From gregor at infomedia.si Sun Jul 1 14:23:40 2018 From: gregor at infomedia.si (Gregor Nanger) Date: Sun, 1 Jul 2018 16:23:40 +0200 Subject: [Freeswitch-users] =?utf-8?b?562U5aSNOiBGcmVlc3dpdGNoIHdpbmRvd3Mg?= =?utf-8?q?build_-_libpng?= In-Reply-To: References: Message-ID: I managed to build successfully with latest Vs2017 version. Latest master built without problem, but 1.6.20, I had to do some manual changes in configuration. I am planning to prepare video tutorial. Best regards, Gregor On Sun, Jul 1, 2018, 14:21 Andrew Keil wrote: > Gregor, > > > > Or you can make the following change to the zlib project for All > Configurations and All Releases - C/C++ - All Options: > > - Disable Specific Warnings (add to end): ;5045 > > - Whole Program Optimization: No > > > > Hope that helps. > > > > Andrew > > > > *From:* FreeSWITCH-users *On > Behalf Of *bob. chen > *Sent:* Tuesday, 26 June 2018 1:23 PM > *To:* FreeSWITCH Users Help > *Subject:* [Freeswitch-users] 答复: Freeswitch windows build - libpng > > > > try vs2015 ;) > > > > > > *发件人**:* FreeSWITCH-users [ > mailto:freeswitch-users-bounces at lists.freeswitch.org > ] *代表 *Gregor Nanger > *发送时间**:* 2018年6月23日 19:08 > *收件人**:* FreeSWITCH Users Help > *主题**:* [Freeswitch-users] Freeswitch windows build - libpng > > > > Hi! > > > > After a year I wanted to build FS on windows and make video tutorial or > for FS wiki. > > > > I am using latest VS2017 and everything builds except libpng project. > Below is error if someone can help me, because I don't have c++ experience: > > > > 1>------ Build started: Project: Download zlib, Configuration: Release > Win32 ------ > > 1>Downloading zlib. > > 2>------ Build started: Project: libpng, Configuration: Release x64 ------ > > 2>Generating pnglibconf.h > > 2> 1 file(s) copied. > > 2>png.c > > 2>pngerror.c > > 2>pngget.c > > 2>pngmem.c > > 2>pngpread.c > > 2>pngread.c > > 2>pngrio.c > > 2>pngrtran.c > > 2>pngrutil.c > > 2>pngset.c > > 2>pngtrans.c > > 2>pngwio.c > > 2>pngwrite.c > > 2>pngwtran.c > > 2>pngwutil.c > > 2>Generating Code... > > 2>zlib.lib(inflate.obj) : MSIL .netmodule or module compiled with /GL > found; restarting link with /LTCG; add /LTCG to the link command line to > improve linker performance > > 2> Creating library D:\Git\freeswitch\x64\Release\libpng16.lib and > object D:\Git\freeswitch\x64\Release\libpng16.exp > > 2>Generating code > > 2>d:\git\freeswitch\libs\zlib\deflate.c(2097): error C2220: warning > treated as error - no 'executable' file generated > > 2>d:\git\freeswitch\libs\zlib\deflate.c(2097): warning C5045: Compiler > will insert Spectre mitigation for memory load if /Qspectre switch specified > > 2>d:\git\freeswitch\libs\zlib\deflate.c(2097) : note: index 'dist' range > checked by comparison on this line > > 2>d:\git\freeswitch\libs\zlib\deflate.c(2097) : note: feeds memory load on > this line > > 2>d:\git\freeswitch\libs\zlib\deflate.c(1987): warning C5045: Compiler > will insert Spectre mitigation for memory load if /Qspectre switch specified > > 2>d:\git\freeswitch\libs\zlib\deflate.c(1987) : note: index 'dist' range > checked by comparison on this line > > 2>d:\git\freeswitch\libs\zlib\deflate.c(1987) : note: feeds memory load on > this line > > 2>d:\git\freeswitch\libs\zlib\deflate.c(1862): warning C5045: Compiler > will insert Spectre mitigation for memory load if /Qspectre switch specified > > 2>d:\git\freeswitch\libs\zlib\deflate.c(1862) : note: index 'dist' range > checked by comparison on this line > > 2>d:\git\freeswitch\libs\zlib\deflate.c(1862) : note: feeds memory load on > this line > > 2>LINK : fatal error LNK1257: code generation failed > > 2>Done building project "libpng.vcxproj" -- FAILED. > > > > Best regards, Gregor > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- 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: From davidswalkabout at gmail.com Sun Jul 1 15:49:34 2018 From: davidswalkabout at gmail.com (David P) Date: Sun, 1 Jul 2018 08:49:34 -0700 Subject: [Freeswitch-users] FireFox 61 + Freeswitch 1.6.20 In-Reply-To: References: Message-ID: We recently started seeing users of both webRTC and SIP, and different browsers, able to connect but they get no audio or video back. Our best guess is that this involves incomplete support for IPv6 on our end, because the same clients can connect to one of our other clouds. We haven't yet found how our IPv6 config differs from what confluence says it should be. Check if the network of the clients is providing them with both IPv4 and IPv6. Only networks that do not provide IPv6 to clients work with all our clouds. On Thu, Jun 28, 2018 at 3:02 PM Geoff Mina wrote: > >> Is anyone else having issues with WebRTC and the latest release of >> FireFox? We are showing one way media - with media flowing properly TO >> the browser, but no media coming back from the browser. >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From davidswalkabout at gmail.com Sun Jul 1 15:51:50 2018 From: davidswalkabout at gmail.com (David P) Date: Sun, 1 Jul 2018 08:51:50 -0700 Subject: [Freeswitch-users] About Verto In-Reply-To: References: <424FFF6C-73B4-4293-A4FE-8F45A24815CD@gmail.com> Message-ID: If you haven't already, look at the min width and height values for 'videoParams' in the verto video sample page. I don't know if this is sufficient. On Sun, 1 Jul 2018, 8:46 am Dom Rumsey, wrote: > Yeah, iOS won't work if you're using VP8 codec, you to connect with h264. > > Also, does anyone know how to change Verto to 16:9 aspect ratio? > > > ------------------------------ > *From:* FreeSWITCH-users > on behalf of Giovanni Maruzzelli > *Sent:* Thursday, June 28, 2018 3:00 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] About Verto > > On 28 June 2018 at 09:30, 王聡 wrote: > > > I had tried to build a WebRTC video call service via mod_verto and Verto > Comunicator. > Now I could make calls from (to) an Android device with Chrome, but it > didn’t work on iOS with Safari. > Is there any point to be modified to run on Safari? > > > be sure to enable H264 codec, both in vars.conf.xml and in verto.conf.xml > > > > > > > Regards. > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com > > > > > -- > > Sincerely, > > Giovanni Maruzzelli > OpenTelecom.IT > cell: +39 347 266 56 18 > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From chad at apartmentlines.com Sun Jul 1 19:12:53 2018 From: chad at apartmentlines.com (Chad Phillips) Date: Sun, 1 Jul 2018 12:12:53 -0700 Subject: [Freeswitch-users] About Verto In-Reply-To: References: <424FFF6C-73B4-4293-A4FE-8F45A24815CD@gmail.com> Message-ID: Yep, H264 required for iOS unfortunately. I wrote up some documentation on how to configure Verto for H264: https://evoluxbr.github.io/verto-docs/tut/adding-h264-support.html On Sun, Jul 1, 2018 at 6:52 AM Giovanni Maruzzelli wrote: > I believe Safari do support webrtc in current ios, it just do not support > vp8, h264 only. > > > On Thu, Jun 28, 2018, 23:38 Francesco Facco de Lagarda < > francesco at delagarda.com> wrote: > >> iOS safari dies not support webrtc yet.... >> Apple keeps promising but.... >> >> Francesco Facco de Lagarda >> >> >> > On 28 Jun 2018, at 09:30, 王聡 wrote: >> > >> > Hey all, >> > >> > I had tried to build a WebRTC video call service via mod_verto and >> Verto Comunicator. >> > Now I could make calls from (to) an Android device with Chrome, but it >> didn’t work on iOS with Safari. >> > Is there any point to be modified to run on Safari? >> > >> > Regards. >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Services >> > sales at freeswitch.com >> > https://freeswitch.com >> > >> > Official FreeSWITCH Sites >> > https://freeswitch.com/oss >> > https://freeswitch.org/confluence >> > https://cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > https://freeswitch.com >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Services >> sales at freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From chad at apartmentlines.com Sun Jul 1 19:17:12 2018 From: chad at apartmentlines.com (Chad Phillips) Date: Sun, 1 Jul 2018 12:17:12 -0700 Subject: [Freeswitch-users] About Verto In-Reply-To: <695562B1-39BB-4816-B7C3-2EA9D1663389@gmail.com> References: <424FFF6C-73B4-4293-A4FE-8F45A24815CD@gmail.com> <695562B1-39BB-4816-B7C3-2EA9D1663389@gmail.com> Message-ID: For interop between all Android devices and iOS devices, I’m pretty sure you need both VP8 and H264 enabled for available codecs. Chrome doesn’t support H264 on all Android devices yet, there’s an open issue to add software H264 encoding in the case where the Android hardware doesn’t have hardware encoding: https://bugs.chromium.org/p/chromium/issues/detail?id=719023 On Sun, Jul 1, 2018 at 8:28 AM 王聡 wrote: > Thanks for your reply. > > I had added H264 codec in verto.conf.xml, and now it’s able to make call > between iOS users. > But it still didn’t work between Android and iOS users, is there any > suggestions? > > Regards. > > 在 2018年6月29日,00:00,Giovanni Maruzzelli 写道: > > On 28 June 2018 at 09:30, 王聡 wrote: > >> >> I had tried to build a WebRTC video call service via mod_verto and Verto >> Comunicator. >> Now I could make calls from (to) an Android device with Chrome, but it >> didn’t work on iOS with Safari. >> Is there any point to be modified to run on Safari? >> > > be sure to enable H264 codec, both in vars.conf.xml and in verto.conf.xml > > > > > >> >> Regards. >> _________________________________________________________________________ >> Professional FreeSWITCH Services >> sales at freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > > > > -- > > Sincerely, > > Giovanni Maruzzelli > OpenTelecom.IT > cell: +39 347 266 56 18 > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com > > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From joelists at tm.net.uk Sun Jul 1 20:50:12 2018 From: joelists at tm.net.uk (Joseph Waite) Date: Sun, 1 Jul 2018 21:50:12 +0100 Subject: [Freeswitch-users] Manipulate the sip: string in the INVITE header on an outgoing Bridge In-Reply-To: References: Message-ID: <8D46C488-0098-4C7D-A601-182DB1E18AF1@tm.net.uk> Hi Alex This doesn’t seem to do anything, or maybe it is just setting the To: field which I have already got set. The problem with the To field is that asterisk doesn’t look at this field without customisation which isn’t great for customer ease of use. Regards > On 29 Jun 2018, at 02:10, Alexey Sibyakin wrote: > > Hi, > > I'm not sure that I got your scenario right but you can try to do something like this: > > (you may need another var, full list is in Confluence) > > Regards, > > Alex > > > On Wed, Jun 27, 2018 at 9:33 PM, Joseph Waite > wrote: > Hi Guys > > Were having issues with Asterisk boxes registering to a trunk on our FreeSwitch box. > > We are sending the DID number called down the trunk in the to: field, however Asterisk doesn’t seem to like this. > > Is there anyway to manipulate the INVITE to be INVITE sip:DID@ <> instead of what is currently INVITE sip:user@ <> > > Im hoping there is something like sip_invite_to_uri, however google is failing me. > > Regards > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com > > > > -- > Alex Sibyakin | Support Engineer > FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 > Email: alex at freeswitch.com > Website: https://www.FreeSWITCH.com > Need commercial support? Contact sales at freeswitch.com for details. > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From mircea.huh at gmail.com Mon Jul 2 05:05:48 2018 From: mircea.huh at gmail.com (Mircea Botoca-Huh) Date: Mon, 2 Jul 2018 08:05:48 +0300 Subject: [Freeswitch-users] Voicemail MWI multiple mailboxes notify one user In-Reply-To: References: Message-ID: Hello, I would like to know if there is any possibility to notify one user when messages are left in multiple mailboxes? I want to have just one registered user which should monitor multiple voicemail mailboxes. Thank you for your time. Best regards, Mircea -------------- next part -------------- An HTML attachment was scrubbed... URL: From gmaruzz at gmail.com Mon Jul 2 06:11:32 2018 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Mon, 2 Jul 2018 08:11:32 +0200 Subject: [Freeswitch-users] About Verto In-Reply-To: References: <424FFF6C-73B4-4293-A4FE-8F45A24815CD@gmail.com> Message-ID: be sure to load mod_av too, and use transcode or mux in conferences On 29 June 2018 at 10:53, Dom Rumsey wrote: > Yeah, iOS won't work if you're using VP8 codec, you to connect with h264. > > Also, does anyone know how to change Verto to 16:9 aspect ratio? > > > ------------------------------ > *From:* FreeSWITCH-users > on behalf of Giovanni Maruzzelli > *Sent:* Thursday, June 28, 2018 3:00 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] About Verto > > On 28 June 2018 at 09:30, 王聡 wrote: > > > I had tried to build a WebRTC video call service via mod_verto and Verto > Comunicator. > Now I could make calls from (to) an Android device with Chrome, but it > didn’t work on iOS with Safari. > Is there any point to be modified to run on Safari? > > > be sure to enable H264 codec, both in vars.conf.xml and in verto.conf.xml > > > > > > > Regards. > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com > > > > > -- > > Sincerely, > > Giovanni Maruzzelli > OpenTelecom.IT > cell: +39 347 266 56 18 > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com > -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: From harangozo.laszlo at tct.hu Mon Jul 2 07:36:37 2018 From: harangozo.laszlo at tct.hu (=?UTF-8?B?SGFyYW5nb3rDsywgTMOhc3psw7M=?