you create. > On Nov 26, 2016, at 7:15 AM, Tezcan ?IRAKO?LU wrote: > > Hi there, > > I am currently trying to develop a management console for freeswitch which is supposed to manage couple of freeswitch instances on different servers. > > So i am using event socket library to manage them remotely. My question is, can i edit configuration xml files (like dialplan, acl, directory etc) with inbound or outbound socket interface? > > Also is there a detailed documentation about commands that i can execute over sockets? > > Thanks for further help > > Tezcan From jm at mayfirst.org Mon Nov 28 19:35:00 2016 From: jm at mayfirst.org (Jamie McClelland) Date: Mon, 28 Nov 2016 11:35:00 -0500 Subject: [Freeswitch-users] permissions for tmute In-Reply-To: <7169CB34-45E4-4E48-962F-A0C489707A74@jerris.com> References: <20161116155457.hrvaf74bdzvk5qpk@mayfirst.org> <4FF85CF6-4E65-460D-B927-EC60849DC13B@jerris.com> <20161116190148.rup63nwasctdhlud@mayfirst.org> <7169CB34-45E4-4E48-962F-A0C489707A74@jerris.com> Message-ID: <20161128163500.ccex3swenvaxmpee@mayfirst.org> Thanks everyone for your help with this problem. Just to close the loop... My mistake was passing in a garbage value for eventChannel when calling vertoHandle.rpcClient.call("verto.broadcast"). Understandably, freeswitch gave me a permission denied. When I set that value to the proper moderator channel it started working. jamie On Wed Nov 16, Michael Jerris wrote: > all you should need to do that i can think of: > > > > > > On Nov 16, 2016, at 2:01 PM, Jamie McClelland wrote: > > > > Thanks Brian and Michael for pointing me to the moderator flag. > > > > However, I'm not having luck with it. > > > > In my web app, everyone logs into via verto using the same username. > > > > I've configured freeswitch to assign one of them the moderator flag (I > > can confirm using fs_cli that the given conference member has the flag > > set). > > > > However, when that member tries to mute another member via verto I'm > > still getting permission denied. > > > > Is it because all members are logging in as the same user (as defined in > > the directory)? > > > > Or, if this really should be working I can test so more to make sure I'm > > not making any other mistakes. > > > > I'm having a hard time figuring out what permissions belong to a logged > > in verto user and which permissions belong to a given member of a given > > conference. > > > > thanks, > > jamie > > > > > > On Wed Nov 16, Michael Jerris wrote: > >> https://freeswitch.org/confluence/display/FREESWITCH/mod_conference > > >> > >> member flag moderator? > >> > >>> On Nov 16, 2016, at 10:54 AM, Jamie McClelland wrote: > >>> > >>> Hi all - Thanks to the devs for the excellent verto code! > >>> > >>> After a year in use, I'm planning to improve my verto web app by > >>> allowing administrators to mute or kick *other* participants from the > >>> call. > >>> > >>> I'm following these instructions: > >>> > >>> https://evoluxbr.github.io/verto-docs/tut/sending-conference-commands.html > >>> > >>> Thankfully, when I tried it as a regular logged in user that has the proper > >>> permissions to join the conference, I received a permission denied message. > >>> > >>> Now I'd like to login as a user with the proper permissions to mute another > >>> conference participant, but I'm not sure how to set those permissions. > >>> > >>> The regular conference participant has the following: > >>> > >>> > >>> > >>> > >>> > >>> > >>> What would I add/change for a user that has permission to mute or kick another > >>> conference participant? > >>> > >>> thanks! > >>> > >>> jamie > >> > > > >> _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://confluence.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > -- > > May First/People Link > > Growing networks to build a just world > > http://www.mayfirst.org > > https://support.mayfirst.org > > > > OpenPGP Key: http://current.workingdirectory.net/pages/identity/ > > xmpp: jamie at mayfirst.org > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://confluence.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > 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 -- May First/People Link Growing networks to build a just world http://www.mayfirst.org https://support.mayfirst.org OpenPGP Key: http://current.workingdirectory.net/pages/identity/ xmpp: jamie at mayfirst.org -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 833 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161128/fa3af53c/attachment.bin From v.zakhozhai at gmail.com Mon Nov 28 19:38:00 2016 From: v.zakhozhai at gmail.com (Vladyslav Zakhozhai) Date: Mon, 28 Nov 2016 18:38:00 +0200 Subject: [Freeswitch-users] all-reg-options-ping don't work In-Reply-To: References: Message-ID: Hi Paul, Do you want to enable it globally? I'm pretty sure that you can do it on per gateway basis. I.e. gateway parameter ping: 2016-11-24 20:02 GMT+02:00 Paul Zillmann : > Hello there, > > I've set the options all-reg-options-ping and > registration-thread-frequency on multiple gateway profile, but they > don't seem to work. > siptrace on those profiles don't show any activity. > > I've put them in the settings part of the profiles. > What do I miss? How do I enable options ping on gateways (for example > every 20 seconds)? > > 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/20161128/9ab65b6a/attachment.html From v.zakhozhai at gmail.com Mon Nov 28 19:42:36 2016 From: v.zakhozhai at gmail.com (Vladyslav Zakhozhai) Date: Mon, 28 Nov 2016 18:42:36 +0200 Subject: [Freeswitch-users] all-reg-options-ping don't work In-Reply-To: References: Message-ID: According to FreeSWITCH'es wiki ( https://wiki.freeswitch.org/wiki/Sofia.conf.xml#all-reg-options-ping) this all-register-options-ping related to registered end points not gateways to which FreeSWITCH is registered. 2016-11-28 18:38 GMT+02:00 Vladyslav Zakhozhai : > Hi Paul, > > Do you want to enable it globally? > > I'm pretty sure that you can do it on per gateway basis. I.e. gateway > parameter ping: > > > > 2016-11-24 20:02 GMT+02:00 Paul Zillmann : > >> Hello there, >> >> I've set the options all-reg-options-ping and >> registration-thread-frequency on multiple gateway profile, but they >> don't seem to work. >> siptrace on those profiles don't show any activity. >> >> I've put them in the settings part of the profiles. >> What do I miss? How do I enable options ping on gateways (for example >> every 20 seconds)? >> >> 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/20161128/01957bff/attachment-0001.html From fernando at softov.com.br Mon Nov 28 21:46:51 2016 From: fernando at softov.com.br (Luiz Fernando Softov) Date: Mon, 28 Nov 2016 15:46:51 -0300 Subject: [Freeswitch-users] Management of freeswitch with event socket library In-Reply-To: <5B73FCFF-7AA4-4337-A871-CBCAB8034613@jerris.com> References: <5B73FCFF-7AA4-4337-A871-CBCAB8034613@jerris.com> Message-ID: I'm using mod_xml_curl + ESL. FS made a simple HTTP request with some params in body POST. You can use C, nodejs, even PHP to receive and reply this requests. So, I create a server in C, process the request and send the XML on-demand. After reload some module, registration, confs, etc. 2016-11-28 13:20 GMT-03:00 Michael Jerris : > We provide no api interface to edit config files. You could do some of > this by way of serving up config dynamically using mod_xml_curl and editing > the underlying store for that information that you create. > > > > On Nov 26, 2016, at 7:15 AM, Tezcan ?IRAKO?LU > wrote: > > > > Hi there, > > > > I am currently trying to develop a management console for freeswitch > which is supposed to manage couple of freeswitch instances on different > servers. > > > > So i am using event socket library to manage them remotely. My question > is, can i edit configuration xml files (like dialplan, acl, directory etc) > with inbound or outbound socket interface? > > > > Also is there a detailed documentation about commands that i can execute > over sockets? > > > > Thanks for further help > > > > Tezcan > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161128/b1e442c8/attachment.html From v.zakhozhai at gmail.com Tue Nov 29 01:31:48 2016 From: v.zakhozhai at gmail.com (Vladyslav Zakhozhai) Date: Tue, 29 Nov 2016 00:31:48 +0200 Subject: [Freeswitch-users] FreeSWITCH Registrar TLS offload In-Reply-To: References: Message-ID: Hi, I'm from ser-userlist with a good news and testing results :) FreeSWITCH do honor path header and will back responses and will originate calls to/through SIP proxy IP address if it is in the path. Before relaying in Kamailio you need put add_path or add_path_received (both worked fine for me). FreeSWITCH will add it to Contact header: Contact: "" No manual manipulations on Contact header is needed from kamailio side (as well as from FreeSWITCH side). But be aware of correct handling SIP requests (i.e. INVITEs) from FreeSWITCHes. For example my FreeSWITCHes backends are in dispatcher table (sip:IP_ADDR:UDP_PORT). And I've checked it with ds_is_from_list in kamailio. But FreeSWITCH originates INVITE to kamailio from IP_ADDR:RANDOM_PORT. In this case ds_is_from_list fails :( Now I'm checking is there mistakes in my configs or this is normal usecase for FreeSWITCH (I did not mention it earlier). 2016-11-25 13:15 GMT+02:00 Vladyslav Zakhozhai : > David, > > yes of course I'll be back with solution here :) But I'm not sure when > exactly. > > 2016-11-24 12:30 GMT+02:00 David Villasmil >: > >> Hello, >> >> Please come back with the solution when you have it. It should be >> interesting for people using kamailio/freeswitch. >> >> Regards, >> >> David >> >> On Wed, Nov 23, 2016 at 10:37 AM Vladyslav Zakhozhai < >> v.zakhozhai at gmail.com> wrote: >> >>> Alexandru, thank you for the answer. I think you've given me right >>> direction to investigate. >>> >>> As you've mentioned this is really kamailio issue/question. So I'm >>> moving to sr-users list. >>> >>> >>> 2016-11-22 13:03 GMT+02:00 Alexandru Covalschi <568691 at gmail.com>: >>> >>> Do you have set_contact_alias or add_contact_alias in Kamailio? Anyways >>> you're doing something wrong as AFAIK Kamailio translates contact header to >>> udp automatically. You should try to post on sr-users list. >>> >>> 2016-11-22 12:33 GMT+02:00 Vladyslav Zakhozhai : >>> >>> Hi, >>> >>> I'm trying to understand what is the best or suitable approach to the >>> following use case. Let me simplify thing a little bit. >>> >>> Suppose we have one FreeSWITCH registrar behind SIP proxy (kamailio). >>> I'd like to offload SSL/TLS encryption/decryption to SIP proxy: >>> >>> REGISTER: >>> >>> Request: UAC == SIP/TLS ==> Kamailio == UDP ==> FreeSWITCH:50 >>> Reply: UAC <== SIP/TLS == Kamailio <== UDP == FreeSWITCH >>> >>> INVITE: >>> UAC1 == SIP/TLS ==> Kamailio == UDP == > FreeSWITCH == UDP ==> Kamailio >>> == SIP/TLS ==> UAC2 >>> >>> (FreeSWITCH uses kamailio as outbound proxy with fs_path tag appended in >>> dialplan). >>> >>> The main problem is in Contact header which contains transport=tls and >>> we can see it in FreeSWITCH console: >>> >>> User: user at domain.com >>> Contact: "" >>> Status: Registered(TLS)(unknown) EXP(2016-11-22 10:16:59) >>> EXPSECS(108) >>> IP: SIP_PROXY_IP >>> Port: 5060 >>> >>> When FreeSWITCH sends INVITE to UAC2 (during call) it tries to establish >>> TLS session to UAC2. It fails because there is no TLS-enabled sofia >>> profiles in the config of FreeSWITCH. >>> >>> I have only one solution in my mind: rewrite transport tag in Contact >>> header on SIP proxy (transport=udp to FreeSWITCH, and transport=tls to UAC). >>> >>> I'd like to know it this solution ok or there is more elegant solutions. >>> >>> I've tried appending tag transport=udp in FreeSWITCH's dialplan but no >>> success. >>> >>> 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-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> >>> -- >>> Alexandru