=) Date: Mon, 2 Jul 2018 09:36:37 +0200 Subject: [Freeswitch-users] freeswitch internal profile hang freequently In-Reply-To: References: Message-ID: Geoff, why did you ask about xml_curl registrations? Did you have any problem with it? I am just about to switch to a setup where the user directory is managed by mod_xml_curl. Might be interesting to hear real-world experiences before I implement the whole workflow. 2018-06-26 3:50 GMT+02:00 Geoff Mina : > We had the exact same issue with MySQL and were assured it was a fault > with the ODBC driver on CentOS. > > I’m concerned to hear this is actually a FS issue and not a DB driver > issue. Concerned. Not surprised. > > Are you using xml_curl registrations by chance? > > On Mon, Jun 25, 2018 at 7:43 PM sukitha jayasinghe > wrote: > >> Dear All, >> >> I have a freeswitch (1.6.20) server in production with db configured to >> postgresql through pgpool2. This system hangs time to time and has become a >> major issue. It happen only to the internal profile where our UACs connects >> through webrtc. After the incident profile stop responding to cli commands, >> only solution is restarting the entire application. I managed to get the >> back trace which indicate system waiting forever for profile database >> mutex. Further, It seems thread #19 is waiting for result from pgsql server >> cause this issue. Please find the back trace link below, >> >> https://pastebin.com/Uqe8HjZn >> >> Has anyone experienced this issue before, what is the solution for this. >> >> Best Regards, >> Sukitha. >> _________________________________________________________________________ >> Professional FreeSWITCH Services >> sales at freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > -- > GEOFF MINA > Chief Executive Officer > Connect First / Contact Center Solutions, Built Better. > > 2545 Central Ave #200, Boulder, CO 80301 > 720.335.5924 > Connect First / Contact Center Solutions, Built Better > www.connectfirst.com > > This email and any files transmitted with it are confidential and are > intended solely for the use of the individual or entity to whom they are > addressed. If you have received this email in error, please notify the > system manager. > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com > -------------- next part -------------- An HTML attachment was scrubbed... URL: From albert.hsueh at cloudigit.com Mon Jul 2 08:34:02 2018 From: albert.hsueh at cloudigit.com (=?UTF-8?B?6Jab5YWJ5a6P?=) Date: Mon, 2 Jul 2018 16:34:02 +0800 Subject: [Freeswitch-users] video on conference with PC and MAC browser works but fail with iPhone safari via webrtc In-Reply-To: References: Message-ID: Hi, Thank you for reply. We tried this before but it still can not work. Is any parameter also to be modified with video-mode=mux? We also tried this Someone suggested this value on iOS but also failed in our case. Sincerely Albert 2018-06-29 10:19 GMT+08:00 Alexey Sibyakin : > Hi, > > try mux instead of transcode > > Regards, > > Alex > > On Tue, Jun 26, 2018 at 12:47 PM, 薛光宏 wrote: > >> Hi All, >> >> here is our simple architecture >> --- >> ----/ >> >> - client A and B are both registered on FreeSWITCH via websocket (using >> sipjs 0.9.2) >> - we enable video-mode= transcode and load mod_av on FreeSWITCH. >> - if client A uses chrome on PC and client B use safari on *MAC*, they >> can see each other, talk to each other in the same conference. >> - if client A uses chrome on PC and client B use safari on *iPhone*(8plus, >> iOS 11.4), they can NOT see each other, talk to each other in the same >> conference. >> >> I though safari on MAC works, then safari on iPhone should work, but not. >> Is it possible to fix that with changing configuration of FreeSWITCH? or >> any suggestion? >> >> Thanks in advance. >> >> FreeSWITCH Version 1.6.20 >> Ubuntu Server Ubuntu 14.04.1 LTS 64bit >> libav-11.4 or libav-12 >> >> See below configurations: >> 1. enable video-mode= transcode >> >> >> >> 2.enable VP8/H264 codec support >> >> >> >> 3.enable mod_av and disable mod_h26x >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Services >> sales at freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com >> > > > > -- > Alex Sibyakin | Support Engineer > FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 > > Email: alex at freeswitch.com > Website: https://www.FreeSWITCH.com > Need commercial support? Contact sales at freeswitch.com for details. > > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com > -------------- next part -------------- An HTML attachment was scrubbed... URL: From paul.muaddib83 at gmail.com Mon Jul 2 08:44:44 2018 From: paul.muaddib83 at gmail.com (Paul Muaddib) Date: Mon, 2 Jul 2018 10:44:44 +0200 Subject: [Freeswitch-users] Failed Registration [908] In-Reply-To: References: Message-ID: Now, I opened all firewall ports. I tried connecting to different SIP Servers with Netcat and it established a connection. But I don't get any response. I read the Technical Specification 1TR118 from Deutsche Telekom and change the gateway setting according to the specifications but no success ;( 2018-06-30 22:57 GMT+02:00 Paul Muaddib : > sofia loglevel all 9 > > ##################################################################### > > 2018-06-30 22:31:43.746032 [NOTICE] sofia_reg.c:448 Registering > sip-trunk.telekom.de > nua.c:622 nua_register() nua: nua_register: entering > nua_stack.c:529 nua_signal() nua(0xb4e2ec50): sent signal r_register > nua_stack.c:569 nua_stack_signal() nua(0xb4e2ec50): recv signal r_register > nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering > soa.c:280 soa_clone() soa_clone(static::0xb6255520, 0x88b30b0, 0xb4e2ec50) > called > soa.c:403 soa_set_params() soa_set_params(static::0xb5923410, ...) called > soa.c:403 soa_set_params() soa_set_params(static::0xb5923410, ...) called > nua_dialog.c:338 nua_dialog_usage_add() nua(0xb4e2ec50): adding register > usage > nta.c:4417 nta_leg_tcreate() nta_leg_tcreate(0xb5906508) > nta.c:2665 nta_tpn_by_url() nta: selecting scheme sip > sres_cache.c:272 sres_cache_get() sres_cache_get(0x88aa8e0, SRV, "_sip._ > tcp.reg.sip-trunk.telekom.de.") called > sres_cache.c:318 sres_cache_get() sres_cache_get(0x88aa8e0, SRV, "_sip._ > tcp.reg.sip-trunk.telekom.de.") returned 3 entries > nta.c:10598 outgoing_query_srv() nta: for "reg.sip-trunk.telekom.de" > query "_sip._tcp.reg.sip-trunk.telekom.de" SRV (cached) > nta.c:10694 outgoing_answer_srv() nta: _sip._tcp.reg.sip-trunk.telekom.de > IN SRV 10 5 5060 d-ipr-a02.sip-trunk.telekom.de. (tcp) > nta.c:10694 outgoing_answer_srv() nta: _sip._tcp.reg.sip-trunk.telekom.de > IN SRV 11 5 5060 b-ipr-a01.sip-trunk.telekom.de. (tcp) > nta.c:10694 outgoing_answer_srv() nta: _sip._tcp.reg.sip-trunk.telekom.de > IN SRV 11 5 5060 b-ipr-a02.sip-trunk.telekom.de. (tcp) > sres_cache.c:272 sres_cache_get() sres_cache_get(0x88aa8e0, A, " > d-ipr-a02.sip-trunk.telekom.de.") called > sres_cache.c:318 sres_cache_get() sres_cache_get(0x88aa8e0, A, " > d-ipr-a02.sip-trunk.telekom.de.") returned 1 entries > nta.c:10803 outgoing_query_a() nta: for "reg.sip-trunk.telekom.de" query " > d-ipr-a02.sip-trunk.telekom.de." A (cached) > nta.c:10856 outgoing_answer_a() nta: d-ipr-a02.sip-trunk.telekom.de. IN A > 217.0.26.133 > tport.c:4588 tport_by_name() tport(0xb625dc58): found 0x880d978 by name > tcp/217.0.26.133:5060 > tport.c:3257 tport_tsend() tport_tsend(0x880d978) tpn = tcp/ > 217.0.26.133:5060 > tport.c:3594 tport_vsend() tport_vsend(0x880d978): 677 bytes of 677 to tcp/ > 217.0.26.133:5060 > tport.c:3492 tport_send_msg() tport_vsend returned 677 > tport.c:2296 tport_set_secondary_timer() tport(0x880d978): reset timer > nta.c:8304 outgoing_send() nta: sent REGISTER (124859287) to tcp/ > 217.0.26.133:5060 > tport.c:4160 tport_pend() tport_pend(0x880d978): pending 0xb6263a08 for > tcp/217.0.26.133:5060 (already 1) > nta.c:8982 outgoing_timer_bf() nta: timer F fired, timeout REGISTER > (124859241) > nta.c:9035 outgoing_timeout() nta(0xb6264318): try next after timeout > nta.c:10200 outgoing_graylist() nta: graylisting > d-ipr-a02.sip-trunk.telekom.de.:5060;transport=tcp > nta.c:10227 outgoing_graylist() nta: reduced priority of 1 _sip._ > tcp.reg.sip-trunk.telekom.de SRV records (increase value to 11) > sres_cache.c:272 sres_cache_get() sres_cache_get(0x88aa8e0, A, " > reg.sip-trunk.telekom.de.") called > nta.c:10803 outgoing_query_a() nta: for "reg.sip-trunk.telekom.de" query " > reg.sip-trunk.telekom.de" A > sres.c:968 sres_query() sres_query(0xb624a8b8, 0xb6264318, A, " > reg.sip-trunk.telekom.de") called > sres.c:2730 sres_send_dns_query() sres_send_dns_query(0xb624a8b8, > 0x88653d0) called > sres.c:2819 sres_send_dns_query() sres_send_dns_query(0xb624a8b8, > 0x88653d0) id=12982 A reg.sip-trunk.telekom.de (to [217.237.148.70]:53) > nta.c:8929 _nta_outgoing_timer() nta_outgoing_timer: 0/0 resent, 1/2 tout, > 0/0 term, 0/2 free > nta.c:1296 agent_timer() nta: timer set next to 29536 ms > sres.c:3467 sres_resolver_receive() sres_resolver_receive(0xb624a8b8, 31) > called > sres.c:3781 sres_create_record() AUTHORITY RR received > sip-trunk.telekom.de. SOA IN 151 rdlen=64 > sres.c:3572 sres_log_response() sres_resolver_receive(0xb624a8b8, > 0x88653d0) id=12982 (from [217.237.148.70]:53) > sres.c:2987 sres_query_report_error() sres(q=0x88653d0): reporting error > RECORD_ERR for A reg.sip-trunk.telekom.de > tport.c:4222 tport_release() tport_release(0x880d978): 0xb623b2e8 by > 0xb6264318 with (nil) > nta.c:8722 outgoing_free() nta: outgoing_free(0xb6264318) > nta.c:8982 outgoing_timer_bf() nta: timer F fired, timeout REGISTER > (124859287) > nta.c:9035 outgoing_timeout() nta(0xb628fb30): try next after timeout > nta.c:10200 outgoing_graylist() nta: graylisting > d-ipr-a02.sip-trunk.telekom.de.:5060;transport=tcp > nta.c:10227 outgoing_graylist() nta: reduced priority of 1 _sip._ > tcp.reg.sip-trunk.telekom.de SRV records (increase value to 12) > sres_cache.c:272 sres_cache_get() sres_cache_get(0x88aa8e0, A, " > b-ipr-a01.sip-trunk.telekom.de.") called > sres_cache.c:318 sres_cache_get() sres_cache_get(0x88aa8e0, A, " > b-ipr-a01.sip-trunk.telekom.de.") returned 1 entries > nta.c:10803 outgoing_query_a() nta: for "reg.sip-trunk.telekom.de" query " > b-ipr-a01.sip-trunk.telekom.de." A (cached) > nta.c:10856 outgoing_answer_a() nta: b-ipr-a01.sip-trunk.telekom.de. IN A > 217.0.26.163 > tport.c:4588 tport_by_name() tport(0xb625dc58): found 0x87fa8a8 by name > tcp/217.0.26.163:5060 > tport.c:4222 tport_release() tport_release(0x880d978): 0xb6263a08 by > 0xb628fb30 with (nil) > tport.c:2296 tport_set_secondary_timer() tport(0x880d978): reset timer > tport.c:3257 tport_tsend() tport_tsend(0x87fa8a8) tpn = tcp/ > 217.0.26.163:5060 > tport.c:3594 tport_vsend() tport_vsend(0x87fa8a8): 677 bytes of 677 to tcp/ > 217.0.26.163:5060 > tport.c:3492 tport_send_msg() tport_vsend returned 677 > tport.c:2296 tport_set_secondary_timer() tport(0x87fa8a8): reset timer > nta.c:8304 outgoing_send() nta: sent REGISTER (124859287) to tcp/ > 217.0.26.163:5060 > tport.c:4160 tport_pend() tport_pend(0x87fa8a8): pending 0xb6263a08 for > tcp/217.0.26.163:5060 (already 0) > nta.c:8929 _nta_outgoing_timer() nta_outgoing_timer: 0/0 resent, 1/1 tout, > 0/0 term, 0/1 free > nta.c:1296 agent_timer() nta: timer set next to 32000 ms > 2018-06-30 22:32:44.083197 [WARNING] sofia_reg.c:484 Timeout Registering > sip-trunk.telekom.de > nua.c:921 nua_handle_destroy() nua: nua_handle_destroy: entering > nua_stack.c:569 nua_stack_signal() nua(0xb4e2ec50): recv signal r_destroy > nua_dialog.c:397 nua_dialog_usage_remove_at() nua(0xb4e2ec50): removing > register usage > nta.c:4470 nta_leg_destroy() nta_leg_destroy(0xb5906508) > soa.c:356 soa_destroy() soa_destroy(static::0xb5923410) called > nua_stack.c:529 nua_signal() nua(0xb4e2ec50): sent signal r_destroy > 2018-06-30 22:32:45.123287 [WARNING] sofia_reg.c:505 sip-trunk.telekom.de > Failed Registration [908], setting retry to 30 seconds. > > ############################################################ > ################# > > 2018-06-30 14:18 GMT+02:00 Paul Muaddib : > >> Hi, >> >> can someone please look at it and tell my what I am doing wrong? >> >> NOTICE] sofia_reg.c:448 Registering sip-trunk.telekom.de >> [WARNING] sofia_reg.c:484 Timeout Registering sip-trunk.telekom.de >> [WARNING] sofia_reg.c:505 sip-trunk.telekom.de Failed Registration >> [908], setting retry to 30 seconds. >> >> My setup: >> ######## >> >> I have a static IP address for my router. My router does not have UPnP or >> NAT-PMP. >> Phones and Freeswitch server are on the same local network. The local >> network is behind NAT >> >> PHONE -> FS -> NAT -> Public Internet >> I only want to connect to the PSTN via my sip gateway provider. >> >> According to this manual I did the following setting: >> https://freeswitch.org/confluence/display/FREESWITCH/NAT+Traversal >> >> vars.xml >> ###### >> >> >> >> internal.xml >> ######### >> >> >> >> external.xml >> ########## >> >> >> >> >> ( What is a little bit strange though is that nat_map status is not >> working >> [ERR] mod_commands.c:751 nat_map API called while NAT not initialized ) >> >> Is the reason why it does not work the port specification in the Contact >> field (sngrep output)? >> Contact: > ip-trunk.telekom.de> >> >> I do not want to setup port forwarding in the firewall. The connection >> should remain open via a keep alive signal. >> >> sngrep output >> ########### >> >> 2018/06/30 13:52:37.211946 10.0.200.2:51813 -> 217.0.26.165:5060 >> REGISTER sip:reg.sip-trunk.telekom.de;transport=tcp SIP/2.0 >> Via: SIP/2.0/TCP X.X.X.X:5080;rport;branch=z2hG4bK588cZXSvrN1De >> Max-Forwards: 70 >> From: ;tag=9aldsj8n67a >> To: >> Call-ID: >> CSeq: 111844514 REGISTER >> Contact: > ip-trunk.telekom.de> >> Expires: 3600 >> User-Agent: FreeSWITCH-mod_sofia/1.6.19+git~20170927T175834Z~38f568d343~ >> 32bit >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, >> REGISTER, REFER, NOTIFY >> Supported: timer, path, replaces >> Content-Length: 0 >> >> Provider settings: >> ############# >> >> output proxy: reg.sip-trunk.telekom.de >> registrar: sip-trunk.telekom.de >> >> sip_profiles/external/telekom_voip.xml >> ############################## >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> Firewall settings >> ############# >> >> # block all traffic in all directions >> block log >> >> # loopback interface is not filtered >> set skip on lo0 >> >> voip_tcp_client_out = "{ssh, ftp, domain, http, https, sip}" >> voip_udp_client_out = "{domain, ntp, sip, 3478}" >> >> match out on $wan_if from any nat-to ($wan_if) >> >> pass on $voip_if inet proto tcp from any to port $voip_tcp_client_out >> pass on $voip_if inet proto udp from any to port $voip_udp_client_out >> >> pass out on $wan_if inet proto tcp from any to port $tcp_client_out >> pass out on $wan_if inet proto udp from any to port $udp_client_out >> >> >> Thank you for helping :) >> >> Regards, >> Paul >> >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From francesco at delagarda.com Mon Jul 2 09:17:29 2018 From: francesco at delagarda.com (Francesco Facco de Lagarda) Date: Mon, 2 Jul 2018 11:17:29 +0200 Subject: [Freeswitch-users] Seperate "new session" and "dial" in JS Message-ID: <042601d411e5$8106b370$83141a50$@delagarda.com> I am trying to make a call progess monitor I need to bind together the answering of incoming session The call to the outgoing session. What I am doing is basically trying to set a user defined variable "t3Id" to both sessions so I "marry then together" from the ESL At present I am doing (just a simplified extract of the code) var t3Id = "##randomly generated##"; session.answer(); session.setVariable("t3Id", t3Id); sessOut=new Session(##my dial string##) sessOut.setVariable("t3Id", t3Id); if(sessout.ready()) { bridge(session, sessOut); } session.hangup(); sessOut.hangup(); Unfortunately I noticed that IF I use sip services (I'm using Messagenet) the outgoing call never gets the variable set! Is it possible to do something like sessOut=new Session() sessOut.setVariable("t3Id", t3Id) sessOut.dial(##my dial string##) ??? I cant find any other parameter common to BOTH sessions to marry the two together!!! -------------- next part -------------- An HTML attachment was scrubbed... URL: From kowalma at gmail.com Mon Jul 2 09:32:12 2018 From: kowalma at gmail.com (Marcin Kowalczyk) Date: Mon, 2 Jul 2018 11:32:12 +0200 Subject: [Freeswitch-users] OPUS recording - playback Message-ID: Hi, I have a call with both legs OPUS encoded. I've enabled call recording with uuid_record file.opus After call was finished I have file.opus but I'm unable to play it back. Tried with VLC but no luck, same goes with opus_decode - it does not recognize file to convert it to wav. Can you guide me how to playback freeswitch recorded .opus file ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: From admin at havesoft.com Mon Jul 2 13:55:13 2018 From: admin at havesoft.com (Jeremy Renner) Date: Mon, 2 Jul 2018 21:55:13 +0800 Subject: [Freeswitch-users] Video call recording and mobile push notifications Message-ID: Hi, I would like to setup up my SIP server / PBX for my business, likes broadsoft, now we have some candidates: 1. Open source solution: - Asterisk PBX, - Freeswitch PBX - Kamailio - OpenSIPS 2. Business solution: - Brekeke PBX(https://www.brekeke.com - Vodia PBX(https://www.vodia.com) - 3CX PBX(https://www.3cx.com) - PortSIP PBX(https://www.portsip.com/portsip-pbx) *Below features are mandatory for our project:* - Video call recording (For the finance industry, the video recording is necessary) - Push notifications for mobile app - Multi-tenant support - Both Linux and Windows support (at 1st stage, we would like to run it on Windows server and migrate it to Linux server in the future if users increased), the Linux support is required, the Windows support is preferred. We have some questions: 1. Does the the Freeswitch can works as the Broadsoft ? 2. If yes, does the Freeswitch support push notifications and video recording ? 3. Does the Freeswitch can works for Multi-tenant ? 4. If Freeswtich doesn't support push notifications, does there has any 3rd plugin supported ? So far according to our research, with the business solution: - The Vodia PBX, PortSIP PBX and brrekeke all are support Multi-tenant, the 3CX is not. - The 3CX and PortSIP support push notifications, - The PortSIP also provide client SDK, with 3CX we only see the 3CX provide client apps, does 3CX has client SDK provided ? - It's seems all these PBX are support video recording ? - The PortSIP PBX and 3CX both support Linux. Please help me to make the decision, base on your experiences, which one (open source or business solution) is good to us ? I'm really new to VoIP... Thanks in advance. Best regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: From krice at freeswitch.org Sun Jul 1 19:04:43 2018 From: krice at freeswitch.org (Ken Rice) Date: Sun, 1 Jul 2018 14:04:43 -0500 Subject: [Freeswitch-users] Debian repos In-Reply-To: <20180630120952.re4t4ujro7vi45yq@eye.freedbms.net> References: <68A1DF24-5BBE-4D6C-95D6-24E99A631B82@gmail.com> <20180630120356.5rotziic4c3j2rlh@eye.freedbms.net> <20180630120952.re4t4ujro7vi45yq@eye.freedbms.net> Message-ID: <340B7D03-9349-4653-AB5D-718AC5A05D47@freeswitch.org> ot sure why you would want such an old verion as there have been 100s of bug fixes in that time frame. from broken functionality to potential security issues. Sent from my iPhone > On Jun 30, 2018, at 07:09, Zenaan Harkness wrote: > > Found this: > https://freeswitch.org/confluence/display/FREESWITCH/Tips+For+Using+Git > > >> On Sat, Jun 30, 2018 at 10:03:56PM +1000, Zenaan Harkness wrote: >> Is there a suggested or (semi?) official "anonymous git" protocol git >> repo - e.g. where the following command would work: >> >> git clone https://freeswitch.org/stash/scm/fs/freeswitch.git >> >> (like how github works...)? >> >> >> >>> On Wed, Nov 29, 2017 at 10:13:36AM +0900, 王聡 wrote: >>> You can compile FS by youself if you want. >>> >>> For FS 1.6.8: >>> >>> git clone https://freeswitch.org/stash/scm/fs/freeswitch.git >>> cd /usr/src/freeswitch/ >>> >>> git checkout -b v1.6.8 refs/tags/v1.6.8 >>> ./bootstrap.sh -j >>> Also you can compile any version as you wish. >>> >>>> 2017/11/28 22:52、Igor Olhovskiy のメール: >>>> >>>> Hi! >>>> >>>> Is there a way to set up different version of FreeSWITCH from repos? >>>> Means for ex I want to use not 1.6.19, but 1.6.8? >>>> Or compile is the only way? >>>> >>>> Regards, Igor >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> -------------- next part -------------- >>> An HTML attachment was scrubbed... >>> URL: > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com From alex at freeswitch.com Tue Jul 3 03:36:49 2018 From: alex at freeswitch.com (Alexey Sibyakin) Date: Tue, 3 Jul 2018 12:36:49 +0900 Subject: [Freeswitch-users] Video call recording and mobile push notifications In-Reply-To: References: Message-ID: Hi, FreeSWITCH is much more than just PBX. All your needs except Push notifications will work out of the box for free. You always can discuss addition of any functionality via sales at freeswitch.com Regards, Alex On Mon, Jul 2, 2018 at 10:55 PM, Jeremy Renner wrote: > Hi, > > I would like to setup up my SIP server / PBX for my business, likes > broadsoft, now we have some candidates: > > 1. Open source solution: > > - Asterisk PBX, > - Freeswitch PBX > - Kamailio > - OpenSIPS > > > 2. Business solution: > > - Brekeke PBX(https://www.brekeke.com > - Vodia PBX(https://www.vodia.com) > - 3CX PBX(https://www.3cx.com) > - PortSIP PBX(https://www.portsip.com/portsip-pbx) > > > *Below features are mandatory for our project:* > > - Video call recording (For the finance industry, the video recording > is necessary) > - Push notifications for mobile app > - Multi-tenant support > - Both Linux and Windows support (at 1st stage, we would like to run > it on Windows server and migrate it to Linux server in the future if users > increased), the Linux support is required, the Windows support is preferred. > > > We have some questions: > > 1. Does the the Freeswitch can works as the Broadsoft ? > 2. If yes, does the Freeswitch support push notifications and video > recording ? > 3. Does the Freeswitch can works for Multi-tenant ? > 4. If Freeswtich doesn't support push notifications, does there has > any 3rd plugin supported ? > > > So far according to our research, with the business solution: > > - The Vodia PBX, PortSIP PBX and brrekeke all are support > Multi-tenant, the 3CX is not. > - The 3CX and PortSIP support push notifications, > - The PortSIP also provide client SDK, with 3CX we only see the 3CX > provide client apps, does 3CX has client SDK provided ? > - It's seems all these PBX are support video recording ? > - The PortSIP PBX and 3CX both support Linux. > > > Please help me to make the decision, base on your experiences, which one > (open source or business solution) is good to us ? I'm really new to > VoIP... > > Thanks in advance. > > Best regards, > > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com > -- Alex Sibyakin | Support Engineer FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 Email: alex at freeswitch.com Website: https://www.FreeSWITCH.com Need commercial support? Contact sales at freeswitch.com for details. -------------- next part -------------- An HTML attachment was scrubbed... URL: From alex at freeswitch.com Tue Jul 3 03:48:02 2018 From: alex at freeswitch.com (Alexey Sibyakin) Date: Tue, 3 Jul 2018 12:48:02 +0900 Subject: [Freeswitch-users] Manipulate the sip: string in the INVITE header on an outgoing Bridge In-Reply-To: <8D46C488-0098-4C7D-A601-182DB1E18AF1@tm.net.uk> References: <8D46C488-0098-4C7D-A601-182DB1E18AF1@tm.net.uk> Message-ID: You probably need proper URI as Nathan suggested and not just $1. Also domain should be something that other side can handle. Regards, Alex On Mon, Jul 2, 2018 at 5:50 AM, Joseph Waite wrote: > Hi Alex > > This doesn’t seem to do anything, or maybe it is just setting the To: > field which I have already got set. > > The problem with the To field is that asterisk doesn’t look at this field > without customisation which isn’t great for customer ease of use. > > Regards > > On 29 Jun 2018, at 02:10, Alexey Sibyakin wrote: > > Hi, > > I'm not sure that I got your scenario right but you can try to do > something like this: > > (you > may need another var, full list is in Confluence) > > Regards, > > Alex > > > > On Wed, Jun 27, 2018 at 9:33 PM, Joseph Waite wrote: > >> Hi Guys >> >> Were having issues with Asterisk boxes registering to a trunk on our >> FreeSwitch box. >> >> We are sending the DID number called down the trunk in the to: field, >> however Asterisk doesn’t seem to like this. >> >> Is there anyway to manipulate the INVITE to be INVITE sip:DID@ instead >> of what is currently INVITE sip:user@ >> >> Im hoping there is something like sip_invite_to_uri, however google is >> failing me. >> >> Regards >> >> _________________________________________________________________________ >> Professional FreeSWITCH Services >> sales at freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com >> > > > > -- > Alex Sibyakin | Support Engineer > FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 > > Email: alex at freeswitch.com > Website: https://www.FreeSWITCH.com > Need commercial support? Contact sales at freeswitch.com for details. > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com > > > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com > -- Alex Sibyakin | Support Engineer FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 Email: alex at freeswitch.com Website: https://www.FreeSWITCH.com Need commercial support? Contact sales at freeswitch.com for details. -------------- next part -------------- An HTML attachment was scrubbed... URL: From andretodd at verizon.net Tue Jul 3 14:15:06 2018 From: andretodd at verizon.net (andretodd at verizon.net) Date: Tue, 3 Jul 2018 10:15:06 -0400 Subject: [Freeswitch-users] FreeSWITCH Conference Message-ID: <0c8a01d412d8$3e56d160$bb047420$@verizon.net> Hi is there a way to bridge a conference to another FreeSWITCH conference? Example FreeSWITCH Conference "A" has 2 people Then OpenSips puts the next person who calls onto FreeSWITCH Conference "B". now we need everyone together. I'm looking for a way to make the conference scalable to many boxes using OpenSips as the Load balancer and RTP Proxy. Any examples of what the dial string would look like would be very helpful Thanks Andre From aqsyounas at gmail.com Mon Jul 2 19:28:47 2018 From: aqsyounas at gmail.com (Aqs Younas) Date: Tue, 3 Jul 2018 00:28:47 +0500 Subject: [Freeswitch-users] Manipulate the sip: string in the INVITE header on an outgoing Bridge In-Reply-To: <8D46C488-0098-4C7D-A601-182DB1E18AF1@tm.net.uk> References: <8D46C488-0098-4C7D-A601-182DB1E18AF1@tm.net.uk> Message-ID: Try this. https://www.youtube.com/watch?v=EVoVmqRWDHc Regards, Aqs On Mon, 2 Jul 2018 at 21:12, Joseph Waite wrote: > Hi Alex > > This doesn’t seem to do anything, or maybe it is just setting the To: > field which I have already got set. > > The problem with the To field is that asterisk doesn’t look at this field > without customisation which isn’t great for customer ease of use. > > Regards > > On 29 Jun 2018, at 02:10, Alexey Sibyakin wrote: > > Hi, > > I'm not sure that I got your scenario right but you can try to do > something like this: > > (you may > need another var, full list is in Confluence) > > Regards, > > Alex > > > > On Wed, Jun 27, 2018 at 9:33 PM, Joseph Waite wrote: > >> Hi Guys >> >> Were having issues with Asterisk boxes registering to a trunk on our >> FreeSwitch box. >> >> We are sending the DID number called down the trunk in the to: field, >> however Asterisk doesn’t seem to like this. >> >> Is there anyway to manipulate the INVITE to be INVITE sip:DID@ instead >> of what is currently INVITE sip:user@ >> >> Im hoping there is something like sip_invite_to_uri, however google is >> failing me. >> >> Regards >> >> _________________________________________________________________________ >> Professional FreeSWITCH Services >> sales at freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com >> > > > > -- > Alex Sibyakin | Support Engineer > FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 > > Email: alex at freeswitch.com > Website: https://www.FreeSWITCH.com > Need commercial support? Contact sales at freeswitch.com for details. > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com > > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From bipin at xbipin.com Mon Jul 2 19:28:09 2018 From: bipin at xbipin.com (Bipin Patel) Date: Mon, 02 Jul 2018 23:28:09 +0400 Subject: [Freeswitch-users] =?utf-8?b?562U5aSNOiBGcmVlc3dpdGNoIHdpbmRvd3Mg?= =?utf-8?q?build_-_libpng?= In-Reply-To: References: Message-ID: <1645c777128.279b.b07ebdf329620b8089087c7205b03f01@xbipin.com> That's great news, would it be possible to change the vs project and config file and merge it to master so can help others as well built in vs2017 as I also currently built using vs2015 only. On July 2, 2018 11:12:17 PM Gregor Nanger wrote: > I managed to build successfully with latest Vs2017 version. Latest master > built without problem, but 1.6.20, I had to do some manual changes in > configuration. I am planning to prepare video tutorial. > > Best regards, Gregor > > > On Sun, Jul 1, 2018, 14:21 Andrew Keil wrote: > Gregor, > > Or you can make the following change to the zlib project for All > Configurations and All Releases - C/C++ - All Options: > - Disable Specific Warnings (add to end): ;5045 > - Whole Program Optimization: No > > Hope that helps. > > Andrew > > From: FreeSWITCH-users > On Behalf Of bob. chen > Sent: Tuesday, 26 June 2018 1:23 PM > To: FreeSWITCH Users Help > Subject: [Freeswitch-users] 答复: Freeswitch windows build - libpng > > try vs2015 ;) > > > 发件人: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] > 代表 > Gregor Nanger > 发送时间: 2018年6月23日 19:08 > 收件人: FreeSWITCH Users Help > 主题: [Freeswitch-users] Freeswitch windows build - libpng > > Hi! > > After a year I wanted to build FS on windows and make video tutorial or for > FS wiki. > > I am using latest VS2017 and everything builds except libpng project. Below > is error if someone can help me, because I don't have c++ experience: > > 1>------ Build started: Project: Download zlib, Configuration: Release > Win32 ------ > 1>Downloading zlib. > 2>------ Build started: Project: libpng, Configuration: Release x64 ------ > 2>Generating pnglibconf.h > 2> 1 file(s) copied. > 2>png.c > 2>pngerror.c > 2>pngget.c > 2>pngmem.c > 2>pngpread.c > 2>pngread.c > 2>pngrio.c > 2>pngrtran.c > 2>pngrutil.c > 2>pngset.c > 2>pngtrans.c > 2>pngwio.c > 2>pngwrite.c > 2>pngwtran.c > 2>pngwutil.c > 2>Generating Code... > 2>zlib.lib(inflate.obj) : MSIL .netmodule or module compiled with /GL > found; restarting link with /LTCG; add /LTCG to the link command line to > improve linker performance > 2> Creating library D:\Git\freeswitch\x64\Release\libpng16.lib and object > D:\Git\freeswitch\x64\Release\libpng16.exp > 2>Generating code > 2>d:\git\freeswitch\libs\zlib\deflate.c(2097): error C2220: warning treated > as error - no 'executable' file generated > 2>d:\git\freeswitch\libs\zlib\deflate.c(2097): warning C5045: Compiler will > insert Spectre mitigation for memory load if /Qspectre switch specified > 2>d:\git\freeswitch\libs\zlib\deflate.c(2097) : note: index 'dist' range > checked by comparison on this line > 2>d:\git\freeswitch\libs\zlib\deflate.c(2097) : note: feeds memory load on > this line > 2>d:\git\freeswitch\libs\zlib\deflate.c(1987): warning C5045: Compiler will > insert Spectre mitigation for memory load if /Qspectre switch specified > 2>d:\git\freeswitch\libs\zlib\deflate.c(1987) : note: index 'dist' range > checked by comparison on this line > 2>d:\git\freeswitch\libs\zlib\deflate.c(1987) : note: feeds memory load on > this line > 2>d:\git\freeswitch\libs\zlib\deflate.c(1862): warning C5045: Compiler will > insert Spectre mitigation for memory load if /Qspectre switch specified > 2>d:\git\freeswitch\libs\zlib\deflate.c(1862) : note: index 'dist' range > checked by comparison on this line > 2>d:\git\freeswitch\libs\zlib\deflate.c(1862) : note: feeds memory load on > this line > 2>LINK : fatal error LNK1257: code generation failed > 2>Done building project "libpng.vcxproj" -- FAILED. > > Best regards, Gregor > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com > -- > > 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 > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From gmaruzz at gmail.com Mon Jul 2 19:40:28 2018 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Mon, 2 Jul 2018 21:40:28 +0200 Subject: [Freeswitch-users] About Verto In-Reply-To: References: <424FFF6C-73B4-4293-A4FE-8F45A24815CD@gmail.com> <695562B1-39BB-4816-B7C3-2EA9D1663389@gmail.com> Message-ID: Aaaand, let's say the truth: this is a battle between Apple and Google, and the result is trillions of cpu.cycles wastqed in transcoding servers... On Mon, Jul 2, 2018, 19:06 Chad Phillips wrote: > For interop between all Android devices and iOS devices, I’m pretty sure > you need both VP8 and H264 enabled for available codecs. > > Chrome doesn’t support H264 on all Android devices yet, there’s an open > issue to add software H264 encoding in the case where the Android hardware > doesn’t have hardware encoding: > https://bugs.chromium.org/p/chromium/issues/detail?id=719023 > > On Sun, Jul 1, 2018 at 8:28 AM 王聡 wrote: > >> Thanks for your reply. >> >> I had added H264 codec in verto.conf.xml, and now it’s able to make call >> between iOS users. >> But it still didn’t work between Android and iOS users, is there any >> suggestions? >> >> Regards. >> >> 在 2018年6月29日,00:00,Giovanni Maruzzelli 写道: >> >> On 28 June 2018 at 09:30, 王聡 wrote: >> >>> >>> I had tried to build a WebRTC video call service via mod_verto and Verto >>> Comunicator. >>> Now I could make calls from (to) an Android device with Chrome, but it >>> didn’t work on iOS with Safari. >>> Is there any point to be modified to run on Safari? >>> >> >> be sure to enable H264 codec, both in vars.conf.xml and in verto.conf.xml >> >> >> >> >> >>> >>> Regards. >>> _________________________________________________________________________ >>> Professional FreeSWITCH Services >>> sales at freeswitch.com >>> https://freeswitch.com >>> >>> Official FreeSWITCH Sites >>> https://freeswitch.com/oss >>> https://freeswitch.org/confluence >>> https://cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> https://freeswitch.com >> >> >> >> >> -- >> >> Sincerely, >> >> Giovanni Maruzzelli >> OpenTelecom.IT >> cell: +39 347 266 56 18 >> _________________________________________________________________________ >> Professional FreeSWITCH Services >> sales at freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Services >> sales at freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From grcamauer at gmail.com Mon Jul 2 20:18:35 2018 From: grcamauer at gmail.com (Guillermo Ruiz Camauer) Date: Mon, 2 Jul 2018 17:18:35 -0300 Subject: [Freeswitch-users] Failed Registration [908] In-Reply-To: References: Message-ID: Are you sure the transport is TCP and not UDP? Guillermo On Mon, Jul 2, 2018 at 3:39 PM Paul Muaddib wrote: > Now, I opened all firewall ports. I tried connecting to different SIP > Servers with Netcat and it established a connection. But I don't get any > response. > I read the Technical Specification 1TR118 from Deutsche Telekom and change > the gateway setting according to the specifications but no success ;( > > > > > > > > > > > > > > > > > > > > > > 2018-06-30 22:57 GMT+02:00 Paul Muaddib : > >> sofia loglevel all 9 >> >> ##################################################################### >> >> 2018-06-30 22:31:43.746032 [NOTICE] sofia_reg.c:448 Registering >> sip-trunk.telekom.de >> nua.c:622 nua_register() nua: nua_register: entering >> nua_stack.c:529 nua_signal() nua(0xb4e2ec50): sent signal r_register >> nua_stack.c:569 nua_stack_signal() nua(0xb4e2ec50): recv signal r_register >> nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: >> entering >> soa.c:280 soa_clone() soa_clone(static::0xb6255520, 0x88b30b0, >> 0xb4e2ec50) called >> soa.c:403 soa_set_params() soa_set_params(static::0xb5923410, ...) called >> soa.c:403 soa_set_params() soa_set_params(static::0xb5923410, ...) called >> nua_dialog.c:338 nua_dialog_usage_add() nua(0xb4e2ec50): adding register >> usage >> nta.c:4417 nta_leg_tcreate() nta_leg_tcreate(0xb5906508) >> nta.c:2665 nta_tpn_by_url() nta: selecting scheme sip >> sres_cache.c:272 sres_cache_get() sres_cache_get(0x88aa8e0, SRV, "_sip._ >> tcp.reg.sip-trunk.telekom.de.") called >> sres_cache.c:318 sres_cache_get() sres_cache_get(0x88aa8e0, SRV, "_sip._ >> tcp.reg.sip-trunk.telekom.de.") returned 3 entries >> nta.c:10598 outgoing_query_srv() nta: for "reg.sip-trunk.telekom.de" >> query "_sip._tcp.reg.sip-trunk.telekom.de" SRV (cached) >> nta.c:10694 outgoing_answer_srv() nta: _sip._tcp.reg.sip-trunk.telekom.de >> IN SRV 10 5 5060 d-ipr-a02.sip-trunk.telekom.de. (tcp) >> nta.c:10694 outgoing_answer_srv() nta: _sip._tcp.reg.sip-trunk.telekom.de >> IN SRV 11 5 5060 b-ipr-a01.sip-trunk.telekom.de. (tcp) >> nta.c:10694 outgoing_answer_srv() nta: _sip._tcp.reg.sip-trunk.telekom.de >> IN SRV 11 5 5060 b-ipr-a02.sip-trunk.telekom.de. (tcp) >> sres_cache.c:272 sres_cache_get() sres_cache_get(0x88aa8e0, A, " >> d-ipr-a02.sip-trunk.telekom.de.") called >> sres_cache.c:318 sres_cache_get() sres_cache_get(0x88aa8e0, A, " >> d-ipr-a02.sip-trunk.telekom.de.") returned 1 entries >> nta.c:10803 outgoing_query_a() nta: for "reg.sip-trunk.telekom.de" query >> "d-ipr-a02.sip-trunk.telekom.de." A (cached) >> nta.c:10856 outgoing_answer_a() nta: d-ipr-a02.sip-trunk.telekom.de. IN >> A 217.0.26.133 >> tport.c:4588 tport_by_name() tport(0xb625dc58): found 0x880d978 by name >> tcp/217.0.26.133:5060 >> tport.c:3257 tport_tsend() tport_tsend(0x880d978) tpn = tcp/ >> 217.0.26.133:5060 >> tport.c:3594 tport_vsend() tport_vsend(0x880d978): 677 bytes of 677 to >> tcp/217.0.26.133:5060 >> tport.c:3492 tport_send_msg() tport_vsend returned 677 >> tport.c:2296 tport_set_secondary_timer() tport(0x880d978): reset timer >> nta.c:8304 outgoing_send() nta: sent REGISTER (124859287) to tcp/ >> 217.0.26.133:5060 >> tport.c:4160 tport_pend() tport_pend(0x880d978): pending 0xb6263a08 for >> tcp/217.0.26.133:5060 (already 1) >> nta.c:8982 outgoing_timer_bf() nta: timer F fired, timeout REGISTER >> (124859241) >> nta.c:9035 outgoing_timeout() nta(0xb6264318): try next after timeout >> nta.c:10200 outgoing_graylist() nta: graylisting >> d-ipr-a02.sip-trunk.telekom.de.:5060;transport=tcp >> nta.c:10227 outgoing_graylist() nta: reduced