Covalschi >>> VoIP engineer and system administrator >>> tel: +37367398493 >>> >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> 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/20161129/dbfe11ec/attachment-0001.html From brian at freeswitch.org Tue Nov 29 01:37:26 2016 From: brian at freeswitch.org (Brian West) Date: Mon, 28 Nov 2016 16:37:26 -0600 Subject: [Freeswitch-users] FreeSWITCH Registrar TLS offload In-Reply-To: References: Message-ID: You're using TLS/TCP the random port is how it happens. /b On Mon, Nov 28, 2016 at 4:31 PM, Vladyslav Zakhozhai wrote: > Hi, I'm from ser-userlist with a good news and testing results :) > > FreeSWITCH do honor path header and will back responses and will originate > calls to/through SIP proxy IP address if it is in the path. > > Before relaying in Kamailio you need put add_path or add_path_received > (both worked fine for me). FreeSWITCH will add it to Contact header: > > Contact: "" 3Akamailio_ip%3Blr> > > No manual manipulations on Contact header is needed from kamailio side (as > well as from FreeSWITCH side). > > But be aware of correct handling SIP requests (i.e. INVITEs) from > FreeSWITCHes. For example my FreeSWITCHes backends are in dispatcher table > (sip:IP_ADDR:UDP_PORT). And I've checked it with ds_is_from_list in > kamailio. But FreeSWITCH originates INVITE to kamailio from > IP_ADDR:RANDOM_PORT. In this case ds_is_from_list fails :( > > Now I'm checking is there mistakes in my configs or this is normal usecase > for FreeSWITCH (I did not mention it earlier). > > > 2016-11-25 13:15 GMT+02:00 Vladyslav Zakhozhai : > >> David, >> >> yes of course I'll be back with solution here :) But I'm not sure when >> exactly. >> >> 2016-11-24 12:30 GMT+02:00 David Villasmil > m>: >> >>> Hello, >>> >>> Please come back with the solution when you have it. It should be >>> interesting for people using kamailio/freeswitch. >>> >>> Regards, >>> >>> David >>> >>> On Wed, Nov 23, 2016 at 10:37 AM Vladyslav Zakhozhai < >>> v.zakhozhai at gmail.com> wrote: >>> >>>> Alexandru, thank you for the answer. I think you've given me right >>>> direction to investigate. >>>> >>>> As you've mentioned this is really kamailio issue/question. So I'm >>>> moving to sr-users list. >>>> >>>> >>>> 2016-11-22 13:03 GMT+02:00 Alexandru Covalschi <568691 at gmail.com>: >>>> >>>> Do you have set_contact_alias or add_contact_alias in Kamailio? Anyways >>>> you're doing something wrong as AFAIK Kamailio translates contact header to >>>> udp automatically. You should try to post on sr-users list. >>>> >>>> 2016-11-22 12:33 GMT+02:00 Vladyslav Zakhozhai : >>>> >>>> Hi, >>>> >>>> I'm trying to understand what is the best or suitable approach to the >>>> following use case. Let me simplify thing a little bit. >>>> >>>> Suppose we have one FreeSWITCH registrar behind SIP proxy (kamailio). >>>> I'd like to offload SSL/TLS encryption/decryption to SIP proxy: >>>> >>>> REGISTER: >>>> >>>> Request: UAC == SIP/TLS ==> Kamailio == UDP ==> FreeSWITCH:50 >>>> Reply: UAC <== SIP/TLS == Kamailio <== UDP == FreeSWITCH >>>> >>>> INVITE: >>>> UAC1 == SIP/TLS ==> Kamailio == UDP == > FreeSWITCH == UDP ==> Kamailio >>>> == SIP/TLS ==> UAC2 >>>> >>>> (FreeSWITCH uses kamailio as outbound proxy with fs_path tag appended >>>> in dialplan). >>>> >>>> The main problem is in Contact header which contains transport=tls and >>>> we can see it in FreeSWITCH console: >>>> >>>> User: user at domain.com >>>> Contact: "" >>>> Status: Registered(TLS)(unknown) EXP(2016-11-22 10:16:59) >>>> EXPSECS(108) >>>> IP: SIP_PROXY_IP >>>> Port: 5060 >>>> >>>> When FreeSWITCH sends INVITE to UAC2 (during call) it tries to >>>> establish TLS session to UAC2. It fails because there is no TLS-enabled >>>> sofia profiles in the config of FreeSWITCH. >>>> >>>> I have only one solution in my mind: rewrite transport tag in Contact >>>> header on SIP proxy (transport=udp to FreeSWITCH, and transport=tls to UAC). >>>> >>>> I'd like to know it this solution ok or there is more elegant solutions. >>>> >>>> I've tried appending tag transport=udp in FreeSWITCH's dialplan but no >>>> success. >>>> >>>> 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-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>> switch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> >>>> -- >>>> Alexandru Covalschi >>>> VoIP engineer and system administrator >>>> tel: +37367398493 >>>> >>>> >>>> ____________________________________________________________ >>>> _____________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>> switch-users >>>> 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/free >>>> switch-users >>>> 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 > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com (50% Discount using code FreeSwitch50) http://www.freeswitchcookbook.com (50% Discount using code FreeSwitch50) https://www.gofundme.com/freeswitch_ubuntu Got Bugs? Report them here ! | Reddit: /r/freeswitch *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/20161128/4f924712/attachment-0001.html From v.zakhozhai at gmail.com Tue Nov 29 01:51:55 2016 From: v.zakhozhai at gmail.com (Vladyslav Zakhozhai) Date: Tue, 29 Nov 2016 00:51:55 +0200 Subject: [Freeswitch-users] FreeSWITCH Registrar TLS offload In-Reply-To: References: Message-ID: Brian, I'm wondering too. First of all thing about my previous mail is not so good. I forgot that I've configured my sofia profile to work with TLS. When I disabled TLS I still have a problem with originating calls with error: [ERR] sofia_glue.c:943 TLS not supported by profile FreeSWITCH still originates calls over TLS. Contact: "" What about random source port. As I have told already on the kamailio side I check source ip and port of dispatcher destination (FS_IP:5060) and make appropriate actions. But originated call from kamailio did not pass this check. When I have looked in kamailio logs I saw that INVITE request is going from FS_IP:RANDOM_PORT Method: URI: SourceIP/Port: :<36378> From/To: [ ] Contact: <> . Here we can see that call was originated over TLS and source port was different than 5061. Here is part of sofia profile: 2016-11-29 0:37 GMT+02:00 Brian West : > You're using TLS/TCP the random port is how it happens. > > /b > > > On Mon, Nov 28, 2016 at 4:31 PM, Vladyslav Zakhozhai < > v.zakhozhai at gmail.com> wrote: > >> Hi, I'm from ser-userlist with a good news and testing results :) >> >> FreeSWITCH do honor path header and will back responses and will >> originate calls to/through SIP proxy IP address if it is in the path. >> >> Before relaying in Kamailio you need put add_path or add_path_received >> (both worked fine for me). FreeSWITCH will add it to Contact header: >> >> Contact: "" > ransport=tls;fs_path=sip%3Akamailio_ip%3Blr> >> >> No manual manipulations on Contact header is needed from kamailio side >> (as well as from FreeSWITCH side). >> >> But be aware of correct handling SIP requests (i.e. INVITEs) from >> FreeSWITCHes. For example my FreeSWITCHes backends are in dispatcher table >> (sip:IP_ADDR:UDP_PORT). And I've checked it with ds_is_from_list in >> kamailio. But FreeSWITCH originates INVITE to kamailio from >> IP_ADDR:RANDOM_PORT. In this case ds_is_from_list fails :( >> >> Now I'm checking is there mistakes in my configs or this is normal >> usecase for FreeSWITCH (I did not mention it earlier). >> >> >> 2016-11-25 13:15 GMT+02:00 Vladyslav Zakhozhai : >> >>> David, >>> >>> yes of course I'll be back with solution here :) But I'm not sure when >>> exactly. >>> >>> 2016-11-24 12:30 GMT+02:00 David Villasmil < >>> david.villasmil.work at gmail.com>: >>> >>>> Hello, >>>> >>>> Please come back with the solution when you have it. It should be >>>> interesting for people using kamailio/freeswitch. >>>> >>>> Regards, >>>> >>>> David >>>> >>>> On Wed, Nov 23, 2016 at 10:37 AM Vladyslav Zakhozhai < >>>> v.zakhozhai at gmail.com> wrote: >>>> >>>>> Alexandru, thank you for the answer. I think you've given me right >>>>> direction to investigate. >>>>> >>>>> As you've mentioned this is really kamailio issue/question. So I'm >>>>> moving to sr-users list. >>>>> >>>>> >>>>> 2016-11-22 13:03 GMT+02:00 Alexandru Covalschi <568691 at gmail.com>: >>>>> >>>>> Do you have set_contact_alias or add_contact_alias in Kamailio? >>>>> Anyways you're doing something wrong as AFAIK Kamailio translates contact >>>>> header to udp automatically. You should try to post on sr-users list. >>>>> >>>>> 2016-11-22 12:33 GMT+02:00 Vladyslav Zakhozhai >>>>> : >>>>> >>>>> Hi, >>>>> >>>>> I'm trying to understand what is the best or suitable approach to the >>>>> following use case. Let me simplify thing a little bit. >>>>> >>>>> Suppose we have one FreeSWITCH registrar behind SIP proxy (kamailio). >>>>> I'd like to offload SSL/TLS encryption/decryption to SIP proxy: >>>>> >>>>> REGISTER: >>>>> >>>>> Request: UAC == SIP/TLS ==> Kamailio == UDP ==> FreeSWITCH:50 >>>>> Reply: UAC <== SIP/TLS == Kamailio <== UDP == FreeSWITCH >>>>> >>>>> INVITE: >>>>> UAC1 == SIP/TLS ==> Kamailio == UDP == > FreeSWITCH == UDP ==> >>>>> Kamailio == SIP/TLS ==> UAC2 >>>>> >>>>> (FreeSWITCH uses kamailio as outbound proxy with fs_path tag appended >>>>> in dialplan). >>>>> >>>>> The main problem is in Contact header which contains transport=tls and >>>>> we can see it in FreeSWITCH console: >>>>> >>>>> User: user at domain.com >>>>> Contact: "" >>>>> Status: Registered(TLS)(unknown) EXP(2016-11-22 10:16:59) >>>>> EXPSECS(108) >>>>> IP: SIP_PROXY_IP >>>>> Port: 5060 >>>>> >>>>> When FreeSWITCH sends INVITE to UAC2 (during call) it tries to >>>>> establish TLS session to UAC2. It fails because there is no TLS-enabled >>>>> sofia profiles in the config of FreeSWITCH. >>>>> >>>>> I have only one solution in my mind: rewrite transport tag in Contact >>>>> header on SIP proxy (transport=udp to FreeSWITCH, and transport=tls to UAC). >>>>> >>>>> I'd like to know it this solution ok or there is more elegant >>>>> solutions. >>>>> >>>>> I've tried appending tag transport=udp in FreeSWITCH's dialplan but no >>>>> success. >>>>> >>>>> 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-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>>> switch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>>> >>>>> -- >>>>> Alexandru Covalschi >>>>> VoIP engineer and system administrator >>>>> tel: +37367398493 >>>>> >>>>> >>>>> ____________________________________________________________ >>>>> _____________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>>> switch-users >>>>> 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/free >>>>> switch-users >>>>> 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/free >>>> switch-users >>>> 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 (50% Discount using code FreeSwitch50) > http://www.freeswitchcookbook.com (50% Discount using code FreeSwitch50) > https://www.gofundme.com/freeswitch_ubuntu > > Got Bugs? Report them here ! | Reddit: > /r/freeswitch > > *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/20161129/d788cb0f/attachment-0001.html From anthony.minessale at gmail.com Tue Nov 29 01:53:55 2016 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 28 Nov 2016 16:53:55 -0600 Subject: [Freeswitch-users] absolute_codec_string not working In-Reply-To: <9EEB3B0B-F454-458B-88E6-38611227751A@jerris.com> References: <40268DB6-3518-4C8E-A022-D67FB585E2D9@jerris.com> <09ed01d24539$9e24ccf0$da6e66d0$@freeswitch.org> <9EEB3B0B-F454-458B-88E6-38611227751A@jerris.com> Message-ID: I answered the question about the escaped comma 6 days ago. That is the answer to why doesn't absolute_codec_string work... On Mon, Nov 28, 2016 at 10:09 AM, Michael Jerris wrote: > FreeSWITCH knows about g729 just fine. you shouldn?t be using proxy. > Check out the page i referenced below for information on how to accomplish > it, Proxy media mode is not the way you want to use for sure. > > > > On Nov 23, 2016, at 11:35 PM, L?i ??ng wrote: > > actually, my FS needs to be in proxy_media mode, since it always deal with > codecs it doesn't know about, g729. > real case: my caller(asterisk) always compose INVITE with > `G729,PCMA,PCMU,GSM` to FS, some of my callee only accept G729, while > others accept G729,PCMA, and so on ... > I want to limit codecs choice for each callee accordingly, instead of > fully pass `G729,PCMA,PCMU,GSM` to every callee, so that they don't know my > full supported codec. > > rgds > > Loi Dang Thanh > Phone : 84.1224.735.448 > Email : loi.dangthanh at gmail.com > > On Thu, Nov 24, 2016 at 12:33 AM, Michael Jerris wrote: > >> I?m not totally sure what you are trying to accomplish but proxy_media is >> completely unnecessary and undesired for what you are doing. It should >> ONLY be used in the case where you are trying to pass codecs we don?t know >> about at all. Take a look at the codec negotiation page on >> freeswitch.org/confluence and I think you will find your answers. I >> don?t think you need a custom mod looking for events with what you have >> described so far. >> >> On Nov 23, 2016, at 5:37 AM, L?i ??ng wrote: >> >> In FS document of media proxy mode: >> > FreeSWITCH has no control or even understanding of other SDP >> parameters. >> Look like I have to find another way, like writing a custom module >> listening on specific event. >> Any suggest? >> >> Thanks to all of you. >> rgds, >> >> Loi Dang Thanh >> Phone : 841224.735.448 >> Email : loi.dangthanh at gmail.com >> >> On Wed, Nov 23, 2016 at 10:28 AM, Ken Rice wrote: >> >>> If you are limiting the calls to specific codecs and avoiding >>> transcoding, proxy media doesn?t really reduce the overhead anymore? that >>> changed a few years ago but the notion its better still hangs on today >>> >>> >>> >>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *L?i ??ng >>> *Sent:* Tuesday, November 22, 2016 9:07 PM >>> *To:* FreeSWITCH Users Help >>> *Subject:* Re: [Freeswitch-users] absolute_codec_string not working >>> >>> >>> >>> Hi @Michael, you were right, I'm intentionally using media_proxy for FS, >>> since I want to reduce CPU usage on FS machine. >>> >>> In this case, I just want to limit the codecs used for each endpoint, >>> and codec negotiation will be handled by them. >>> >>> e.g: caller use PCMA, PCMU, GSM by its own in INVITE, I want to limit >>> the callee to only use PCMA,GSM. >>> >>> Look like `absolute_codec_string` is not what I'm looking for right? Any >>> way out? >>> >>> >>> Loi Dang Thanh >>> >>> Phone : 01224.735.448 >>> >>> Email : loi.dangthanh at gmail.com >>> >>> >>> >>> On Tue, Nov 22, 2016 at 9:57 PM, Michael Jerris wrote: >>> >>> using proxy_media is my best guess but can?t tell with this little info. >>> >>> >>> >>> On Nov 22, 2016, at 5:27 AM, L?i ??ng wrote: >>> >>> >>> >>> >>> >>> Hi List, I got some trouble with using `absolute_codec_string` param. >>> >>> My call scenario is pretty simple: caller <--> FS <--> callee. >>> >>> My caller compose `m=audio 7078 RTP/AVP 8 0 101` in its INVITE, and I'm >>> doing `` in the dialplan. >>> >>> But FS still use `m=audio 22952 RTP/AVP 8 0 101` in the INVITE to the >>> callee. >>> >>> not sure what I'm missing, helps would be appreciated. >>> >>> Note that when I'm using `originate` application in fs_cli, things are >>> good. >>> >>> `originate {absolute_codec_string=PCMU}sofia/gateway/caller/100 >>> &bridge({absolute_codec_string=PCMA}sofia/gateway/callee/100`. >>> >>> I have FS with proper behavior in transcoding, caller has `m=audio 31184 >>> RTP/AVP 0 101` received, and callee has `m=audio 21922 RTP/AVP 8 101` >>> received. >>> >>> rgds, >>> >>> Loi Dang Thanh >>> >>> Phone : 84.1224.735.448 >>> >>> Email : loi.dangthanh 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 > -- 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 https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161128/895f64ab/attachment.html From v.zakhozhai at gmail.com Tue Nov 29 01:54:22 2016 From: v.zakhozhai at gmail.com (Vladyslav Zakhozhai) Date: Tue, 29 Nov 2016 00:54:22 +0200 Subject: [Freeswitch-users] FreeSWITCH Registrar TLS offload In-Reply-To: References: Message-ID: P.S. In kamailio's dispatcher the freeswitch destination is as follows sip:FS_IP:5060 2016-11-29 0:51 GMT+02:00 Vladyslav Zakhozhai : > Brian, I'm wondering too. > > First of all thing about my previous mail is not so good. I forgot that > I've configured my sofia profile to work with TLS. When I disabled TLS I > still have a problem with originating calls with error: > > [ERR] sofia_glue.c:943 TLS not supported by profile > > FreeSWITCH still originates calls over TLS. > > Contact: "" 3Asip_proxy_ip%3Blr> > > What about random source port. > > As I have told already on the kamailio side I check source ip and port of > dispatcher destination (FS_IP:5060) and make appropriate actions. But > originated call from kamailio did not pass this check. When I have looked > in kamailio logs I saw that INVITE request is going from FS_IP:RANDOM_PORT > > Method: URI: > SourceIP/Port: :<36378> From/To: [ > ] Contact: < :5061;transport=tls>> . > > Here we can see that call was originated over TLS and source port was > different than 5061. > > Here is part of sofia profile: > > > > > > > > > > > > > > > > 2016-11-29 0:37 GMT+02:00 Brian West : > >> You're using TLS/TCP the random port is how it happens. >> >> /b >> >> >> On Mon, Nov 28, 2016 at 4:31 PM, Vladyslav Zakhozhai < >> v.zakhozhai at gmail.com> wrote: >> >>> Hi, I'm from ser-userlist with a good news and testing results :) >>> >>> FreeSWITCH do honor path header and will back responses and will >>> originate calls to/through SIP proxy IP address if it is in the path. >>> >>> Before relaying in Kamailio you need put add_path or add_path_received >>> (both worked fine for me). FreeSWITCH will add it to Contact header: >>> >>> Contact: "" >> ransport=tls;fs_path=sip%3Akamailio_ip%3Blr> >>> >>> No manual manipulations on Contact header is needed from kamailio side >>> (as well as from FreeSWITCH side). >>> >>> But be aware of correct handling SIP requests (i.e. INVITEs) from >>> FreeSWITCHes. For example my FreeSWITCHes backends are in dispatcher table >>> (sip:IP_ADDR:UDP_PORT). And I've checked it with ds_is_from_list in >>> kamailio. But FreeSWITCH originates INVITE to kamailio from >>> IP_ADDR:RANDOM_PORT. In this case ds_is_from_list fails :( >>> >>> Now I'm checking is there mistakes in my configs or this is normal >>> usecase for FreeSWITCH (I did not mention it earlier). >>> >>> >>> 2016-11-25 13:15 GMT+02:00 Vladyslav Zakhozhai : >>> >>>> David, >>>> >>>> yes of course I'll be back with solution here :) But I'm not sure when >>>> exactly. >>>> >>>> 2016-11-24 12:30 GMT+02:00 David Villasmil < >>>> david.villasmil.work at gmail.com>: >>>> >>>>> Hello, >>>>> >>>>> Please come back with the solution when you have it. It should be >>>>> interesting for people using kamailio/freeswitch. >>>>> >>>>> Regards, >>>>> >>>>> David >>>>> >>>>> On Wed, Nov 23, 2016 at 10:37 AM Vladyslav Zakhozhai < >>>>> v.zakhozhai at gmail.com> wrote: >>>>> >>>>>> Alexandru, thank you for the answer. I think you've given me right >>>>>> direction to investigate. >>>>>> >>>>>> As you've mentioned this is really kamailio issue/question. So I'm >>>>>> moving to sr-users list. >>>>>> >>>>>> >>>>>> 2016-11-22 13:03 GMT+02:00 Alexandru Covalschi <568691 at gmail.com>: >>>>>> >>>>>> Do you have set_contact_alias or add_contact_alias in Kamailio? >>>>>> Anyways you're doing something wrong as AFAIK Kamailio translates contact >>>>>> header to udp automatically. You should try to post on sr-users list. >>>>>> >>>>>> 2016-11-22 12:33 GMT+02:00 Vladyslav Zakhozhai >>>>> >: >>>>>> >>>>>> Hi, >>>>>> >>>>>> I'm trying to understand what is the best or suitable approach to the >>>>>> following use case. Let me simplify thing a little bit. >>>>>> >>>>>> Suppose we have one FreeSWITCH registrar behind SIP proxy (kamailio). >>>>>> I'd like to offload SSL/TLS encryption/decryption to SIP proxy: >>>>>> >>>>>> REGISTER: >>>>>> >>>>>> Request: UAC == SIP/TLS ==> Kamailio == UDP ==> FreeSWITCH:50 >>>>>> Reply: UAC <== SIP/TLS == Kamailio <== UDP == FreeSWITCH >>>>>> >>>>>> INVITE: >>>>>> UAC1 == SIP/TLS ==> Kamailio == UDP == > FreeSWITCH == UDP ==> >>>>>> Kamailio == SIP/TLS ==> UAC2 >>>>>> >>>>>> (FreeSWITCH uses kamailio as outbound proxy with fs_path tag appended >>>>>> in dialplan). >>>>>> >>>>>> The main problem is in Contact header which contains transport=tls >>>>>> and we can see it in FreeSWITCH console: >>>>>> >>>>>> User: user at domain.com >>>>>> Contact: "" >>>>>> Status: Registered(TLS)(unknown) EXP(2016-11-22 10:16:59) >>>>>> EXPSECS(108) >>>>>> IP: SIP_PROXY_IP >>>>>> Port: 5060 >>>>>> >>>>>> When FreeSWITCH sends INVITE to UAC2 (during call) it tries to >>>>>> establish TLS session to UAC2. It fails because there is no TLS-enabled >>>>>> sofia profiles in the config of FreeSWITCH. >>>>>> >>>>>> I have only one solution in my mind: rewrite transport tag in Contact >>>>>> header on SIP proxy (transport=udp to FreeSWITCH, and transport=tls to UAC). >>>>>> >>>>>> I'd like to know it this solution ok or there is more elegant >>>>>> solutions. >>>>>> >>>>>> I've tried appending tag transport=udp in FreeSWITCH's dialplan but >>>>>> no success. >>>>>> >>>>>> 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-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>>>> switch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> Alexandru Covalschi >>>>>> VoIP engineer and system administrator >>>>>> tel: +37367398493 >>>>>> >>>>>> >>>>>> ____________________________________________________________ >>>>>> _____________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://confluence.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>>>> switch-users >>>>>> 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/free >>>>>> switch-users >>>>>> 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/free >>>>> switch-users >>>>> 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 (50% Discount using code FreeSwitch50) >> http://www.freeswitchcookbook.com (50% Discount using code FreeSwitch50) >> https://www.gofundme.com/freeswitch_ubuntu >> >> Got Bugs? Report them here ! | Reddit: >> /r/freeswitch >> >> *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/20161129/c8b9956a/attachment-0001.html From loi.dangthanh at gmail.com Tue Nov 29 05:57:48 2016 From: loi.dangthanh at gmail.com (=?UTF-8?B?TOG7o2kgxJDhurduZw==?=) Date: Tue, 29 Nov 2016 09:57:48 +0700 Subject: [Freeswitch-users] absolute_codec_string not working In-Reply-To: References: <40268DB6-3518-4C8E-A022-D67FB585E2D9@jerris.com> <09ed01d24539$9e24ccf0$da6e66d0$@freeswitch.org> <9EEB3B0B-F454-458B-88E6-38611227751A@jerris.com> Message-ID: @Anthony, in inbound-proxy-media="true" in sip profile conf, I don't think setting absolute_codec_string can handle codec list anymore. I tried and it doesn't work. @Michael, referred to your link, I've done what I'm trying to accomplish, by setting switch_r_sdp variable, so many thanks. Anyway, I'm still not sure why I shouldn't be using media proxy? since I have another point in my voice stack for codec negotiation? rgds, Loi Dang Thanh Phone : +84.1224.735.448 Email : loi.dangthanh at gmail.com On Tue, Nov 29, 2016 at 5:53 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > I answered the question about the escaped comma 6 days ago. That is the > answer to why doesn't absolute_codec_string work... > > > On Mon, Nov 28, 2016 at 10:09 AM, Michael Jerris wrote: > >> FreeSWITCH knows about g729 just fine. you shouldn?t be using proxy. >> Check out the page i referenced below for information on how to accomplish >> it, Proxy media mode is not the way you want to use for sure. >> >> >> >> On Nov 23, 2016, at 11:35 PM, L?i ??ng wrote: >> >> actually, my FS needs to be in proxy_media mode, since it always deal >> with codecs it doesn't know about, g729. >> real case: my caller(asterisk) always compose INVITE with >> `G729,PCMA,PCMU,GSM` to FS, some of my callee only accept G729, while >> others accept G729,PCMA, and so on ... >> I want to limit codecs choice for each callee accordingly, instead of >> fully pass `G729,PCMA,PCMU,GSM` to every callee, so that they don't know my >> full supported codec. >> >> rgds >> >> Loi Dang Thanh >> Phone : 84.1224.735.448 >> Email : loi.dangthanh at gmail.com >> >> On Thu, Nov 24, 2016 at 12:33 AM, Michael Jerris wrote: >> >>> I?m not totally sure what you are trying to accomplish but proxy_media >>> is completely unnecessary and undesired for what you are doing. It should >>> ONLY be used in the case where you are trying to pass codecs we don?t know >>> about at all. Take a look at the codec negotiation page on >>> freeswitch.org/confluence and I think you will find your answers. I >>> don?t think you need a custom mod looking for events with what you have >>> described so far. >>> >>> On Nov 23, 2016, at 5:37 AM, L?i ??ng wrote: >>> >>> In FS document of media proxy mode: >>> > FreeSWITCH has no control or even understanding of other SDP >>> parameters. >>> Look like I have to find another way, like writing a custom module >>> listening on specific event. >>> Any suggest? >>> >>> Thanks to all of you. >>> rgds, >>> >>> Loi Dang Thanh >>> Phone : 841224.735.448 >>> Email : loi.dangthanh at gmail.com >>> >>> On Wed, Nov 23, 2016 at 10:28 AM, Ken Rice wrote: >>> >>>> If you are limiting the calls to specific codecs and avoiding >>>> transcoding, proxy media doesn?t really reduce the overhead anymore? that >>>> changed a few years ago but the notion its better still hangs on today >>>> >>>> >>>> >>>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >>>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *L?i ??ng >>>> *Sent:* Tuesday, November 22, 2016 9:07 PM >>>> *To:* FreeSWITCH Users Help >>>> *Subject:* Re: [Freeswitch-users] absolute_codec_string not working >>>> >>>> >>>> >>>> Hi @Michael, you were right, I'm intentionally using media_proxy for >>>> FS, since I want to reduce CPU usage on FS machine. >>>> >>>> In this case, I just want to limit the codecs used for each endpoint, >>>> and codec negotiation will be handled by them. >>>> >>>> e.g: caller use PCMA, PCMU, GSM by its own in INVITE, I want to limit >>>> the callee to only use PCMA,GSM. >>>> >>>> Look like `absolute_codec_string` is not what I'm looking for right? >>>> Any way out? >>>> >>>> >>>> Loi Dang Thanh >>>> >>>> Phone : 01224.735.448 >>>> >>>> Email : loi.dangthanh at gmail.com >>>> >>>> >>>> >>>> On Tue, Nov 22, 2016 at 9:57 PM, Michael Jerris >>>> wrote: >>>> >>>> using proxy_media is my best guess but can?t tell with this little info. >>>> >>>> >>>> >>>> On Nov 22, 2016, at 5:27 AM, L?i ??ng wrote: >>>> >>>> >>>> >>>> >>>> >>>> Hi List, I got some trouble with using `absolute_codec_string` param. >>>> >>>> My call scenario is pretty simple: caller <--> FS <--> callee. >>>> >>>> My caller compose `m=audio 7078 RTP/AVP 8 0 101` in its INVITE, and I'm >>>> doing `` in the dialplan. >>>> >>>> But FS still use `m=audio 22952 RTP/AVP 8 0 101` in the INVITE to the >>>> callee. >>>> >>>> not sure what I'm missing, helps would be appreciated. >>>> >>>> Note that when I'm using `originate` application in fs_cli, things are >>>> good. >>>> >>>> `originate {absolute_codec_string=PCMU}sofia/gateway/caller/100 >>>> &bridge({absolute_codec_string=PCMA}sofia/gateway/callee/100`. >>>> >>>> I have FS with proper behavior in transcoding, caller has `m=audio >>>> 31184 RTP/AVP 0 101` received, and callee has `m=audio 21922 RTP/AVP 8 101` >>>> received. >>>> >>>> rgds, >>>> >>>> Loi Dang Thanh >>>> >>>> Phone : 84.1224.735.448 >>>> >>>> Email : loi.dangthanh 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 >> > > > > -- > 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 > > https://www.youtube.com/watch?v=9XXgW34t40s > https://www.youtube.com/watch?v=NLaDpGQuZDA > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161129/a28949f8/attachment.html From bilaln018 at gmail.com Tue Nov 29 10:42:28 2016 From: bilaln018 at gmail.com (Bilal Abbasi) Date: Tue, 29 Nov 2016 12:42:28 +0500 Subject: [Freeswitch-users] [Unable to get VAD event TALK] In-Reply-To: References: Message-ID: Can anybody help me out on this please. anyone on user list? Regards Abbasi On Mon, Nov 28, 2016 at 12:41 PM, Bilal Abbasi wrote: > Just for users info i have activated following event. > > self.eventsocket.registerEvent("TALK", True, self.channelAnswer) > > But no luck. > > Regards > Abbasi > > > On Sun, Nov 27, 2016 at 1:08 PM, Bilal Abbasi wrote: > >> Hi users, >> >> I am currently playing with freeswitch to detect TALK event, i have the >> following set at profile level, >> >> Plus i am exporting fire_talk_events=true while originate as well. >> Using python pyswitch >> self.eventsocket.apiOriginate("{execute_on_answer='transfer 05123456789 >> XML default',originate_timeout='120',origination_caller_id_name= >> '123456789',origination_caller_id_number='123456789'}sofia/ >> external/"+NUMBER+"@X.X.X.X",cidname="godson", cidnum="123" >> ,channelvars={'rtp_enable_vad_in':'True','rtp_enable_vad_out >> ':'True','origination_uuid':uid,'DTMF':UUID,'fire_talk_eve >> nts':'true','fire_not_talk_events':'true'}) >> >> I am getting this on fs_cli >> 2016-11-27 09:49:45.160380 [DEBUG] switch_rtp.c:7962 Activate VAD codec >> G722 20ms >> 2016-11-27 09:49:45.160380 [DEBUG] switch_core_media.c:6754 AUDIO RTP >> Engage VAD for sofia/external/05123456789 at X.X.X.X ( in out ) >> >> I am getting all other registered events(CHANNEL_ANSWER, CHANNEL_DESTROY >> etc), but i am not able to get TALK event. >> >> Can someone please help me out to get the talk event. >> >> Regards >> Abbasi >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161129/3d9b186a/attachment-0001.html From roman at dissauer.net Tue Nov 29 11:26:24 2016 From: roman at dissauer.net (Roman Dissauer) Date: Tue, 29 Nov 2016 09:26:24 +0100 Subject: [Freeswitch-users] log X-AUTH-IP instead of source IP Message-ID: Hi, I?m using Kamailio as a transparent SIP Proxy in front of my Freeswitch which is configured to use Digest Auth for the clients. I already set up X-AUTH-IP headers on Kamailio and apply-proxy-acl in internal FS profile. To use fail2ban I need to get X-AUTH-IP information instead of source IP in the log messages: --> "SIP auth challenge (INVITE|REGISTER) on sofia profile 'internal' for [xxxxxx] from ip x.x.x.x" (proxy IP) Is there a way to get the original source IP into the SIP auth challenge log message? Thank you for your help!! Best Regards, Roman From abalashov at evaristesys.com Tue Nov 29 11:28:53 2016 From: abalashov at evaristesys.com (Alex Balashov) Date: Tue, 29 Nov 2016 03:28:53 -0500 Subject: [Freeswitch-users] log X-AUTH-IP instead of source IP In-Reply-To: References: Message-ID: <20161129082853.GA2139@saurus> Hi Roman, It sounds like you should just attach the original source IP as a custom SIP header on the Kamailio side: append_hf("X-Orig-SRC-IP: $si\r\n"); You can then recover that in FreeSWITCH with ${sip_h_X-Orig-SRC-IP}. -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/ From roman at dissauer.net Tue Nov 29 11:37:26 2016 From: roman at dissauer.net (Roman Dissauer) Date: Tue, 29 Nov 2016 09:37:26 +0100 Subject: [Freeswitch-users] log X-AUTH-IP instead of source IP In-Reply-To: <20161129082853.GA2139@saurus> References: <20161129082853.GA2139@saurus> Message-ID: <4B787B29-13CC-498D-AB59-B5C2F7636747@dissauer.net> Hi Alex, I already get the original source IP into FS but I need to log unsuccessful auth attempts with original IP to block them with fail2ban. The default log message ?SIP auth challenge?? does only log the Proxy IP Roman > Am 29.11.2016 um 09:28 schrieb Alex Balashov : > > Hi Roman, > > It sounds like you should just attach the original source IP as a custom SIP header on the Kamailio side: > > append_hf("X-Orig-SRC-IP: $si\r\n"); > > You can then recover that in FreeSWITCH with ${sip_h_X-Orig-SRC-IP}. > > -- Alex > > -- > Alex Balashov | Principal | Evariste Systems LLC > > Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free) > Web: http://www.evaristesys.com/, http://www.csrpswitch.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 mail at paulzillmann.de Tue Nov 29 11:41:24 2016 From: mail at paulzillmann.de (Paul Zillmann) Date: Tue, 29 Nov 2016 09:41:24 +0100 Subject: [Freeswitch-users] all-reg-options-ping don't work In-Reply-To: References: Message-ID: <17eb9eea-d6a5-8d5d-991e-0b2159707bcc@paulzillmann.de> Hello Vladyslav, ah I see. That's the option I've searched for. Thanks. Param ping works now great for me. Am 28.11.2016 um 17:42 schrieb Vladyslav Zakhozhai: > According to FreeSWITCH'es wiki > (https://wiki.freeswitch.org/wiki/Sofia.conf.xml#all-reg-options-ping) > this all-register-options-ping related to registered end points not > gateways to which FreeSWITCH is registered. > > > 2016-11-28 18:38 GMT+02:00 Vladyslav Zakhozhai >: > > Hi Paul, > > Do you want to enable it globally? > > I'm pretty sure that you can do it on per gateway basis. I.e. > gateway parameter ping: > > > > 2016-11-24 20:02 GMT+02:00 Paul Zillmann >: > > Hello there, > > I've set the options all-reg-options-ping and > registration-thread-frequency on multiple gateway profile, but > they > don't seem to work. > siptrace on those profiles don't show any activity. > > I've put them in the settings part of the profiles. > What do I miss? How do I enable options ping on gateways (for > example > every 20 seconds)? > > 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 > > > > > -- > ? ?????????, > ????????? ??????? > > > > > -- > ? ?????????, > ????????? ??????? > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161129/457ed1c6/attachment.html From abalashov at evaristesys.com Tue Nov 29 11:50:10 2016 From: abalashov at evaristesys.com (Alex Balashov) Date: Tue, 29 Nov 2016 03:50:10 -0500 Subject: [Freeswitch-users] log X-AUTH-IP instead of source IP In-Reply-To: <4B787B29-13CC-498D-AB59-B5C2F7636747@dissauer.net> References: <20161129082853.GA2139@saurus> <4B787B29-13CC-498D-AB59-B5C2F7636747@dissauer.net> Message-ID: <20161129085010.GC2139@saurus> On Tue, Nov 29, 2016 at 09:37:26AM +0100, Roman Dissauer wrote: > I already get the original source IP into FS but I need to log unsuccessful auth attempts with original IP to block them with fail2ban. > The default log message ?SIP auth challenge?? does only log the Proxy IP Ah, I see. I must have misunderstood your question, then. It sounds like what you actually want to know is how to issue an arbitrary log statement? I think this page answers that question: https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_log Or did I miss something still? -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/ From v.zakhozhai at gmail.com Tue Nov 29 12:22:25 2016 From: v.zakhozhai at gmail.com (Vladyslav Zakhozhai) Date: Tue, 29 Nov 2016 11:22:25 +0200 Subject: [Freeswitch-users] log X-AUTH-IP instead of source IP In-Reply-To: <20161129085010.GC2139@saurus> References: <20161129082853.GA2139@saurus> <4B787B29-13CC-498D-AB59-B5C2F7636747@dissauer.net> <20161129085010.GC2139@saurus> Message-ID: Hi Roman, I think that more elegant solution for your task is cut off bruteforce on Kamailio side rather than on FreeSWITCH. You do not need (and must not) pass malicious traffic to backends. It is best practice. Is every inbound requests (REGISTER, INVITE) passes Kamailio? If Kamailio work as stateful SIP proxy then you can pay attention to unsuccessful authentication attempts in reply routes and manages it, for example, with pike or something like that. Or event configure fail2ban on Kamailio server rather than on FreeSWITCH. This is just my opinion. 2016-11-29 10:50 GMT+02:00 Alex Balashov : > On Tue, Nov 29, 2016 at 09:37:26AM +0100, Roman Dissauer wrote: > > > I already get the original source IP into FS but I need to log > unsuccessful auth attempts with original IP to block them with fail2ban. > > The default log message ?SIP auth challenge?? does only log the Proxy IP > > Ah, I see. > > I must have misunderstood your question, then. It sounds like what you > actually want to know is how to issue an arbitrary log statement? > > I think this page answers that question: > > https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_log > > Or did I miss something still? > > -- Alex > > -- > Alex Balashov | Principal | Evariste Systems LLC > > Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free) > Web: http://www.evaristesys.com/, http://www.csrpswitch.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/20161129/d17e41ce/attachment-0001.html From shaun.stokes at itec-support.co.uk Tue Nov 29 12:32:42 2016 From: shaun.stokes at itec-support.co.uk (Shaun Stokes) Date: Tue, 29 Nov 2016 09:32:42 +0000 Subject: [Freeswitch-users] Best Way To Chat / Presence / File Xfer In-Reply-To: <003401d24744$d60621c0$82126540$@successos.com> References: <003501d2466c$c11a6330$434f2990$@successos.com> <6FD2F8B5BB72834E9939AEDF9FB802A901E86132D4@mbx-01.sysconfig.co.uk> <003401d24744$d60621c0$82126540$@successos.com> Message-ID: <6FD2F8B5BB72834E9939AEDF9FB802A901E86148E2@mbx-01.sysconfig.co.uk> I?ve just come across this, it?s a fork of the Rocket Chat platform which integrates with FreeSWITCH using Verto: https://github.com/VoiSmart/Rocket.Chat/tree/verto_phone According to this discussion it looks like they haven?t had as much interest as they?d like and so are no longer looking at developing this as a generic solution: https://github.com/RocketChat/Rocket.Chat/issues/3488 I?d like to try and get some encouragement to keep this project going as this has a lot of potential, it would be great if they merged the fork required for FreeSWITCH Verto integration with the current Rocket Chat project. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Adrian Thompson Sent: 25 November 2016 17:54 To: 'FreeSWITCH Users Help' Subject: Re: [Freeswitch-users] Best Way To Chat / Presence / File Xfer Thanks for the advice ? I ended up installing openfire and jitsi (desktop) and Bria (Mobile). Seems to be the most painless in the short term. Both clients accept SIP and XMPP with video, chat and presence. The only drawback is that Bria doesn?t seem to support the video bridging or XMPP file transfers. In the long run, I am going to play with verto and see what I come up with. Cheers Thanks! From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Vladyslav Zakhozhai Sent: November 25, 2016 9:49 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Best Way To Chat / Presence / File Xfer Many things depend on client capabilities. I.e. jitsi may act as SIP cleint and XMPP client. Linphone can act as SIP client (including SIP SIMPLE), etc. In the first example we can setup separate XMPP server and configure client app. But there is some draft - https://tools.ietf.org/html/draft-ietf-mmusic-file-transfer-mech-11 (sorry for maybe obsoleted doc, I beleive there is more fresh standard). Kamailio supports MSRP for example. And I think that it can be solution of this issue. From the other side Kamailio can act as XMPP gateway also. In this use case there is SIP proxy and XMPP server still separate. But one entry point for client app. Sorry for mess of thoughts. I'm investigating this topic too. 2016-11-25 16:54 GMT+02:00 Shaun Stokes >: This is something we?ve also been looking into. I believe the FreeSWITCH Verto module can support these features over WebRTC and it looks very promising but you would need to develop the client app or web page to facilitate this, you?re best bet would be to chat with FreeSWITCH Solutions. We have some fairly tight deadlines so we?re planning to re-visit Verto at a later date, for the time being we intend to use the Jitsi softphone client (javascript based) with a combination of SIP (FreeSWITCH) and XMPP you might be able to package this into a phone app. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Adrian Thompson Sent: 24 November 2016 16:07 To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Best Way To Chat / Presence / File Xfer Hello, Been a user and fan of Freeswitch for two years now ? thank you for the great software! I?ve done days of research and would like to reach out to see if anyone else has some ideas here: We are looking into giving all of our techs a softphone for their mobile device so they can communicate back to base and to each other while out in the field. Our desired features are as follows: - Video (for troubleshooting equipment with other remote techs) - Chat (for quick updates) - Group Chat (for group reminders) - Page (reach out to a busy individual with an important message) - Group Page (reach out to a busy group of workers with an important message) - File Transfer (send each other marketing material and tech manuals on the fly) - Presence (so employees can let everyone else know if they are available) Video ? Check, Working Chat ? Check, SIMPLE, Working Group Chat ? Not an option with SIMPLE?, Not working Page ? Check, Working Group Page ? Check, Working File Transfer ? Not an option with SIMPLE?, Not working Presence ? I have BLF working great, but what about extended presence such as ?busy? or ?I?m a leprechaun? Do I have to start using XMPP for these things? It could complicate the setup three-fold if I have to go that route and I would rather not?. But if I have to, is there documents supporting a configuration that includes Freeswitch for voice/video and XMPP for everything else? Also, does anyone have a suggestion for a mobile softphone with video, chat, file xfer and can be provisioned server side? (I?ve tried the Grandstream Wave, but it cannot use an HTTPS server with authentication for some reason so I?m looking for more options) Thanks in advance for advice! Adrian ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ [http://www.itec-support.co.uk/wp-content/uploads/2016/07/email_logo.jpg] Shaun Stokes - Infrastructure Analyst T : 01453 700713 E : shaun.stokes at itec-support.co.uk W : www.itec-support.co.uk Registered Address :- ITEC Support, Suite 2 Prospect House, Bath Road, Stroud, Gloucestershire GL5 3QF Company No. 06908001 CONFIDENTIALITY NOTICE This communication and the information it contains are intended for the person or organisation to which it is addressed. Its contents are confidential and may be protected in law. Unauthorised use, copying or disclosure of any of it may be unlawful. If you are not the intended recipient, please contact us immediately. The contents of any attachments in this e-mail may contain software viruses, which could damage your own computer system. While ITEC Support has taken every reasonable precaution to minimise this risk, we cannot accept liability for any damage which you sustain as a result of software viruses. You should carry out your own virus checking procedure before opening any attachment. ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- ? ?????????, ????????? ??????? ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ [http://www.itec-support.co.uk/wp-content/uploads/2016/07/email_logo.jpg] Shaun Stokes - Infrastructure Analyst T : 01453 700713 E : shaun.stokes at itec-support.co.uk W : www.itec-support.co.uk Registered Address :- ITEC Support, Suite 2 Prospect House, Bath Road, Stroud, Gloucestershire GL5 3QF Company No. 06908001 CONFIDENTIALITY NOTICE This communication and the information it contains are intended for the person or organisation to which it is addressed. Its contents are confidential and may be protected in law. Unauthorised use, copying or disclosure of any of it may be unlawful. If you are not the intended recipient, please contact us immediately. The contents of any attachments in this e-mail may contain software viruses, which could damage your own computer system. While ITEC Support has taken every reasonable precaution to minimise this risk, we cannot accept liability for any damage which you sustain as a result of software viruses. You should carry out your own virus checking procedure before opening any attachment. ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161129/c43cf79c/attachment-0001.html From abalashov at evaristesys.com Tue Nov 29 12:40:59 2016 From: abalashov at evaristesys.com (Alex Balashov) Date: Tue, 29 Nov 2016 04:40:59 -0500 Subject: [Freeswitch-users] log X-AUTH-IP instead of source IP In-Reply-To: References: <20161129082853.GA2139@saurus> <4B787B29-13CC-498D-AB59-B5C2F7636747@dissauer.net> <20161129085010.GC2139@saurus> Message-ID: <20161129094059.GD2139@saurus> On Tue, Nov 29, 2016 at 11:22:25AM +0200, Vladyslav Zakhozhai wrote: > I think that more elegant solution for your task is cut off bruteforce on > Kamailio side rather than on FreeSWITCH. You do not need (and must not) > pass malicious traffic to backends. It is best practice. I would agree with that. Kamailio makes a far better "condom" than Freeswitch. I just assumed there was something in his use-case that compelled relaying traffic forward uncritically. -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/ From v.zakhozhai at gmail.com Tue Nov 29 13:43:40 2016 From: v.zakhozhai at gmail.com (Vladyslav Zakhozhai) Date: Tue, 29 Nov 2016 12:43:40 +0200 Subject: [Freeswitch-users] FreeSWITCH Registrar TLS offload In-Reply-To: References: Message-ID: Hi, Here is SIP REGISTER message which goes UAC => Kamailio => FreeSWITCH: REGISTER sip:DOMAIN_NAME SIP/2.0 Via: SIP/2.0/UDP KAMAILIO_IP;branch=z9hG4bK95f8. b6cff139a89c58ea38df4e2f8d375039.0;i=9 Via: SIP/2.0/TLS USER_IP:34913;received=USER_IP;alias;branch=z9hG4bK.KAL7~ HJ2E;rport=34913 From: ;tag=EbEqf28Bb To: sip:USER_NAME at DOMAIN_NAME CSeq: 22 REGISTER Call-ID: QHttR-2N4V Max-Forwards: 69 Supported: outbound Accept: application/sdp Accept: text/plain Accept: application/vnd.gsma.rcs-ft-http+xml Contact: ;+sip.instance=" " Expires: 60 User-Agent: Linphone/3.9.0 (belle-sip/1.4.2) Content-Length: 0 Path: Looks good. Isn't it? Call origination from FreeSWITCH => Kamailio => UAC INVITE sip:TO_USER at TO_USER_IP:56408;transport=tls SIP/2.0 Via: SIP/2.0/TLS FS_IP;branch=z9hG4bKS4Dr1pBa4NB1K Route: ;lr;received=sip:TO_USER_IP:56408;transport=tls Max-Forwards: 68 From: "vlakas" ;tag=91r5XtyZa62Bj To: Call-ID: 7a17700d-30ae-1235-8bbb-005056b9778d CSeq: 99867524 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.6.12-20-b91a0a6~64bit 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: 246 X-FS-Support: update_display,send_info Remote-Party-ID: "TO_USER" ;party= calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1480390787 1480390788 IN IP4 FS_IP s=FreeSWITCH c=IN IP4 FS_IP t=0 0 m=audio 16390 RTP/AVP 8 101 13 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 This is looks good too I guess... I can't understand why FreeSWITCH tries to originate call over TLS. What did I miss? 2016-11-29 0:54 GMT+02:00 Vladyslav Zakhozhai : > P.S. In kamailio's dispatcher the freeswitch destination is as follows > > sip:FS_IP:5060 > > 2016-11-29 0:51 GMT+02:00 Vladyslav Zakhozhai : > >> Brian, I'm wondering too. >> >> First of all thing about my previous mail is not so good. I forgot that >> I've configured my sofia profile to work with TLS. When I disabled TLS I >> still have a problem with originating calls with error: >> >> [ERR] sofia_glue.c:943 TLS not supported by profile >> >> FreeSWITCH still originates calls over TLS. >> >> Contact: "" > ransport=tls;fs_path=sip%3Asip_proxy_ip%3Blr> >> >> What about random source port. >> >> As I have told already on the kamailio side I check source ip and port of >> dispatcher destination (FS_IP:5060) and make appropriate actions. But >> originated call from kamailio did not pass this check. When I have looked >> in kamailio logs I saw that INVITE request is going from FS_IP:RANDOM_PORT >> >> Method: URI: >> SourceIP/Port: :<36378> From/To: [ >> ] Contact: <> :5061;transport=tls>> . >> >> Here we can see that call was originated over TLS and source port was >> different than 5061. >> >> Here is part of sofia profile: >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> 2016-11-29 0:37 GMT+02:00 Brian West : >> >>> You're using TLS/TCP the random port is how it happens. >>> >>> /b >>> >>> >>> On Mon, Nov 28, 2016 at 4:31 PM, Vladyslav Zakhozhai < >>> v.zakhozhai at gmail.com> wrote: >>> >>>> Hi, I'm from ser-userlist with a good news and testing results :) >>>> >>>> FreeSWITCH do honor path header and will back responses and will >>>> originate calls to/through SIP proxy IP address if it is in the path. >>>> >>>> Before relaying in Kamailio you need put add_path or add_path_received >>>> (both worked fine for me). FreeSWITCH will add it to Contact header: >>>> >>>> Contact: "" >>> ransport=tls;fs_path=sip%3Akamailio_ip%3Blr> >>>> >>>> No manual manipulations on Contact header is needed from kamailio side >>>> (as well as from FreeSWITCH side). >>>> >>>> But be aware of correct handling SIP requests (i.e. INVITEs) from >>>> FreeSWITCHes. For example my FreeSWITCHes backends are in dispatcher table >>>> (sip:IP_ADDR:UDP_PORT). And I've checked it with ds_is_from_list in >>>> kamailio. But FreeSWITCH originates INVITE to kamailio from >>>> IP_ADDR:RANDOM_PORT. In this case ds_is_from_list fails :( >>>> >>>> Now I'm checking is there mistakes in my configs or this is normal >>>> usecase for FreeSWITCH (I did not mention it earlier). >>>> >>>> >>>> 2016-11-25 13:15 GMT+02:00 Vladyslav Zakhozhai : >>>> >>>>> David, >>>>> >>>>> yes of course I'll be back with solution here :) But I'm not sure when >>>>> exactly. >>>>> >>>>> 2016-11-24 12:30 GMT+02:00 David Villasmil < >>>>> david.villasmil.work at gmail.com>: >>>>> >>>>>> Hello, >>>>>> >>>>>> Please come back with the solution when you have it. It should be >>>>>> interesting for people using kamailio/freeswitch. >>>>>> >>>>>> Regards, >>>>>> >>>>>> David >>>>>> >>>>>> On Wed, Nov 23, 2016 at 10:37 AM Vladyslav Zakhozhai < >>>>>> v.zakhozhai at gmail.com> wrote: >>>>>> >>>>>>> Alexandru, thank you for the answer. I think you've given me right >>>>>>> direction to investigate. >>>>>>> >>>>>>> As you've mentioned this is really kamailio issue/question. So I'm >>>>>>> moving to sr-users list. >>>>>>> >>>>>>> >>>>>>> 2016-11-22 13:03 GMT+02:00 Alexandru Covalschi <568691 at gmail.com>: >>>>>>> >>>>>>> Do you have set_contact_alias or add_contact_alias in Kamailio? >>>>>>> Anyways you're doing something wrong as AFAIK Kamailio translates contact >>>>>>> header to udp automatically. You should try to post on sr-users list. >>>>>>> >>>>>>> 2016-11-22 12:33 GMT+02:00 Vladyslav Zakhozhai < >>>>>>> v.zakhozhai at gmail.com>: >>>>>>> >>>>>>> Hi, >>>>>>> >>>>>>> I'm trying to understand what is the best or suitable approach to >>>>>>> the following use case. Let me simplify thing a little bit. >>>>>>> >>>>>>> Suppose we have one FreeSWITCH registrar behind SIP proxy >>>>>>> (kamailio). I'd like to offload SSL/TLS encryption/decryption to SIP proxy: >>>>>>> >>>>>>> REGISTER: >>>>>>> >>>>>>> Request: UAC == SIP/TLS ==> Kamailio == UDP ==> FreeSWITCH:50 >>>>>>> Reply: UAC <== SIP/TLS == Kamailio <== UDP == FreeSWITCH >>>>>>> >>>>>>> INVITE: >>>>>>> UAC1 == SIP/TLS ==> Kamailio == UDP == > FreeSWITCH == UDP ==> >>>>>>> Kamailio == SIP/TLS ==> UAC2 >>>>>>> >>>>>>> (FreeSWITCH uses kamailio as outbound proxy with fs_path tag >>>>>>> appended in dialplan). >>>>>>> >>>>>>> The main problem is in Contact header which contains transport=tls >>>>>>> and we can see it in FreeSWITCH console: >>>>>>> >>>>>>> User: user at domain.com >>>>>>> Contact: "" >>>>>>> Status: Registered(TLS)(unknown) EXP(2016-11-22 10:16:59) >>>>>>> EXPSECS(108) >>>>>>> IP: SIP_PROXY_IP >>>>>>> Port: 5060 >>>>>>> >>>>>>> When FreeSWITCH sends INVITE to UAC2 (during call) it tries to >>>>>>> establish TLS session to UAC2. It fails because there is no TLS-enabled >>>>>>> sofia profiles in the config of FreeSWITCH. >>>>>>> >>>>>>> I have only one solution in my mind: rewrite transport tag in >>>>>>> Contact header on SIP proxy (transport=udp to FreeSWITCH, and transport=tls >>>>>>> to UAC). >>>>>>> >>>>>>> I'd like to know it this solution ok or there is more elegant >>>>>>> solutions. >>>>>>> >>>>>>> I've tried appending tag transport=udp in FreeSWITCH's dialplan but >>>>>>> no success. >>>>>>> >>>>>>> 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-users mailing list >>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>>>>> switch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> -- >>>>>>> Alexandru Covalschi >>>>>>> VoIP engineer and system administrator >>>>>>> tel: +37367398493 >>>>>>> >>>>>>> >>>>>>> ____________________________________________________________ >>>>>>> _____________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://confluence.freeswitch.org >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>>>>> switch-users >>>>>>> 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/free >>>>>>> switch-users >>>>>>> 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/free >>>>>> switch-users >>>>>> 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/free >>>> switch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> >>> -- >>> >>> *Brian West* >>> brian at freeswitch.org >>> >>> >>> *Twitter: @FreeSWITCH , @briankwest* >>> http://www.freeswitchbook.com (50% Discount using code FreeSwitch50) >>> http://www.freeswitchcookbook.com (50% Discount using code FreeSwitch50) >>> https://www.gofundme.com/freeswitch_ubuntu >>> >>> Got Bugs? Report them here ! | Reddit: >>> /r/freeswitch >>> >>> *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/20161129/4689f064/attachment-0001.html From lists at telefaks.de Tue Nov 29 14:32:30 2016 From: lists at telefaks.de (Peter Steinbach) Date: Tue, 29 Nov 2016 12:32:30 +0100 Subject: [Freeswitch-users] Force one way audio? Message-ID: <583D674E.2010009@telefaks.de> Hello, we want to setup a system where a larger numer of callers (500-1000) are listening to MOH. In order to reduce bandwidth, I would like to force Freeswitch to negociate only one way audio (outbound audio only, MOH). Is there a way to do this in Freeswitch, maybe? -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de From roman at dissauer.net Tue Nov 29 16:08:47 2016 From: roman at dissauer.net (Roman Dissauer) Date: Tue, 29 Nov 2016 14:08:47 +0100 Subject: [Freeswitch-users] log X-AUTH-IP instead of source IP In-Reply-To: <20161129094059.GD2139@saurus> References: <20161129082853.GA2139@saurus> <4B787B29-13CC-498D-AB59-B5C2F7636747@dissauer.net> <20161129085010.GC2139@saurus> <20161129094059.GD2139@saurus> Message-ID: <66A76739-0E66-4956-B819-7847E7E1E4AC@dissauer.net> Thanks guys, I?m sure that it is best practice to prevent brute force attacks on Kamailio. I also do that on another system where Kamailio handles full registration/authentication. Due to the fact that my SIP Proxy is as basic as possible forwarding all packets to the backend Freeswitch (just for load balancing), I thought it would be easier to solve that on Freeswitch side. I?ll try to get that done in the reply route and will post the results here. Roman > Am 29.11.2016 um 10:40 schrieb Alex Balashov : > > On Tue, Nov 29, 2016 at 11:22:25AM +0200, Vladyslav Zakhozhai wrote: > >> I think that more elegant solution for your task is cut off bruteforce on >> Kamailio side rather than on FreeSWITCH. You do not need (and must not) >> pass malicious traffic to backends. It is best practice. > > I would agree with that. Kamailio makes a far better "condom" than Freeswitch. > > I just assumed there was something in his use-case that compelled relaying traffic forward uncritically. > > -- > Alex Balashov | Principal | Evariste Systems LLC > > Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free) > Web: http://www.evaristesys.com/, http://www.csrpswitch.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 v.zakhozhai at gmail.com Tue Nov 29 17:08:13 2016 From: v.zakhozhai at gmail.com (Vladyslav Zakhozhai) Date: Tue, 29 Nov 2016 16:08:13 +0200 Subject: [Freeswitch-users] log X-AUTH-IP instead of source IP In-Reply-To: <66A76739-0E66-4956-B819-7847E7E1E4AC@dissauer.net> References: <20161129082853.GA2139@saurus> <4B787B29-13CC-498D-AB59-B5C2F7636747@dissauer.net> <20161129085010.GC2139@saurus> <20161129094059.GD2139@saurus> <66A76739-0E66-4956-B819-7847E7E1E4AC@dissauer.net> Message-ID: Roman, My point of view is the following. Fail2ban bans requests based on logs output. Right? You can keep kamailio's config as simple as possible but add logging of failed auth attempts. I.e. reply_route[FROM_BACKEND] { if(status == 403) { xlog("L_WARN", "Frobidden from $si:$sp blah...\n"); } } And then you just need make appropriate filter. This is just my opinion. In the case of FreeSWITCH I do not know is it possible to show original IP address due failed auth attempts. sorry. 2016-11-29 15:08 GMT+02:00 Roman Dissauer : > Thanks guys, > > I?m sure that it is best practice to prevent brute force attacks on > Kamailio. I also do that on another system where Kamailio handles full > registration/authentication. Due to the fact that my SIP Proxy is as basic > as possible forwarding all packets to the backend Freeswitch (just for load > balancing), I thought it would be easier to solve that on Freeswitch side. > > I?ll try to get that done in the reply route and will post the results > here. > > Roman > > > > > Am 29.11.2016 um 10:40 schrieb Alex Balashov >: > > > > On Tue, Nov 29, 2016 at 11:22:25AM +0200, Vladyslav Zakhozhai wrote: > > > >> I think that more elegant solution for your task is cut off bruteforce > on > >> Kamailio side rather than on FreeSWITCH. You do not need (and must not) > >> pass malicious traffic to backends. It is best practice. > > > > I would agree with that. Kamailio makes a far better "condom" than > Freeswitch. > > > > I just assumed there was something in his use-case that compelled > relaying traffic forward uncritically. > > > > -- > > Alex Balashov | Principal | Evariste Systems LLC > > > > Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free) > > Web: http://www.evaristesys.com/, http://www.csrpswitch.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/20161129/470df390/attachment.html From aqsyounas at gmail.com Tue Nov 29 17:43:08 2016 From: aqsyounas at gmail.com (Aqs Younas) Date: Tue, 29 Nov 2016 19:43:08 +0500 Subject: [Freeswitch-users] freeswitch logs A-leg and B-leg json_cdr in different directories Message-ID: Hi, I am using mod_json_cdr to log files in a directory. I am setting *json_cdr_base* in my javascript to dynamically set log directory. I have observed freeswitch is writing only A-leg files in directory set by *json_cdr_base *and whereas B-leg files always goes to *log-dir *set in json_cdr.conf.xml If I do not set *json_cdr_base in *script both A-leg and B-leg goes to the same directory set by *log-dir *in json_cdr.conf.xml I need to set directory dynamically for json_cdr that is why I am using *json_cdr_base*. I tried to use log-dir like this but freeswitch is not evaluating the expression. Instead, it is treating this as a string. param name="log-dir" value="/tmp/${strftime(%Y-%m-%d)}"/> I need to set directory dyonically having both A-leg and B-leg json_cdrs. Any suggestion is much appreciated. Best Regards. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161129/a183de8a/attachment.html From brian at freeswitch.org Tue Nov 29 17:47:47 2016 From: brian at freeswitch.org (Brian West) Date: Tue, 29 Nov 2016 14:47:47 +0000 Subject: [Freeswitch-users] freeswitch logs A-leg and B-leg json_cdr in different directories In-Reply-To: References: Message-ID: It would have to be patched to allow that. It's a simple patch really, file a JIRA and I'll look at it. /b On Tue, Nov 29, 2016 at 8:44 AM Aqs Younas wrote: > Hi, > > I am using mod_json_cdr to log files in a directory. I am setting > *json_cdr_base* in my javascript to dynamically set log directory. I have > observed freeswitch is writing only A-leg files in directory set by *json_cdr_base > *and whereas B-leg files always goes to *log-dir *set in json_cdr.conf.xml > > If I do not set *json_cdr_base in *script both A-leg and B-leg goes to > the same directory set by *log-dir *in json_cdr.conf.xml > > I need to set directory dynamically for json_cdr that is why I am using > *json_cdr_base*. > > I tried to use log-dir like this but freeswitch is not evaluating the > expression. Instead, it is treating this as a string. > > param name="log-dir" value="/tmp/${strftime(%Y-%m-%d)}"/> > > > I need to set directory dyonically having both A-leg and B-leg json_cdrs. > > Any suggestion is much appreciated. > > Best 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/20161129/8c1eafe2/attachment-0001.html From mike at jerris.com Tue Nov 29 18:02:00 2016 From: mike at jerris.com (Michael Jerris) Date: Tue, 29 Nov 2016 10:02:00 -0500 Subject: [Freeswitch-users] absolute_codec_string not working In-Reply-To: References: <40268DB6-3518-4C8E-A022-D67FB585E2D9@jerris.com> <09ed01d24539$9e24ccf0$da6e66d0$@freeswitch.org> <9EEB3B0B-F454-458B-88E6-38611227751A@jerris.com> Message-ID: <95E35AD0-3FD0-4350-BB16-5548FB2B7604@jerris.com> This is EASY to do without setting proxy mode? You are doing things the very hard way for zero benefit. Yet another reason you don?t want to use proxy. > On Nov 28, 2016, at 9:57 PM, L?i ??ng wrote: > > @Anthony, in inbound-proxy-media="true" in sip profile conf, I don't think setting absolute_codec_string can handle codec list anymore. I tried and it doesn't work. > @Michael, referred to your link, I've done what I'm trying to accomplish, by setting switch_r_sdp variable, so many thanks. > Anyway, I'm still not sure why I shouldn't be using media proxy? since I have another point in my voice stack for codec negotiation? > > rgds, > > Loi Dang Thanh > Phone : +84.1224.735.448 > Email : loi.dangthanh at gmail.com > > On Tue, Nov 29, 2016 at 5:53 AM, Anthony Minessale > wrote: > I answered the question about the escaped comma 6 days ago. That is the answer to why doesn't absolute_codec_string work... > > > On Mon, Nov 28, 2016 at 10:09 AM, Michael Jerris > wrote: > FreeSWITCH knows about g729 just fine. you shouldn?t be using proxy. Check out the page i referenced below for information on how to accomplish it, Proxy media mode is not the way you want to use for sure. > > > >> On Nov 23, 2016, at 11:35 PM, L?i ??ng > wrote: >> >> actually, my FS needs to be in proxy_media mode, since it always deal with codecs it doesn't know about, g729. >> real case: my caller(asterisk) always compose INVITE with `G729,PCMA,PCMU,GSM` to FS, some of my callee only accept G729, while others accept G729,PCMA, and so on ... >> I want to limit codecs choice for each callee accordingly, instead of fully pass `G729,PCMA,PCMU,GSM` to every callee, so that they don't know my full supported codec. >> >> rgds >> >> Loi Dang Thanh >> Phone : 84.1224.735.448 >> Email : loi.dangthanh at gmail.com >> >> On Thu, Nov 24, 2016 at 12:33 AM, Michael Jerris > wrote: >> I?m not totally sure what you are trying to accomplish but proxy_media is completely unnecessary and undesired for what you are doing. It should ONLY be used in the case where you are trying to pass codecs we don?t know about at all. Take a look at the codec negotiation page on freeswitch.org/confluence and I think you will find your answers. I don?t think you need a custom mod looking for events with what you have described so far. >> >>> On Nov 23, 2016, at 5:37 AM, L?i ??ng > wrote: >>> >>> In FS document of media proxy mode: >>> > FreeSWITCH has no control or even understanding of other SDP parameters. >>> Look like I have to find another way, like writing a custom module listening on specific event. >>> Any suggest? >>> >>> Thanks to all of you. >>> rgds, >>> >>> Loi Dang Thanh >>> Phone : 841224.735.448 >>> Email : loi.dangthanh at gmail.com >>> >>> On Wed, Nov 23, 2016 at 10:28 AM, Ken Rice > wrote: >>> If you are limiting the calls to specific codecs and avoiding transcoding, proxy media doesn?t really reduce the overhead anymore? that changed a few years ago but the notion its better still hangs on today >>> >>> >>> >>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org ] On Behalf Of L?i ??ng >>> Sent: Tuesday, November 22, 2016 9:07 PM >>> To: FreeSWITCH Users Help > >>> Subject: Re: [Freeswitch-users] absolute_codec_string not working >>> >>> >>> >>> Hi @Michael, you were right, I'm intentionally using media_proxy for FS, since I want to reduce CPU usage on FS machine. >>> >>> In this case, I just want to limit the codecs used for each endpoint, and codec negotiation will be handled by them. >>> >>> e.g: caller use PCMA, PCMU, GSM by its own in INVITE, I want to limit the callee to only use PCMA,GSM. >>> >>> Look like `absolute_codec_string` is not what I'm looking for right? Any way out? >>> >>> >>> >>> Loi Dang Thanh >>> >>> Phone : 01224.735.448 >>> >>> Email : loi.dangthanh at gmail.com >>> >>> >>> On Tue, Nov 22, 2016 at 9:57 PM, Michael Jerris > wrote: >>> >>> using proxy_media is my best guess but can?t tell with this little info. >>> >>> >>> >>> On Nov 22, 2016, at 5:27 AM, L?i ??ng > wrote: >>> >>> >>> >>> >>> >>> Hi List, I got some trouble with using `absolute_codec_string` param. >>> >>> My call scenario is pretty simple: caller <--> FS <--> callee. >>> >>> My caller compose `m=audio 7078 RTP/AVP 8 0 101` in its INVITE, and I'm doing `` in the dialplan. >>> >>> But FS still use `m=audio 22952 RTP/AVP 8 0 101` in the INVITE to the callee. >>> >>> not sure what I'm missing, helps would be appreciated. >>> >>> Note that when I'm using `originate` application in fs_cli, things are good. >>> >>> `originate {absolute_codec_string=PCMU}sofia/gateway/caller/100 &bridge({absolute_codec_string=PCMA}sofia/gateway/callee/100`. >>> >>> I have FS with proper behavior in transcoding, caller has `m=audio 31184 RTP/AVP 0 101` received, and callee has `m=audio 21922 RTP/AVP 8 101` received. >>> >>> rgds, >>> >>> Loi Dang Thanh >>> >>> Phone : 84.1224.735.448 >>> >>> Email : loi.dangthanh 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 > > > > -- > 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 > > https://www.youtube.com/watch?v=9XXgW34t40s > https://www.youtube.com/watch?v=NLaDpGQuZDA > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > 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/20161129/e62ac4da/attachment-0001.html From evoip at ukr.net Tue Nov 29 18:03:08 2016 From: evoip at ukr.net (Eugene) Date: Tue, 29 Nov 2016 17:03:08 +0200 Subject: [Freeswitch-users] Problem with CALL PICKUP (intercept) in QUEUES (mod_callcenter) In-Reply-To: <3aa89065-7ecf-4ee3-2cf0-00f30beea831@ukr.net> References: <7a9f66b1-cca1-9d01-369c-a2c6d26ceb16@ukr.net> <2a6bef5e-9238-dc62-fc43-59063e37d4fb@ukr.net> <3aa89065-7ecf-4ee3-2cf0-00f30beea831@ukr.net> Message-ID: Hi ! is that possible to increase the BUG priority ? That's very important for our system. Regards Ewgeny. 17.11.16 15:40, Ewgeny ?????: > > the Bug report created - https://freeswitch.org/jira/browse/FS-9752 > > Regards Ewgeny. > > > 16.11.2016 22:56, ?talo Rossi ?????: >> Bug reports go to JIRA. >> >> Don't forget to put debug logs (/log 7 siptrace etc) and your >> configurations. >> >> On Wed, Nov 16, 2016 at 9:02 AM, Ewgeny > > wrote: >> >> P.S. >> >> see attached file with call-flows. >> >> >> >> >> Thanks in advance for any help. >> >> Regards Ewgeny. >> >> >> >> 16.11.2016 13:56, Ewgeny ?????: >>> >>> Some more information about the issue with group pickup. >>> >>> Why intercept doesn't work with Queues ? >>> >>> >>> 15.11.2016 17:47, Ewgeny ?????: >>>> >>>> Hi ! >>>> >>>> First about terminology: *Call Pickup*the ability to pull a >>>> ringing call to the phone you are currently on. >>>> >>>> Call Pickup = Call Intercept = Call group pickup. >>>> >>>> >>>> We're using FreeSWITCH version: 1.6.8~64bit in our complex >>>> telephony system with Kamailio and other SIP services. >>>> >>>> The problem with call pickup (intercept) when using a Queue >>>> (mod_callcenter). >>>> >>>> The group pickup scheme we are using described here: >>>> http://www.tech-invite.com/fo-sip/tinv-fo-sip-service-16.html >>>> . >>>> >>>> We do SUBSCRIBE (3 see link) to some our service - that return >>>> information about the call (call legs) in NOTIFY with XML body (5). >>>> >>>> Then we do an INVITE with REPLACES (7) that actually do the >>>> Intercept. >>>> >>>> This works on normal calls, but doesn't work with Queues. >>>> >>>> In the queue a few agents simultaneously calling, and when >>>> INVITE w Replace headers comes it didn't intercept the call. >>>> >>>> The question is: how to implement Call Group Pickup (Call >>>> Intercept) with mod_callcenter ? >>>> >>>> If it necessary i can add call sip traces for more details. >>>> >>>> Thanks in advance for any help. >>>> >>>> >>>> Regards Ewgeny. >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> >>>> 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 >> >> -- >> ?talo Rossi >> italo at freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161129/c3270d3a/attachment.html From mike at jerris.com Tue Nov 29 18:10:06 2016 From: mike at jerris.com (Michael Jerris) Date: Tue, 29 Nov 2016 10:10:06 -0500 Subject: [Freeswitch-users] Problem with CALL PICKUP (intercept) in QUEUES (mod_callcenter) In-Reply-To: References: <7a9f66b1-cca1-9d01-369c-a2c6d26ceb16@ukr.net> <2a6bef5e-9238-dc62-fc43-59063e37d4fb@ukr.net> <3aa89065-7ecf-4ee3-2cf0-00f30beea831@ukr.net> Message-ID: <9A6EC4A7-B661-48A8-BC77-C8EA69294E63@jerris.com> If you would like to engage commercial services on this, feel free to contact consulting at freeswitch.org . > On Nov 29, 2016, at 10:03 AM, Eugene wrote: > > Hi ! > > is that possible to increase the BUG priority ? > > That's very important for our system. > > > Regards Ewgeny. > > > 17.11.16 15:40, Ewgeny ?????: >> the Bug report created - https://freeswitch.org/jira/browse/FS-9752 Regards Ewgeny. >> >> >> 16.11.2016 22:56, ?talo Rossi ?????: >>> Bug reports go to JIRA. >>> >>> Don't forget to put debug logs (/log 7 siptrace etc) and your configurations. >>> >>> On Wed, Nov 16, 2016 at 9:02 AM, Ewgeny > wrote: >>> P.S. >>> >>> see attached file with call-flows. >>> >>> >>> >>> >>> Thanks in advance for any help. >>> Regards Ewgeny. >>> >>> >>> 16.11.2016 13:56, Ewgeny ?????: >>>> Some more information about the issue with group pickup. >>>> >>>> Why intercept doesn't work with Queues ? >>>> >>>> >>>> 15.11.2016 17:47, Ewgeny ?????: >>>>> Hi ! >>>>> >>>>> First about terminology: Call Pickup the ability to pull a ringing call to the phone you are currently on. >>>>> >>>>> Call Pickup = Call Intercept = Call group pickup. >>>>> >>>>> >>>>> We're using FreeSWITCH version: 1.6.8~64bit in our complex telephony system with Kamailio and other SIP services. >>>>> >>>>> The problem with call pickup (intercept) when using a Queue (mod_callcenter). >>>>> >>>>> The group pickup scheme we are using described here: http://www.tech-invite.com/fo-sip/tinv-fo-sip-service-16.html . >>>>> >>>>> We do SUBSCRIBE (3 see link) to some our service - that return information about the call (call legs) in NOTIFY with XML body (5). >>>>> >>>>> Then we do an INVITE with REPLACES (7) that actually do the Intercept. >>>>> >>>>> This works on normal calls, but doesn't work with Queues. >>>>> In the queue a few agents simultaneously calling, and when INVITE w Replace headers comes it didn't intercept the call. >>>>> >>>>> The question is: how to implement Call Group Pickup (Call Intercept) with mod_callcenter ? >>>>> >>>>> If it necessary i can add call sip traces for more details. >>>>> Thanks in advance for any help. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161129/50bf32d8/attachment-0001.html From aqsyounas at gmail.com Tue Nov 29 18:43:44 2016 From: aqsyounas at gmail.com (Aqs Younas) Date: Tue, 29 Nov 2016 20:43:44 +0500 Subject: [Freeswitch-users] freeswitch logs A-leg and B-leg json_cdr in different directories In-Reply-To: References: Message-ID: Filed https://freeswitch.org/jira/browse/FS-9781 Thank You. On 29 November 2016 at 19:47, Brian West wrote: > It would have to be patched to allow that. It's a simple patch really, > file a JIRA and I'll look at it. > > /b > > On Tue, Nov 29, 2016 at 8:44 AM Aqs Younas wrote: > >> Hi, >> >> I am using mod_json_cdr to log files in a directory. I am setting >> *json_cdr_base* in my javascript to dynamically set log directory. I >> have observed freeswitch is writing only A-leg files in directory set by *json_cdr_base >> *and whereas B-leg files always goes to *log-dir *set >> in json_cdr.conf.xml >> >> If I do not set *json_cdr_base in *script both A-leg and B-leg goes to >> the same directory set by *log-dir *in json_cdr.conf.xml >> >> I need to set directory dynamically for json_cdr that is why I am using >> *json_cdr_base*. >> >> I tried to use log-dir like this but freeswitch is not evaluating the >> expression. Instead, it is treating this as a string. >> >> param name="log-dir" value="/tmp/${strftime(%Y-%m-%d)}"/> >> >> >> I need to set directory dyonically having both A-leg and B-leg json_cdrs. >> >> >> Any suggestion is much appreciated. >> >> Best 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/20161129/269d24ec/attachment.html From kamil.nigmatullin at gmail.com Tue Nov 29 19:55:49 2016 From: kamil.nigmatullin at gmail.com (Kamil Nigmatullin) Date: Tue, 29 Nov 2016 22:55:49 +0600 Subject: [Freeswitch-users] Problem with CALL PICKUP (intercept) in QUEUES (mod_callcenter) In-Reply-To: <9A6EC4A7-B661-48A8-BC77-C8EA69294E63@jerris.com> References: <7a9f66b1-cca1-9d01-369c-a2c6d26ceb16@ukr.net> <2a6bef5e-9238-dc62-fc43-59063e37d4fb@ukr.net> <3aa89065-7ecf-4ee3-2cf0-00f30beea831@ukr.net> <9A6EC4A7-B661-48A8-BC77-C8EA69294E63@jerris.com> Message-ID: You can do this by using loopback endpoint. It worked for me with queues. Although this may be not the best idea. 2016-11-29 21:10 GMT+06:00 Michael Jerris : > If you would like to engage commercial services on this, feel free to > contact consulting at freeswitch.org. > > > On Nov 29, 2016, at 10:03 AM, Eugene wrote: > > Hi ! > > is that possible to increase the BUG priority ? > > That's very important for our system. > > > Regards Ewgeny. > > 17.11.16 15:40, Ewgeny ?????: > > the Bug report created - https://freeswitch.org/jira/browse/FS-9752 > Regards Ewgeny. > > > 16.11.2016 22:56, ?talo Rossi ?????: > > Bug reports go to JIRA. > > Don't forget to put debug logs (/log 7 siptrace etc) and your > configurations. > > On Wed, Nov 16, 2016 at 9:02 AM, Ewgeny wrote: > >> P.S. >> >> see attached file with call-flows. >> >> >> >> >> Thanks in advance for any help. >> Regards Ewgeny. >> >> >> >> 16.11.2016 13:56, Ewgeny ?????: >> >> Some more information about the issue with group pickup. >> >> Why intercept doesn't work with Queues ? >> >> >> 15.11.2016 17:47, Ewgeny ?????: >> >> Hi ! >> >> First about terminology: *Call Pickup* the ability to pull a ringing >> call to the phone you are currently on. >> >> Call Pickup = Call Intercept = Call group pickup. >> >> >> We're using FreeSWITCH version: 1.6.8~64bit in our complex telephony >> system with Kamailio and other SIP services. >> >> The problem with call pickup (intercept) when using a Queue >> (mod_callcenter). >> >> The group pickup scheme we are using described here: >> http://www.tech-invite.com/fo-sip/tinv-fo-sip-service-16.html. >> >> We do SUBSCRIBE (3 see link) to some our service - that return >> information about the call (call legs) in NOTIFY with XML body (5). >> >> Then we do an INVITE with REPLACES (7) that actually do the Intercept. >> >> This works on normal calls, but doesn't work with Queues. >> >> In the queue a few agents simultaneously calling, and when INVITE w >> Replace headers comes it didn't intercept the call. >> >> The question is: how to implement Call Group Pickup (Call Intercept) with >> mod_callcenter ? >> >> If it necessary i can add call sip traces for more details. >> >> Thanks in advance 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 > -- Kamil Nigmatullin Tel: 77272323748 mob: 7 (707) 2517003 Skype: kamil.nigmatullin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161129/a6b2cf62/attachment.html From steveayre at gmail.com Wed Nov 30 00:19:18 2016 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 29 Nov 2016 21:19:18 +0000 Subject: [Freeswitch-users] log X-AUTH-IP instead of source IP In-Reply-To: <4B787B29-13CC-498D-AB59-B5C2F7636747@dissauer.net> References: <20161129082853.GA2139@saurus> <4B787B29-13CC-498D-AB59-B5C2F7636747@dissauer.net> Message-ID: Are you using apply-proxy-acl? If not see if that helps, otherwise perhaps file a Jira to request it. Regards, Steve On 29 November 2016 at 08:37, Roman Dissauer wrote: > Hi Alex, > > I already get the original source IP into FS but I need to log > unsuccessful auth attempts with original IP to block them with fail2ban. > The default log message ?SIP auth challenge?? does only log the Proxy IP > > Roman > > > > Am 29.11.2016 um 09:28 schrieb Alex Balashov >: > > > > Hi Roman, > > > > It sounds like you should just attach the original source IP as a custom > SIP header on the Kamailio side: > > > > append_hf("X-Orig-SRC-IP: $si\r\n"); > > > > You can then recover that in FreeSWITCH with ${sip_h_X-Orig-SRC-IP}. > > > > -- Alex > > > > -- > > Alex Balashov | Principal | Evariste Systems LLC > > > > Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free) > > Web: http://www.evaristesys.com/, http://www.csrpswitch.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/20161129/dcca2bf9/attachment-0001.html From s.safarov at gmail.com Wed Nov 30 06:52:24 2016 From: s.safarov at gmail.com (Sergey Safarov) Date: Wed, 30 Nov 2016 03:52:24 +0000 Subject: [Freeswitch-users] log X-AUTH-IP instead of source IP In-Reply-To: References: <20161129082853.GA2139@saurus> <4B787B29-13CC-498D-AB59-B5C2F7636747@dissauer.net> Message-ID: When you use kamailio then close sip interface port via iptables rules like "block all except kamailio IP where destination port 5060" fail2ban is not required in your case. Failed registration and failed INVITE authorization can be logged at kamailio configuration. ??, 30 ????. 2016, 0:21 Steven Ayre : > Are you using apply-proxy-acl? > > If not see if that helps, otherwise perhaps file a Jira to request it. > > Regards, > Steve > > > On 29 November 2016 at 08:37, Roman Dissauer wrote: > > Hi Alex, > > I already get the original source IP into FS but I need to log > unsuccessful auth attempts with original IP to block them with fail2ban. > The default log message ?SIP auth challenge?? does only log the Proxy IP > > Roman > > > > Am 29.11.2016 um 09:28 schrieb Alex Balashov >: > > > > Hi Roman, > > > > It sounds like you should just attach the original source IP as a custom > SIP header on the Kamailio side: > > > > append_hf("X-Orig-SRC-IP: $si\r\n"); > > > > You can then recover that in FreeSWITCH with ${sip_h_X-Orig-SRC-IP}. > > > > -- Alex > > > > -- > > Alex Balashov | Principal | Evariste Systems LLC > > > > Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free) > > Web: http://www.evaristesys.com/, http://www.csrpswitch.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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161130/f157f7be/attachment.html From aubalde at presenceco.com Wed Nov 30 18:43:13 2016 From: aubalde at presenceco.com (=?UTF-8?Q?Agust=C3=AD_Ubalde?=) Date: Wed, 30 Nov 2016 16:43:13 +0100 Subject: [Freeswitch-users] TLS configuration Message-ID: Hi all, I've an Asterisk Server with TLS enabled (wildcard certificate). I'm trying to connect from FreeSWITCH but I don't know how. I need specified CA file? Thanks, PRESENCE TECHNOLOGY An ENGHOUSE INTERACTIVE Company Agust? Ubalde Bellot Chief Developer C/ Comte Urgell 240 3?-A Barcelona 08036 aubalde at presenceco.com Ph: +34 93 10 10 322 Enghouse is listed on the Toronto Stock Exchange, stock symbol: ENGH -- *Presence Technology - DisclaimerThis message, its content and any file attached thereto is for the intended recipient only and is confidential and /or privileged. If you have received this e-mail in error or had access to it, you should note that the information in it is private and any use thereof is unauthorized. In such an event please notify us by e-mail or by telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by whatsoever means and any transmission or dissemination thereof to other persons is prohibited. It should be deleted immediately from your system. Presence Technology reserves the right to take legal action against any persons unlawfully gaining access to the content of any external message it has emitted.* *For additional information, please visit our website **www.presenceco.com * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161130/703914e7/attachment.html From brian at freeswitch.org Wed Nov 30 19:55:36 2016 From: brian at freeswitch.org (Brian West) Date: Wed, 30 Nov 2016 10:55:36 -0600 Subject: [Freeswitch-users] TLS configuration In-Reply-To: References: Message-ID: Have you read this? https://freeswitch.org/confluence/display/FREESWITCH/SIP+TLS While it appears to have some broken images that the docs team can fix when they get time. /b On Wed, Nov 30, 2016 at 9:43 AM, Agust? Ubalde wrote: > Hi all, > > I've an Asterisk Server with TLS enabled (wildcard certificate). > I'm trying to connect from FreeSWITCH but I don't know how. I need > specified CA file? > > > Thanks, > > PRESENCE TECHNOLOGY > > An ENGHOUSE INTERACTIVE Company > > Agust? Ubalde Bellot > > Chief Developer > > C/ Comte Urgell 240 3?-A > > Barcelona 08036 > > aubalde at presenceco.com > > Ph: +34 93 10 10 322 > > Enghouse is listed on the Toronto Stock Exchange, stock symbol: ENGH > > > > *Presence Technology - DisclaimerThis message, its content and any file > attached thereto is for the intended recipient only and is confidential and > /or privileged. If you have received this e-mail in error or had access to > it, you should note that the information in it is private and any use > thereof is unauthorized. In such an event please notify us by e-mail or by > telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by > whatsoever means and any transmission or dissemination thereof to other > persons is prohibited. It should be deleted immediately from your system. > Presence Technology reserves the right to take legal action against any > persons unlawfully gaining access to the content of any external message it > has emitted.* > > *For additional information, please visit our website **www.presenceco.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 (50% Discount using code FreeSwitch50) http://www.freeswitchcookbook.com (50% Discount using code FreeSwitch50) https://www.gofundme.com/freeswitch_ubuntu Got Bugs? Report them here ! | Reddit: /r/freeswitch *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/20161130/046c77fa/attachment-0001.html