priority of 1 _sip._ >> tcp.reg.sip-trunk.telekom.de SRV records (increase value to 11) >> sres_cache.c:272 sres_cache_get() sres_cache_get(0x88aa8e0, A, " >> reg.sip-trunk.telekom.de.") called >> nta.c:10803 outgoing_query_a() nta: for "reg.sip-trunk.telekom.de" query >> "reg.sip-trunk.telekom.de" A >> sres.c:968 sres_query() sres_query(0xb624a8b8, 0xb6264318, A, " >> reg.sip-trunk.telekom.de") called >> sres.c:2730 sres_send_dns_query() sres_send_dns_query(0xb624a8b8, >> 0x88653d0) called >> sres.c:2819 sres_send_dns_query() sres_send_dns_query(0xb624a8b8, >> 0x88653d0) id=12982 A reg.sip-trunk.telekom.de (to [217.237.148.70]:53) >> nta.c:8929 _nta_outgoing_timer() nta_outgoing_timer: 0/0 resent, 1/2 >> tout, 0/0 term, 0/2 free >> nta.c:1296 agent_timer() nta: timer set next to 29536 ms >> sres.c:3467 sres_resolver_receive() sres_resolver_receive(0xb624a8b8, 31) >> called >> sres.c:3781 sres_create_record() AUTHORITY RR received >> sip-trunk.telekom.de. SOA IN 151 rdlen=64 >> sres.c:3572 sres_log_response() sres_resolver_receive(0xb624a8b8, >> 0x88653d0) id=12982 (from [217.237.148.70]:53) >> sres.c:2987 sres_query_report_error() sres(q=0x88653d0): reporting error >> RECORD_ERR for A reg.sip-trunk.telekom.de >> tport.c:4222 tport_release() tport_release(0x880d978): 0xb623b2e8 by >> 0xb6264318 with (nil) >> nta.c:8722 outgoing_free() nta: outgoing_free(0xb6264318) >> nta.c:8982 outgoing_timer_bf() nta: timer F fired, timeout REGISTER >> (124859287) >> nta.c:9035 outgoing_timeout() nta(0xb628fb30): try next after timeout >> nta.c:10200 outgoing_graylist() nta: graylisting >> d-ipr-a02.sip-trunk.telekom.de.:5060;transport=tcp >> nta.c:10227 outgoing_graylist() nta: reduced priority of 1 _sip._ >> tcp.reg.sip-trunk.telekom.de SRV records (increase value to 12) >> sres_cache.c:272 sres_cache_get() sres_cache_get(0x88aa8e0, A, " >> b-ipr-a01.sip-trunk.telekom.de.") called >> sres_cache.c:318 sres_cache_get() sres_cache_get(0x88aa8e0, A, " >> b-ipr-a01.sip-trunk.telekom.de.") returned 1 entries >> nta.c:10803 outgoing_query_a() nta: for "reg.sip-trunk.telekom.de" query >> "b-ipr-a01.sip-trunk.telekom.de." A (cached) >> nta.c:10856 outgoing_answer_a() nta: b-ipr-a01.sip-trunk.telekom.de. IN >> A 217.0.26.163 >> tport.c:4588 tport_by_name() tport(0xb625dc58): found 0x87fa8a8 by name >> tcp/217.0.26.163:5060 >> tport.c:4222 tport_release() tport_release(0x880d978): 0xb6263a08 by >> 0xb628fb30 with (nil) >> tport.c:2296 tport_set_secondary_timer() tport(0x880d978): reset timer >> tport.c:3257 tport_tsend() tport_tsend(0x87fa8a8) tpn = tcp/ >> 217.0.26.163:5060 >> tport.c:3594 tport_vsend() tport_vsend(0x87fa8a8): 677 bytes of 677 to >> tcp/217.0.26.163:5060 >> tport.c:3492 tport_send_msg() tport_vsend returned 677 >> tport.c:2296 tport_set_secondary_timer() tport(0x87fa8a8): reset timer >> nta.c:8304 outgoing_send() nta: sent REGISTER (124859287) to tcp/ >> 217.0.26.163:5060 >> tport.c:4160 tport_pend() tport_pend(0x87fa8a8): pending 0xb6263a08 for >> tcp/217.0.26.163:5060 (already 0) >> nta.c:8929 _nta_outgoing_timer() nta_outgoing_timer: 0/0 resent, 1/1 >> tout, 0/0 term, 0/1 free >> nta.c:1296 agent_timer() nta: timer set next to 32000 ms >> 2018-06-30 22:32:44.083197 [WARNING] sofia_reg.c:484 Timeout Registering >> sip-trunk.telekom.de >> nua.c:921 nua_handle_destroy() nua: nua_handle_destroy: entering >> nua_stack.c:569 nua_stack_signal() nua(0xb4e2ec50): recv signal r_destroy >> nua_dialog.c:397 nua_dialog_usage_remove_at() nua(0xb4e2ec50): removing >> register usage >> nta.c:4470 nta_leg_destroy() nta_leg_destroy(0xb5906508) >> soa.c:356 soa_destroy() soa_destroy(static::0xb5923410) called >> nua_stack.c:529 nua_signal() nua(0xb4e2ec50): sent signal r_destroy >> 2018-06-30 22:32:45.123287 [WARNING] sofia_reg.c:505 sip-trunk.telekom.de >> Failed Registration [908], setting retry to 30 seconds. >> >> >> ############################################################################# >> >> 2018-06-30 14:18 GMT+02:00 Paul Muaddib : >> >>> Hi, >>> >>> can someone please look at it and tell my what I am doing wrong? >>> >>> NOTICE] sofia_reg.c:448 Registering sip-trunk.telekom.de >>> [WARNING] sofia_reg.c:484 Timeout Registering sip-trunk.telekom.de >>> [WARNING] sofia_reg.c:505 sip-trunk.telekom.de Failed Registration >>> [908], setting retry to 30 seconds. >>> >>> My setup: >>> ######## >>> >>> I have a static IP address for my router. My router does not have UPnP >>> or NAT-PMP. >>> Phones and Freeswitch server are on the same local network. The local >>> network is behind NAT >>> >>> PHONE -> FS -> NAT -> Public Internet >>> I only want to connect to the PSTN via my sip gateway provider. >>> >>> According to this manual I did the following setting: >>> https://freeswitch.org/confluence/display/FREESWITCH/NAT+Traversal >>> >>> vars.xml >>> ###### >>> >>> >>> >>> internal.xml >>> ######### >>> >>> >>> >>> external.xml >>> ########## >>> >>> >>> >>> >>> ( What is a little bit strange though is that nat_map status is not >>> working >>> [ERR] mod_commands.c:751 nat_map API called while NAT not initialized ) >>> >>> Is the reason why it does not work the port specification in the Contact >>> field (sngrep output)? >>> Contact: >> sip-trunk.telekom.de> >>> >>> I do not want to setup port forwarding in the firewall. The connection >>> should remain open via a keep alive signal. >>> >>> sngrep output >>> ########### >>> >>> 2018/06/30 13:52:37.211946 10.0.200.2:51813 -> 217.0.26.165:5060 >>> REGISTER sip:reg.sip-trunk.telekom.de;transport=tcp SIP/2.0 >>> Via: SIP/2.0/TCP X.X.X.X:5080;rport;branch=z2hG4bK588cZXSvrN1De >>> Max-Forwards: 70 >>> From: ;tag=9aldsj8n67a >>> To: >>> Call-ID: >>> CSeq: 111844514 REGISTER >>> Contact: >> sip-trunk.telekom.de> >>> Expires: 3600 >>> User-Agent: >>> FreeSWITCH-mod_sofia/1.6.19+git~20170927T175834Z~38f568d343~32bit >>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, >>> REGISTER, REFER, NOTIFY >>> Supported: timer, path, replaces >>> Content-Length: 0 >>> >>> Provider settings: >>> ############# >>> >>> output proxy: reg.sip-trunk.telekom.de >>> registrar: sip-trunk.telekom.de >>> >>> sip_profiles/external/telekom_voip.xml >>> ############################## >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> Firewall settings >>> ############# >>> >>> # block all traffic in all directions >>> block log >>> >>> # loopback interface is not filtered >>> set skip on lo0 >>> >>> voip_tcp_client_out = "{ssh, ftp, domain, http, https, sip}" >>> voip_udp_client_out = "{domain, ntp, sip, 3478}" >>> >>> match out on $wan_if from any nat-to ($wan_if) >>> >>> pass on $voip_if inet proto tcp from any to port $voip_tcp_client_out >>> pass on $voip_if inet proto udp from any to port $voip_udp_client_out >>> >>> pass out on $wan_if inet proto tcp from any to port $tcp_client_out >>> pass out on $wan_if inet proto udp from any to port $udp_client_out >>> >>> >>> Thank you for helping :) >>> >>> Regards, >>> Paul >>> >>> >>> >> > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Guillermo Ruiz Camauer -------------- next part -------------- An HTML attachment was scrubbed... URL: From mario_fs at mgtech.com Mon Jul 2 22:30:02 2018 From: mario_fs at mgtech.com (Mario) Date: Mon, 2 Jul 2018 15:30:02 -0700 Subject: [Freeswitch-users] Voicemail MWI multiple mailboxes notify one user In-Reply-To: References: Message-ID: Aha… something I know (a little) about. This is how an extension can monitor another mailbox: In the user/extension definition you would add in “params” section: Where 100 is the VMs extension to monitor, like 2001, etc. This will monitor only one mailbox, see below for some ideas on multiple MBs. Now for the issues: Be very careful during testing because of https://freeswitch.org/jira/browse/FS-10683 , I am putting a bounty up for this once 1.8 hits the streets. The syntax requires the domain or IP address after the extension, IMO it should not require the domain but make it optional. Why? Because you can’t use IP V6, see https://freeswitch.org/jira/browse/FS-9428 , also offering a bounty after 1.8 hits. If there is no way to monitor multiple boxes, here are a couple of suggestions: Have the other extensions send VM to a common MB, in their “variable” section use: There is no need to set any tcp params, the DNS SRV record does the job. Good luck, Henning > Am 29.06.2018 um 19:38 schrieb Paul Muaddib : > > Hi, > > how do I fix this problem: > Failed Registration with status DNS Error [503]? > > I already search the net but couldn't find a proper answer. The only thing that I found was something about DNS SRV Resource Record but this was fixed in freeswitch a long time ago. > > My Setup: > FreeSWITCH Version 1.6.19 > Debian 7 wheezy > Freeswitch(Debian) -> Router(OpenBSD) -> Internet > > Failed Registration with status DNS Error [503] > 2018-06-29 19:27:21.732868 [NOTICE] sofia_reg.c:448 Registering telekom_voip > nua.c:622 nua_register() nua: nua_register: entering > nua_stack.c:529 nua_signal() nua(0x8c8c0c8): sent signal r_register > nua_stack.c:569 nua_stack_signal() nua(0x8c8c0c8): recv signal r_register > nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering > soa.c:403 soa_set_params() soa_set_params(static::0xb5c1a298, ...) called > nua_dialog.c:338 nua_dialog_usage_add() nua(0x8c8c0c8): adding register usage > nta.c:4417 nta_leg_tcreate() nta_leg_tcreate(0xb6279478) > nta.c:2665 nta_tpn_by_url() nta: selecting scheme sip > sres_cache.c:272 sres_cache_get() sres_cache_get(0x8ccd7b0, NAPTR, "sip-trunk.telekom.de.") called > sres_cache.c:318 sres_cache_get() sres_cache_get(0x8ccd7b0, NAPTR, "sip-trunk.telekom.de.") returned 1 entries > nta.c:10398 outgoing_query_naptr() nta: for "sip-trunk.telekom.de" query "sip-trunk.telekom.de" NAPTR (cached) > sres_cache.c:272 sres_cache_get() sres_cache_get(0x8ccd7b0, SRV, "_sip._udp.sip-trunk.telekom.de.") called > sres_cache.c:318 sres_cache_get() sres_cache_get(0x8ccd7b0, SRV, "_sip._udp.sip-trunk.telekom.de.") returned 1 entries > nta.c:10598 outgoing_query_srv() nta: for "sip-trunk.telekom.de" query "_sip._udp.sip-trunk.telekom.de" SRV (cached) > sres_cache.c:272 sres_cache_get() sres_cache_get(0x8ccd7b0, A, "sip-trunk.telekom.de.") called > sres_cache.c:318 sres_cache_get() sres_cache_get(0x8ccd7b0, A, "sip-trunk.telekom.de.") returned 1 entries > nta.c:10803 outgoing_query_a() nta: for "sip-trunk.telekom.de" query "sip-trunk.telekom.de" A (cached) > nta.c:1350 set_timeout() nta: timer set to 32000 ms > nta.c:1348 set_timeout() nta: timer shortened to 5000 ms > nua_stack.c:271 nua_stack_event() nua(0x8c8c0c8): event r_register 503 DNS Error > nua_dialog.c:397 nua_dialog_usage_remove_at() nua(0x8c8c0c8): removing register usage > nta.c:4470 nta_leg_destroy() nta_leg_destroy(0xb6279478) > nua_stack.c:359 nua_application_event() nua: nua_application_event: entering > nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering > 2018-06-29 19:27:21.732868 [ERR] sofia_reg.c:2447 telekom_voip Failed Registration with status DNS Error [503]. failure #34 > nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering > 2018-06-29 19:27:22.772957 [WARNING] sofia_reg.c:505 telekom_voip Failed Registration [503], setting retry to 30 seconds. > nta.c:9101 outgoing_timer_dk() nta: timer K fired, terminate REGISTER (124810556) > nta.c:8799 outgoing_reclaim_queued() outgoing_reclaim_all((nil), (nil), 0xb60911e0) > nta.c:8929 _nta_outgoing_timer() nta_outgoing_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/1 free > nta.c:1289 agent_timer() nta: timer not set > > sofia_dig --tcp reg.sip-trunk.telekom.de > Preference Weight Transport Port Address > ================================================================================ > 1 1.000 tcp 5060 217.0.26.165 > 2 1.000 tcp 5060 217.0.26.163 > 3 1.000 tcp 5060 217.0.26.133 > > Gateway settings > > > > > > > > > > > > > > > Regards, > Paul -------------- next part -------------- An HTML attachment was scrubbed... URL: From vma at vallimamod.org Mon Jul 2 16:27:03 2018 From: vma at vallimamod.org (Vallimamod Abdullah) Date: Mon, 2 Jul 2018 18:27:03 +0200 Subject: [Freeswitch-users] Manipulate the sip: string in the INVITE header on an outgoing Bridge In-Reply-To: <8D46C488-0098-4C7D-A601-182DB1E18AF1@tm.net.uk> References: <8D46C488-0098-4C7D-A601-182DB1E18AF1@tm.net.uk> Message-ID: <5E8184E6-6EAE-4829-9553-DC0A3289A1CF@vallimamod.org> Hi, If I understand correctly your requirement: - Asterisk is registered as asterisk at IP - You want to send invites with sip-uri sip:DID at IP instead of the asterisk registration uri (sip:asterisk at IP) Then, you can set the dial-string to the following: So when you bridge with user/xxx, fsw will put the DID in the sip-uri and add a Route header sip asterisk reg uri for the next hop. Hope this helps. Best Regards, -- Vallimamod Abdullah SIP Solutions vma at sip.solutions linkedin.com/in/vallimamod . > On 1 Jul 2018, at 22:50, Joseph Waite wrote: > > Hi Alex > > This doesn’t seem to do anything, or maybe it is just setting the To: field which I have already got set. > > The problem with the To field is that asterisk doesn’t look at this field without customisation which isn’t great for customer ease of use. > > Regards >> On 29 Jun 2018, at 02:10, Alexey Sibyakin wrote: >> >> Hi, >> >> I'm not sure that I got your scenario right but you can try to do something like this: >> >> (you may need another var, full list is in Confluence) >> >> Regards, >> >> Alex >> >> >> >> On Wed, Jun 27, 2018 at 9:33 PM, Joseph Waite wrote: >> Hi Guys >> >> Were having issues with Asterisk boxes registering to a trunk on our FreeSwitch box. >> >> We are sending the DID number called down the trunk in the to: field, however Asterisk doesn’t seem to like this. >> >> Is there anyway to manipulate the INVITE to be INVITE sip:DID@ instead of what is currently INVITE sip:user@ >> >> Im hoping there is something like sip_invite_to_uri, however google is failing me. >> >> Regards >> >> _________________________________________________________________________ >> Professional FreeSWITCH Services >> sales at freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com >> >> >> >> -- >> Alex Sibyakin | Support Engineer >> FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 >> Email: alex at freeswitch.com >> Website: https://www.FreeSWITCH.com >> Need commercial support? Contact sales at freeswitch.com for details. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Services >> sales at freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com From paul.muaddib83 at gmail.com Tue Jul 3 20:27:06 2018 From: paul.muaddib83 at gmail.com (Paul Muaddib) Date: Tue, 3 Jul 2018 22:27:06 +0200 Subject: [Freeswitch-users] 481 Call does not Exit Message-ID: Hi, if I call someone, who accepts the call and then hangs up, the call does not end on my side Error message: 481 Call does not Exit What is the reason for this? Best regards, Paul -------------- next part -------------- An HTML attachment was scrubbed... URL: From paul.muaddib83 at gmail.com Tue Jul 3 20:34:27 2018 From: paul.muaddib83 at gmail.com (Paul Muaddib) Date: Tue, 3 Jul 2018 22:34:27 +0200 Subject: [Freeswitch-users] Failed Registration [908] In-Reply-To: References: Message-ID: My router was configured wrong. The MTU size was set incorrectly. This resulted in strange dropouts. Some things, e.g. websites worked and others didn't answer. That was a nasty mistake. That took me ages to figure out. Right now I'm connecting with siproxd to the outside. Unfortunately siproxd does not support DNS SRV. Does anyone have another idea how I make my installation work without forwarding ports in the firewall? I already read the NAT descriptions in the documentation. But many of the settings don't tell me anything and the description is not very understandable. Setup: Phone -> FS -> NAT -> Gateway. 2018-07-02 10:44 GMT+02:00 Paul Muaddib : > Now, I opened all firewall ports. I tried connecting to different SIP > Servers with Netcat and it established a connection. But I don't get any > response. > I read the Technical Specification 1TR118 from Deutsche Telekom and change > the gateway setting according to the specifications but no success ;( > > > > > > > > > > > > > > > > > > > > > > 2018-06-30 22:57 GMT+02:00 Paul Muaddib : > >> sofia loglevel all 9 >> >> ##################################################################### >> >> 2018-06-30 22:31:43.746032 [NOTICE] sofia_reg.c:448 Registering >> sip-trunk.telekom.de >> nua.c:622 nua_register() nua: nua_register: entering >> nua_stack.c:529 nua_signal() nua(0xb4e2ec50): sent signal r_register >> nua_stack.c:569 nua_stack_signal() nua(0xb4e2ec50): recv signal r_register >> nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: >> entering >> soa.c:280 soa_clone() soa_clone(static::0xb6255520, 0x88b30b0, >> 0xb4e2ec50) called >> soa.c:403 soa_set_params() soa_set_params(static::0xb5923410, ...) called >> soa.c:403 soa_set_params() soa_set_params(static::0xb5923410, ...) called >> nua_dialog.c:338 nua_dialog_usage_add() nua(0xb4e2ec50): adding register >> usage >> nta.c:4417 nta_leg_tcreate() nta_leg_tcreate(0xb5906508) >> nta.c:2665 nta_tpn_by_url() nta: selecting scheme sip >> sres_cache.c:272 sres_cache_get() sres_cache_get(0x88aa8e0, SRV, "_sip._ >> tcp.reg.sip-trunk.telekom.de.") called >> sres_cache.c:318 sres_cache_get() sres_cache_get(0x88aa8e0, SRV, "_sip._ >> tcp.reg.sip-trunk.telekom.de.") returned 3 entries >> nta.c:10598 outgoing_query_srv() nta: for "reg.sip-trunk.telekom.de" >> query "_sip._tcp.reg.sip-trunk.telekom.de" SRV (cached) >> nta.c:10694 outgoing_answer_srv() nta: _sip._tcp.reg.sip-trunk.telekom.de >> IN SRV 10 5 5060 d-ipr-a02.sip-trunk.telekom.de. (tcp) >> nta.c:10694 outgoing_answer_srv() nta: _sip._tcp.reg.sip-trunk.telekom.de >> IN SRV 11 5 5060 b-ipr-a01.sip-trunk.telekom.de. (tcp) >> nta.c:10694 outgoing_answer_srv() nta: _sip._tcp.reg.sip-trunk.telekom.de >> IN SRV 11 5 5060 b-ipr-a02.sip-trunk.telekom.de. (tcp) >> sres_cache.c:272 sres_cache_get() sres_cache_get(0x88aa8e0, A, " >> d-ipr-a02.sip-trunk.telekom.de.") called >> sres_cache.c:318 sres_cache_get() sres_cache_get(0x88aa8e0, A, " >> d-ipr-a02.sip-trunk.telekom.de.") returned 1 entries >> nta.c:10803 outgoing_query_a() nta: for "reg.sip-trunk.telekom.de" query >> "d-ipr-a02.sip-trunk.telekom.de." A (cached) >> nta.c:10856 outgoing_answer_a() nta: d-ipr-a02.sip-trunk.telekom.de. IN >> A 217.0.26.133 >> tport.c:4588 tport_by_name() tport(0xb625dc58): found 0x880d978 by name >> tcp/217.0.26.133:5060 >> tport.c:3257 tport_tsend() tport_tsend(0x880d978) tpn = tcp/ >> 217.0.26.133:5060 >> tport.c:3594 tport_vsend() tport_vsend(0x880d978): 677 bytes of 677 to >> tcp/217.0.26.133:5060 >> tport.c:3492 tport_send_msg() tport_vsend returned 677 >> tport.c:2296 tport_set_secondary_timer() tport(0x880d978): reset timer >> nta.c:8304 outgoing_send() nta: sent REGISTER (124859287) to tcp/ >> 217.0.26.133:5060 >> tport.c:4160 tport_pend() tport_pend(0x880d978): pending 0xb6263a08 for >> tcp/217.0.26.133:5060 (already 1) >> nta.c:8982 outgoing_timer_bf() nta: timer F fired, timeout REGISTER >> (124859241) >> nta.c:9035 outgoing_timeout() nta(0xb6264318): try next after timeout >> nta.c:10200 outgoing_graylist() nta: graylisting >> d-ipr-a02.sip-trunk.telekom.de.:5060;transport=tcp >> nta.c:10227 outgoing_graylist() nta: reduced priority of 1 _sip._ >> tcp.reg.sip-trunk.telekom.de SRV records (increase value to 11) >> sres_cache.c:272 sres_cache_get() sres_cache_get(0x88aa8e0, A, " >> reg.sip-trunk.telekom.de.") called >> nta.c:10803 outgoing_query_a() nta: for "reg.sip-trunk.telekom.de" query >> "reg.sip-trunk.telekom.de" A >> sres.c:968 sres_query() sres_query(0xb624a8b8, 0xb6264318, A, " >> reg.sip-trunk.telekom.de") called >> sres.c:2730 sres_send_dns_query() sres_send_dns_query(0xb624a8b8, >> 0x88653d0) called >> sres.c:2819 sres_send_dns_query() sres_send_dns_query(0xb624a8b8, >> 0x88653d0) id=12982 A reg.sip-trunk.telekom.de (to [217.237.148.70]:53) >> nta.c:8929 _nta_outgoing_timer() nta_outgoing_timer: 0/0 resent, 1/2 >> tout, 0/0 term, 0/2 free >> nta.c:1296 agent_timer() nta: timer set next to 29536 ms >> sres.c:3467 sres_resolver_receive() sres_resolver_receive(0xb624a8b8, >> 31) called >> sres.c:3781 sres_create_record() AUTHORITY RR received >> sip-trunk.telekom.de. SOA IN 151 rdlen=64 >> sres.c:3572 sres_log_response() sres_resolver_receive(0xb624a8b8, >> 0x88653d0) id=12982 (from [217.237.148.70]:53) >> sres.c:2987 sres_query_report_error() sres(q=0x88653d0): reporting error >> RECORD_ERR for A reg.sip-trunk.telekom.de >> tport.c:4222 tport_release() tport_release(0x880d978): 0xb623b2e8 by >> 0xb6264318 with (nil) >> nta.c:8722 outgoing_free() nta: outgoing_free(0xb6264318) >> nta.c:8982 outgoing_timer_bf() nta: timer F fired, timeout REGISTER >> (124859287) >> nta.c:9035 outgoing_timeout() nta(0xb628fb30): try next after timeout >> nta.c:10200 outgoing_graylist() nta: graylisting >> d-ipr-a02.sip-trunk.telekom.de.:5060;transport=tcp >> nta.c:10227 outgoing_graylist() nta: reduced priority of 1 _sip._ >> tcp.reg.sip-trunk.telekom.de SRV records (increase value to 12) >> sres_cache.c:272 sres_cache_get() sres_cache_get(0x88aa8e0, A, " >> b-ipr-a01.sip-trunk.telekom.de.") called >> sres_cache.c:318 sres_cache_get() sres_cache_get(0x88aa8e0, A, " >> b-ipr-a01.sip-trunk.telekom.de.") returned 1 entries >> nta.c:10803 outgoing_query_a() nta: for "reg.sip-trunk.telekom.de" query >> "b-ipr-a01.sip-trunk.telekom.de." A (cached) >> nta.c:10856 outgoing_answer_a() nta: b-ipr-a01.sip-trunk.telekom.de. IN >> A 217.0.26.163 >> tport.c:4588 tport_by_name() tport(0xb625dc58): found 0x87fa8a8 by name >> tcp/217.0.26.163:5060 >> tport.c:4222 tport_release() tport_release(0x880d978): 0xb6263a08 by >> 0xb628fb30 with (nil) >> tport.c:2296 tport_set_secondary_timer() tport(0x880d978): reset timer >> tport.c:3257 tport_tsend() tport_tsend(0x87fa8a8) tpn = tcp/ >> 217.0.26.163:5060 >> tport.c:3594 tport_vsend() tport_vsend(0x87fa8a8): 677 bytes of 677 to >> tcp/217.0.26.163:5060 >> tport.c:3492 tport_send_msg() tport_vsend returned 677 >> tport.c:2296 tport_set_secondary_timer() tport(0x87fa8a8): reset timer >> nta.c:8304 outgoing_send() nta: sent REGISTER (124859287) to tcp/ >> 217.0.26.163:5060 >> tport.c:4160 tport_pend() tport_pend(0x87fa8a8): pending 0xb6263a08 for >> tcp/217.0.26.163:5060 (already 0) >> nta.c:8929 _nta_outgoing_timer() nta_outgoing_timer: 0/0 resent, 1/1 >> tout, 0/0 term, 0/1 free >> nta.c:1296 agent_timer() nta: timer set next to 32000 ms >> 2018-06-30 22:32:44.083197 [WARNING] sofia_reg.c:484 Timeout Registering >> sip-trunk.telekom.de >> nua.c:921 nua_handle_destroy() nua: nua_handle_destroy: entering >> nua_stack.c:569 nua_stack_signal() nua(0xb4e2ec50): recv signal r_destroy >> nua_dialog.c:397 nua_dialog_usage_remove_at() nua(0xb4e2ec50): removing >> register usage >> nta.c:4470 nta_leg_destroy() nta_leg_destroy(0xb5906508) >> soa.c:356 soa_destroy() soa_destroy(static::0xb5923410) called >> nua_stack.c:529 nua_signal() nua(0xb4e2ec50): sent signal r_destroy >> 2018-06-30 22:32:45.123287 [WARNING] sofia_reg.c:505 sip-trunk.telekom.de >> Failed Registration [908], setting retry to 30 seconds. >> >> ############################################################ >> ################# >> >> 2018-06-30 14:18 GMT+02:00 Paul Muaddib : >> >>> Hi, >>> >>> can someone please look at it and tell my what I am doing wrong? >>> >>> NOTICE] sofia_reg.c:448 Registering sip-trunk.telekom.de >>> [WARNING] sofia_reg.c:484 Timeout Registering sip-trunk.telekom.de >>> [WARNING] sofia_reg.c:505 sip-trunk.telekom.de Failed Registration >>> [908], setting retry to 30 seconds. >>> >>> My setup: >>> ######## >>> >>> I have a static IP address for my router. My router does not have UPnP >>> or NAT-PMP. >>> Phones and Freeswitch server are on the same local network. The local >>> network is behind NAT >>> >>> PHONE -> FS -> NAT -> Public Internet >>> I only want to connect to the PSTN via my sip gateway provider. >>> >>> According to this manual I did the following setting: >>> https://freeswitch.org/confluence/display/FREESWITCH/NAT+Traversal >>> >>> vars.xml >>> ###### >>> >>> >>> >>> internal.xml >>> ######### >>> >>> >>> >>> external.xml >>> ########## >>> >>> >>> >>> >>> ( What is a little bit strange though is that nat_map status is not >>> working >>> [ERR] mod_commands.c:751 nat_map API called while NAT not initialized ) >>> >>> Is the reason why it does not work the port specification in the Contact >>> field (sngrep output)? >>> Contact: >> ip-trunk.telekom.de> >>> >>> I do not want to setup port forwarding in the firewall. The connection >>> should remain open via a keep alive signal. >>> >>> sngrep output >>> ########### >>> >>> 2018/06/30 13:52:37.211946 10.0.200.2:51813 -> 217.0.26.165:5060 >>> REGISTER sip:reg.sip-trunk.telekom.de;transport=tcp SIP/2.0 >>> Via: SIP/2.0/TCP X.X.X.X:5080;rport;branch=z2hG4bK588cZXSvrN1De >>> Max-Forwards: 70 >>> From: ;tag=9aldsj8n67a >>> To: >>> Call-ID: >>> CSeq: 111844514 REGISTER >>> Contact: >> ip-trunk.telekom.de> >>> Expires: 3600 >>> User-Agent: FreeSWITCH-mod_sofia/1.6.19+git~20170927T175834Z~38f568d343~ >>> 32bit >>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, >>> REGISTER, REFER, NOTIFY >>> Supported: timer, path, replaces >>> Content-Length: 0 >>> >>> Provider settings: >>> ############# >>> >>> output proxy: reg.sip-trunk.telekom.de >>> registrar: sip-trunk.telekom.de >>> >>> sip_profiles/external/telekom_voip.xml >>> ############################## >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> Firewall settings >>> ############# >>> >>> # block all traffic in all directions >>> block log >>> >>> # loopback interface is not filtered >>> set skip on lo0 >>> >>> voip_tcp_client_out = "{ssh, ftp, domain, http, https, sip}" >>> voip_udp_client_out = "{domain, ntp, sip, 3478}" >>> >>> match out on $wan_if from any nat-to ($wan_if) >>> >>> pass on $voip_if inet proto tcp from any to port $voip_tcp_client_out >>> pass on $voip_if inet proto udp from any to port $voip_udp_client_out >>> >>> pass out on $wan_if inet proto tcp from any to port $tcp_client_out >>> pass out on $wan_if inet proto udp from any to port $udp_client_out >>> >>> >>> Thank you for helping :) >>> >>> Regards, >>> Paul >>> >>> >>> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From tculjaga at gmail.com Mon Jul 2 18:21:28 2018 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Mon, 2 Jul 2018 20:21:28 +0200 Subject: [Freeswitch-users] OPUS recording - playback In-Reply-To: References: Message-ID: coz its not an opus file :=) there is no support in FS for opus recording. On 2 July 2018 at 11:32, Marcin Kowalczyk wrote: > Hi, > > I have a call with both legs OPUS encoded. I've enabled call recording > with > uuid_record file.opus > After call was finished I have file.opus but I'm unable to play it back. > Tried with VLC but no luck, same goes with opus_decode - it does not > recognize file to convert it to wav. > Can you guide me how to playback freeswitch recorded .opus file ? > > Regards > > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com > -------------- next part -------------- An HTML attachment was scrubbed... URL: From tculjaga at gmail.com Mon Jul 2 18:24:56 2018 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Mon, 2 Jul 2018 20:24:56 +0200 Subject: [Freeswitch-users] Recording calls In-Reply-To: References: Message-ID: hi Gregor, that was always a 2 step process for me. 1st dump it to a file system ( ram disk if you can ) 2nd on record_stop event, move that file to a storage location by a separate application outside freeswitch. T. On 27 June 2018 at 16:38, Gregor Nanger wrote: > Hi! > > One question. > > Can I record calls with http_api or http_cache modules and post it to > webserver? > > If yes, does anyone one have any example? > > Best regards, Gregor > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com > -------------- next part -------------- An HTML attachment was scrubbed... URL: From blackholdmailer at gmail.com Mon Jul 2 20:20:14 2018 From: blackholdmailer at gmail.com (Blackhold) Date: Mon, 2 Jul 2018 22:20:14 +0200 Subject: [Freeswitch-users] problem loading fs_curl (contrib) Message-ID: Hi, I'm adding mod fs_curl from intralanman[1]. I have freeswitch installed from git[2]. I followed this manual[3] to install fs_curl. It seems that pgsql connection is ok, but when configuring users to DB, list_clients doesn't shows users from database (I also tried to load FS without users file configuration). Looking at logs it appears that: 2018-07-02 19:20:49.250075 [CRIT] switch_loadable_module.c:1522 Error Loading module /usr/local/freeswitch/mod/mod_xml_curl.so **/usr/local/freeswitch/mod/mod_xml_curl.so: cannot open shared object file: No such file or directory** Oviously, I could not install freeswitch-mod-xml-curl package, so I simply downloaded the .deb package[4] and extracted file mod_xml_curl.so to /usr/local/freeswitch/mod/ directory. Then it appeared an other error that I can't lead 2018-07-02 21:16:47.207969 [CRIT] switch_loadable_module.c:1522 Error Loading module /usr/local/freeswitch/mod/mod_xml_curl.so **libssl.so.1.0.0: cannot open shared object file: No such file or directory** I tried to create symlinks calling libssl.so.1.0.0 but it seems it have some problem to read it. The libraries that I have are libssl.so.1.1 and libssl.so.1.0.2 and are located at /usr/lib/x86_64-linux-gnu the error continues, then I tried to install libssl that contains libssl.so.1.0.0[5] and finally get mod_xml-curl activates, then appeared and other error: 2018-07-02 21:52:23.411646 [ERR] mod_xml_curl.c:315 Received HTTP error 500 trying to fetch http://localhost/fs_curl/ data: [hostname=XXX§ion=directory&tag_name=&key_name=&key_value=&Event-Name=REQUEST_PARAMS&Core-UUID=XXX&FreeSWITCH-Hostname=XXX&FreeSWITCH-Switchname=XXX&FreeSWITCH-IPv4= X XX &FreeSWITCH-IPv6=%3A%3A1&Event-Date-Local=2018-07-02%2021%3A52%3A23&Event-Date-GMT=Mon,%2002%20Jul%202018%2019%3A52%3A23%20GMT&Event-Date-Timestamp=1530561143405060&Event-Calling-File=sofia.c&Event-Calling-Function=launch_sofia_worker_thread&Event-Calling-Line-Number=3048&Event-Sequence=35&purpose=gateways&profile=external] The apache saids me that got this error: [Mon Jul 02 21:55:23.817938 2018] [:error] [pid 27216] [client 10.228.192.219:38874] PHP Fatal error: Uncaught Error: Call to undefined function ereg_replace() in /var/www/html/fs_curl/index.php:13\nStack trace:\n#0 {main}\n thrown in /var/www/html/fs_curl/index.php on line 13 I fixed with preg_replace() and gives this error: [Mon Jul 02 22:05:53.353522 2018] [:error] [pid 27218] [client ::1:32960] PHP Warning: preg_replace(): No ending delimiter '^' found in /var/www/html/fs_curl/index.php on line 13 Changing ereg for preg, it only gives a warning and freeswitch don't throw any errors* [Mon Jul 02 22:05:53.353522 2018] [:error] [pid 27218] [client ::1:32960] PHP Warning: preg_replace(): No ending delimiter '^' found in /var/www/html/fs_curl/index.php on line 13 * yep it do, but on trying to create table schemas to db for example: switch_pgsql.c:415 Query (create index sr_call_id on sip_registrations (call_id)) returned PGRES_FATAL_ERROR Finally I executed fs_cli and listed users, but it appears empty. In database I have created 5 users. freeswitch at freeswitch> list_users userid|context|domain|group|contact|callgroup|effective_caller_id_name|effective_caller_id_number +OK # IN DB freeswitch_db=# select * from directory_domains; id | domain_name ----+-------------------- 4 | 10.XXX 1 | crustaci.mydomain.com (2 rows) freeswitch_db=# select * from directory; id | username | domain_id | cache ----+----------+-----------+-------- 1 | 1010 | 1 | 300000 2 | 1011 | 1 | 300000 3 | 20000 | 4 | 0 4 | 20001 | 4 | 0 5 | 20002 | 4 | 0 (4 rows) freeswitch_db=# select * from directory_params; id | directory_id | param_name | param_value ----+--------------+------------+------------- 1 | 3 | password | XXX 2 | 4 | password | XXX 3 | 4 | password | XXX (2 rows) Some ideas for why it is not working? Thanks you much! [1] https://freeswitch.org/stash/projects/FS/repos/freeswitch-contrib/browse/intralanman [2] http://blackhold.nusepas.com/2018/05/26/freeswitch-1-9-en-debian-8-psql/ [3] https://saevolgo.blogspot.com/2012/07/freeswitch-with-sip-users-in-mysql-mod.html [4] http://files.freeswitch.org/repo/deb/debian-unstable/pool/main/f/freeswitch/freeswitch-mod-xml-curl_1.9.0~1832~25e9376-1~jessie+1_amd64.deb [5] https://packages.debian.org/wheezy/amd64/libssl1.0.0/download - Blackhold http://blackhold.nusepas.com @blackhold_ ~> cal lluitar contra el fort per deixar de ser febles, i contra nosaltres mateixos quan siguem forts (Xirinacs) <°((( >< -------------- next part -------------- An HTML attachment was scrubbed... URL: From blackholdmailer at gmail.com Mon Jul 2 23:11:22 2018 From: blackholdmailer at gmail.com (Blackhold) Date: Tue, 3 Jul 2018 01:11:22 +0200 Subject: [Freeswitch-users] problem loading fs_curl (contrib) In-Reply-To: References: Message-ID: Hi! I finally resolved! I explain for those who could have similar problems :) I have fixed two things in fs_curl code: [index.php:13] //define('START_TIME', ereg_replace('^0\.([0-9]+) ([0-9]+)$', '\2.\1', microtime())); define('START_TIME', preg_replace('/^0\.(\d+) (\d+)$/', '\2.\1', microtime())); ereg is deprecated for php7[6], instead use preg_replace, and you have to change the string format [fs_directory.php:91] //$where_array[] = sprintf( "domain='%s'", $domain['id'] ); $where_array[] = sprintf( "domain_id='%s'", $domain['id'] ); the easyiest way is to change here in the code. When inserting table schemas the row in table is called domain_id, not domain. Changing that you can fix this warning message 2018-07-02 19:20:53.322313 [WARNING] switch_core.c:1611 Cannot locate domain 10.XXX Now I can enter to fs_cli and see users that are defined in database (none users config file has loaded) freeswitch at freeswitch-capa8> list_users userid|context|domain|group|contact|callgroup|effective_caller_id_name|effective_caller_id_number 20000||10.XXX|default|error/user_not_registered||| 20001||10.XXX|default|error/user_not_registered||| 1010||crustaci.domain.com|default|error/user_not_registered||| 1011||crustaci.domain.com|default|error/user_not_registered||| +OK That's all! :) [6] http://php.net/ereg_replace - Blackhold http://blackhold.nusepas.com @blackhold_ ~> cal lluitar contra el fort per deixar de ser febles, i contra nosaltres mateixos quan siguem forts (Xirinacs) <°((( >< 2018-07-02 22:20 GMT+02:00 Blackhold : > Hi, > I'm adding mod fs_curl from intralanman[1]. I have freeswitch installed > from git[2]. I followed this manual[3] to install fs_curl. > > It seems that pgsql connection is ok, but when configuring users to DB, > list_clients doesn't shows users from database (I also tried to load FS > without users file configuration). Looking at logs it appears that: > > 2018-07-02 19:20:49.250075 [CRIT] switch_loadable_module.c:1522 Error > Loading module /usr/local/freeswitch/mod/mod_xml_curl.so > **/usr/local/freeswitch/mod/mod_xml_curl.so: cannot open shared object > file: No such file or directory** > > Oviously, I could not install freeswitch-mod-xml-curl package, so I > simply downloaded the .deb package[4] and extracted file mod_xml_curl.so > to /usr/local/freeswitch/mod/ directory. > > Then it appeared an other error that I can't lead > > 2018-07-02 21:16:47.207969 [CRIT] switch_loadable_module.c:1522 Error > Loading module /usr/local/freeswitch/mod/mod_xml_curl.so > **libssl.so.1.0.0: cannot open shared object file: No such file or > directory** > > I tried to create symlinks calling libssl.so.1.0.0 but it seems it have > some problem to read it. The libraries that I have are libssl.so.1.1 and > libssl.so.1.0.2 and are located at /usr/lib/x86_64-linux-gnu > > the error continues, then I tried to install libssl that contains > libssl.so.1.0.0[5] and finally get mod_xml-curl activates, then appeared > and other error: > > 2018-07-02 21:52:23.411646 [ERR] mod_xml_curl.c:315 Received HTTP error > 500 trying to fetch http://localhost/fs_curl/ > data: [hostname=XXX§ion=directory&tag_name=&key_name=& > key_value=&Event-Name=REQUEST_PARAMS&Core-UUID=XXX& > FreeSWITCH-Hostname=XXX&FreeSWITCH-Switchname=XXX&FreeSWITCH-IPv4=X > XX&FreeSWITCH-IPv6=%3A%3A1&Event- > Date-Local=2018-07-02%2021%3A52%3A23&Event-Date-GMT=Mon,% > 2002%20Jul%202018%2019%3A52%3A23%20GMT&Event-Date- > Timestamp=1530561143405060&Event-Calling-File=sofia.c& > Event-Calling-Function=launch_sofia_worker_thread&Event- > Calling-Line-Number=3048&Event-Sequence=35&purpose= > gateways&profile=external] > > The apache saids me that got this error: > > [Mon Jul 02 21:55:23.817938 2018] [:error] [pid 27216] [client > 10.228.192.219:38874] PHP Fatal error: Uncaught Error: Call to undefined > function ereg_replace() in /var/www/html/fs_curl/index.php:13\nStack > trace:\n#0 {main}\n thrown in /var/www/html/fs_curl/index.php on line 13 > > I fixed with preg_replace() and gives this error: > > [Mon Jul 02 22:05:53.353522 2018] [:error] [pid 27218] [client ::1:32960] > PHP Warning: preg_replace(): No ending delimiter '^' found in > /var/www/html/fs_curl/index.php on line 13 > > Changing ereg for preg, it only gives a warning and freeswitch don't throw > any errors* > > [Mon Jul 02 22:05:53.353522 2018] [:error] [pid 27218] [client ::1:32960] > PHP Warning: preg_replace(): No ending delimiter '^' found in > /var/www/html/fs_curl/index.php on line 13 > > * yep it do, but on trying to create table schemas to db for example: > switch_pgsql.c:415 Query (create index sr_call_id on sip_registrations > (call_id)) returned PGRES_FATAL_ERROR > > Finally I executed fs_cli and listed users, but it appears empty. In > database I have created 5 users. > > freeswitch at freeswitch> list_users userid|context|domain|group| > contact|callgroup|effective_caller_id_name|effective_caller_id_number +OK > > # IN DB > > freeswitch_db=# select * from directory_domains; id | domain_name > ----+-------------------- 4 | 10.XXX 1 | crustaci.mydomain.com (2 rows) > > > freeswitch_db=# select * from directory; id | username | domain_id | cache > ----+----------+-----------+-------- 1 | 1010 | 1 | 300000 2 | 1011 | 1 | > 300000 3 | 20000 | 4 | 0 4 | 20001 | 4 | 0 > 5 | 20002 | 4 | 0 (4 rows) > > freeswitch_db=# select * from directory_params; > id | directory_id | param_name | param_value > ----+--------------+------------+------------- > 1 | 3 | password | XXX > 2 | 4 | password | XXX > 3 | 4 | password | XXX > (2 rows) > > Some ideas for why it is not working? > > Thanks you much! > > [1] https://freeswitch.org/stash/projects/FS/repos/ > freeswitch-contrib/browse/intralanman > [2] http://blackhold.nusepas.com/2018/05/26/freeswitch-1-9- > en-debian-8-psql/ > [3] https://saevolgo.blogspot.com/2012/07/freeswitch-with- > sip-users-in-mysql-mod.html > [4] http://files.freeswitch.org/repo/deb/debian-unstable/ > pool/main/f/freeswitch/freeswitch-mod-xml-curl_1.9.0~ > 1832~25e9376-1~jessie+1_amd64.deb > [5] https://packages.debian.org/wheezy/amd64/libssl1.0.0/download > > - Blackhold > http://blackhold.nusepas.com > @blackhold_ > ~> cal lluitar contra el fort per deixar de ser febles, i contra nosaltres > mateixos quan siguem forts (Xirinacs) > <°((( >< > -------------- next part -------------- An HTML attachment was scrubbed... URL: From blackholdmailer at gmail.com Tue Jul 3 00:27:56 2018 From: blackholdmailer at gmail.com (Blackhold) Date: Tue, 3 Jul 2018 02:27:56 +0200 Subject: [Freeswitch-users] problem loading fs_curl (contrib) In-Reply-To: References: Message-ID: here you have the manual http://blackhold.nusepas.com/2018/07/03/freeswitch-1-9-continuacion-mod-fs_curl-usuarios-almacenados-en-bbdd/ - Blackhold http://blackhold.nusepas.com @blackhold_ ~> cal lluitar contra el fort per deixar de ser febles, i contra nosaltres mateixos quan siguem forts (Xirinacs) <°((( >< 2018-07-03 1:11 GMT+02:00 Blackhold : > Hi! I finally resolved! > > I explain for those who could have similar problems :) > > I have fixed two things in fs_curl code: > > [index.php:13] > //define('START_TIME', ereg_replace('^0\.([0-9]+) ([0-9]+)$', '\2.\1', > microtime())); > define('START_TIME', preg_replace('/^0\.(\d+) (\d+)$/', '\2.\1', > microtime())); > > ereg is deprecated for php7[6], instead use preg_replace, and you have to > change the string format > > [fs_directory.php:91] > //$where_array[] = sprintf( "domain='%s'", $domain['id'] ); > $where_array[] = sprintf( "domain_id='%s'", $domain['id'] ); > > the easyiest way is to change here in the code. When inserting table > schemas the row in table is called domain_id, not domain. Changing that you > can fix this warning message > > 2018-07-02 19:20:53.322313 [WARNING] switch_core.c:1611 Cannot locate > domain 10.XXX > > Now I can enter to fs_cli and see users that are defined in database (none > users config file has loaded) > > freeswitch at freeswitch-capa8> list_users > userid|context|domain|group|contact|callgroup|effective_ > caller_id_name|effective_caller_id_number > 20000||10.XXX|default|error/user_not_registered||| > 20001||10.XXX|default|error/user_not_registered||| > 1010||crustaci.domain.com|default|error/user_not_registered||| > 1011||crustaci.domain.com|default|error/user_not_registered||| > > +OK > > That's all! :) > > [6] http://php.net/ereg_replace > > > - Blackhold > http://blackhold.nusepas.com > @blackhold_ > ~> cal lluitar contra el fort per deixar de ser febles, i contra nosaltres > mateixos quan siguem forts (Xirinacs) > <°((( >< > > 2018-07-02 22:20 GMT+02:00 Blackhold : > >> Hi, >> I'm adding mod fs_curl from intralanman[1]. I have freeswitch installed >> from git[2]. I followed this manual[3] to install fs_curl. >> >> It seems that pgsql connection is ok, but when configuring users to DB, >> list_clients doesn't shows users from database (I also tried to load FS >> without users file configuration). Looking at logs it appears that: >> >> 2018-07-02 19:20:49.250075 [CRIT] switch_loadable_module.c:1522 Error >> Loading module /usr/local/freeswitch/mod/mod_xml_curl.so >> **/usr/local/freeswitch/mod/mod_xml_curl.so: cannot open shared object >> file: No such file or directory** >> >> Oviously, I could not install freeswitch-mod-xml-curl package, so I >> simply downloaded the .deb package[4] and extracted file mod_xml_curl.so >> to /usr/local/freeswitch/mod/ directory. >> >> Then it appeared an other error that I can't lead >> >> 2018-07-02 21:16:47.207969 [CRIT] switch_loadable_module.c:1522 Error >> Loading module /usr/local/freeswitch/mod/mod_xml_curl.so >> **libssl.so.1.0.0: cannot open shared object file: No such file or >> directory** >> >> I tried to create symlinks calling libssl.so.1.0.0 but it seems it have >> some problem to read it. The libraries that I have are libssl.so.1.1 and >> libssl.so.1.0.2 and are located at /usr/lib/x86_64-linux-gnu >> >> the error continues, then I tried to install libssl that contains >> libssl.so.1.0.0[5] and finally get mod_xml-curl activates, then appeared >> and other error: >> >> 2018-07-02 21:52:23.411646 [ERR] mod_xml_curl.c:315 Received HTTP error >> 500 trying to fetch http://localhost/fs_curl/ >> data: [hostname=XXX§ion=directory&tag_name=&key_name=&key_ >> value=&Event-Name=REQUEST_PARAMS&Core-UUID=XXX&FreeSWITCH- >> Hostname=XXX&FreeSWITCH-Switchname=XXX&FreeSWITCH-IPv4=X >> XX&FreeSWITCH-IPv6=%3A% >> 3A1&Event-Date-Local=2018-07-02%2021%3A52%3A23&Event-Date- >> GMT=Mon,%2002%20Jul%202018%2019%3A52%3A23%20GMT&Event-Date-Timestamp= >> 1530561143405060&Event-Calling-File=sofia.c&Event- >> Calling-Function=launch_sofia_worker_thread&Event-Calling- >> Line-Number=3048&Event-Sequence=35&purpose=gateways&profile=external] >> >> The apache saids me that got this error: >> >> [Mon Jul 02 21:55:23.817938 2018] [:error] [pid 27216] [client >> 10.228.192.219:38874] PHP Fatal error: Uncaught Error: Call to undefined >> function ereg_replace() in /var/www/html/fs_curl/index.php:13\nStack >> trace:\n#0 {main}\n thrown in /var/www/html/fs_curl/index.php on line 13 >> >> I fixed with preg_replace() and gives this error: >> >> [Mon Jul 02 22:05:53.353522 2018] [:error] [pid 27218] [client ::1:32960] >> PHP Warning: preg_replace(): No ending delimiter '^' found in >> /var/www/html/fs_curl/index.php on line 13 >> >> Changing ereg for preg, it only gives a warning and freeswitch don't >> throw any errors* >> >> [Mon Jul 02 22:05:53.353522 2018] [:error] [pid 27218] [client ::1:32960] >> PHP Warning: preg_replace(): No ending delimiter '^' found in >> /var/www/html/fs_curl/index.php on line 13 >> >> * yep it do, but on trying to create table schemas to db for example: >> switch_pgsql.c:415 Query (create index sr_call_id on sip_registrations >> (call_id)) returned PGRES_FATAL_ERROR >> >> Finally I executed fs_cli and listed users, but it appears empty. In >> database I have created 5 users. >> >> freeswitch at freeswitch> list_users userid|context|domain|group|co >> ntact|callgroup|effective_caller_id_name|effective_caller_id_number +OK >> >> # IN DB >> >> freeswitch_db=# select * from directory_domains; id | domain_name >> ----+-------------------- 4 | 10.XXX 1 | crustaci.mydomain.com (2 rows) >> >> >> freeswitch_db=# select * from directory; id | username | domain_id | >> cache ----+----------+-----------+-------- 1 | 1010 | 1 | 300000 2 | >> 1011 | 1 | 300000 3 | 20000 | 4 | 0 4 | 20001 | 4 | 0 >> 5 | 20002 | 4 | 0 (4 rows) >> >> freeswitch_db=# select * from directory_params; >> id | directory_id | param_name | param_value >> ----+--------------+------------+------------- >> 1 | 3 | password | XXX >> 2 | 4 | password | XXX >> 3 | 4 | password | XXX >> (2 rows) >> >> Some ideas for why it is not working? >> >> Thanks you much! >> >> [1] https://freeswitch.org/stash/projects/FS/repos/freeswitc >> h-contrib/browse/intralanman >> [2] http://blackhold.nusepas.com/2018/05/26/freeswitch-1-9-e >> n-debian-8-psql/ >> [3] https://saevolgo.blogspot.com/2012/07/freeswitch-with-si >> p-users-in-mysql-mod.html >> [4] http://files.freeswitch.org/repo/deb/debian-unstable/poo >> l/main/f/freeswitch/freeswitch-mod-xml-curl_1.9.0~1832~ >> 25e9376-1~jessie+1_amd64.deb >> [5] https://packages.debian.org/wheezy/amd64/libssl1.0.0/download >> >> - Blackhold >> http://blackhold.nusepas.com >> @blackhold_ >> ~> cal lluitar contra el fort per deixar de ser febles, i contra >> nosaltres mateixos quan siguem forts (Xirinacs) >> <°((( >< >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From s.safarov at gmail.com Thu Jul 5 07:57:00 2018 From: s.safarov at gmail.com (Sergey Safarov) Date: Thu, 5 Jul 2018 10:57:00 +0300 Subject: [Freeswitch-users] ignore_early_media Message-ID: Could anybody help me understand difference between "true" and "consume" vales of "ignore_early_media". Sure confluence description is not correct about "consume" value. -------------- next part -------------- An HTML attachment was scrubbed... URL: From avi at avimarcus.net Wed Jul 4 05:37:56 2018 From: avi at avimarcus.net (Avi Marcus) Date: Wed, 4 Jul 2018 05:37:56 +0000 Subject: [Freeswitch-users] Limiting call length In-Reply-To: References: Message-ID: <0100016463cc145a-171fb8a3-f05c-4a86-bdd7-d1f815a7d9ea-000000@email.amazonses.com> I'm not sure how it works with originate -- but there's two versions as per the docs . There's sched_hangup and calling it as an api. If you call it as an API, then you need to put the ${uuid} in, which I don't know if that's available in an originate unless you specify it. What do the logs say when the time expires? I have an api_on_answer in my bridge string, to only cut off the b leg: -Avi Marcus BestFone On Wed, Jul 4, 2018 at 1:12 AM Rick Jarvis wrote: > Hey all > > Looking for a way to limit all calls to a maximum of 10 minutes, when > originating a call using the event socket. > > I’ve tried using sched_hangup like this: > > originate > {origination_caller_id_number=01234567890,execute_on_answer=sched_hangup > +600 alloted_timeout}sofia/. . . . > > But no dice… > > R > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From chad at apartmentlines.com Wed Jul 4 00:27:52 2018 From: chad at apartmentlines.com (Chad Phillips) Date: Tue, 3 Jul 2018 17:27:52 -0700 Subject: [Freeswitch-users] About Verto In-Reply-To: References: <424FFF6C-73B4-4293-A4FE-8F45A24815CD@gmail.com> <695562B1-39BB-4816-B7C3-2EA9D1663389@gmail.com> Message-ID: At least Google is *trying* to follow the spec. I don’t think Apple gives a shit. Once the Chromium issue referenced in my previous post is fixed, I think they’ll be fully spec compliant on the codecs. On Tue, Jul 3, 2018 at 3:08 PM Giovanni Maruzzelli wrote: > Aaaand, let's say the truth: this is a battle between Apple and Google, > and the result is trillions of cpu.cycles wastqed in transcoding servers... > > On Mon, Jul 2, 2018, 19:06 Chad Phillips wrote: > >> For interop between all Android devices and iOS devices, I’m pretty sure >> you need both VP8 and H264 enabled for available codecs. >> >> Chrome doesn’t support H264 on all Android devices yet, there’s an open >> issue to add software H264 encoding in the case where the Android hardware >> doesn’t have hardware encoding: >> https://bugs.chromium.org/p/chromium/issues/detail?id=719023 >> >> On Sun, Jul 1, 2018 at 8:28 AM 王聡 wrote: >> >>> Thanks for your reply. >>> >>> I had added H264 codec in verto.conf.xml, and now it’s able to make call >>> between iOS users. >>> But it still didn’t work between Android and iOS users, is there any >>> suggestions? >>> >>> Regards. >>> >>> 在 2018年6月29日,00:00,Giovanni Maruzzelli 写道: >>> >>> On 28 June 2018 at 09:30, 王聡 wrote: >>> >>>> >>>> I had tried to build a WebRTC video call service via mod_verto and >>>> Verto Comunicator. >>>> Now I could make calls from (to) an Android device with Chrome, but it >>>> didn’t work on iOS with Safari. >>>> Is there any point to be modified to run on Safari? >>>> >>> >>> be sure to enable H264 codec, both in vars.conf.xml and in >>> verto.conf.xml >>> >>> >>> >>> >>> >>>> >>>> Regards. >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Services >>>> sales at freeswitch.com >>>> https://freeswitch.com >>>> >>>> Official FreeSWITCH Sites >>>> https://freeswitch.com/oss >>>> https://freeswitch.org/confluence >>>> https://cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> https://freeswitch.com >>> >>> >>> >>> >>> -- >>> >>> Sincerely, >>> >>> Giovanni Maruzzelli >>> OpenTelecom.IT >>> cell: +39 347 266 56 18 >>> _________________________________________________________________________ >>> Professional FreeSWITCH Services >>> sales at freeswitch.com >>> https://freeswitch.com >>> >>> Official FreeSWITCH Sites >>> https://freeswitch.com/oss >>> https://freeswitch.org/confluence >>> https://cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> https://freeswitch.com >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Services >>> sales at freeswitch.com >>> https://freeswitch.com >>> >>> Official FreeSWITCH Sites >>> https://freeswitch.com/oss >>> https://freeswitch.org/confluence >>> https://cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> https://freeswitch.com >> >> _________________________________________________________________________ >> Professional FreeSWITCH Services >> sales at freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From cong.wang.itsherpa at gmail.com Mon Jul 9 01:26:35 2018 From: cong.wang.itsherpa at gmail.com (=?utf-8?B?546L6IGh?=) Date: Mon, 9 Jul 2018 10:26:35 +0900 Subject: [Freeswitch-users] About Verto In-Reply-To: References: <424FFF6C-73B4-4293-A4FE-8F45A24815CD@gmail.com> <695562B1-39BB-4816-B7C3-2EA9D1663389@gmail.com> Message-ID: <24AF3D1D-A7B0-43BD-8A8F-C70AD98D05BF@gmail.com> Thanks for all advices, now I could make calls between Chrome and Safari successfully on official Verto Community client. However, the video size is not correct, it seems too large to display on mobile, is there any settings for video size? Besides, I had tried to make a custom verto client based on verto-min.js, but the SDP info is null when I want to make a call. Any suggestions? Error Message: [Error] ERROR: FSRTC audioEnabled: true constraints: {offerToReceiveAudio: true, offerToReceiveVideo: true} mediaData: {SDP: null, profile: {}, candidateList: []} options: {useVideo: