From jungleboogie0 at gmail.com Sat Dec 1 01:53:48 2018 From: jungleboogie0 at gmail.com (jungle boogie) Date: Fri, 30 Nov 2018 17:53:48 -0800 Subject: [Freeswitch-users] Effective_Caller_ID_Name set to Dialed name for internal Ext. In-Reply-To: References: Message-ID: <5c123512-6980-e9ec-c579-c537f16fec78@gmail.com> Thus said Abaci B on Fri, 23 Nov 2018 09:49:41 -0500 > Just add it to the dialplan you use to call users. > How would you have this work on users who are not in your directory? For example, calling voicemail at 4000 still shows the digits. From imfanee at gmail.com Sat Dec 1 06:41:33 2018 From: imfanee at gmail.com (Faisal Hanif) Date: Sat, 1 Dec 2018 11:41:33 +0500 Subject: [Freeswitch-users] Conference Video Mux Option doesn't display video but Video Muted (but audio not affected) instead In-Reply-To: References: Message-ID: Hi Alex, Thanks a lot for reply. I have adopted the suggestion and installed FreeSWITCH 1.8.2 from package on Debian 9 but I still have the same issue, FreeSWITCH Version 1.8.2-3-a98a958ac3~64bit (-3-a98a958ac3 64bit) Distributor ID: Debian Description: Debian GNU/Linux 9.6 (stretch) Release: 9.6 Codename: stretch This error I got in logs while joining the conference, 2018-12-01 06:33:43.020210 [NOTICE] switch_core_media.c:15575 Activating write resampler 2018-12-01 06:33:43.050211 [NOTICE] switch_vpx.c:486 VPX encoder reset (WxH/BW) from 0x0/0 to 1280x480/1024 2018-12-01 06:33:43.070188 [NOTICE] switch_vpx.c:486 VPX encoder reset (WxH/BW) from 1280x480/1024 to 1280x480/645 2018-12-01 06:33:43.070188 [ERR] switch_vpx.c:841 VPX encode error 8:Invalid parameter:(null) I really appreciate your cooperation. Regards, Faisal On Fri, 30 Nov 2018 at 16:02, Alexey Sibyakin wrote: > Hi, > > Try Debian 9 and FreeSWITCH 1.8.2 from official packages. If you are going > to use dev version on nonsupported OS you have to handle it yourself. > > Regards, > > Alex > > On Thu, Nov 29, 2018 at 1:16 AM Faisal Hanif wrote: > >> Hi Geeks, >> >> I am trying to implement a conference in mux mode and FreeSWITCH send >> canvas properly but never show video on but a pic "Video Muted (but audio >> not affected)" pic in place of every member's video on canvas. I tried a >> lot with no success :( >> >> OS : Ubuntu 14.04.5 LTS trusty >> FreeSWITCH Version 1.9.0+git~20181120T210412Z~968c76b29c~64bit (git >> 968c76b 2018-11-20 21:04:12Z 64bit) >> >> My conference profile is >> >> >> >> >> >> >> >> >> > value="tone_stream://%(200,0,500,600,700)"/> >> > value="tone_stream://%(500,0,300,200,100,50,25)"/> >> >> >> >> >> > value="audio-always|livearray-json-status"/> >> >> >> >> >> >> >> >> >> >> >> > value="/usr/local/freeswitch/conf/images/video-muted.png"/> >> > value="/usr/local/freeswitch/conf/images/video-muted.png"/> >> >> >> >> can anyone please help me. >> Regards, >> >> Faisal >> >> _________________________________________________________________________ >> 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 -- Regards, Faisal Hanif -------------- next part -------------- An HTML attachment was scrubbed... URL: From krice at freeswitch.org Sat Dec 1 12:05:39 2018 From: krice at freeswitch.org (Ken Rice) Date: Sat, 1 Dec 2018 06:05:39 -0600 Subject: [Freeswitch-users] early media + sip trunk In-Reply-To: References: <8CADAD5B-2D85-4B50-9C49-BA3C6CCB3FCB@freeswitch.org> Message-ID: You can not detect the machine before answer. That’s just not how voicemail works. You want to use mod_com_amd (available at the freeswitch.com website) for doing proper voicemail detection. From: FreeSWITCH-users On Behalf Of Tiago Galvão Gomes de Souza Sent: Friday, November 30, 2018 9:33 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] early media + sip trunk some can help me? Em ter, 27 de nov de 2018 às 20:36, Tiago Galvão Gomes de Souza > escreveu: I'm from brazil and here we have a lot of early media calls with machine answer(voicemail), I want to detect early media + machine answer and hangup this call , I read about avmd mod and it is a way to detect after answer call and i would like to detect machine before answer, it is possible with freeswitch? I tought that monitor_early_media_fail was a better way to do it but I don't know how use it because when i tried to use doesn't work, problably I'm using in a wrong way, I used AMD from asterisk to detect Answer machine after Answer... some can help me? Em seg, 26 de nov de 2018 às 08:00, Sergey Safarov > escreveu: Need to try detect voicemail tone. https://freeswitch.org/confluence/display/FREESWITCH/mod_avmd Or try play announcement after answer and check voice activity. If voice activity is high, then voicemail https://github.com/seanbright/mod_amd пн, 26 нояб. 2018 г. в 03:34, Ken Rice >: Voicemail is not earlier media. the call is answed. early media is things like ringing or other in band signaling methods like fast busy or ‘not in service’ recordings. Sent from my iPhone On Nov 24, 2018, at 09:03, Tiago Galvão Gomes de Souza > wrote: Hello friends, I would like to know how is the better way to block outbound calls with early media and sip trunk, I have a system for callcenter and I have a big problem with calls in voicemail entering to agents , I tried to use monitor_early_media_fail with ignore_early_media but I didn't have a sucessfull, it is possible to use this function to block it? How can i do it? My wish is to stop the call when i knew that is a voicemail call. -- Atenciosamente, Tiago Galvão Gomes de Souza. _________________________________________________________________________ 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 -- Atenciosamente, Tiago Galvão Gomes de Souza. -- Atenciosamente, Tiago Galvão Gomes de Souza. -------------- next part -------------- An HTML attachment was scrubbed... URL: From fs at voice2net.ca Sun Dec 2 02:49:59 2018 From: fs at voice2net.ca (Darcy Primrose) Date: Sat, 1 Dec 2018 21:49:59 -0500 Subject: [Freeswitch-users] Mutli-lingual ivr In-Reply-To: References: <1dd84c16-9337-711c-c2a3-344a3774a347@voice2net.ca> Message-ID: <96c1efc5-2b03-e8b9-4e9c-a2421387c588@voice2net.ca> It seems this only partially works, I had it set up correctly but was trying to play the invalid_entry, that plays in english, but if you do a call transfer, it plays your call is being transferred, that plays in french, so a mix and match on what works, or is there an error in my setup.  We have been using freeswitch pretty well since it's inception so we are pretty proficient with it. Darcy On 2018-11-30 5:44 a.m., Faisal Hanif wrote: > This as simple you just set channel variable > language={the-language-shortcode} before playing IVR and install that > language sound files. > > On Fri, Nov 30, 2018, 3:42 PM Darcy Primrose wrote: > > We are long time freeswitch users in an area of  Canada where > there is a > strong english french mix.  We have for some time be providing a > multi-lingual experience for the extensions/users quite > successfully but > have not been able to get the ivr to switch to a second > language.   Has > anyone had experience or success with this. > > We use en as our prime language code but when specific DIDs are > dialed, > we would like the IVR responses to be in french.  First, is this > possible in freeswitch and if so, does anyone have any hints on > how to > accomplish this. > > > Thanks > > Darcy Primrose > > Voice2Net > > > _________________________________________________________________________ > 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 imfanee at gmail.com Sun Dec 2 03:38:38 2018 From: imfanee at gmail.com (Faisal Hanif) Date: Sun, 2 Dec 2018 08:38:38 +0500 Subject: [Freeswitch-users] Mutli-lingual ivr In-Reply-To: <96c1efc5-2b03-e8b9-4e9c-a2421387c588@voice2net.ca> References: <1dd84c16-9337-711c-c2a3-344a3774a347@voice2net.ca> <96c1efc5-2b03-e8b9-4e9c-a2421387c588@voice2net.ca> Message-ID: Hmm.. I ideally it shouldn't be the situation. If you are playing "invalid_entry" from dialplan then please check console logs if it points to correct language folder but if this from some conference PIN then may need to use language variable in conference audio files path configuration parameters. On Sun, 2 Dec 2018 at 07:51, Darcy Primrose wrote: > It seems this only partially works, I had it set up correctly but was > trying to play the invalid_entry, that plays in english, but if you do a > call transfer, it plays your call is being transferred, that plays in > french, so a mix and match on what works, or is there an error in my > setup. We have been using freeswitch pretty well since it's inception so > we are pretty proficient with it. > > Darcy > > On 2018-11-30 5:44 a.m., Faisal Hanif wrote: > > This as simple you just set channel variable > language={the-language-shortcode} before playing IVR and install that > language sound files. > > On Fri, Nov 30, 2018, 3:42 PM Darcy Primrose >> We are long time freeswitch users in an area of Canada where there is a >> strong english french mix. We have for some time be providing a >> multi-lingual experience for the extensions/users quite successfully but >> have not been able to get the ivr to switch to a second language. Has >> anyone had experience or success with this. >> >> We use en as our prime language code but when specific DIDs are dialed, >> we would like the IVR responses to be in french. First, is this >> possible in freeswitch and if so, does anyone have any hints on how to >> accomplish this. >> >> >> Thanks >> >> Darcy Primrose >> >> Voice2Net >> >> >> _________________________________________________________________________ >> 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 Servicessales at freeswitch.comhttps://freeswitch.com > > Official FreeSWITCH Siteshttps://freeswitch.com/osshttps://freeswitch.org/confluencehttps://cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttps://freeswitch.com > > -- Regards, Faisal Hanif -------------- next part -------------- An HTML attachment was scrubbed... URL: From fs at voice2net.ca Sun Dec 2 03:58:29 2018 From: fs at voice2net.ca (Darcy Primrose) Date: Sat, 1 Dec 2018 22:58:29 -0500 Subject: [Freeswitch-users] Mutli-lingual ivr In-Reply-To: References: <1dd84c16-9337-711c-c2a3-344a3774a347@voice2net.ca> <96c1efc5-2b03-e8b9-4e9c-a2421387c588@voice2net.ca> Message-ID: <3bd3cd15-6b89-c248-7c06-a471f136d6fc@voice2net.ca> Message from log does point to en/us.  What led me astray was I was using that message to test the language and it seems it is the only one not working, I have gone thru the code etc and cannot see why it is rejecting it.  I am not playing it from the dial plan, when I go to an ivr, press an invalid digit it plays that message, the others appear to be ok. Darcy switch_ivr_play_say.c:1749 done playing file /usr/local/freeswitch/sounds/en/us/callie/ivr/ivr-that_was_an_invalid_entry.wav On 2018-12-01 10:38 p.m., Faisal Hanif wrote: > Hmm.. I ideally it shouldn't be the situation. If you are playing > "invalid_entry" from dialplan then please check console logs if it > points to correct language folder but if this from some conference PIN > then may need to use language variable in conference audio files path > configuration parameters. > > On Sun, 2 Dec 2018 at 07:51, Darcy Primrose > wrote: > > It seems this only partially works, I had it set up correctly but > was trying to play the invalid_entry, that plays in english, but > if you do a call transfer, it plays your call is being > transferred, that plays in french, so a mix and match on what > works, or is there an error in my setup.  We have been using > freeswitch pretty well since it's inception so we are pretty > proficient with it. > > Darcy > > > On 2018-11-30 5:44 a.m., Faisal Hanif wrote: >> This as simple you just set channel variable >> language={the-language-shortcode} before playing IVR and install >> that language sound files. >> >> On Fri, Nov 30, 2018, 3:42 PM Darcy Primrose > wrote: >> >> We are long time freeswitch users in an area of  Canada where >> there is a >> strong english french mix.  We have for some time be providing a >> multi-lingual experience for the extensions/users quite >> successfully but >> have not been able to get the ivr to switch to a second >> language.   Has >> anyone had experience or success with this. >> >> We use en as our prime language code but when specific DIDs >> are dialed, >> we would like the IVR responses to be in french. First, is this >> possible in freeswitch and if so, does anyone have any hints >> on how to >> accomplish this. >> >> >> Thanks >> >> Darcy Primrose >> >> Voice2Net >> >> >> _________________________________________________________________________ >> 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 > > > > -- > Regards, > > Faisal Hanif -------------- next part -------------- An HTML attachment was scrubbed... URL: From fs at voice2net.ca Sun Dec 2 04:00:27 2018 From: fs at voice2net.ca (Darcy Primrose) Date: Sat, 1 Dec 2018 23:00:27 -0500 Subject: [Freeswitch-users] Mutli-lingual ivr In-Reply-To: References: <1dd84c16-9337-711c-c2a3-344a3774a347@voice2net.ca> <96c1efc5-2b03-e8b9-4e9c-a2421387c588@voice2net.ca> Message-ID: Further, it does this in both the fusionpbx and in our own voice2net version, so something within the system for it seems. Darcy On 2018-12-01 10:38 p.m., Faisal Hanif wrote: > Hmm.. I ideally it shouldn't be the situation. If you are playing > "invalid_entry" from dialplan then please check console logs if it > points to correct language folder but if this from some conference PIN > then may need to use language variable in conference audio files path > configuration parameters. > > On Sun, 2 Dec 2018 at 07:51, Darcy Primrose > wrote: > > It seems this only partially works, I had it set up correctly but > was trying to play the invalid_entry, that plays in english, but > if you do a call transfer, it plays your call is being > transferred, that plays in french, so a mix and match on what > works, or is there an error in my setup.  We have been using > freeswitch pretty well since it's inception so we are pretty > proficient with it. > > Darcy > > > On 2018-11-30 5:44 a.m., Faisal Hanif wrote: >> This as simple you just set channel variable >> language={the-language-shortcode} before playing IVR and install >> that language sound files. >> >> On Fri, Nov 30, 2018, 3:42 PM Darcy Primrose > wrote: >> >> We are long time freeswitch users in an area of  Canada where >> there is a >> strong english french mix.  We have for some time be providing a >> multi-lingual experience for the extensions/users quite >> successfully but >> have not been able to get the ivr to switch to a second >> language.   Has >> anyone had experience or success with this. >> >> We use en as our prime language code but when specific DIDs >> are dialed, >> we would like the IVR responses to be in french. First, is this >> possible in freeswitch and if so, does anyone have any hints >> on how to >> accomplish this. >> >> >> Thanks >> >> Darcy Primrose >> >> Voice2Net >> >> >> _________________________________________________________________________ >> 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 > > > > -- > Regards, > > Faisal Hanif -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Sun Dec 2 13:21:20 2018 From: david.villasmil.work at gmail.com (David Villasmil) Date: Sun, 2 Dec 2018 13:21:20 +0000 Subject: [Freeswitch-users] FS not un-registering at shutdown In-Reply-To: References: Message-ID: Hello guys, Sorry for not updating this. Indeed, Fs does the unregister properly. But the ASG simply shuts down the instances without going through the shutdown process (ungrateful shutdown). I had to setup a notification pointing to a lambda which shuts it down properly. If anyone has a better way, I’d appreciate suggestions. David On Thu, 15 Nov 2018 at 15:50, David Villasmil < david.villasmil.work at gmail.com> wrote: > I’m double checking this, will let you know. > Thanks for replying > > On Thu, 15 Nov 2018 at 14:15, Brian West wrote: > >> I see it doing it, what are you registering to? and what transport is in >> use? >> >> sofia loglevel all 9, and I bet it tries and fails >> >> >> /b >> >> >> On Thu, Nov 15, 2018 at 7:10 AM David Villasmil < >> david.villasmil.work at gmail.com> wrote: >> >>> Hello guys, >>> >>> I was under the impression FS would un-register from gateways when >>> shutting down, is this not the case? I don't see it sending a REGISTER with >>> expires 0. >>> >>> Regards, >>> >>> David Villasmil >>> email: david.villasmil.work at gmail.com >>> phone: +34669448337 >>> ᐧ >>> >> _________________________________________________________________________ >>> 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 >> >> >> >> -- >> >> Brian West | Co-founder and Developer >> >> Need Commercial support? email sales at freeswitch.com >> >> FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 >> >> >> Email: brian at freeswitch.com >> >> Mobile: 918-424-9378 >> >> Website: https://www.FreeSWITCH.com >> >> [image: https://www.facebook.com/signalwireinc?src=email] >> [image: >> https://twitter.com/freeswitch] >> _________________________________________________________________________ >> 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 alex at freeswitch.com Mon Dec 3 03:06:38 2018 From: alex at freeswitch.com (Alexey Sibyakin) Date: Mon, 3 Dec 2018 12:06:38 +0900 Subject: [Freeswitch-users] Diversion Header in Dialplan In-Reply-To: References: Message-ID: Hello, Try *info* dialplan application, it will help you to examine all the channel variables. Also, Vanilla dialplan has a lot of examples of variable manipulations. Regards, Alex On Sat, Dec 1, 2018 at 2:31 AM Igor Potjevlesch wrote: > Hello freeswitch users, > > > > I’m discovering freeswitch, and I want to understand how to extract and > use the variable « sip_h_Diversion » in the Dialplan. > > I know this question has been asked more times in the mailing-list but I > didn’t understand the issue of the diverse topic who evocate this variable. > > Also I read the information on the wiki but I wasn’t able to know what I > have to do in my case. > > > > In my Dialplan , I tried to put : > > > > field="${sip_h_Diversion}"> > > > > > > > > to apply a modification on the Diversion Header, but this condition is > unrecognized. > > I hope someone could explain me, if I have to extract this variable before > to use it and how to simply do it ? > > > > Thanks, > Bests Regards, > Igor > _________________________________________________________________________ > 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 Mon Dec 3 03:15:24 2018 From: alex at freeswitch.com (Alexey Sibyakin) Date: Mon, 3 Dec 2018 12:15:24 +0900 Subject: [Freeswitch-users] Change Gateway Context to receive calls In-Reply-To: References: Message-ID: Hello, That's expected behavior. If you need extra flexibility with ACLs you can check them real-time in the Dialplan (e.g. in Public context) and to Transfer on match. https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools:+check_acl https://freeswitch.org/confluence/display/FREESWITCH/ACL (check example at very bottom) Regards, Alex On Sat, Dec 1, 2018 at 3:34 AM Caio Assis wrote: > Hello! > > I have to configure a Gateway to receive calls from it and process it > correctly. The problem is that I haven't found a way to change the context, > and, when I receive a call from a Peer with the same IP as the Gateway, > FreeSwitch ignores the authentication from the Peer and checks if the IP is > allowed in ACL domains. > > Call comes from authenticated Peer 1000 though context CUSTOM -> OK > I have to configure a Gateway which has the same IP as the Peer 1000, so I > allow it in ACL. > Then, all calls from Peer 1000 come through PUBLIC context. > > Thanks. > _________________________________________________________________________ > 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 Mon Dec 3 03:18:37 2018 From: alex at freeswitch.com (Alexey Sibyakin) Date: Mon, 3 Dec 2018 12:18:37 +0900 Subject: [Freeswitch-users] Effective_Caller_ID_Name set to Dialed name for internal Ext. In-Reply-To: <5c123512-6980-e9ec-c579-c537f16fec78@gmail.com> References: <5c123512-6980-e9ec-c579-c537f16fec78@gmail.com> Message-ID: Try this one: https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools%3A+send+display Worked fine for me with Yealink devices. Regards, Alex On Sun, Dec 2, 2018 at 6:59 AM jungle boogie wrote: > Thus said Abaci B on Fri, 23 Nov 2018 09:49:41 -0500 > > Just add it to the dialplan you use to call users. > > > > How would you have this work on users who are not in your directory? For > example, calling voicemail at 4000 still shows the digits. > > _________________________________________________________________________ > 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 Mon Dec 3 03:22:25 2018 From: alex at freeswitch.com (Alexey Sibyakin) Date: Mon, 3 Dec 2018 12:22:25 +0900 Subject: [Freeswitch-users] Conference Video Mux Option doesn't display video but Video Muted (but audio not affected) instead In-Reply-To: References: Message-ID: That's odd. Please issue a Jira. https://freeswitch.org/confluence/display/FREESWITCH/Reporting+Bugs+to+JIRA Regards, Alex On Sun, Dec 2, 2018 at 7:40 AM Faisal Hanif wrote: > Hi Alex, > > Thanks a lot for reply. I have adopted the suggestion and installed > FreeSWITCH 1.8.2 from package on Debian 9 but I still have the same issue, > > FreeSWITCH Version 1.8.2-3-a98a958ac3~64bit (-3-a98a958ac3 64bit) > Distributor ID: Debian > Description: Debian GNU/Linux 9.6 (stretch) > Release: 9.6 > Codename: stretch > > This error I got in logs while joining the conference, > > 2018-12-01 06:33:43.020210 [NOTICE] switch_core_media.c:15575 Activating > write resampler > 2018-12-01 06:33:43.050211 [NOTICE] switch_vpx.c:486 VPX encoder reset > (WxH/BW) from 0x0/0 to 1280x480/1024 > 2018-12-01 06:33:43.070188 [NOTICE] switch_vpx.c:486 VPX encoder reset > (WxH/BW) from 1280x480/1024 to 1280x480/645 > 2018-12-01 06:33:43.070188 [ERR] switch_vpx.c:841 VPX encode error > 8:Invalid parameter:(null) > > > I really appreciate your cooperation. > > Regards, > > Faisal > > On Fri, 30 Nov 2018 at 16:02, Alexey Sibyakin wrote: > >> Hi, >> >> Try Debian 9 and FreeSWITCH 1.8.2 from official packages. If you are >> going to use dev version on nonsupported OS you have to handle it yourself. >> >> Regards, >> >> Alex >> >> On Thu, Nov 29, 2018 at 1:16 AM Faisal Hanif wrote: >> >>> Hi Geeks, >>> >>> I am trying to implement a conference in mux mode and FreeSWITCH send >>> canvas properly but never show video on but a pic "Video Muted (but audio >>> not affected)" pic in place of every member's video on canvas. I tried a >>> lot with no success :( >>> >>> OS : Ubuntu 14.04.5 LTS trusty >>> FreeSWITCH Version 1.9.0+git~20181120T210412Z~968c76b29c~64bit (git >>> 968c76b 2018-11-20 21:04:12Z 64bit) >>> >>> My conference profile is >>> >>> >>> >>> >>> >>> >>> >>> >>> >> value="tone_stream://%(200,0,500,600,700)"/> >>> >> value="tone_stream://%(500,0,300,200,100,50,25)"/> >>> >>> >>> >>> >>> >> value="audio-always|livearray-json-status"/> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >> value="/usr/local/freeswitch/conf/images/video-muted.png"/> >>> >> value="/usr/local/freeswitch/conf/images/video-muted.png"/> >>> >>> >>> >>> can anyone please help me. >>> Regards, >>> >>> Faisal >>> >>> _________________________________________________________________________ >>> 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 > > > > -- > Regards, > > Faisal Hanif > _________________________________________________________________________ > 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 udy786 at gmail.com Mon Dec 3 10:59:56 2018 From: udy786 at gmail.com (Uday kumar) Date: Mon, 3 Dec 2018 16:29:56 +0530 Subject: [Freeswitch-users] Call recording start if call answer time more than 30sec Message-ID: Hello All, I know that we can start call recording *execute_on_answer* but can we start call recording after 30sec once call answered? Reason behind to implement this, extensions getting more than 50 calls in a day and some are not important for agent so they hangup in less 30sec so recording is also not required for such calls. They do normally introduction in 30sec so that is not important. So this is the reason that we would like to start call recording after 30sec only. *Is this possible with Freeswitch?* -- Thanks & Regard Uday. Mobile:- +91-9377579349 -------------- next part -------------- An HTML attachment was scrubbed... URL: From tiagoggsouza at gmail.com Mon Dec 3 15:46:12 2018 From: tiagoggsouza at gmail.com (=?UTF-8?Q?Tiago_Galv=C3=A3o_Gomes_de_Souza?=) Date: Mon, 3 Dec 2018 13:46:12 -0200 Subject: [Freeswitch-users] early media + sip trunk In-Reply-To: References: <8CADAD5B-2D85-4B50-9C49-BA3C6CCB3FCB@freeswitch.org> Message-ID: I would like to detect any audio from early media and hangup, could be more simple is this way.. do you know how? Em sáb, 1 de dez de 2018 às 20:26, Ken Rice escreveu: > You can not detect the machine before answer. That’s just not how > voicemail works. > > > > You want to use mod_com_amd (available at the freeswitch.com website) for > doing proper voicemail detection. > > > > *From:* FreeSWITCH-users *On > Behalf Of *Tiago Galvão Gomes de Souza > *Sent:* Friday, November 30, 2018 9:33 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] early media + sip trunk > > > > some can help me? > > > > Em ter, 27 de nov de 2018 às 20:36, Tiago Galvão Gomes de Souza < > tiagoggsouza at gmail.com> escreveu: > > I'm from brazil and here we have a lot of early media calls with machine > answer(voicemail), I want to detect early media + machine answer and hangup > this call , I read about avmd mod and it is a way to detect after answer > call and i would like to detect machine before answer, it is possible with > freeswitch? I tought that monitor_early_media_fail was a better way to do > it but I don't know how use it because when i tried to use doesn't work, > problably I'm using in a wrong way, I used AMD from asterisk to detect > Answer machine after Answer... some can help me? > > > > > > > > Em seg, 26 de nov de 2018 às 08:00, Sergey Safarov > escreveu: > > Need to try detect voicemail tone. > https://freeswitch.org/confluence/display/FREESWITCH/mod_avmd > > Or try play announcement after answer and check voice activity. If voice > activity is high, then voicemail > https://github.com/seanbright/mod_amd > > > > > > пн, 26 нояб. 2018 г. в 03:34, Ken Rice : > > Voicemail is not earlier media. the call is answed. early media is things > like ringing or other in band signaling methods like fast busy or ‘not in > service’ recordings. > > Sent from my iPhone > > > On Nov 24, 2018, at 09:03, Tiago Galvão Gomes de Souza < > tiagoggsouza at gmail.com> wrote: > > Hello friends, > > > > I would like to know how is the better way to block outbound calls with > early media and sip trunk, I have a system for callcenter and I have a big > problem with calls in voicemail entering to agents , I tried to > use monitor_early_media_fail with ignore_early_media but I didn't have a > sucessfull, it is possible to use this function to block it? How can i do > it? > > > > My wish is to stop the call when i knew that is a voicemail call. > > > > > -- > > Atenciosamente, > > Tiago Galvão Gomes de Souza. > > _________________________________________________________________________ > 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 > > > > > -- > > Atenciosamente, > > Tiago Galvão Gomes de Souza. > > > > > -- > > Atenciosamente, > > Tiago Galvão Gomes de Souza. > _________________________________________________________________________ > 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 -- Atenciosamente, Tiago Galvão Gomes de Souza. -------------- next part -------------- An HTML attachment was scrubbed... URL: From ap at gen-ip.fr Mon Dec 3 16:11:23 2018 From: ap at gen-ip.fr (Alexis) Date: Mon, 3 Dec 2018 17:11:23 +0100 Subject: [Freeswitch-users] Call recording start if call answer time more than 30sec In-Reply-To: References: Message-ID: <338746b3-e0e3-24b2-a2ef-60819aecf722@gen-ip.fr> Hi, Check this : https://freeswitch.org/confluence/display/FREESWITCH/RECORD_MIN_SEC Alexis Le 03/12/2018 à 11:59, Uday kumar a écrit : > Hello All, > > I know that we can start call recording *execute_on_answer* but can we > start call recording after 30sec once call answered? > > Reason behind to implement this, extensions getting more than 50 calls > in a day and some are not important for agent so they hangup in less > 30sec so recording is also not required for such calls. They do > normally introduction in 30sec so that is not important. So this is > the reason that we would like to start call recording after 30sec only. > > *Is this possible with Freeswitch?* > > -- > Thanks & Regard > Uday. > Mobile:- +91-9377579349 -------------- next part -------------- An HTML attachment was scrubbed... URL: From udy786 at gmail.com Mon Dec 3 16:25:36 2018 From: udy786 at gmail.com (Uday kumar) Date: Mon, 3 Dec 2018 21:55:36 +0530 Subject: [Freeswitch-users] Call recording start if call answer time more than 30sec In-Reply-To: <338746b3-e0e3-24b2-a2ef-60819aecf722@gen-ip.fr> References: <338746b3-e0e3-24b2-a2ef-60819aecf722@gen-ip.fr> Message-ID: Thank you for reply. I will try as you suggested. On Mon, Dec 3, 2018 at 9:43 PM Alexis wrote: > Hi, > > Check this : > https://freeswitch.org/confluence/display/FREESWITCH/RECORD_MIN_SEC > > Alexis > > Le 03/12/2018 à 11:59, Uday kumar a écrit : > > Hello All, > > I know that we can start call recording *execute_on_answer* but can we > start call recording after 30sec once call answered? > > Reason behind to implement this, extensions getting more than 50 calls in > a day and some are not important for agent so they hangup in less 30sec so > recording is also not required for such calls. They do normally > introduction in 30sec so that is not important. So this is the reason that > we would like to start call recording after 30sec only. > > *Is this possible with Freeswitch?* > > -- > Thanks & Regard > Uday. > Mobile:- +91-9377579349 > > > _________________________________________________________________________ > 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 -- Thanks & Regard Uday. Mobile:- +91-9377579349 -------------- next part -------------- An HTML attachment was scrubbed... URL: From imfanee at gmail.com Mon Dec 3 19:20:07 2018 From: imfanee at gmail.com (Faisal Hanif) Date: Tue, 4 Dec 2018 00:20:07 +0500 Subject: [Freeswitch-users] Call recording start if call answer time more than 30sec In-Reply-To: References: Message-ID: As I know there is not a streight way to do it however an easy trick is to use a small script in hangup_hook which checks call duration and remove short duration recording. Regards, Faisal On Mon, Dec 3, 2018, 9:17 PM Uday kumar Hello All, > > I know that we can start call recording *execute_on_answer* but can we > start call recording after 30sec once call answered? > > Reason behind to implement this, extensions getting more than 50 calls in > a day and some are not important for agent so they hangup in less 30sec so > recording is also not required for such calls. They do normally > introduction in 30sec so that is not important. So this is the reason that > we would like to start call recording after 30sec only. > > *Is this possible with Freeswitch?* > > -- > Thanks & Regard > Uday. > Mobile:- +91-9377579349 > _________________________________________________________________________ > 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 joel at textplus.com Mon Dec 3 23:39:50 2018 From: joel at textplus.com (Joel Serrano) Date: Mon, 3 Dec 2018 15:39:50 -0800 Subject: [Freeswitch-users] Call recording start if call answer time more than 30sec In-Reply-To: References: Message-ID: You can also schedule the task of “start recording” to +30s after the “call has been answered”. That should achieve nicely exactly what you need. On Mon, Dec 3, 2018 at 14:10 Faisal Hanif wrote: > As I know there is not a streight way to do it however an easy trick is to > use a small script in hangup_hook which checks call duration and remove > short duration recording. > > Regards, > > Faisal > > On Mon, Dec 3, 2018, 9:17 PM Uday kumar >> Hello All, >> >> I know that we can start call recording *execute_on_answer* but can we >> start call recording after 30sec once call answered? >> >> Reason behind to implement this, extensions getting more than 50 calls in >> a day and some are not important for agent so they hangup in less 30sec so >> recording is also not required for such calls. They do normally >> introduction in 30sec so that is not important. So this is the reason that >> we would like to start call recording after 30sec only. >> >> *Is this possible with Freeswitch?* >> >> -- >> Thanks & Regard >> Uday. >> Mobile:- +91-9377579349 >> _________________________________________________________________________ >> 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 joel at textplus.com Tue Dec 4 02:43:13 2018 From: joel at textplus.com (Joel Serrano) Date: Mon, 3 Dec 2018 18:43:13 -0800 Subject: [Freeswitch-users] Call recording start if call answer time more than 30sec In-Reply-To: References: Message-ID: Hit send too soon... (on phone sorry) Check for “sched_api” docs in mod_commands confluence page... combine that with “execute_on_answer” On Mon, Dec 3, 2018 at 15:39 Joel Serrano wrote: > You can also schedule the task of “start recording” to +30s after the > “call has been answered”. > > That should achieve nicely exactly what you need. > > On Mon, Dec 3, 2018 at 14:10 Faisal Hanif wrote: > >> As I know there is not a streight way to do it however an easy trick is >> to use a small script in hangup_hook which checks call duration and remove >> short duration recording. >> >> Regards, >> >> Faisal >> >> On Mon, Dec 3, 2018, 9:17 PM Uday kumar > >>> Hello All, >>> >>> I know that we can start call recording *execute_on_answer* but can we >>> start call recording after 30sec once call answered? >>> >>> Reason behind to implement this, extensions getting more than 50 calls >>> in a day and some are not important for agent so they hangup in less 30sec >>> so recording is also not required for such calls. They do normally >>> introduction in 30sec so that is not important. So this is the reason that >>> we would like to start call recording after 30sec only. >>> >>> *Is this possible with Freeswitch?* >>> >>> -- >>> Thanks & Regard >>> Uday. >>> Mobile:- +91-9377579349 >>> _________________________________________________________________________ >>> 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 josedavid at zennio.com Tue Dec 4 08:29:14 2018 From: josedavid at zennio.com (Jose David Jurado Alonso) Date: Tue, 4 Dec 2018 09:29:14 +0100 Subject: [Freeswitch-users] Source device doesn't hangup videocall when the target device hangs Message-ID: Hi! I call from a intercom (video door) to a mobile but when the mobile hangup the call, the intercom continues with the open call (it does not hang) and you have to press the button to hangup. That is, the video door phone does not hang when the call is hung from the mobile. I am using the external profile, intercom and mobile devices are in the same network but the server is in the cloud. The dialplan extension: ---------------------------------------------------------------------- ---------------------------------------------------------------------- Thanks! Regards, José David Jurado Alonso -------------- next part -------------- An HTML attachment was scrubbed... URL: From vishalrigel1784 at gmail.com Tue Dec 4 10:02:20 2018 From: vishalrigel1784 at gmail.com (Vishal Dalsania) Date: Tue, 4 Dec 2018 15:32:20 +0530 Subject: [Freeswitch-users] Session Timers Message-ID: I am trying to make session timers work by adding below in internal.xml and all of my external profiles However i dont see any REINVITE being sent from FreeSwitch after 90 seconds. How can i make it work ? -------------- next part -------------- An HTML attachment was scrubbed... URL: From kowalma at gmail.com Tue Dec 4 13:12:43 2018 From: kowalma at gmail.com (Marcin Kowalczyk) Date: Tue, 4 Dec 2018 14:12:43 +0100 Subject: [Freeswitch-users] SIP-Info message - how to forward it inside profile Message-ID: Hi, I'm using custom SIP-Info message to send some info: uuid_setvar ec139363-a68d-4df0-9b23-a84d7811e12a fs_send_unsupported_info true uuid_send_info ec139363-a68d-4df0-9b23-a84d7811e12a text plain record and SIP info message is send out. It's easy to forward this INFO on Kamailio (as it's in-dialog message) but I'm unable to forward it inside freeswitch to B-leg. FS1 ----> INFO ---> FS2 ----> KAMAILIO FS2 is transcoder box ALAW -> OPUS and I'can force it to forward info message from FS1 back to Kamailio. Is there any way to do so? Regards. Marcin -------------- next part -------------- An HTML attachment was scrubbed... URL: From imfanee at gmail.com Tue Dec 4 14:06:09 2018 From: imfanee at gmail.com (Faisal Hanif) Date: Tue, 4 Dec 2018 19:06:09 +0500 Subject: [Freeswitch-users] SIP-Info message - how to forward it inside profile In-Reply-To: References: Message-ID: The most easy is to handle event of info message and trigger same on B-Leg On Tue, Dec 4, 2018, 6:40 PM Marcin Kowalczyk Hi, > > I'm using custom SIP-Info message to send some info: > > uuid_setvar ec139363-a68d-4df0-9b23-a84d7811e12a fs_send_unsupported_info > true > uuid_send_info ec139363-a68d-4df0-9b23-a84d7811e12a text plain record > > and SIP info message is send out. It's easy to forward this INFO on > Kamailio (as it's in-dialog message) but I'm unable to forward it inside > freeswitch to B-leg. > > FS1 ----> INFO ---> FS2 ----> KAMAILIO > > FS2 is transcoder box ALAW -> OPUS and I'can force it to forward info > message from FS1 back to Kamailio. Is there any way to do so? > > Regards. > Marcin > _________________________________________________________________________ > 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 imfanee at gmail.com Tue Dec 4 14:39:43 2018 From: imfanee at gmail.com (Faisal Hanif) Date: Tue, 4 Dec 2018 19:39:43 +0500 Subject: [Freeswitch-users] Source device doesn't hangup videocall when the target device hangs In-Reply-To: References: Message-ID: You need to check SIP traces to make sure if BYE received from mobile and sent to door phone On Tue, Dec 4, 2018, 6:40 PM Jose David Jurado Alonso Hi! > > I call from a intercom (video door) to a mobile but when the mobile hangup > the call, the intercom continues with the open call (it does not hang) and > you have to press the button to hangup. > > That is, the video door phone does not hang when the call is hung from the > mobile. > > I am using the external profile, intercom and mobile devices are in the > same network but the server is in the cloud. > > The dialplan extension: > ---------------------------------------------------------------------- > > > > > > > > > > > > > ---------------------------------------------------------------------- > > Thanks! > > Regards, > > José David Jurado Alonso > _________________________________________________________________________ > 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 denys.pozniak at gmail.com Tue Dec 4 16:24:19 2018 From: denys.pozniak at gmail.com (Denys Pozniak) Date: Tue, 4 Dec 2018 18:24:19 +0200 Subject: [Freeswitch-users] SIP-Info message - how to forward it inside profile In-Reply-To: References: Message-ID: Hello! I would try to use вт, 4 дек. 2018 г. в 17:05, Marcin Kowalczyk : > Hi, > > I'm using custom SIP-Info message to send some info: > > uuid_setvar ec139363-a68d-4df0-9b23-a84d7811e12a fs_send_unsupported_info > true > uuid_send_info ec139363-a68d-4df0-9b23-a84d7811e12a text plain record > > and SIP info message is send out. It's easy to forward this INFO on > Kamailio (as it's in-dialog message) but I'm unable to forward it inside > freeswitch to B-leg. > > FS1 ----> INFO ---> FS2 ----> KAMAILIO > > FS2 is transcoder box ALAW -> OPUS and I'can force it to forward info > message from FS1 back to Kamailio. Is there any way to do so? > > Regards. > Marcin > _________________________________________________________________________ > 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 -- BR, Denys Pozniak -------------- next part -------------- An HTML attachment was scrubbed... URL: From kaiduanx at yahoo.ca Tue Dec 4 19:48:24 2018 From: kaiduanx at yahoo.ca (kaiduan xie) Date: Tue, 4 Dec 2018 19:48:24 +0000 (UTC) Subject: Build DEB installer on Ubuntu from source References: <2060020577.1922923.1543952904961.ref@mail.yahoo.com> Message-ID: <2060020577.1922923.1543952904961@mail.yahoo.com> Hi, I managed to build, install and run Freeswitch (with some local changes) from source code on Ubuntu, what is the way to build an DEB installer from the source code so that I can install Freeswitch on other Ubuntu box without building/installing from source code? Thanks for help, /Kaiduan -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Tue Dec 4 20:52:32 2018 From: mike at jerris.com (Michael Jerris) Date: Tue, 4 Dec 2018 15:52:32 -0500 Subject: [Freeswitch-users] Build DEB installer on Ubuntu from source In-Reply-To: References: <2060020577.1922923.1543952904961.ref@mail.yahoo.com> Message-ID: If you look at the Debian docs you should be able to see the process, but no idea what needs to be done to make it work ob ubuntu, probably a bunch of dependency stuff. If you have patches that are required, please submit pull requests so we can get those merged. They need to be done in such a way that they dont break other platforms > On Dec 4, 2018, at 2:52 PM, kaiduan xie via FreeSWITCH-users wrote: > > > From: kaiduan xie > Subject: Build DEB installer on Ubuntu from source > Date: December 4, 2018 at 2:48:24 PM EST > To: Freeswitch Dev , FreeSWITCH Users Help > > > Hi, > > I managed to build, install and run Freeswitch (with some local changes) from source code on Ubuntu, what is the way to build an DEB installer from the source code so that I can install Freeswitch on other Ubuntu box without building/installing from source code? > > Thanks for help, > > /Kaiduan -------------- next part -------------- An HTML attachment was scrubbed... URL: From kaiduanx at yahoo.ca Tue Dec 4 21:05:05 2018 From: kaiduanx at yahoo.ca (kaiduan xie) Date: Tue, 4 Dec 2018 21:05:05 +0000 (UTC) Subject: [Freeswitch-users] Build DEB installer on Ubuntu from source In-Reply-To: References: <2060020577.1922923.1543952904961.ref@mail.yahoo.com> Message-ID: <1731937356.86252.1543957505836@mail.yahoo.com> Michael, Thanks a lot for the quick response, I will read the Debian docs first. /Kaiduan On Tuesday, December 4, 2018, 3:52:36 p.m. EST, Michael Jerris wrote: If you look at the Debian docs you should be able to see the process, but no idea what needs to be done to make it work ob ubuntu, probably a bunch of dependency stuff.  If you have patches that are required, please submit pull requests so we can get those merged.  They need to be done in such a way that they dont break other platforms On Dec 4, 2018, at 2:52 PM, kaiduan xie via FreeSWITCH-users wrote: From: kaiduan xie Subject: Build DEB installer on Ubuntu from source Date: December 4, 2018 at 2:48:24 PM EST To: Freeswitch Dev , FreeSWITCH Users Help Hi, I managed to build, install and run Freeswitch (with some local changes) from source code on Ubuntu, what is the way to build an DEB installer from the source code so that I can install Freeswitch on other Ubuntu box without building/installing from source code? Thanks for help, /Kaiduan -------------- next part -------------- An HTML attachment was scrubbed... URL: From asilva at wirelessmundi.com Wed Dec 5 11:02:42 2018 From: asilva at wirelessmundi.com (=?UTF-8?Q?Ant=c3=b3nio_Silva?=) Date: Wed, 5 Dec 2018 12:02:42 +0100 Subject: [Freeswitch-users] Memory increase because of multiple instances of freeswitch In-Reply-To: <3362de9e-1e03-85f4-b1b6-3721a16d81a6@wirelessmundi.com> References: <04fc4086-d678-2f65-ca06-059f40b89e22@wirelessmundi.com> <3362de9e-1e03-85f4-b1b6-3721a16d81a6@wirelessmundi.com> Message-ID: <158833cc-7981-421e-3a06-60e14b0f7b21@wirelessmundi.com> update on this issue, setting the parameter, threaded_system_exec = true, solved my problem, no more forks from fs and memory usage is stable. On 20/11/2018 17:23, António Silva wrote: > > Hi, > > I was getting the process list from the command: "ps auxf | grep > freeswitch" > > Still waiting to see it again on the server, i can't reproduce it, > when it happens again i post here the output. Not sure if enabling > threaded_system_exec solved the issue. > > On 19/11/2018 19:11, David Villasmil wrote: >> Totally true, i didn't think of that... @antonio, can you try `ps -ef`? >> Regards, >> >> David Villasmil >> email: david.villasmil.work at gmail.com >> >> phone: +34669448337 >> >> ᐧ >> >> On Mon, Nov 19, 2018 at 3:49 PM Ken Rice > > wrote: >> >> well also keep in mind that depending on how he is looking for >> freeswitch processes he may be seeing multiple as the threads >> will show up ad their individual psuedo PID for example in htop >> >> Sent from my iPhone >> >> On Nov 17, 2018, at 17:01, David Villasmil >> > > wrote: >> >>> Also, regarding memory usage, that's how FS works. It takes >>> memory AS NEEDED and simply doesn't returns it, it holds it for >>> future use. >>> Regards, >>> >>> David Villasmil >>> email: david.villasmil.work at gmail.com >>> >>> phone: +34669448337 >>> >>> ᐧ >>> >>> On Sat, Nov 17, 2018 at 11:00 PM David Villasmil >>> >> > wrote: >>> >>> I've never seen this... >>> >>> Is it possible the process is being started multiple times? >>> You _can_ run fs multiple times, but only the first would >>> start properly as the rest would try to bind to a port >>> already in use by the first process. >>> There's a parameter on all profiles to shutdown if it can't >>> start, try setting that and see what happens. >>> You should also check your crontab... This is NOT normal FS >>> behaviour, as far as i know. >>> >>> >>> Regards, >>> >>> David Villasmil >>> email: david.villasmil.work at gmail.com >>> >>> phone: +34669448337 >>> >>> ᐧ >>> >>> On Fri, Nov 9, 2018 at 7:48 PM António Silva via >>> FreeSWITCH-users >> > wrote: >>> >>> >>> >>> >>> ---------- Forwarded message ---------- >>> From: "António Silva" >> > >>> To: FreeSWITCH Users Help >>> >> > >>> Cc: >>> Bcc: >>> Date: Thu, 8 Nov 2018 17:53:03 +0100 >>> Subject: Re: Memory increase because of multiple >>> instances of freeswitch >>> Forget to put my unit configuration: >>> >>> [Service] >>> ; service >>> Type=forking >>> PIDFile=/run/freeswitch/freeswitch.pid >>> ExecStart=/usr/bin/freeswitch -ncwait -nonat -scripts >>> /scripts/fs >>> TimeoutSec=300s >>> Restart=on-failure >>> RestartSec=500ms >>> ; exec >>> User=root >>> Group=daemon >>> LimitCORE=infinity >>> LimitNOFILE=100000 >>> LimitNPROC=60000 >>> LimitSTACK=250000 >>> LimitRTPRIO=infinity >>> LimitRTTIME=infinity >>> IOSchedulingClass=realtime >>> IOSchedulingPriority=2 >>> CPUSchedulingPolicy=rr >>> CPUSchedulingPriority=89 >>> UMask=0007 >>> >>> >>> >>> On 08/11/2018 17:11, António Silva wrote: >>> > Hi all, >>> > >>> > I notice a strange behaviour on machine due to >>> increase of memory, >>> > when i went to see that was the process consuming the >>> memory i notice >>> > that freeswitch have multiple  process running: >>> > >>> > root      2543 90.4  4.8 2178716 1587484 ?     S>> Sep27 36619:46 >>> > /usr/bin/freeswitch -ncwait -nonat -scripts >>> /opt/commsmundi/scripts/fs >>> > root      7858  0.0  5.0 2289392 1649484 ?     SN   >>> Oct19   0:00 \_ >>> > /usr/bin/freeswitch -ncwait -nonat -scripts >>> /opt/commsmundi/scripts/fs >>> > root     30626  0.0  4.4 2172632 1464684 ?     SN   >>> Oct23   0:00 \_ >>> > /usr/bin/freeswitch -ncwait -nonat -scripts >>> /opt/commsmundi/scripts/fs >>> > root      4505  0.0  4.9 2336544 1621768 ?     SN   >>> 10:43   0:00 \_ >>> > /usr/bin/freeswitch -ncwait -nonat -scripts >>> /opt/commsmundi/scripts/fs >>> > root     22557  0.0  4.9 2336548 1636604 ?     SN   >>> 11:02   0:00 \_ >>> > /usr/bin/freeswitch -ncwait -nonat -scripts >>> /opt/commsmundi/scripts/fs >>> > >>> > >>> > and today: >>> > >>> > root     21823 93.9  4.8 2154128 1581064 ?     S>> Oct26 17781:45 >>> > /usr/bin/freeswitch -ncwait -nonat -scripts >>> /opt/commsmundi/scripts/fs >>> > root     18417  0.0  4.9 2247264 1610108 ?     SN   >>> 11:09   0:00 \_ >>> > /usr/bin/freeswitch -ncwait -nonat -scripts >>> /opt/commsmundi/scripts/fs >>> > >>> > >>> > what could be the cause of this? anyone experience the >>> same? >>> > >>> > As for the service everything is running ok, is just >>> that the memory >>> > keeps increasing.. i restart freewitch when i start to >>> reach my memory >>> > limit. >>> > >>> > >>> > I'm running fs 1.8.2 on debian jessie. >>> > >>> > >>> -- >>> Saludos / Regards / Cumprimentos >>> António Silva >>> >>> >>> >>> >>> >>> ---------- Forwarded message ---------- >>> From: "António Silva via FreeSWITCH-users" >>> >> > >>> To: FreeSWITCH Users Help >>> >> > >>> Cc: >>> Bcc: >>> Date: Fri, 09 Nov 2018 11:48:33 -0800 (PST) >>> Subject: Re: [Freeswitch-users] Memory increase because >>> of multiple instances of freeswitch >>> _________________________________________________________________________ >>> 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 > -- > Saludos / Regards / Cumprimentos > António Silva -- Saludos / Regards / Cumprimentos António Silva -------------- next part -------------- An HTML attachment was scrubbed... URL: From igor.potjevlesch at gmail.com Wed Dec 5 17:01:02 2018 From: igor.potjevlesch at gmail.com (igor.potjevlesch at gmail.com) Date: Wed, 5 Dec 2018 18:01:02 +0100 Subject: [Freeswitch-users] P-Asserted-Identity transparency between leg A and leg B Message-ID: <003501d48cbc$1ab66b90$502342b0$@gmail.com> Hello, In the scenario where the From is different than the P-Asserted-Identity on leg A, FS put the PAI in the From on leg B. I don't find how to change this behaviour. I would like to keep the From and the P-Asserted-Identity as they come on the leg A except some manipulations. Someone knows which parameters must be changed? Currently, the leg B is configured with "pass-callee-id" to false, "caller-id-in-from" to true and "caller-id-type" to "pid". Regards, Igor. -------------- next part -------------- An HTML attachment was scrubbed... URL: From ch.chhatra at gmail.com Thu Dec 6 00:33:23 2018 From: ch.chhatra at gmail.com (Chhorm Chhatra) Date: Thu, 6 Dec 2018 07:33:23 +0700 Subject: [Freeswitch-users] Get Active User In-Reply-To: References: Message-ID: Dear Michael, Thank you for your answer. Best regards, Chhatra Chhorm On Fri, 30 Nov 2018 at 04:29, Michael Jerris wrote: > > https://freeswitch.org/confluence/display/FREESWITCH/mod_commands?focusedCommentId=16351380#mod_commands-show > > > On Nov 23, 2018, at 9:51 AM, Chhorm Chhatra > wrote: > > > > Hi, > > I'd like to know if there is any possible way to get all active users > (users that are reachable and being to be bridged to). > > Any help would be appreciated. > > Best regards, > > Chhatra Chhorm > > > _________________________________________________________________________ > 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 ch.chhatra at gmail.com Thu Dec 6 00:34:10 2018 From: ch.chhatra at gmail.com (Chhorm Chhatra) Date: Thu, 6 Dec 2018 07:34:10 +0700 Subject: [Freeswitch-users] Setup Verto in Local to Replace SIP In-Reply-To: <77108710-F3BE-4D83-86D6-229220E98FE2@jerris.com> References: <77108710-F3BE-4D83-86D6-229220E98FE2@jerris.com> Message-ID: Dear Michael, Thank you for the answer. Best regards, Chhatra Chhorm On Fri, 30 Nov 2018 at 03:24, Michael Jerris wrote: > The browsers generally require secure connections or significant extra > hoops to jump through to start a call (approving connection on every call > for example). Even for local you will want to use certs. > > > > On Nov 23, 2018, at 9:48 AM, Chhorm Chhatra > wrote: > > > > Hi, > > I am not sure if this approach is still being used these days, but I'd > like to know if there is any guide out there to set up Verto in Local (No > SSL certificates required). > > My use case is that we don't want to involve with the Internet yet we > still want the call to be established via Verto rather than SIP since SIP > is not supported by the latest version of Android out of the box. > > Any help would be appreciated. > > Best regards, > > Chhatra Chhorm > > > _________________________________________________________________________ > 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 ch.chhatra at gmail.com Thu Dec 6 00:39:34 2018 From: ch.chhatra at gmail.com (Chhorm Chhatra) Date: Thu, 6 Dec 2018 07:39:34 +0700 Subject: [Freeswitch-users] Forward/Transfer Leg A to another extension after bridge success Message-ID: Hi, I'd like to know if there is any possible way to build a dialplan to forward/transfer Leg A to another extension based on B input from the keyboard. For example, Leg A call to Leg B (extension 1080), then Leg B answers and eventually, based on the conversation, Leg B may transfer Leg A to another extension (says, another department with extension 1090 for example). Best regards, Chhatra Chhorm -------------- next part -------------- An HTML attachment was scrubbed... URL: From sebastian_ml at gmx.net Thu Dec 6 09:40:32 2018 From: sebastian_ml at gmx.net (Sebastian Kemper) Date: Thu, 6 Dec 2018 10:40:32 +0100 Subject: [Freeswitch-users] P-Asserted-Identity transparency between leg A and leg B In-Reply-To: <003501d48cbc$1ab66b90$502342b0$@gmail.com> References: <003501d48cbc$1ab66b90$502342b0$@gmail.com> Message-ID: <20181206094031.GA856@darth.lan> On Wed, Dec 05, 2018 at 06:01:02PM +0100, igor.potjevlesch at gmail.com wrote: > Hello, > > In the scenario where the From is different than the > P-Asserted-Identity on leg A, FS put the PAI in the From on leg B. > > I don't find how to change this behaviour. I would like to keep the > From and the P-Asserted-Identity as they come on the leg A except some > manipulations. > > Someone knows which parameters must be changed? Currently, the leg B > is configured with "pass-callee-id" to false, "caller-id-in-from" to > true and "caller-id-type" to "pid". Hello Igor, Thanks for posting this. I find this an interesting question, too. I use this currently on the leg to my local phone and ATA: When there is a "clip no screening" call coming my way, I also get both numbers, the calling party in PAI and the generic number in From, from the provider. I didn't bother looking further into this because I actually don't know which number the phone/ATA would display if they received both numbers. Maybe they are not aware of "clip no screening", which wouldn't be a surprise because this feature is not supported in all countries (I think in the US they don't support it). Kind regards, Seb From sebastian_ml at gmx.net Thu Dec 6 11:36:47 2018 From: sebastian_ml at gmx.net (Sebastian Kemper) Date: Thu, 06 Dec 2018 11:36:47 +0000 Subject: [Freeswitch-users] P-Asserted-Identity transparency between leg A and leg B In-Reply-To: <000001d48d56$671f5ec0$355e1c40$@gmail.com> References: <003501d48cbc$1ab66b90$502342b0$@gmail.com> <20181206094031.GA856@darth.lan> <000001d48d56$671f5ec0$355e1c40$@gmail.com> Message-ID: <657F9B74-4C35-4668-9667-B38182B909D8@gmx.net> Am December 6, 2018 11:25:33 AM UTC schrieb igor.potjevlesch at gmail.com: >Hello Sebastian, > >In your case the local phone/ATA is the leg B right? >With that settings you get transparency between the two legs? >How ${sip_from_user} is defined? Hello Igor, Phone/ATA are B leg. I don't get transparency with this. Transparency would be if I sent PAI untouched to B leg and the From would have the same user as in A leg. I only get the From user from A leg to B leg this way, the PAI is completely dropped. Which is no problem for my situation. I'm not sure if my workaround is also suitable for anonymous calls by the way. I'll try that out next week. Kind regards, Seb From sebastian_ml at gmx.net Thu Dec 6 11:39:20 2018 From: sebastian_ml at gmx.net (Sebastian Kemper) Date: Thu, 06 Dec 2018 11:39:20 +0000 Subject: [Freeswitch-users] P-Asserted-Identity transparency between leg A and leg B In-Reply-To: <657F9B74-4C35-4668-9667-B38182B909D8@gmx.net> References: <003501d48cbc$1ab66b90$502342b0$@gmail.com> <20181206094031.GA856@darth.lan> <000001d48d56$671f5ec0$355e1c40$@gmail.com> <657F9B74-4C35-4668-9667-B38182B909D8@gmx.net> Message-ID: Oh, I forgot. ${sip_from_user} is a channel var. I found it in the CDRs. I think it's also mentioned on the wiki. From sebastian_ml at gmx.net Thu Dec 6 11:46:20 2018 From: sebastian_ml at gmx.net (Sebastian Kemper) Date: Thu, 06 Dec 2018 11:46:20 +0000 Subject: [Freeswitch-users] P-Asserted-Identity transparency between leg A and leg B In-Reply-To: References: <003501d48cbc$1ab66b90$502342b0$@gmail.com> <20181206094031.GA856@darth.lan> <000001d48d56$671f5ec0$355e1c40$@gmail.com> <657F9B74-4C35-4668-9667-B38182B909D8@gmx.net> Message-ID: <85ABF421-6ED3-40D8-AF63-64469FEEFA56@gmx.net> Removing the caller id _name_ is just a personal preference of mine, btw. From fs at voice2net.ca Thu Dec 6 12:57:52 2018 From: fs at voice2net.ca (Darcy Primrose) Date: Thu, 6 Dec 2018 07:57:52 -0500 Subject: [Freeswitch-users] Mod FIFO In-Reply-To: References: <1dd84c16-9337-711c-c2a3-344a3774a347@voice2net.ca> <96c1efc5-2b03-e8b9-4e9c-a2421387c588@voice2net.ca> Message-ID: Doing a follow up on a fifo issue to ringall as shown below.  I cannot find anything that tells me if this will indeed work.  I am using the latest build to test with.  We provide service to small medical clinics and they are asking for the ability to ring two or three sets simultaneously, cannot seem to get it working on mod fifo.     {call_timeout=30,fifo_member_wait=nowait}user/223@$${domain}     {call_timeout=30,fifo_member_wait=nowait}user/224@$${domain} Thanks Darcy -------------- next part -------------- An HTML attachment was scrubbed... URL: From igor.potjevlesch at gmail.com Thu Dec 6 11:25:33 2018 From: igor.potjevlesch at gmail.com (igor.potjevlesch at gmail.com) Date: Thu, 6 Dec 2018 12:25:33 +0100 Subject: [Freeswitch-users] P-Asserted-Identity transparency between leg A and leg B In-Reply-To: <20181206094031.GA856@darth.lan> References: <003501d48cbc$1ab66b90$502342b0$@gmail.com> <20181206094031.GA856@darth.lan> Message-ID: <000001d48d56$671f5ec0$355e1c40$@gmail.com> Hello Sebastian, In your case the local phone/ATA is the leg B right? With that settings you get transparency between the two legs? How ${sip_from_user} is defined? Regards, Igor. -----Message d'origine----- De : Sebastian Kemper Envoyé : jeudi 6 décembre 2018 10:41 À : FreeSWITCH Users Help Cc : igor.potjevlesch at gmail.com Objet : Re: [Freeswitch-users] P-Asserted-Identity transparency between leg A and leg B On Wed, Dec 05, 2018 at 06:01:02PM +0100, igor.potjevlesch at gmail.com wrote: > Hello, > > In the scenario where the From is different than the > P-Asserted-Identity on leg A, FS put the PAI in the From on leg B. > > I don't find how to change this behaviour. I would like to keep the > From and the P-Asserted-Identity as they come on the leg A except some > manipulations. > > Someone knows which parameters must be changed? Currently, the leg B > is configured with "pass-callee-id" to false, "caller-id-in-from" to > true and "caller-id-type" to "pid". Hello Igor, Thanks for posting this. I find this an interesting question, too. I use this currently on the leg to my local phone and ATA: When there is a "clip no screening" call coming my way, I also get both numbers, the calling party in PAI and the generic number in From, from the provider. I didn't bother looking further into this because I actually don't know which number the phone/ATA would display if they received both numbers. Maybe they are not aware of "clip no screening", which wouldn't be a surprise because this feature is not supported in all countries (I think in the US they don't support it). Kind regards, Seb From ch.chhatra at gmail.com Fri Dec 7 04:17:24 2018 From: ch.chhatra at gmail.com (Chhorm Chhatra) Date: Fri, 7 Dec 2018 11:17:24 +0700 Subject: [Freeswitch-users] Transfer Call to another extension from mod_callcenter Message-ID: Hi, Is it possible for mod_callcenter agent to transfer the call to another extension based on DTMF keyboard input from the agent? For example, a user calls to the agent, the agent then answers it. After some conversation, the agent transfers the user to another department by entering the keyboard input from the phone. Any help would be appreciated. For example Best regards, Chhatra Chhorm -------------- next part -------------- An HTML attachment was scrubbed... URL: From melekoktay at gmail.com Fri Dec 7 07:29:01 2018 From: melekoktay at gmail.com (Melek Oktay) Date: Fri, 7 Dec 2018 08:29:01 +0100 Subject: [Freeswitch-users] FreeSwitch blocked In-Reply-To: References: Message-ID: Hi, After deeply research about this issue, we understand *Freeswitch-Core* mechanism and three possible solutions for fixing these issue. Here the link https://stackoverflow.com/questions/53609817/freeswitch-blocked Actually it would be be better third solution supported by *Freeswitch*, before deliver event to module, it create new thread for not waiting consumer thread (event hendler) On Fri, Dec 22, 2017 at 8:35 AM Melek Oktay wrote: > > Hi, > > FreeSwitch software working well in a few days (~3 - 5 days), then new > incoming call requests are accepted since FreeSwitch is blocked !! Ongoing > calls continue their session, their calls seems not effected, but new calls > are not accepted. I got FreeSwitch snapshot and analyzed it in GDB. > > I have 601 therads & most of them are waiting > > Thread 0x7f16bc55f700 (LWP 28544) pthread_cond_wait@@GLIBC_2.3.2 () at > ../nptl/sysdeps/unix/sysv/linux/x86_64/pthread_cond_wait.S:185 > > When i apply "*thread apply all bt*" in gdb, I see most of the threads > try to push events into queue (*switch_queue_push *) > > Thread 600 (Thread 0x7f16bc55f700 (LWP 28544)): > #0 pthread_cond_wait@@GLIBC_2.3.2 () at > ../nptl/sysdeps/unix/sysv/linux/x86_64/pthread_cond_wait.S:185 > #1 0x00007f180cf9b87d in apr_thread_cond_wait (cond=, > mutex=) at locks/unix/thread_cond.c:68 > #2 0x00007f180cf92dd0 in apr_queue_push (queue=queue at entry=0x7f180db157a8, > data=data at entry=0x7f16d3d5ec20) at misc/apr_queue.c:166 > #3 0x00007f180cc958fb in *switch_queue_push *(queue=*0x7f180db157a8*, > data=data at entry=*0x7f16d3d5ec20*) at src/switch_apr.c:1134 > #4 0x00007f180cd17850 in switch_event_queue_dispatch_event > (eventp=0x7f16bc55ec48) at src/switch_event.c:384 > #5 switch_event_fire_detailed (file=file at entry=0x7f180cfb07ea > "src/switch_channel.c", func=func at entry=0x7f180cfb2ba0 <__func__.18348> > "switch_channel_perform_set_running_state", line=line at entry=2260, > event=event at entry=0x7f16bc55ec48, user_data=user_data at entry=0x0) at > src/switch_event.c:1986 > #6 0x00007f180cc9f118 in switch_channel_perform_set_running_state > (channel=0x7f17e3e7de00, state=CS_NEW, file=0x7f180cfbc590 > "src/switch_core_state_machine.c", func=, line=543) at > src/switch_channel.c:2260 > #7 0x00007f180ccc87d0 in switch_core_session_run (session=0x7f17e3e7fd28) > at src/switch_core_state_machine.c:543 > #8 0x00007f180ccc36de in switch_core_session_thread (thread= out>, obj=0x7f17e3e7fd28) at src/switch_core_session.c:1629 > #9 0x00007f180ccbf47d in switch_core_session_thread_pool_worker > (thread=0x7f17e3e9abb0, obj=0x80) at src/switch_core_session.c:1692 > #10 0x00007f180cfa1910 in dummy_worker (opaque=0x7f17e3e9abb0) at > threadproc/unix/thread.c:151 > #11 0x00007f180c1e0064 in start_thread (arg=0x7f16bc55f700) at > pthread_create.c:309 > #12 0x00007f180b8b862d in clone () at > ../sysdeps/unix/sysv/linux/x86_64/clone.S:111 > > > More interesting thing is below, when I look up event type, approximately > all of them are "SWITCH_EVENT_CHANNEL_STATE" and switch_queue (i think > sofia_module queue is used in this scenario ) *become full* !!! *nelts* > (number of elements ) and *bounds *values are equal, and there are 553 > (full_waiters) waiters try to push , but no body try to consume it > (empty_waiters = 0) > > (gdb) print *(switch_queue_t *) *0x7f180db157a8* > $1 = { > data = 0x7f1805cfe038, > nelts = 50000, > in = 43000, > out = 43000, > bounds = 50000, > full_waiters = 553, > empty_waiters = 0, > one_big_mutex = 0x7f180db157e8, > not_empty = 0x7f180db15838, > not_full = 0x7f180db15890, > terminated = 0 > } > > (gdb) print *(switch_event_t *) *0x7f16d3d5ec20* > $1 = { > event_id = SWITCH_EVENT_CHANNEL_STATE, > priority = SWITCH_PRIORITY_NORMAL, > owner = 0x0, > subclass_name = 0x0, > headers = 0x7f16d3d5f750, > last_header = 0x7f16d3d601d0, > body = 0x0, > bind_user_data = 0x0, > event_user_data = 0x0, > key = 0, > next = 0x0, > flags = 0 > } > > > Why i am gonna getting this state? > > Any thoughts, tips, tricks would be much appreciated. > > Regards, > > Angel > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From krice at freeswitch.org Sat Dec 8 09:03:30 2018 From: krice at freeswitch.org (Ken Rice) Date: Sat, 8 Dec 2018 03:03:30 -0600 Subject: [Freeswitch-users] FreeSwitch blocked In-Reply-To: References: Message-ID: the problem isnt freeswitch, but your event consumer isnt consuming the events fast enough. you should adjust your side of the equation. the event queue is what drives everything Sent from my iPhone > On Dec 7, 2018, at 01:29, Melek Oktay wrote: > > Hi, > > After deeply research about this issue, we understand Freeswitch-Core mechanism and three possible solutions for fixing these issue. > > Here the link > https://stackoverflow.com/questions/53609817/freeswitch-blocked > > Actually it would be be better third solution supported by Freeswitch, before deliver event to module, it create new thread for not waiting consumer thread (event hendler) > >> On Fri, Dec 22, 2017 at 8:35 AM Melek Oktay wrote: >> >> Hi, >> >> FreeSwitch software working well in a few days (~3 - 5 days), then new incoming call requests are accepted since FreeSwitch is blocked !! Ongoing calls continue their session, their calls seems not effected, but new calls are not accepted. I got FreeSwitch snapshot and analyzed it in GDB. >> >> I have 601 therads & most of them are waiting >> >> Thread 0x7f16bc55f700 (LWP 28544) pthread_cond_wait@@GLIBC_2.3.2 () at ../nptl/sysdeps/unix/sysv/linux/x86_64/pthread_cond_wait.S:185 >> >> When i apply "thread apply all bt" in gdb, I see most of the threads try to push events into queue (switch_queue_push ) >> >> Thread 600 (Thread 0x7f16bc55f700 (LWP 28544)): >> #0 pthread_cond_wait@@GLIBC_2.3.2 () at ../nptl/sysdeps/unix/sysv/linux/x86_64/pthread_cond_wait.S:185 >> #1 0x00007f180cf9b87d in apr_thread_cond_wait (cond=, mutex=) at locks/unix/thread_cond.c:68 >> #2 0x00007f180cf92dd0 in apr_queue_push (queue=queue at entry=0x7f180db157a8, data=data at entry=0x7f16d3d5ec20) at misc/apr_queue.c:166 >> #3 0x00007f180cc958fb in switch_queue_push (queue=0x7f180db157a8, data=data at entry=0x7f16d3d5ec20) at src/switch_apr.c:1134 >> #4 0x00007f180cd17850 in switch_event_queue_dispatch_event (eventp=0x7f16bc55ec48) at src/switch_event.c:384 >> #5 switch_event_fire_detailed (file=file at entry=0x7f180cfb07ea "src/switch_channel.c", func=func at entry=0x7f180cfb2ba0 <__func__.18348> "switch_channel_perform_set_running_state", line=line at entry=2260, event=event at entry=0x7f16bc55ec48, user_data=user_data at entry=0x0) at src/switch_event.c:1986 >> #6 0x00007f180cc9f118 in switch_channel_perform_set_running_state (channel=0x7f17e3e7de00, state=CS_NEW, file=0x7f180cfbc590 "src/switch_core_state_machine.c", func=, line=543) at src/switch_channel.c:2260 >> #7 0x00007f180ccc87d0 in switch_core_session_run (session=0x7f17e3e7fd28) at src/switch_core_state_machine.c:543 >> #8 0x00007f180ccc36de in switch_core_session_thread (thread=, obj=0x7f17e3e7fd28) at src/switch_core_session.c:1629 >> #9 0x00007f180ccbf47d in switch_core_session_thread_pool_worker (thread=0x7f17e3e9abb0, obj=0x80) at src/switch_core_session.c:1692 >> #10 0x00007f180cfa1910 in dummy_worker (opaque=0x7f17e3e9abb0) at threadproc/unix/thread.c:151 >> #11 0x00007f180c1e0064 in start_thread (arg=0x7f16bc55f700) at pthread_create.c:309 >> #12 0x00007f180b8b862d in clone () at ../sysdeps/unix/sysv/linux/x86_64/clone.S:111 >> >> >> More interesting thing is below, when I look up event type, approximately all of them are "SWITCH_EVENT_CHANNEL_STATE" and switch_queue (i think sofia_module queue is used in this scenario ) become full !!! nelts (number of elements ) and bounds values are equal, and there are 553 (full_waiters) waiters try to push , but no body try to consume it (empty_waiters = 0) >> >> (gdb) print *(switch_queue_t *) 0x7f180db157a8 >> $1 = { >> data = 0x7f1805cfe038, >> nelts = 50000, >> in = 43000, >> out = 43000, >> bounds = 50000, >> full_waiters = 553, >> empty_waiters = 0, >> one_big_mutex = 0x7f180db157e8, >> not_empty = 0x7f180db15838, >> not_full = 0x7f180db15890, >> terminated = 0 >> } >> >> (gdb) print *(switch_event_t *) 0x7f16d3d5ec20 >> $1 = { >> event_id = SWITCH_EVENT_CHANNEL_STATE, >> priority = SWITCH_PRIORITY_NORMAL, >> owner = 0x0, >> subclass_name = 0x0, >> headers = 0x7f16d3d5f750, >> last_header = 0x7f16d3d601d0, >> body = 0x0, >> bind_user_data = 0x0, >> event_user_data = 0x0, >> key = 0, >> next = 0x0, >> flags = 0 >> } >> >> >> Why i am gonna getting this state? >> Any thoughts, tips, tricks would be much appreciated. >> Regards, >> Angel > _________________________________________________________________________ > 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 sagarmalam at gmail.com Sun Dec 9 18:17:24 2018 From: sagarmalam at gmail.com (sagar malam) Date: Sun, 9 Dec 2018 23:47:24 +0530 Subject: [Freeswitch-users] Disabling 200 OK with contact headers of all the lines Message-ID: Hello , I am using FS SLA feature and it works very well.However i am facing an issue of registrations getting dropped as explained below : There are 3 phones registered with same extension number.Sofia is configured with "sip-expires-max-deviation" to randomise registration through expiry header in contact header.All the phones are Polycom. In case of shared lines FS adds contact header of all the lines(or phones with same extension) in 200 OK as shown below due to which all the phones are reading expiry timer from first contact header only.So in below example,Phone re registers after 379 seconds(first contact header) instead of 88 seconds(third contact header) leading to registration expiry on FS. Reason why phones are always reading expiry from first contact header is same Public IP(contact is re written by Proxy in front of FS) for all three phones which is confusing phone to identify its own contact header. Is there any way to configure FS to not send contact headers of all the registrations but only one that belongs to the line itself ? or any other way to fix it. ============================200 OK for register packet ============= 2018/12/08 13:36:35.426099 10.50.7.251:5070 -> 10.50.7.253:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 66.160.237.253:5060 ;branch=z9hG4bK9b46.f9b7a68125108f411537617d02bd48f8.0;received=10.50.7.253 Via: SIP/2.0/UDP 198.136.236.1:5060 ;rport=5060;received=10.50.8.1;branch=z9hG4bK9b46.e6f1aa3817a238f499aa3aafa4217425.0 Via: SIP/2.0/UDP 172.16.1.11;rport=1426;received=71.239.113.14;branch=z9hG4bK1df2302cD062D28F From: "Main Line" ;tag=22CE54C0-84BCCF23 To: ;tag=e8Fa2tND4U80D Call-ID: d66f0f0592c09746b903406f312eb0c2 CSeq: 590 REGISTER Contact: ;expires=379 Contact: ;expires=245 Contact: ;expires=88 Date: Sat, 08 Dec 2018 08:06:35 GMT User-Agent: FreeSWITCH-mod_sofia/1.6.17~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: path, replaces Content-Length: 0 =========================================================== -- Thanks, Sagar -------------- next part -------------- An HTML attachment was scrubbed... URL: From infos at madovsky.org Sun Dec 9 21:46:57 2018 From: infos at madovsky.org (Madovsky) Date: Sun, 9 Dec 2018 13:46:57 -0800 Subject: [Freeswitch-users] FS and gcc 8.x.x Message-ID: Just FYI FS does not compile anymore with gcc 8.x.x major changes have been made with this new version of gcc, especially strncpy and strncat. From kbdfck at gmail.com Sun Dec 9 22:39:15 2018 From: kbdfck at gmail.com (Dmitry Sytchev) Date: Mon, 10 Dec 2018 01:39:15 +0300 Subject: [Freeswitch-users] Replace User-Agent and Server headers In-Reply-To: <94A74EA3-A0EA-4F1E-9CFF-A685C5F0CFC0@gmx.net> References: <94A74EA3-A0EA-4F1E-9CFF-A685C5F0CFC0@gmx.net> Message-ID: Besides UA and session owner fields there are special headers, methods list, callid generation, contact field format, headers sequence and so on. So I think if somebody really wants to dig some info about your switch, there are many other ways to detect it's using FreeSwitch. пт, 23 нояб. 2018 г. в 17:44, Markus Bönke : > > You can set: > > > > in the sipprofile config. > > > > Am 23.11.2018 um 09:44 schrieb Kevin Olbrich : > > > > Hi! > > > > How can I replace User-Agent and Server headers in freeswitch? > > This is some kind of hardening (why tell people which version I use if > > the don't need to know...). > > > > Sure, I use the latest releases but I would like to hide the version I > > use or that I use freeswitch at all. > > > > Kind regards > > Kevin > > > > _________________________________________________________________________ > > 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 -- Best regards, Dmitry Sytchev, IT Engineer From ryandelgrosso at gmail.com Mon Dec 10 03:04:01 2018 From: ryandelgrosso at gmail.com (Ryan Delgrosso) Date: Sun, 9 Dec 2018 19:04:01 -0800 Subject: [Freeswitch-users] Issue compiling v 1.8 on CentOS 7 Message-ID: Hey all, Im working on building v1.8 on CentOS but im having trouble getting to recognize that libemp3lame-dev is present. Any suggestions on how I can get configure to locate the installed version and get mod_shout to build? ------------------------------------------------------------------------ making all mod_shout make[4]: Entering directory `/usr/local/src/freeswitch/src/mod/formats/mod_shout' Makefile:904: *** You must install libmp3lame-dev to build mod_shout.  Stop. make[4]: Leaving directory `/usr/local/src/freeswitch/src/mod/formats/mod_shout' make[3]: *** [mod_shout-all] Error 1 make[3]: Leaving directory `/usr/local/src/freeswitch/src/mod' make[2]: *** [all-recursive] Error 1 make[2]: Leaving directory `/usr/local/src/freeswitch/src' make[1]: *** [all-recursive] Error 1 make[1]: Leaving directory `/usr/local/src/freeswitch' make: *** [all] Error 2 # yum list | grep lame lame.x86_64 3.100-1.el7                    @epel lame-devel.x86_64 3.100-1.el7                    @epel lame-libs.x86_64 3.100-1.el7                    @epel lame-mp3x.x86_64 3.100-1.el7                    @epel ------------------------------------------------------------------------ Thanks in advance! -Ryan -------------- next part -------------- An HTML attachment was scrubbed... URL: From alex at freeswitch.com Mon Dec 10 05:11:44 2018 From: alex at freeswitch.com (Alexey Sibyakin) Date: Mon, 10 Dec 2018 14:11:44 +0900 Subject: [Freeswitch-users] Forward/Transfer Leg A to another extension after bridge success In-Reply-To: References: Message-ID: Hi, Try to check out an example of transfer in the *features* context of vanilla dialplan. Regards, Alex On Fri, Dec 7, 2018 at 1:37 AM Chhorm Chhatra wrote: > Hi, > I'd like to know if there is any possible way to build a dialplan to > forward/transfer Leg A to another extension based on B input from the > keyboard. > > For example, Leg A call to Leg B (extension 1080), then Leg B answers and > eventually, based on the conversation, Leg B may transfer Leg A to another > extension (says, another department with extension 1090 for example). > > Best regards, > Chhatra Chhorm > _________________________________________________________________________ > 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 ch.chhatra at gmail.com Mon Dec 10 09:11:41 2018 From: ch.chhatra at gmail.com (Chhorm Chhatra) Date: Mon, 10 Dec 2018 16:11:41 +0700 Subject: [Freeswitch-users] Forward/Transfer Leg A to another extension after bridge success In-Reply-To: References: Message-ID: Thank you for your suggested solution. On Mon, 10 Dec 2018 at 16:09, Alexey Sibyakin wrote: > Hi, > > Try to check out an example of transfer in the *features* context of > vanilla dialplan. > > Regards, > > Alex > > On Fri, Dec 7, 2018 at 1:37 AM Chhorm Chhatra > wrote: > >> Hi, >> I'd like to know if there is any possible way to build a dialplan to >> forward/transfer Leg A to another extension based on B input from the >> keyboard. >> >> For example, Leg A call to Leg B (extension 1080), then Leg B answers and >> eventually, based on the conversation, Leg B may transfer Leg A to another >> extension (says, another department with extension 1090 for example). >> >> Best regards, >> Chhatra Chhorm >> _________________________________________________________________________ >> 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 s.safarov at gmail.com Mon Dec 10 10:16:14 2018 From: s.safarov at gmail.com (Sergey Safarov) Date: Mon, 10 Dec 2018 13:16:14 +0300 Subject: [Freeswitch-users] FS and gcc 8.x.x In-Reply-To: References: Message-ID: Relate https://freeswitch.org/jira/browse/FS-11345 пн, 10 дек. 2018 г. в 12:49, Madovsky : > Just FYI FS does not compile anymore with gcc 8.x.x > > major changes have been made with this new version of gcc, especially > strncpy and strncat. > > > _________________________________________________________________________ > 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 s.safarov at gmail.com Mon Dec 10 10:33:27 2018 From: s.safarov at gmail.com (Sergey Safarov) Date: Mon, 10 Dec 2018 13:33:27 +0300 Subject: [Freeswitch-users] Issue compiling v 1.8 on CentOS 7 In-Reply-To: References: Message-ID: Update for SPEC file to compile on CentOS 7 is located here https://freeswitch.org/stash/projects/FS/repos/freeswitch/pull-requests/1561/overview пн, 10 дек. 2018 г. в 13:19, Ryan Delgrosso : > Hey all, > > Im working on building v1.8 on CentOS but im having trouble getting to > recognize that libemp3lame-dev is present. Any suggestions on how I can get > configure to locate the installed version and get mod_shout to build? > > > ------------------------------ > > making all mod_shout > make[4]: Entering directory > `/usr/local/src/freeswitch/src/mod/formats/mod_shout' > Makefile:904: *** You must install libmp3lame-dev to build mod_shout. > Stop. > make[4]: Leaving directory > `/usr/local/src/freeswitch/src/mod/formats/mod_shout' > make[3]: *** [mod_shout-all] Error 1 > make[3]: Leaving directory `/usr/local/src/freeswitch/src/mod' > make[2]: *** [all-recursive] Error 1 > make[2]: Leaving directory `/usr/local/src/freeswitch/src' > make[1]: *** [all-recursive] Error 1 > make[1]: Leaving directory `/usr/local/src/freeswitch' > make: *** [all] Error 2 > > # yum list | grep lame > lame.x86_64 3.100-1.el7 > @epel > lame-devel.x86_64 3.100-1.el7 > @epel > lame-libs.x86_64 3.100-1.el7 > @epel > lame-mp3x.x86_64 3.100-1.el7 > @epel > ------------------------------ > > Thanks in advance! > > -Ryan > > > _________________________________________________________________________ > 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 social at bohboh.info Mon Dec 10 12:52:19 2018 From: social at bohboh.info (Social Boh) Date: Mon, 10 Dec 2018 07:52:19 -0500 Subject: [Freeswitch-users] Issue compiling v 1.8 on CentOS 7 In-Reply-To: References: Message-ID: <4377a69b-2e40-22f4-6328-052f88c97714@bohboh.info> Hello, you have to install: *yum install libvorbis libvorbis-devel vorbis-tools libogg libogg-devel mpg123-devel mpg123-libs libshout-devel -y* Regards ** --- I'm SoCIaL, MayBe El 09/12/2018 a las 22:04, Ryan Delgrosso escribió: > > Hey all, > > Im working on building v1.8 on CentOS but im having trouble getting to > recognize that libemp3lame-dev is present. Any suggestions on how I > can get configure to locate the installed version and get mod_shout to > build? > > > ------------------------------------------------------------------------ > > making all mod_shout > make[4]: Entering directory > `/usr/local/src/freeswitch/src/mod/formats/mod_shout' > Makefile:904: *** You must install libmp3lame-dev to build mod_shout.  > Stop. > make[4]: Leaving directory > `/usr/local/src/freeswitch/src/mod/formats/mod_shout' > make[3]: *** [mod_shout-all] Error 1 > make[3]: Leaving directory `/usr/local/src/freeswitch/src/mod' > make[2]: *** [all-recursive] Error 1 > make[2]: Leaving directory `/usr/local/src/freeswitch/src' > make[1]: *** [all-recursive] Error 1 > make[1]: Leaving directory `/usr/local/src/freeswitch' > make: *** [all] Error 2 > > # yum list | grep lame > lame.x86_64 3.100-1.el7                    @epel > lame-devel.x86_64 3.100-1.el7                    @epel > lame-libs.x86_64 3.100-1.el7                    @epel > lame-mp3x.x86_64 3.100-1.el7                    @epel > > ------------------------------------------------------------------------ > > Thanks in advance! > > -Ryan > > > > _________________________________________________________________________ > 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 josedavid at zennio.com Mon Dec 10 13:08:54 2018 From: josedavid at zennio.com (Jose David Jurado Alonso) Date: Mon, 10 Dec 2018 14:08:54 +0100 Subject: [Freeswitch-users] Source device doesn't hangup videocall when the target device hangs In-Reply-To: References: Message-ID: Hi, In principle it seems that the BYE is sent correctly. This is the record fragment that appears when I hang up from my mobile: ------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------- 27203df 2018-12-10 12:53:40.751007 [DEBUG] switch_ivr_bridge.c:848 Ending video thread. 127203df 2018-12-10 12:53:40.751007 [DEBUG] switch_ivr_bridge.c:906 Ending video thread. 127203df 2018-12-10 12:53:40.751007 [DEBUG] switch_ivr_bridge.c:255 sofia/external/MOBILE01 at 88.22.1.17:42697 video thread ended. 127203df 2018-12-10 12:53:40.751007 [DEBUG] switch_ivr_bridge.c:257 sofia/external/MOBILE01 at 88.22.1.17:42697 skip receive message [DISPLAY] (channel is hungup already) 70c48dd9 2018-12-10 12:53:40.751007 [DEBUG] switch_ivr_bridge.c:255 sofia/external/DOOR01 at sip.mytestdomain.com:5080 video thread ended. 2018-12-10 12:53:40.751007 [DEBUG] switch_core_media.c:7470 sofia/external/ MOBILE01 at 88.22.1.17:42697 Video thread ended 127203df 2018-12-10 12:53:40.761014 [DEBUG] switch_ivr_bridge.c:917 BRIDGE THREAD DONE [sofia/external/MOBILE01 at 88.22.1.17:42697] 127203df 2018-12-10 12:53:40.761014 [DEBUG] switch_core_state_machine.c:653 (sofia/external/MOBILE01 at 88.22.1.17:42697) State EXCHANGE_MEDIA going to sleep 127203df 2018-12-10 12:53:40.761014 [DEBUG] switch_core_state_machine.c:584 (sofia/external/MOBILE01 at 88.22.1.17:42697) Running State Change CS_HANGUP (Cur 2 Tot 652) 127203df 2018-12-10 12:53:40.761014 [DEBUG] switch_core_state_machine.c:847 (sofia/external/MOBILE01 at 88.22.1.17:42697) Callstate Change ACTIVE -> HANGUP 127203df 2018-12-10 12:53:40.761014 [DEBUG] switch_core_state_machine.c:849 (sofia/external/MOBILE01 at 88.22.1.17:42697) State HANGUP 127203df 2018-12-10 12:53:40.761014 [DEBUG] mod_sofia.c:449 Channel sofia/external/MOBILE01 at 88.22.1.17:42697 hanging up, cause: NORMAL_CLEARING 127203df 2018-12-10 12:53:40.761014 [DEBUG] switch_core_state_machine.c:60 sofia/external/MOBILE01 at 88.22.1.17:42697 Standard HANGUP, cause: NORMAL_CLEARING 127203df 2018-12-10 12:53:40.761014 [DEBUG] switch_core_state_machine.c:849 (sofia/external/MOBILE01 at 88.22.1.17:42697) State HANGUP going to sleep 127203df 2018-12-10 12:53:40.761014 [DEBUG] switch_core_state_machine.c:619 (sofia/external/MOBILE01 at 88.22.1.17:42697) State Change CS_HANGUP -> CS_REPORTING 127203df 2018-12-10 12:53:40.761014 [DEBUG] switch_core_state_machine.c:584 (sofia/external/MOBILE01 at 88.22.1.17:42697) Running State Change CS_REPORTING (Cur 2 Tot 652) 127203df 2018-12-10 12:53:40.761014 [DEBUG] switch_core_state_machine.c:935 (sofia/external/MOBILE01 at 88.22.1.17:42697) State REPORTING 127203df 2018-12-10 12:53:40.761014 [DEBUG] switch_core_state_machine.c:174 sofia/external/MOBILE01 at 88.22.1.17:42697 Standard REPORTING, cause: NORMAL_CLEARING 127203df 2018-12-10 12:53:40.761014 [DEBUG] switch_core_state_machine.c:935 (sofia/external/MOBILE01 at 88.22.1.17:42697) State REPORTING going to sleep 127203df 2018-12-10 12:53:40.761014 [DEBUG] switch_core_state_machine.c:610 (sofia/external/MOBILE01 at 88.22.1.17:42697) State Change CS_REPORTING -> CS_DESTROY 127203df 2018-12-10 12:53:40.761014 [DEBUG] switch_core_session.c:1714 Session 652 (sofia/external/MOBILE01 at 88.22.1.17:42697) Locked, Waiting on external entities 70c48dd9 2018-12-10 12:53:40.761014 [DEBUG] switch_ivr_bridge.c:825 sofia/external/MOBILE01 at 88.22.1.17:42697 ending bridge by request from write function � 2018-12-10 12:53:40.761014 [DEBUG] switch_ivr_bridge.c:825 sofia/external/ MOBILE01 at 88.22.1.17:42697 ending bridge by request from write function 70c48dd9 2018-12-10 12:53:40.761014 [DEBUG] switch_ivr_bridge.c:917 BRIDGE THREAD DONE [sofia/external/DOOR01 at sip.mytestdomain.com:5080] 70c48dd9 2018-12-10 12:53:40.761014 [NOTICE] switch_ivr_bridge.c:1933 Hangup sofia/external/DOOR01 at sip.mytestdomain.com:5080 [CS_EXECUTE] [NORMAL_CLEARING] 127203df 2018-12-10 12:53:40.771024 [NOTICE] switch_core_session.c:1732 Session 652 (sofia/external/MOBILE01 at 88.22.1.17:42697) Ended 127203df 2018-12-10 12:53:40.771024 [NOTICE] switch_core_session.c:1736 Close Channel sofia/external/MOBILE01 at 88.22.1.17:42697 [CS_DESTROY] 70c48dd9 2018-12-10 12:53:40.771024 [DEBUG] switch_core_session.c:2886 sofia/external/DOOR01 at sip.mytestdomain.com:5080 skip receive message [PHONE_EVENT] (channel is hungup already) 70c48dd9 2018-12-10 12:53:40.771024 [DEBUG] switch_core_state_machine.c:650 (sofia/external/DOOR01 at sip.mytestdomain.com:5080) State EXECUTE going to sleep 70c48dd9 2018-12-10 12:53:40.771024 [DEBUG] switch_core_state_machine.c:584 (sofia/external/DOOR01 at sip.mytestdomain.com:5080) Running State Change CS_HANGUP (Cur 1 Tot 652) 70c48dd9 2018-12-10 12:53:40.771024 [DEBUG] switch_core_state_machine.c:847 (sofia/external/DOOR01 at sip.mytestdomain.com:5080) Callstate Change ACTIVE -> HANGUP 127203df 2018-12-10 12:53:40.771024 [DEBUG] switch_core_state_machine.c:738 (sofia/external/MOBILE01 at 88.22.1.17:42697) Running State Change CS_DESTROY (Cur 1 Tot 652) 70c48dd9 2018-12-10 12:53:40.771024 [DEBUG] switch_core_state_machine.c:849 (sofia/external/DOOR01 at sip.mytestdomain.com:5080) State HANGUP 70c48dd9 2018-12-10 12:53:40.771024 [DEBUG] mod_sofia.c:443 sofia/external/ DOOR01 at sip.mytestdomain.com:5080 Overriding SIP cause 480 with 200 from the other leg 70c48dd9 2018-12-10 12:53:40.771024 [DEBUG] mod_sofia.c:449 Channel sofia/external/DOOR01 at sip.mytestdomain.com:5080 hanging up, cause: NORMAL_CLEARING 70c48dd9 2018-12-10 12:53:40.771024 [DEBUG] mod_sofia.c:502 Sending BYE to sofia/external/DOOR01 at sip.mytestdomain.com:5080 127203df 2018-12-10 12:53:40.771024 [DEBUG] switch_core_state_machine.c:748 (sofia/external/MOBILE01 at 88.22.1.17:42697) State DESTROY 127203df 2018-12-10 12:53:40.771024 [DEBUG] mod_sofia.c:354 sofia/external/ MOBILE01 at 88.22.1.17:42697 SOFIA DESTROY 70c48dd9 2018-12-10 12:53:40.771024 [DEBUG] switch_core_state_machine.c:60 sofia/external/DOOR01 at sip.mytestdomain.com:5080 Standard HANGUP, cause: NORMAL_CLEARING 70c48dd9 2018-12-10 12:53:40.771024 [DEBUG] switch_core_state_machine.c:849 (sofia/external/DOOR01 at sip.mytestdomain.com:5080) State HANGUP going to sleep 70c48dd9 2018-12-10 12:53:40.771024 [DEBUG] switch_core_state_machine.c:619 (sofia/external/DOOR01 at sip.mytestdomain.com:5080) State Change CS_HANGUP -> CS_REPORTING 127203df 2018-12-10 12:53:40.771024 [DEBUG] switch_core_state_machine.c:181 sofia/external/MOBILE01 at 88.22.1.17:42697 Standard DESTROY 127203df 2018-12-10 12:53:40.771024 [DEBUG] switch_core_state_machine.c:748 (sofia/external/MOBILE01 at 88.22.1.17:42697) State DESTROY going to sleep 70c48dd9 2018-12-10 12:53:40.771024 [DEBUG] switch_core_state_machine.c:584 (sofia/external/DOOR01 at sip.mytestdomain.com:5080) Running State Change CS_REPORTING (Cur 1 Tot 652) 70c48dd9 2018-12-10 12:53:40.771024 [DEBUG] switch_core_state_machine.c:935 (sofia/external/DOOR01 at sip.mytestdomain.com:5080) State REPORTING 70c48dd9 2018-12-10 12:53:40.771024 [DEBUG] switch_core_state_machine.c:174 sofia/external/DOOR01 at sip.mytestdomain.com:5080 Standard REPORTING, cause: NORMAL_CLEARING 70c48dd9 2018-12-10 12:53:40.771024 [DEBUG] switch_core_state_machine.c:935 (sofia/external/DOOR01 at sip.mytestdomain.com:5080) State REPORTING going to sleep 70c48dd9 2018-12-10 12:53:40.771024 [DEBUG] switch_core_state_machine.c:610 (sofia/external/DOOR01 at sip.mytestdomain.com:5080) State Change CS_REPORTING -> CS_DESTROY 70c48dd9 2018-12-10 12:53:40.771024 [DEBUG] switch_core_session.c:1714 Session 650 (sofia/external/DOOR01 at sip.mytestdomain.com:5080) Locked, Waiting on external entities 2018-12-10 12:53:40.781009 [DEBUG] switch_core_media.c:7470 sofia/external/ DOOR01 at sip.mytestdomain.com:5080 Video thread ended 70c48dd9 2018-12-10 12:53:40.781009 [NOTICE] switch_core_session.c:1732 Session 650 (sofia/external/DOOR01 at sip.mytestdomain.com:5080) Ended 70c48dd9 2018-12-10 12:53:40.781009 [NOTICE] switch_core_session.c:1736 Close Channel sofia/external/DOOR01 at sip.mytestdomain.com:5080 [CS_DESTROY] 70c48dd9 2018-12-10 12:53:40.781009 [DEBUG] switch_core_state_machine.c:738 (sofia/external/DOOR01 at sip.mytestdomain.com:5080) Running State Change CS_DESTROY (Cur 0 Tot 652) 70c48dd9 2018-12-10 12:53:40.781009 [DEBUG] switch_core_state_machine.c:748 (sofia/external/DOOR01 at sip.mytestdomain.com:5080) State DESTROY 70c48dd9 2018-12-10 12:53:40.781009 [DEBUG] mod_sofia.c:354 sofia/external/ DOOR01 at sip.mytestdomain.com:5080 SOFIA DESTROY 70c48dd9 2018-12-10 12:53:40.781009 [DEBUG] switch_core_state_machine.c:181 sofia/external/DOOR01 at sip.mytestdomain.com:5080 Standard DESTROY 70c48dd9 2018-12-10 12:53:40.781009 [DEBUG] switch_core_state_machine.c:748 (sofia/external/DOOR01 at sip.mytestdomain.com:5080) State DESTROY going to sleep 2018-12-10 12:53:41.251007 [DEBUG] switch_scheduler.c:144 Deleting task 70 switch_ivr_schedule_hangup (70c48dd9-8e87-46af-be8d-1aa0b61b2da1) ------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------- I have several video intercoms in the same network and under the same switch, I do not know if that can influence. In addition, I have seen this trace that is a bit strange: *Overriding SIP cause 480 with 200 from the other leg.* Regards, José David Jurado Alonso El mar., 4 dic. 2018 a las 18:07, Faisal Hanif () escribió: > You need to check SIP traces to make sure if BYE received from mobile and > sent to door phone > > On Tue, Dec 4, 2018, 6:40 PM Jose David Jurado Alonso < > josedavid at zennio.com wrote: > >> Hi! >> >> I call from a intercom (video door) to a mobile but when the mobile >> hangup the call, the intercom continues with the open call (it does not >> hang) and you have to press the button to hangup. >> >> That is, the video door phone does not hang when the call is hung from >> the mobile. >> >> I am using the external profile, intercom and mobile devices are in the >> same network but the server is in the cloud. >> >> The dialplan extension: >> ---------------------------------------------------------------------- >> >> >> >> >> >> >> >> >> >> >> >> >> ---------------------------------------------------------------------- >> >> Thanks! >> >> Regards, >> >> José David Jurado Alonso >> _________________________________________________________________________ >> 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 brian at freeswitch.com Mon Dec 10 14:02:43 2018 From: brian at freeswitch.com (Brian West) Date: Mon, 10 Dec 2018 08:02:43 -0600 Subject: [Freeswitch-users] FS and gcc 8.x.x In-Reply-To: References: Message-ID: Can you please send a bug report to JIRA? Thanks, /b On Mon, Dec 10, 2018 at 3:09 AM Madovsky wrote: > Just FYI FS does not compile anymore with gcc 8.x.x > > major changes have been made with this new version of gcc, especially > strncpy and strncat. > > > _________________________________________________________________________ > 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 -- Brian West | Co-founder and Developer Need Commercial support? email sales at freeswitch.com FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 Email: brian at freeswitch.com Mobile: 918-424-9378 Website: https://www.FreeSWITCH.com [image: https://www.facebook.com/signalwireinc?src=email] [image: https://twitter.com/freeswitch] -------------- next part -------------- An HTML attachment was scrubbed... URL: From infos at madovsky.org Mon Dec 10 20:57:15 2018 From: infos at madovsky.org (Madovsky) Date: Mon, 10 Dec 2018 12:57:15 -0800 Subject: [Freeswitch-users] FS and PHP Message-ID: <66d30c45-01be-f2c5-b7b0-85eaeb9e7ba0@madovsky.org> I'm looking for a way to compile FS with a custom PHP folder (path is /usr/local/php-5.6) did not find any way to force FS to look into this folder rather than the system ones. any clue? Thanks From shaun.stokes at itec-support.co.uk Tue Dec 11 15:28:33 2018 From: shaun.stokes at itec-support.co.uk (Shaun Stokes) Date: Tue, 11 Dec 2018 15:28:33 +0000 Subject: [Freeswitch-users] FreeSWITCH 1.8.2 - 1-2 second dropped Audio\RTP at the start of a call In-Reply-To: References: <16D901F8-1A94-4FF3-AAC9-BF37E073BEE4@gmail.com> <97EE2A01-D430-421A-90B6-7CFA71DAF34B@gmail.com>, Message-ID: Hi All, Since we've been using FreeSWITCH 1.8.2 we've noticed that the first 1-2 seconds of Audio\RTP at the start of the call when the call is answered is now dropped\missing but this doesn't occur on 1.6.20. When comparing the examples we've noticed the call flow is slightly different, as follows. FreeSWITCH 1.8.2 Leg A: switch_channel.c:2249 (sofia/internal/SRC_EXT at DOMAIN:PORT) Callstate Change DOWN -> RINGING Leg B: sofia.c:7291 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [calling][0] Leg B: sofia.c:7291 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [proceeding][180] Leg B: sofia.c:7401 Ring-Ready sofia/internal/DST_EXT at LAN_IP:PORT! Leg B: switch_channel.c:3354 (sofia/internal/DST_EXT at LAN_IP:PORT) Callstate Change DOWN -> RINGING Leg A: switch_ivr_originate.c:1246 Sending early media Leg A: switch_channel.c:3482 (sofia/internal/SRC_EXT at DOMAIN:PORT) Callstate Change RINGING -> EARLY Leg A: sofia.c:7291 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering state [early][183] Leg B: sofia.c:7291 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [completing][200] Leg B: switch_channel.c:3482 (sofia/internal/DST_EXT at LAN_IP:PORT) Callstate Change RINGING -> EARLY Leg B: sofia.c:7291 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [ready][200] Leg B: sofia.c:8429 Channel [sofia/internal/DST_EXT at LAN_IP:PORT] has been answered Leg B: switch_channel.c:3781 (sofia/internal/DST_EXT at LAN_IP:PORT) Callstate Change EARLY -> ACTIVE Leg A: switch_ivr_bridge.c:766 Channel [sofia/internal/SRC_EXT at DOMAIN:PORT] has been answered Leg A: sofia.c:7291 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering state [completed][200] Leg A: switch_channel.c:3781 (sofia/internal/SRC_EXT at DOMAIN:PORT) Callstate Change EARLY -> ACTIVE Leg A: sofia.c:7291 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering state [ready][200] Leg A: switch_ivr_async.c:1500 No silence detection configured; assuming start of speech Leg B: sofia.c:7291 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [calling][0] Leg A: sofia.c:7291 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering state [calling][0] Leg A: sofia.c:7291 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering state [ready][200] Leg A: sofia.c:8272 Processing updated SDP Leg B: sofia.c:7291 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [ready][200] FreeSWITCH 1.6.20 Leg A: switch_channel.c:2249 (sofia/internal/SRC_EXT at DOMAIN:PORT) Callstate Change DOWN -> RINGING Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [calling][0] Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [proceeding][180] Leg B: sofia.c:7192 Ring-Ready sofia/internal/DST_EXT at LAN_IP:PORT! Leg B: switch_channel.c:3346 (sofia/internal/DST_EXT at LAN_IP:PORT) Callstate Change DOWN -> RINGING Leg A: switch_ivr_originate.c:1215 Sending early media Leg A: switch_channel.c:3474 (sofia/internal/SRC_EXT at DOMAIN:PORT) Callstate Change RINGING -> EARLY Leg A: sofia.c:7084 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering state [early][183] Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [completing][200] Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [ready][200] Leg A: switch_channel.c:3773 (sofia/internal/SRC_EXT at DOMAIN:PORT) Callstate Change EARLY -> ACTIVE Leg A: sofia.c:7084 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering state [completed][200] Leg A: switch_ivr_originate.c:3705 Originate Resulted in Success: [sofia/internal/DST_EXT at LAN_IP:PORT] Leg B: switch_channel.c:3773 (sofia/internal/DST_EXT at LAN_IP:PORT) Callstate Change RINGING -> ACTIVE Leg A: sofia.c:7084 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering state [ready][200] Leg B: Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [ready][200] Leg A: switch_ivr_async.c:1500 No silence detection configured; assuming start of speech Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [calling][0] Leg A: sofia.c:7084 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering state [calling][0] Leg A: sofia.c:7084 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering state [ready][200] Leg A: sofia.c:8061 Processing updated SDP Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [completing][200] Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [ready][200] On 1.6.20 Leg B changes straight from RINGING to ACTIVE, but on 1.8.2 Leg B first changes from RINGING to EARLY then EARLY to ACTIVE, not sure if this could be related. We've experimented with the following to no avail. rtp-rewrite-timestamps send_silence_when_idle fsctl sync_clock suppress_cng ignore_early_media As per: https://freeswitch.org/confluence/display/FREESWITCH/RTP+Issues https://freeswitch.org/confluence/display/FREESWITCH/VAD+and+CNG https://freeswitch.org/confluence/display/FREESWITCH/send_silence_when_idle https://freeswitch.org/confluence/display/FREESWITCH/Early+Media The calls are local between two extensions\endpoints on the same FreeSWITCH instance and the same SIP profile, the SIP profiles on both servers (1.6.20 and 1.8.2) are identical. Does anyone have any ideas? Thanks, Shaun -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Tue Dec 11 19:24:35 2018 From: mike at jerris.com (Michael Jerris) Date: Tue, 11 Dec 2018 14:24:35 -0500 Subject: [Freeswitch-users] Replace User-Agent and Server headers In-Reply-To: References: <94A74EA3-A0EA-4F1E-9CFF-A685C5F0CFC0@gmx.net> Message-ID: That is correct, and that is intentional. There will never be away to hide all signs that you are using freeswitch. If you really feel the need to hide that, you’ll have to inspect the code and figure out the changes you need to do on your own. There are probably 100 ways I can tell you are using FreeSWITCH remotely, most have no configuration way to disable. > On Dec 9, 2018, at 5:39 PM, Dmitry Sytchev wrote: > > Besides UA and session owner fields there are special headers, methods > list, callid generation, contact field format, headers sequence and so > on. > > So I think if somebody really wants to dig some info about your > switch, there are many other ways to detect it's using FreeSwitch. > пт, 23 нояб. 2018 г. в 17:44, Markus Bönke : >> >> You can set: >> >> >> >> in the sipprofile config. >> >> >>> Am 23.11.2018 um 09:44 schrieb Kevin Olbrich : >>> >>> Hi! >>> >>> How can I replace User-Agent and Server headers in freeswitch? >>> This is some kind of hardening (why tell people which version I use if >>> the don't need to know...). >>> >>> Sure, I use the latest releases but I would like to hide the version I >>> use or that I use freeswitch at all. From mike at jerris.com Tue Dec 11 19:26:21 2018 From: mike at jerris.com (Michael Jerris) Date: Tue, 11 Dec 2018 14:26:21 -0500 Subject: [Freeswitch-users] Disabling 200 OK with contact headers of all the lines In-Reply-To: References: Message-ID: These are all valid current registrations, there are params you can adjust for multi reg that will replace previous registrations instead of allowing multiple. > On Dec 9, 2018, at 1:17 PM, sagar malam wrote: > > Hello , > > I am using FS SLA feature and it works very well.However i am facing an issue of registrations getting dropped as explained below : > There are 3 phones registered with same extension number.Sofia is configured with "sip-expires-max-deviation" to randomise registration through expiry header in contact header.All the phones are Polycom. In case of shared lines FS adds contact header of all the lines(or phones with same extension) in 200 OK as shown below due to which all the phones are reading expiry timer from first contact header only.So in below example,Phone re registers after 379 seconds(first contact header) instead of 88 seconds(third contact header) leading to registration expiry on FS. > > Reason why phones are always reading expiry from first contact header is same Public IP(contact is re written by Proxy in front of FS) for all three phones which is confusing phone to identify its own contact header. > Is there any way to configure FS to not send contact headers of all the registrations but only one that belongs to the line itself ? or any other way to fix it. > ============================200 OK for register packet ============= > > 2018/12/08 13:36:35.426099 10.50.7.251:5070 -> 10.50.7.253:5060 > SIP/2.0 200 OK > Via: SIP/2.0/UDP 66.160.237.253:5060;branch=z9hG4bK9b46.f9b7a68125108f411537617d02bd48f8.0;received=10.50.7.253 > Via: SIP/2.0/UDP 198.136.236.1:5060;rport=5060;received=10.50.8.1;branch=z9hG4bK9b46.e6f1aa3817a238f499aa3aafa4217425.0 > Via: SIP/2.0/UDP 172.16.1.11;rport=1426;received=71.239.113.14;branch=z9hG4bK1df2302cD062D28F > From: "Main Line" >;tag=22CE54C0-84BCCF23 > To: >;tag=e8Fa2tND4U80D > Call-ID: d66f0f0592c09746b903406f312eb0c2 > CSeq: 590 REGISTER > Contact: t=tcp>;expires=379 > Contact: ort=tcp>;expires=245 > Contact: eb0c2>;expires=88 > Date: Sat, 08 Dec 2018 08:06:35 GMT > User-Agent: FreeSWITCH-mod_sofia/1.6.17~64bit > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: path, replaces > Content-Length: 0 -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Tue Dec 11 19:28:25 2018 From: mike at jerris.com (Michael Jerris) Date: Tue, 11 Dec 2018 14:28:25 -0500 Subject: [Freeswitch-users] FS and gcc 8.x.x In-Reply-To: References: Message-ID: <0B4A3051-6878-45AB-9934-EB4A65C6CAEF@jerris.com> Looking for community contributions to help fix these issues properly (not just disabling errors, but actually fixing the problems). > On Dec 10, 2018, at 5:16 AM, Sergey Safarov wrote: > > Relate > https://freeswitch.org/jira/browse/FS-11345 > > пн, 10 дек. 2018 г. в 12:49, Madovsky >: > Just FYI FS does not compile anymore with gcc 8.x.x > > major changes have been made with this new version of gcc, especially > strncpy and strncat. > -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Tue Dec 11 19:30:10 2018 From: mike at jerris.com (Michael Jerris) Date: Tue, 11 Dec 2018 14:30:10 -0500 Subject: [Freeswitch-users] Issue compiling v 1.8 on CentOS 7 In-Reply-To: References: Message-ID: <2C2DDF76-4888-4F4C-A9E7-8E41CC66FBEA@jerris.com> This pull request is in conflict, can you please update it, and remove any unrelated changes. Should ONLY add what is necessary to get it to build right, NOT any other changes in this pr, if you want to further change how things work, please move those changes to another PR. > On Dec 10, 2018, at 5:33 AM, Sergey Safarov wrote: > > Update for SPEC file to compile on CentOS 7 is located here > https://freeswitch.org/stash/projects/FS/repos/freeswitch/pull-requests/1561/overview > > > пн, 10 дек. 2018 г. в 13:19, Ryan Delgrosso >: > Hey all, > > Im working on building v1.8 on CentOS but im having trouble getting to recognize that libemp3lame-dev is present. Any suggestions on how I can get configure to locate the installed version and get mod_shout to build? > > > > making all mod_shout > make[4]: Entering directory `/usr/local/src/freeswitch/src/mod/formats/mod_shout' > Makefile:904: *** You must install libmp3lame-dev to build mod_shout. Stop. > make[4]: Leaving directory `/usr/local/src/freeswitch/src/mod/formats/mod_shout' > make[3]: *** [mod_shout-all] Error 1 > make[3]: Leaving directory `/usr/local/src/freeswitch/src/mod' > make[2]: *** [all-recursive] Error 1 > make[2]: Leaving directory `/usr/local/src/freeswitch/src' > make[1]: *** [all-recursive] Error 1 > make[1]: Leaving directory `/usr/local/src/freeswitch' > make: *** [all] Error 2 > > > # yum list | grep lame > lame.x86_64 3.100-1.el7 @epel > lame-devel.x86_64 3.100-1.el7 @epel > lame-libs.x86_64 3.100-1.el7 @epel > lame-mp3x.x86_64 3.100-1.el7 @epel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Tue Dec 11 19:32:36 2018 From: mike at jerris.com (Michael Jerris) Date: Tue, 11 Dec 2018 14:32:36 -0500 Subject: [Freeswitch-users] FS and PHP In-Reply-To: <66d30c45-01be-f2c5-b7b0-85eaeb9e7ba0@madovsky.org> References: <66d30c45-01be-f2c5-b7b0-85eaeb9e7ba0@madovsky.org> Message-ID: <7D4D25C8-52B8-46CA-862E-4369616B4825@jerris.com> Configure uses the php-config program to locate the right version. Setting shell vars to find your version of that command will cause it to use that > On Dec 10, 2018, at 3:57 PM, Madovsky wrote: > > I'm looking for a way to compile FS with a custom PHP folder (path is /usr/local/php-5.6) > > did not find any way to force FS to look into this folder rather than the system ones. From infos at madovsky.org Tue Dec 11 21:47:10 2018 From: infos at madovsky.org (Madovsky) Date: Tue, 11 Dec 2018 13:47:10 -0800 Subject: [Freeswitch-users] FS and PHP In-Reply-To: <7D4D25C8-52B8-46CA-862E-4369616B4825@jerris.com> References: <66d30c45-01be-f2c5-b7b0-85eaeb9e7ba0@madovsky.org> <7D4D25C8-52B8-46CA-862E-4369616B4825@jerris.com> Message-ID: Should I use PHP_CONFIG="...." ./configure? On 12/11/2018 11:32 AM, Michael Jerris wrote: > Configure uses the php-config program to locate the right version. Setting shell vars to find your version of that command will cause it to use that > > >> On Dec 10, 2018, at 3:57 PM, Madovsky wrote: >> >> I'm looking for a way to compile FS with a custom PHP folder (path is /usr/local/php-5.6) >> >> did not find any way to force FS to look into this folder rather than the system ones. > > _________________________________________________________________________ > 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 krice at tollfreegateway.com Wed Dec 12 00:14:06 2018 From: krice at tollfreegateway.com (Ken Rice) Date: Tue, 11 Dec 2018 18:14:06 -0600 Subject: [Freeswitch-users] Replace User-Agent and Server headers In-Reply-To: References: <94A74EA3-A0EA-4F1E-9CFF-A685C5F0CFC0@gmx.net> Message-ID: <3B2C1B0C-569E-4247-8A91-4FB1A978DE85@tollfreegateway.com> not to mention this is security by obscurity and that rarely if ever actually works Sent from my iPhone > On Dec 11, 2018, at 13:24, Michael Jerris wrote: > > That is correct, and that is intentional. There will never be away to hide all signs that you are using freeswitch. If you really feel the need to hide that, you’ll have to inspect the code and figure out the changes you need to do on your own. There are probably 100 ways I can tell you are using FreeSWITCH remotely, most have no configuration way to disable. > >> On Dec 9, 2018, at 5:39 PM, Dmitry Sytchev wrote: >> >> Besides UA and session owner fields there are special headers, methods >> list, callid generation, contact field format, headers sequence and so >> on. >> >> So I think if somebody really wants to dig some info about your >> switch, there are many other ways to detect it's using FreeSwitch. >> пт, 23 нояб. 2018 г. в 17:44, Markus Bönke : >>> >>> You can set: >>> >>> >>> >>> in the sipprofile config. >>> >>> >>>> Am 23.11.2018 um 09:44 schrieb Kevin Olbrich : >>>> >>>> Hi! >>>> >>>> How can I replace User-Agent and Server headers in freeswitch? >>>> This is some kind of hardening (why tell people which version I use if >>>> the don't need to know...). >>>> >>>> Sure, I use the latest releases but I would like to hide the version I >>>> use or that I use freeswitch at all. > > > _________________________________________________________________________ > 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 sat at calgaryit.com Wed Dec 12 04:02:25 2018 From: sat at calgaryit.com (George) Date: Tue, 11 Dec 2018 21:02:25 -0700 (MST) Subject: [Freeswitch-users] calls to local 311 line with two IVRs Message-ID: <26233380.38805.1544587345787.JavaMail.zimbra@calgaryit.com> when I call a local 311 number, the call goes through, I hear the first announcement, after with time the call is supposed to get transferred to a second IVR which should give the choices of departments, the call just goes silent and eventually gets hung on on. FreeSWITCH Version 1.8.2-3-a98a958ac3~64bit (-3-a98a958ac3 64bit) Debian 9.6 this is what my FS sends back to the provider: v=0 o=FreeSWITCH 1544570427 1544570429 IN IP4 10.185.45.238 s=FreeSWITCH c=IN IP4 10.185.45.238 t=0 0 m=audio 16506 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendonly a=ptime:20 the provider says its because the a=sendonly I am not receiving the announcements any help on this? Thank You, George From s.safarov at gmail.com Wed Dec 12 05:11:00 2018 From: s.safarov at gmail.com (Sergey Safarov) Date: Wed, 12 Dec 2018 08:11:00 +0300 Subject: [Freeswitch-users] Issue compiling v 1.8 on CentOS 7 In-Reply-To: <2C2DDF76-4888-4F4C-A9E7-8E41CC66FBEA@jerris.com> References: <2C2DDF76-4888-4F4C-A9E7-8E41CC66FBEA@jerris.com> Message-ID: PR is recreated at https://freeswitch.org/stash/projects/FS/repos/freeswitch/pull-requests/1638 Sergey ср, 12 дек. 2018 г. в 02:01, Michael Jerris : > This pull request is in conflict, can you please update it, and remove any > unrelated changes. Should ONLY add what is necessary to get it to build > right, NOT any other changes in this pr, if you want to further change how > things work, please move those changes to another PR. > > > On Dec 10, 2018, at 5:33 AM, Sergey Safarov wrote: > > Update for SPEC file to compile on CentOS 7 is located here > > https://freeswitch.org/stash/projects/FS/repos/freeswitch/pull-requests/1561/overview > > > пн, 10 дек. 2018 г. в 13:19, Ryan Delgrosso : > >> Hey all, >> >> Im working on building v1.8 on CentOS but im having trouble getting to >> recognize that libemp3lame-dev is present. Any suggestions on how I can get >> configure to locate the installed version and get mod_shout to build? >> >> >> ------------------------------ >> >> making all mod_shout >> make[4]: Entering directory >> `/usr/local/src/freeswitch/src/mod/formats/mod_shout' >> Makefile:904: *** You must install libmp3lame-dev to build mod_shout. >> Stop. >> make[4]: Leaving directory >> `/usr/local/src/freeswitch/src/mod/formats/mod_shout' >> make[3]: *** [mod_shout-all] Error 1 >> make[3]: Leaving directory `/usr/local/src/freeswitch/src/mod' >> make[2]: *** [all-recursive] Error 1 >> make[2]: Leaving directory `/usr/local/src/freeswitch/src' >> make[1]: *** [all-recursive] Error 1 >> make[1]: Leaving directory `/usr/local/src/freeswitch' >> make: *** [all] Error 2 >> >> # yum list | grep lame >> lame.x86_64 3.100-1.el7 >> @epel >> lame-devel.x86_64 3.100-1.el7 >> @epel >> lame-libs.x86_64 3.100-1.el7 >> @epel >> lame-mp3x.x86_64 3.100-1.el7 >> @epel >> ------------------------------ >> > > _________________________________________________________________________ > 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 filipe at redseven.tech Wed Dec 12 10:41:57 2018 From: filipe at redseven.tech (Filipe Cunha) Date: Wed, 12 Dec 2018 10:41:57 +0000 Subject: [Freeswitch-users] FS 1.8 Docker image Message-ID: Hi all, I'm trying to create a docker image with the latest version of FreeSWITCH. But I'm not able to do it. These are the steps that I did to try to do it. 1. Download debian stretch image *docker pull debian:stretch * 2. Build FreeSWITCH from source code following the guide in confluence step by step. I follow the production version 3. The step 2 went very well, and then I follow the post-installation instructions and this was when it stop working. I set up owners and permissions and when I try to do the systemd I was not able to do it. I copied the file *cp /usr/src/freeswitch/debian/freeswitch-systemd.freeswitch.service /etc/systemd/system/freeswitch.service *the content of the file was not exactly the same as the one in the tutorial, so I copied the content of the file on the tutorial and replace it. And when I tried to do *systemctl deamon-reload* I got this error message, *Failed to connect to bus: No such file or directory*. After trying to find a solution online I found this answer on SO Do you guys know if there is a solution for this problem? We plan to use the latest version of FreeSwitch in our production environment, but we have a normal linux server to do it, and the devops team got everything working fine, we only have this problem when we try to use docker. I'm trying to use docker so my team could test everything locally before pushing something to the test environment, so far we have been deploying our code to test environment to test our code 😖 I found this image that works fine in our local environment, but the version is 1.6.16. Is there any braking change between 1.8.2 and 1.6.16? My concern it that we start developing against an older version, and when we deploy to production it doesn't work, or maybe the latest version have a super useful command that we are not taking advantage of. We control FreeSWITCH mainly by events, we use java to do it. Thanks for the help, Filipe Cunha -------------- next part -------------- An HTML attachment was scrubbed... URL: From tg-maillistings at level5.de Wed Dec 12 15:52:18 2018 From: tg-maillistings at level5.de (=?UTF-8?Q?Thorsten_G=c3=b6llner?=) Date: Wed, 12 Dec 2018 16:52:18 +0100 Subject: [Freeswitch-users] 3CX-Sip-Trunk / Freeswitch cannot reach 3CX Message-ID: <34d3fea8-1a28-b495-386a-a787469e99e3@level5.de> Hi, has anybody experience with 3CX-SIP-Trunks? I have installed Freeswitch and added an extension (trunk-1). On 3CX-Side I added a SIP-Trunk (generic sip provider) and I can see a succesfull registration. Now I make a call to Freeswitch and want to pass it to 3CX-Sip-Trunk. But when bridging to user/trunk-1 I get a "Originate Resulted in Error Cause: 41 [NORMAL_TEMPORARY_FAILURE]" error (with no delay). 3CX is installed as a VM in a Azure-Cloud behind NAT. I disabled local iptables and the Firewall-Test on 3CX is all fine. Freeswitch is on bare metal with a public ip. When making a call, wireshark on 3CX-machine shows the INVITE followed by "Destination unreachable (Port unreachable)". But I am not sure, if this is a firewall-issue or something else. Any idea? Thanks in advance, Thorsten From mike at jerris.com Wed Dec 12 16:56:04 2018 From: mike at jerris.com (Michael Jerris) Date: Wed, 12 Dec 2018 11:56:04 -0500 Subject: [Freeswitch-users] FS and PHP In-Reply-To: References: <66d30c45-01be-f2c5-b7b0-85eaeb9e7ba0@madovsky.org> <7D4D25C8-52B8-46CA-862E-4369616B4825@jerris.com> Message-ID: You’ll need to do it with PATH > On Dec 11, 2018, at 4:47 PM, Madovsky wrote: > > Should I use PHP_CONFIG="...." ./configure? > > On 12/11/2018 11:32 AM, Michael Jerris wrote: >> Configure uses the php-config program to locate the right version. Setting shell vars to find your version of that command will cause it to use that >> >> >>> On Dec 10, 2018, at 3:57 PM, Madovsky wrote: >>> >>> I'm looking for a way to compile FS with a custom PHP folder (path is /usr/local/php-5.6) >>> >>> did not find any way to force FS to look into this folder rather than the system ones. From mike at jerris.com Wed Dec 12 16:57:09 2018 From: mike at jerris.com (Michael Jerris) Date: Wed, 12 Dec 2018 11:57:09 -0500 Subject: [Freeswitch-users] calls to local 311 line with two IVRs In-Reply-To: <26233380.38805.1544587345787.JavaMail.zimbra@calgaryit.com> References: <26233380.38805.1544587345787.JavaMail.zimbra@calgaryit.com> Message-ID: Would need to see the full sip trace to answer this. A debug log with sip trace is most useful > On Dec 11, 2018, at 11:02 PM, George wrote: > > when I call a local 311 number, the call goes through, I hear the first announcement, after with time the call is supposed to get transferred to a second IVR which should give the choices of departments, the call just goes silent and eventually gets hung on on. > > FreeSWITCH Version 1.8.2-3-a98a958ac3~64bit (-3-a98a958ac3 64bit) > Debian 9.6 > > this is what my FS sends back to the provider: > > v=0 > o=FreeSWITCH 1544570427 1544570429 IN IP4 10.185.45.238 > s=FreeSWITCH > c=IN IP4 10.185.45.238 > t=0 0 > m=audio 16506 RTP/AVP 0 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=sendonly > a=ptime:20 > > the provider says its because the a=sendonly I am not receiving the announcements > > any help on this? > > Thank You, > George From mike at jerris.com Wed Dec 12 17:05:06 2018 From: mike at jerris.com (Michael Jerris) Date: Wed, 12 Dec 2018 12:05:06 -0500 Subject: [Freeswitch-users] Issue compiling v 1.8 on CentOS 7 In-Reply-To: References: <2C2DDF76-4888-4F4C-A9E7-8E41CC66FBEA@jerris.com> Message-ID: <41FFA8CA-E347-4D04-A292-2C8E384F6E52@jerris.com> The PR is not formatted properly to be merged but we will take a look and see what of these changes we can manually merge instead. In the future please make sure to follow the guidelines for commit notes and PR’s./ > On Dec 12, 2018, at 12:11 AM, Sergey Safarov wrote: > > PR is recreated at https://freeswitch.org/stash/projects/FS/repos/freeswitch/pull-requests/1638 > > Sergey > > ср, 12 дек. 2018 г. в 02:01, Michael Jerris >: > This pull request is in conflict, can you please update it, and remove any unrelated changes. Should ONLY add what is necessary to get it to build right, NOT any other changes in this pr, if you want to further change how things work, please move those changes to another PR. > > >> On Dec 10, 2018, at 5:33 AM, Sergey Safarov > wrote: >> >> Update for SPEC file to compile on CentOS 7 is located here >> https://freeswitch.org/stash/projects/FS/repos/freeswitch/pull-requests/1561/overview >> >> >> пн, 10 дек. 2018 г. в 13:19, Ryan Delgrosso >: >> Hey all, >> >> Im working on building v1.8 on CentOS but im having trouble getting to recognize that libemp3lame-dev is present. Any suggestions on how I can get configure to locate the installed version and get mod_shout to build? >> >> >> >> making all mod_shout >> make[4]: Entering directory `/usr/local/src/freeswitch/src/mod/formats/mod_shout' >> Makefile:904: *** You must install libmp3lame-dev to build mod_shout. Stop. >> make[4]: Leaving directory `/usr/local/src/freeswitch/src/mod/formats/mod_shout' >> make[3]: *** [mod_shout-all] Error 1 >> make[3]: Leaving directory `/usr/local/src/freeswitch/src/mod' >> make[2]: *** [all-recursive] Error 1 >> make[2]: Leaving directory `/usr/local/src/freeswitch/src' >> make[1]: *** [all-recursive] Error 1 >> make[1]: Leaving directory `/usr/local/src/freeswitch' >> make: *** [all] Error 2 >> >> >> # yum list | grep lame >> lame.x86_64 3.100-1.el7 @epel >> lame-devel.x86_64 3.100-1.el7 @epel >> lame-libs.x86_64 3.100-1.el7 @epel >> lame-mp3x.x86_64 3.100-1.el7 @epel >> -------------- next part -------------- An HTML attachment was scrubbed... URL: From rmarand at prosodie.com Wed Dec 12 17:42:10 2018 From: rmarand at prosodie.com (MARAND, Remi) Date: Wed, 12 Dec 2018 17:42:10 +0000 Subject: [Freeswitch-users] One way audio case between FS and Chrome on WebRTC In-Reply-To: <21c60857-88ba-c77e-f624-57d6661a12f2@m4x.org> References: <21c60857-88ba-c77e-f624-57d6661a12f2@m4x.org> Message-ID: Hello, Sorry, i come back with this case. I understand that the browsers behaviour is to use DTLS... but in my case it is not the case. I have to set this constraints in my client javascript, and i do not find how to code it with sip.js.... Should you please give me a way to set correctly that on the Freeswitch side, a configuration option to specify « WebRTC Media » on connection ?? I tried with in the dialplan/default.xml file, but it does not change, i tried before with {media_webrtc=true} in the extension file without success. Thanks for help. Rémi Marand. Capgemini/Prosodie. De : FreeSWITCH-users De la part de Jeremy Lainé Envoyé : lundi 19 novembre 2018 09:18 À : freeswitch-users at lists.freeswitch.org Objet : Re: [Freeswitch-users] One way audio case between FS and Chrome on WebRTC That is very odd, the default behaviour in browsers *is* to use DTLS for SRTP keying instead of the legacy SDES. In fact SDES's status is "MUST NOT implement" since 2013, and the DtlsSrtpKeyAgreement constraint is scheduled for retirement: https://bugs.chromium.org/p/chromium/issues/detail?id=804275 Concerning SIP.js I have never experienced the need to set this constraint, it works fine against freeswitch without it. Jeremy On 11/13/18 8:55 AM, MARAND, Remi wrote: Hello, I finally was able to solve my problem. For information, on this trouble, the parameter : RTCConstraints: {"optional": [{'DtlsSrtpKeyAgreement': 'true'}]} Is to add at the good place in jquery.FSRTC.js (or verto-min.js (I think that's already the case in this script)) i did not find how to code in the sip.js version but it should be possible. Thanks to those who answered my question, and sorry for the 3 Mb of pcap file i sent to the user-list !!! Perhaps this DtlsSrtpKeyAgreement parameter role should be added and explain in the Verto/WebRTC examples availables on Websites, i suppose that in 2014, it was not mandatory but now with the lasts versions of Chrome and FF it seems to be. Regards. Remi Marand. rmarand at prosodie.com +33687725325. De : MARAND, Remi Envoyé : lundi 5 novembre 2018 18:01 À : freeswitch-users at lists.freeswitch.org Objet : One way audio case between FS and Chrome on WebRTC Hello, I am trying to validate FS as a SIP to WebRTC Gateway in our lab environment. I started in middle October and have good result on it, but i cannot understand this One Way Audio trouble. I must thank the Freeswitch team and contributors for this very impressive work. FreeSWITCH Version 1.8.1-2-4f54cff36a~64bit (-2-4f54cff36a 64bit) On system: SMP Debian 4.9.110-3+deb9u6 (2018-10-08) x86_64 GNU/Linux Openssl version : OpenSSL 1.1.0f 25 May 2017 Chrome version: 69.0 (I tried with different version and with Firefox with the same trouble). The wss part is ok with sip.js and verto.js The Ice negotiation is ok, I use sometimes local networks and sometimes web, I have had to authorize networks in the candidate ACL and domain ACL (acl.conf.xml) The result is the same on both topology. DTLS negotiation is OK, and there is UDP streams between Chrome (or Firefox) and FS in both ways. There is no "audible" audio in the direction from FS to Chrome, the other direction is OK. The simplest test is to call the 5000 number from the Chrome client, I send you a paste bin and pcap trace for this call. Should you give me information element to progress on this, what is really mandatory in the sip_profile/internal.xml and external.xml files, and in directory/default/1000.xml for a WebRTC call ?? What should be the good options in fs_cli to see if the encryption of RTP packets is ok or not.. ? Do you think that I have to reinstall a Freeswitch from the current branch ? @IP for FS : 192.168.145.67 @IP for Chrome : 10.70.54.43 Link on the pastebin : https://pastebin.freeswitch.org/view/09a72087 I have a pcap on the same call that I can provide (3 Mb) if necessary.. Thank you for helping me !! Best regards. [Prosodie-signature] Rémi Marand - Product Owner - Pod Connect. PROSODIE - Marketing & Produit Tél. : +33 (0)1.46.84.12.77 / 06.87.72.53.25 rmarand at prosodie.com This message contains information that may be privileged or confidential and is the property of the Capgemini Group. It is intended only for the person to whom it is addressed. If you are not the intended recipient, you are not authorized to read, print, retain, copy, disseminate, distribute, or use this message or any part thereof. If you receive this message in error, please notify the sender immediately and delete all copies of this message. _________________________________________________________________________ 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: -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.gif Type: image/gif Size: 3368 bytes Desc: image001.gif URL: From s.safarov at gmail.com Wed Dec 12 17:47:43 2018 From: s.safarov at gmail.com (Sergey Safarov) Date: Wed, 12 Dec 2018 20:47:43 +0300 Subject: [Freeswitch-users] FS 1.8 Docker image In-Reply-To: References: Message-ID: you can use safarov/freeswitch:1.8.2 image repo of this is located here https://bitbucket.org/sergey-safarov/freeswitch/commits/branch/v1.8 ср, 12 дек. 2018 г. в 18:14, Filipe Cunha : > Hi all, > > I'm trying to create a docker image with the latest version of FreeSWITCH. > But I'm not able to do it. These are the steps that I did to try to do it. > > 1. Download debian stretch image > *docker pull debian:stretch * > 2. Build FreeSWITCH from source code following the guide > > in confluence step by step. I follow the production version > 3. The step 2 went very well, and then I follow the post-installation > instructions > > and this was when it stop working. I set up owners and permissions and when > I try to do the systemd I was not able to do it. I copied the file *cp > /usr/src/freeswitch/debian/freeswitch-systemd.freeswitch.service > /etc/systemd/system/freeswitch.service *the content of the file was not > exactly the same as the one in the tutorial, so I copied the content of the > file on the tutorial and replace it. And when I tried to do *systemctl > deamon-reload* I got this error message, *Failed to connect to bus: No > such file or directory*. After trying to find a solution online I found > this answer on SO > > > Do you guys know if there is a solution for this problem? > We plan to use the latest version of FreeSwitch in our production > environment, but we have a normal linux server to do it, and the devops > team got everything working fine, we only have this problem when we try to > use docker. > I'm trying to use docker so my team could test everything locally before > pushing something to the test environment, so far we have been deploying > our code to test environment to test our code 😖 > I found this image > that works fine in our local environment, but the version is 1.6.16. > Is there any braking change between 1.8.2 and 1.6.16? My concern it that > we start developing against an older version, and when we deploy to > production it doesn't work, or maybe the latest version have a super useful > command that we are not taking advantage of. > We control FreeSWITCH mainly by events, we use java to do it. > > Thanks for the help, > Filipe Cunha > > > > > _________________________________________________________________________ > 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 sat at calgaryit.com Wed Dec 12 18:34:16 2018 From: sat at calgaryit.com (George) Date: Wed, 12 Dec 2018 11:34:16 -0700 (MST) Subject: [Freeswitch-users] calls to local 311 line with two IVRs In-Reply-To: References: <26233380.38805.1544587345787.JavaMail.zimbra@calgaryit.com> Message-ID: <501702402.43110.1544639656116.JavaMail.zimbra@calgaryit.com> here it is: https://freeswitch.org/jira/secure/attachment/28046/cap20181212.pcap Thank You, George ----- Original Message ----- From: "Michael Jerris" To: "freeswitch-users" Sent: Wednesday, December 12, 2018 9:57:09 AM Subject: Re: [Freeswitch-users] calls to local 311 line with two IVRs Would need to see the full sip trace to answer this. A debug log with sip trace is most useful > On Dec 11, 2018, at 11:02 PM, George wrote: > > when I call a local 311 number, the call goes through, I hear the first announcement, after with time the call is supposed to get transferred to a second IVR which should give the choices of departments, the call just goes silent and eventually gets hung on on. > > FreeSWITCH Version 1.8.2-3-a98a958ac3~64bit (-3-a98a958ac3 64bit) > Debian 9.6 > > this is what my FS sends back to the provider: > > v=0 > o=FreeSWITCH 1544570427 1544570429 IN IP4 10.185.45.238 > s=FreeSWITCH > c=IN IP4 10.185.45.238 > t=0 0 > m=audio 16506 RTP/AVP 0 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=sendonly > a=ptime:20 > > the provider says its because the a=sendonly I am not receiving the announcements > > any help on this? > > Thank You, > George _________________________________________________________________________ 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 tahir at ictinnovations.com Thu Dec 13 18:16:36 2018 From: tahir at ictinnovations.com (Tahir Almas) Date: Thu, 13 Dec 2018 10:16:36 -0800 Subject: [Freeswitch-users] ICTBroadcast Enterprise Edition free licenses available Message-ID: On completion of 10 years of ICTBroadcast , the unified telemarketing software, We are happy to offer free licenses of ICTBroadcast Enterprise Edition 2 channels , 5 channels , 10 channels and 50 channels licenses. Limited licenses Register now https://service.ictinnovations.com/cart.php?gid=1 http://www.ictbroadcast.com Regards *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT -------------- next part -------------- An HTML attachment was scrubbed... URL: From alex at freeswitch.com Fri Dec 14 02:30:18 2018 From: alex at freeswitch.com (Alexey Sibyakin) Date: Fri, 14 Dec 2018 11:30:18 +0900 Subject: [Freeswitch-users] FS 1.8 Docker image In-Reply-To: References: Message-ID: Hi, Try to install dbus into your Stretch. Regards, Alex On Thu, Dec 13, 2018 at 3:40 AM Sergey Safarov wrote: > you can use safarov/freeswitch:1.8.2 > image > repo of this is located here > https://bitbucket.org/sergey-safarov/freeswitch/commits/branch/v1.8 > > > ср, 12 дек. 2018 г. в 18:14, Filipe Cunha : > >> Hi all, >> >> I'm trying to create a docker image with the latest version of >> FreeSWITCH. But I'm not able to do it. These are the steps that I did to >> try to do it. >> >> 1. Download debian stretch image >> *docker pull debian:stretch * >> 2. Build FreeSWITCH from source code following the guide >> >> in confluence step by step. I follow the production version >> 3. The step 2 went very well, and then I follow the post-installation >> instructions >> >> and this was when it stop working. I set up owners and permissions and when >> I try to do the systemd I was not able to do it. I copied the file *cp >> /usr/src/freeswitch/debian/freeswitch-systemd.freeswitch.service >> /etc/systemd/system/freeswitch.service *the content of the file was not >> exactly the same as the one in the tutorial, so I copied the content of the >> file on the tutorial and replace it. And when I tried to do *systemctl >> deamon-reload* I got this error message, *Failed to connect to bus: No >> such file or directory*. After trying to find a solution online I found >> this answer on SO >> >> >> Do you guys know if there is a solution for this problem? >> We plan to use the latest version of FreeSwitch in our production >> environment, but we have a normal linux server to do it, and the devops >> team got everything working fine, we only have this problem when we try to >> use docker. >> I'm trying to use docker so my team could test everything locally before >> pushing something to the test environment, so far we have been deploying >> our code to test environment to test our code 😖 >> I found this image >> that works fine in our local environment, but the version is 1.6.16. >> Is there any braking change between 1.8.2 and 1.6.16? My concern it that >> we start developing against an older version, and when we deploy to >> production it doesn't work, or maybe the latest version have a super useful >> command that we are not taking advantage of. >> We control FreeSWITCH mainly by events, we use java to do it. >> >> Thanks for the help, >> Filipe Cunha >> >> >> >> >> _________________________________________________________________________ >> 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 alex at freeswitch.com Fri Dec 14 02:33:34 2018 From: alex at freeswitch.com (Alexey Sibyakin) Date: Fri, 14 Dec 2018 11:33:34 +0900 Subject: [Freeswitch-users] 3CX-Sip-Trunk / Freeswitch cannot reach 3CX In-Reply-To: <34d3fea8-1a28-b495-386a-a787469e99e3@level5.de> References: <34d3fea8-1a28-b495-386a-a787469e99e3@level5.de> Message-ID: Hi, "Port unreachable" is a network (routing/NAT/firewall) issue. Regards, Alex On Thu, Dec 13, 2018 at 2:00 AM Thorsten Göllner wrote: > Hi, > > has anybody experience with 3CX-SIP-Trunks? I have installed Freeswitch > and added an extension (trunk-1). On 3CX-Side I added a SIP-Trunk > (generic sip provider) and I can see a succesfull registration. Now I > make a call to Freeswitch and want to pass it to 3CX-Sip-Trunk. But when > bridging to user/trunk-1 I get a "Originate Resulted in Error Cause: 41 > [NORMAL_TEMPORARY_FAILURE]" error (with no delay). > > 3CX is installed as a VM in a Azure-Cloud behind NAT. I disabled local > iptables and the Firewall-Test on 3CX is all fine. > > Freeswitch is on bare metal with a public ip. > > When making a call, wireshark on 3CX-machine shows the INVITE followed > by "Destination unreachable (Port unreachable)". But I am not sure, if > this is a firewall-issue or something else. > > Any idea? > > Thanks in advance, > Thorsten > > > _________________________________________________________________________ > 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 bilaln018 at gmail.com Fri Dec 14 11:22:46 2018 From: bilaln018 at gmail.com (Bilal Abbasi) Date: Fri, 14 Dec 2018 16:22:46 +0500 Subject: [Freeswitch-users] [Zombie Ports Listening] Message-ID: Hi Users, I just installed FS 1.8.2, I enabled the bare minimal configurations needed for my platform, but i can see see two ports listening. I searched them and found out that its msrp ports, but i do not have any configurations/modules for these ports and they are still there, can somebody guide me how to put them down. tcp 0 0 0.0.0.0:2855 0.0.0.0:* LISTEN 26243/freeswitch tcp 0 0 0.0.0.0:2856 0.0.0.0:* LISTEN 26243/freeswitch I could not find any configuration where i can exclude them. Regards Abbasi -------------- next part -------------- An HTML attachment was scrubbed... URL: From shaun.stokes at itec-support.co.uk Fri Dec 14 11:44:18 2018 From: shaun.stokes at itec-support.co.uk (Shaun Stokes) Date: Fri, 14 Dec 2018 11:44:18 +0000 Subject: [Freeswitch-users] FreeSWITCH 1.8.2 - 1-2 second dropped Audio\RTP at the start of a call In-Reply-To: References: <16D901F8-1A94-4FF3-AAC9-BF37E073BEE4@gmail.com> <97EE2A01-D430-421A-90B6-7CFA71DAF34B@gmail.com>, , Message-ID: We have built two test servers side by side on the same hardware with the same configuration, as follows. Server 1: Debian 8 with FreeSWITCH 1.6.20 Server 2: Debian 9 with FreeSWITCH 1.8.2 We can replicate the 1-2 second delay on Server 2 only, whereas Server 1 provides near instant RTP in both directions upon answer. Interestingly, if we move FreeSWITCH 1.8.2 from Server 2 to Server 1 there are still no issues with delay on Server 1, the problem is only observable on the Server 2 running Debian 9 so the problem is not specifically related to FreeSWITCH 1.8.2. At this stage it seems likely the issue lies with Debian 9 or the change in packages on Debian 9. Thanks, Shaun ________________________________ From: FreeSWITCH-users on behalf of Shaun Stokes Sent: 11 December 2018 15:28:33 To: FreeSWITCH Users Help Subject: [Freeswitch-users] FreeSWITCH 1.8.2 - 1-2 second dropped Audio\RTP at the start of a call Hi All, Since we've been using FreeSWITCH 1.8.2 we've noticed that the first 1-2 seconds of Audio\RTP at the start of the call when the call is answered is now dropped\missing but this doesn't occur on 1.6.20. When comparing the examples we've noticed the call flow is slightly different, as follows. FreeSWITCH 1.8.2 Leg A: switch_channel.c:2249 (sofia/internal/SRC_EXT at DOMAIN:PORT) Callstate Change DOWN -> RINGING Leg B: sofia.c:7291 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [calling][0] Leg B: sofia.c:7291 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [proceeding][180] Leg B: sofia.c:7401 Ring-Ready sofia/internal/DST_EXT at LAN_IP:PORT! Leg B: switch_channel.c:3354 (sofia/internal/DST_EXT at LAN_IP:PORT) Callstate Change DOWN -> RINGING Leg A: switch_ivr_originate.c:1246 Sending early media Leg A: switch_channel.c:3482 (sofia/internal/SRC_EXT at DOMAIN:PORT) Callstate Change RINGING -> EARLY Leg A: sofia.c:7291 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering state [early][183] Leg B: sofia.c:7291 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [completing][200] Leg B: switch_channel.c:3482 (sofia/internal/DST_EXT at LAN_IP:PORT) Callstate Change RINGING -> EARLY Leg B: sofia.c:7291 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [ready][200] Leg B: sofia.c:8429 Channel [sofia/internal/DST_EXT at LAN_IP:PORT] has been answered Leg B: switch_channel.c:3781 (sofia/internal/DST_EXT at LAN_IP:PORT) Callstate Change EARLY -> ACTIVE Leg A: switch_ivr_bridge.c:766 Channel [sofia/internal/SRC_EXT at DOMAIN:PORT] has been answered Leg A: sofia.c:7291 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering state [completed][200] Leg A: switch_channel.c:3781 (sofia/internal/SRC_EXT at DOMAIN:PORT) Callstate Change EARLY -> ACTIVE Leg A: sofia.c:7291 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering state [ready][200] Leg A: switch_ivr_async.c:1500 No silence detection configured; assuming start of speech Leg B: sofia.c:7291 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [calling][0] Leg A: sofia.c:7291 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering state [calling][0] Leg A: sofia.c:7291 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering state [ready][200] Leg A: sofia.c:8272 Processing updated SDP Leg B: sofia.c:7291 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [ready][200] FreeSWITCH 1.6.20 Leg A: switch_channel.c:2249 (sofia/internal/SRC_EXT at DOMAIN:PORT) Callstate Change DOWN -> RINGING Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [calling][0] Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [proceeding][180] Leg B: sofia.c:7192 Ring-Ready sofia/internal/DST_EXT at LAN_IP:PORT! Leg B: switch_channel.c:3346 (sofia/internal/DST_EXT at LAN_IP:PORT) Callstate Change DOWN -> RINGING Leg A: switch_ivr_originate.c:1215 Sending early media Leg A: switch_channel.c:3474 (sofia/internal/SRC_EXT at DOMAIN:PORT) Callstate Change RINGING -> EARLY Leg A: sofia.c:7084 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering state [early][183] Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [completing][200] Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [ready][200] Leg A: switch_channel.c:3773 (sofia/internal/SRC_EXT at DOMAIN:PORT) Callstate Change EARLY -> ACTIVE Leg A: sofia.c:7084 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering state [completed][200] Leg A: switch_ivr_originate.c:3705 Originate Resulted in Success: [sofia/internal/DST_EXT at LAN_IP:PORT] Leg B: switch_channel.c:3773 (sofia/internal/DST_EXT at LAN_IP:PORT) Callstate Change RINGING -> ACTIVE Leg A: sofia.c:7084 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering state [ready][200] Leg B: Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [ready][200] Leg A: switch_ivr_async.c:1500 No silence detection configured; assuming start of speech Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [calling][0] Leg A: sofia.c:7084 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering state [calling][0] Leg A: sofia.c:7084 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering state [ready][200] Leg A: sofia.c:8061 Processing updated SDP Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [completing][200] Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [ready][200] On 1.6.20 Leg B changes straight from RINGING to ACTIVE, but on 1.8.2 Leg B first changes from RINGING to EARLY then EARLY to ACTIVE, not sure if this could be related. We've experimented with the following to no avail. rtp-rewrite-timestamps send_silence_when_idle fsctl sync_clock suppress_cng ignore_early_media As per: https://freeswitch.org/confluence/display/FREESWITCH/RTP+Issues https://freeswitch.org/confluence/display/FREESWITCH/VAD+and+CNG https://freeswitch.org/confluence/display/FREESWITCH/send_silence_when_idle https://freeswitch.org/confluence/display/FREESWITCH/Early+Media The calls are local between two extensions\endpoints on the same FreeSWITCH instance and the same SIP profile, the SIP profiles on both servers (1.6.20 and 1.8.2) are identical. Does anyone have any ideas? Thanks, Shaun -------------- next part -------------- An HTML attachment was scrubbed... URL: From shaun.stokes at itec-support.co.uk Fri Dec 14 13:16:14 2018 From: shaun.stokes at itec-support.co.uk (Shaun Stokes) Date: Fri, 14 Dec 2018 13:16:14 +0000 Subject: [Freeswitch-users] FreeSWITCH 1.8.2 - 1-2 second dropped Audio\RTP at the start of a call In-Reply-To: References: <16D901F8-1A94-4FF3-AAC9-BF37E073BEE4@gmail.com> <97EE2A01-D430-421A-90B6-7CFA71DAF34B@gmail.com>, , , Message-ID: Correction, we had moved FreeSWITCH 1.4 (not 1.8) to Server 1 which worked without any audio delays. Upon testing FreeSWITCH 1.8 on Server 1 there is a 1-2 second delay before RTP is established once the call is answered. This is a FreeSWITCH 1.8.2 issue, not a Debian 9 specific (also occurs on Debian 8). FreeSWITCH 1.6 and 1.4 are not effected using the same configuration through-out. ________________________________ From: Shaun Stokes Sent: 14 December 2018 11:44:18 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] FreeSWITCH 1.8.2 - 1-2 second dropped Audio\RTP at the start of a call We have built two test servers side by side on the same hardware with the same configuration, as follows. Server 1: Debian 8 with FreeSWITCH 1.6.20 Server 2: Debian 9 with FreeSWITCH 1.8.2 We can replicate the 1-2 second delay on Server 2 only, whereas Server 1 provides near instant RTP in both directions upon answer. Interestingly, if we move FreeSWITCH 1.8.2 from Server 2 to Server 1 there are still no issues with delay on Server 1, the problem is only observable on the Server 2 running Debian 9 so the problem is not specifically related to FreeSWITCH 1.8.2. At this stage it seems likely the issue lies with Debian 9 or the change in packages on Debian 9. Thanks, Shaun ________________________________ From: FreeSWITCH-users on behalf of Shaun Stokes Sent: 11 December 2018 15:28:33 To: FreeSWITCH Users Help Subject: [Freeswitch-users] FreeSWITCH 1.8.2 - 1-2 second dropped Audio\RTP at the start of a call Hi All, Since we've been using FreeSWITCH 1.8.2 we've noticed that the first 1-2 seconds of Audio\RTP at the start of the call when the call is answered is now dropped\missing but this doesn't occur on 1.6.20. When comparing the examples we've noticed the call flow is slightly different, as follows. FreeSWITCH 1.8.2 Leg A: switch_channel.c:2249 (sofia/internal/SRC_EXT at DOMAIN:PORT) Callstate Change DOWN -> RINGING Leg B: sofia.c:7291 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [calling][0] Leg B: sofia.c:7291 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [proceeding][180] Leg B: sofia.c:7401 Ring-Ready sofia/internal/DST_EXT at LAN_IP:PORT! Leg B: switch_channel.c:3354 (sofia/internal/DST_EXT at LAN_IP:PORT) Callstate Change DOWN -> RINGING Leg A: switch_ivr_originate.c:1246 Sending early media Leg A: switch_channel.c:3482 (sofia/internal/SRC_EXT at DOMAIN:PORT) Callstate Change RINGING -> EARLY Leg A: sofia.c:7291 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering state [early][183] Leg B: sofia.c:7291 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [completing][200] Leg B: switch_channel.c:3482 (sofia/internal/DST_EXT at LAN_IP:PORT) Callstate Change RINGING -> EARLY Leg B: sofia.c:7291 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [ready][200] Leg B: sofia.c:8429 Channel [sofia/internal/DST_EXT at LAN_IP:PORT] has been answered Leg B: switch_channel.c:3781 (sofia/internal/DST_EXT at LAN_IP:PORT) Callstate Change EARLY -> ACTIVE Leg A: switch_ivr_bridge.c:766 Channel [sofia/internal/SRC_EXT at DOMAIN:PORT] has been answered Leg A: sofia.c:7291 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering state [completed][200] Leg A: switch_channel.c:3781 (sofia/internal/SRC_EXT at DOMAIN:PORT) Callstate Change EARLY -> ACTIVE Leg A: sofia.c:7291 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering state [ready][200] Leg A: switch_ivr_async.c:1500 No silence detection configured; assuming start of speech Leg B: sofia.c:7291 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [calling][0] Leg A: sofia.c:7291 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering state [calling][0] Leg A: sofia.c:7291 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering state [ready][200] Leg A: sofia.c:8272 Processing updated SDP Leg B: sofia.c:7291 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [ready][200] FreeSWITCH 1.6.20 Leg A: switch_channel.c:2249 (sofia/internal/SRC_EXT at DOMAIN:PORT) Callstate Change DOWN -> RINGING Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [calling][0] Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [proceeding][180] Leg B: sofia.c:7192 Ring-Ready sofia/internal/DST_EXT at LAN_IP:PORT! Leg B: switch_channel.c:3346 (sofia/internal/DST_EXT at LAN_IP:PORT) Callstate Change DOWN -> RINGING Leg A: switch_ivr_originate.c:1215 Sending early media Leg A: switch_channel.c:3474 (sofia/internal/SRC_EXT at DOMAIN:PORT) Callstate Change RINGING -> EARLY Leg A: sofia.c:7084 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering state [early][183] Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [completing][200] Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [ready][200] Leg A: switch_channel.c:3773 (sofia/internal/SRC_EXT at DOMAIN:PORT) Callstate Change EARLY -> ACTIVE Leg A: sofia.c:7084 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering state [completed][200] Leg A: switch_ivr_originate.c:3705 Originate Resulted in Success: [sofia/internal/DST_EXT at LAN_IP:PORT] Leg B: switch_channel.c:3773 (sofia/internal/DST_EXT at LAN_IP:PORT) Callstate Change RINGING -> ACTIVE Leg A: sofia.c:7084 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering state [ready][200] Leg B: Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [ready][200] Leg A: switch_ivr_async.c:1500 No silence detection configured; assuming start of speech Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [calling][0] Leg A: sofia.c:7084 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering state [calling][0] Leg A: sofia.c:7084 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering state [ready][200] Leg A: sofia.c:8061 Processing updated SDP Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [completing][200] Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [ready][200] On 1.6.20 Leg B changes straight from RINGING to ACTIVE, but on 1.8.2 Leg B first changes from RINGING to EARLY then EARLY to ACTIVE, not sure if this could be related. We've experimented with the following to no avail. rtp-rewrite-timestamps send_silence_when_idle fsctl sync_clock suppress_cng ignore_early_media As per: https://freeswitch.org/confluence/display/FREESWITCH/RTP+Issues https://freeswitch.org/confluence/display/FREESWITCH/VAD+and+CNG https://freeswitch.org/confluence/display/FREESWITCH/send_silence_when_idle https://freeswitch.org/confluence/display/FREESWITCH/Early+Media The calls are local between two extensions\endpoints on the same FreeSWITCH instance and the same SIP profile, the SIP profiles on both servers (1.6.20 and 1.8.2) are identical. Does anyone have any ideas? Thanks, Shaun -------------- next part -------------- An HTML attachment was scrubbed... URL: From sebastian_ml at gmx.net Fri Dec 14 18:52:39 2018 From: sebastian_ml at gmx.net (Sebastian Kemper) Date: Fri, 14 Dec 2018 19:52:39 +0100 Subject: [Freeswitch-users] [Zombie Ports Listening] In-Reply-To: References: Message-ID: <20181214185238.GA25158@darth.lan> On Fri, Dec 14, 2018 at 04:22:46PM +0500, Bilal Abbasi wrote: > Hi Users, > I just installed FS 1.8.2, I enabled the bare minimal configurations needed > for my platform, but i can see see two ports listening. > I searched them and found out that its msrp ports, but i do not have any > configurations/modules for these ports and they are still there, can > somebody guide me how to put them down. > > tcp 0 0 0.0.0.0:2855 0.0.0.0:* LISTEN > 26243/freeswitch > > tcp 0 0 0.0.0.0:2856 0.0.0.0:* LISTEN > 26243/freeswitch > > > > I could not find any configuration where i can exclude them. Hello Abbasi, You can add /etc/freeswitch/autoload_configs/msrp.conf.xml. This doesn't disable msrp. But it only listens on localhost afterward. I haven't found a way to disable msrp. This is not perfect IMHO. Maybe somebody else knows how to completely turn it off. Kind regards, Seb From imfanee at gmail.com Sat Dec 15 06:18:00 2018 From: imfanee at gmail.com (Faisal Hanif) Date: Sat, 15 Dec 2018 11:18:00 +0500 Subject: [Freeswitch-users] 3CX-Sip-Trunk / Freeswitch cannot reach 3CX In-Reply-To: <34d3fea8-1a28-b495-386a-a787469e99e3@level5.de> References: <34d3fea8-1a28-b495-386a-a787469e99e3@level5.de> Message-ID: Share SIP traces of call On Wed, Dec 12, 2018, 9:55 PM Thorsten Göllner Hi, > > has anybody experience with 3CX-SIP-Trunks? I have installed Freeswitch > and added an extension (trunk-1). On 3CX-Side I added a SIP-Trunk > (generic sip provider) and I can see a succesfull registration. Now I > make a call to Freeswitch and want to pass it to 3CX-Sip-Trunk. But when > bridging to user/trunk-1 I get a "Originate Resulted in Error Cause: 41 > [NORMAL_TEMPORARY_FAILURE]" error (with no delay). > > 3CX is installed as a VM in a Azure-Cloud behind NAT. I disabled local > iptables and the Firewall-Test on 3CX is all fine. > > Freeswitch is on bare metal with a public ip. > > When making a call, wireshark on 3CX-machine shows the INVITE followed > by "Destination unreachable (Port unreachable)". But I am not sure, if > this is a firewall-issue or something else. > > Any idea? > > Thanks in advance, > Thorsten > > > _________________________________________________________________________ > 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 imfanee at gmail.com Sat Dec 15 06:19:28 2018 From: imfanee at gmail.com (Faisal Hanif) Date: Sat, 15 Dec 2018 11:19:28 +0500 Subject: [Freeswitch-users] calls to local 311 line with two IVRs In-Reply-To: <26233380.38805.1544587345787.JavaMail.zimbra@calgaryit.com> References: <26233380.38805.1544587345787.JavaMail.zimbra@calgaryit.com> Message-ID: You need to share config and SIP traces. On Wed, Dec 12, 2018, 7:39 PM George when I call a local 311 number, the call goes through, I hear the first > announcement, after with time the call is supposed to get transferred to a > second IVR which should give the choices of departments, the call just goes > silent and eventually gets hung on on. > > FreeSWITCH Version 1.8.2-3-a98a958ac3~64bit (-3-a98a958ac3 64bit) > Debian 9.6 > > this is what my FS sends back to the provider: > > v=0 > o=FreeSWITCH 1544570427 1544570429 IN IP4 10.185.45.238 > s=FreeSWITCH > c=IN IP4 10.185.45.238 > t=0 0 > m=audio 16506 RTP/AVP 0 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=sendonly > a=ptime:20 > > the provider says its because the a=sendonly I am not receiving the > announcements > > any help on this? > > Thank You, > George > > _________________________________________________________________________ > 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 imfanee at gmail.com Sat Dec 15 06:23:20 2018 From: imfanee at gmail.com (Faisal Hanif) Date: Sat, 15 Dec 2018 11:23:20 +0500 Subject: [Freeswitch-users] FreeSWITCH 1.8.2 - 1-2 second dropped Audio\RTP at the start of a call In-Reply-To: References: <16D901F8-1A94-4FF3-AAC9-BF37E073BEE4@gmail.com> <97EE2A01-D430-421A-90B6-7CFA71DAF34B@gmail.com> Message-ID: Seems like your issue could be related IP Config, ICE & NAT which can cause delay in media port identification on different servers on different version of FS. On Fri, Dec 14, 2018, 6:59 PM Shaun Stokes Correction, we had moved FreeSWITCH 1.4 (not 1.8) to Server 1 which worked > without any audio delays. Upon testing FreeSWITCH 1.8 on Server 1 there is > a 1-2 second delay before RTP is established once the call is answered. > > This is a FreeSWITCH 1.8.2 issue, not a Debian 9 specific (also occurs on > Debian 8). FreeSWITCH 1.6 and 1.4 are not effected using the same > configuration through-out. > ------------------------------ > *From:* Shaun Stokes > *Sent:* 14 December 2018 11:44:18 > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] FreeSWITCH 1.8.2 - 1-2 second dropped > Audio\RTP at the start of a call > > We have built two test servers side by side on the same hardware with the > same configuration, as follows. > Server 1: Debian 8 with FreeSWITCH 1.6.20 > Server 2: Debian 9 with FreeSWITCH 1.8.2 > > We can replicate the 1-2 second delay on Server 2 only, whereas Server 1 > provides near instant RTP in both directions upon answer. Interestingly, if > we move FreeSWITCH 1.8.2 from Server 2 to Server 1 there are still no > issues with delay on Server 1, the problem is only observable on the Server > 2 running Debian 9 so the problem is not specifically related to FreeSWITCH > 1.8.2. > > At this stage it seems likely the issue lies with Debian 9 or the change > in packages on Debian 9. > > Thanks, > Shaun > ------------------------------ > *From:* FreeSWITCH-users > on behalf of Shaun Stokes > *Sent:* 11 December 2018 15:28:33 > *To:* FreeSWITCH Users Help > *Subject:* [Freeswitch-users] FreeSWITCH 1.8.2 - 1-2 second dropped > Audio\RTP at the start of a call > > > Hi All, > > > Since we've been using FreeSWITCH 1.8.2 we've noticed that the first 1-2 > seconds of Audio\RTP at the start of the call when the call is answered is > now dropped\missing but this doesn't occur on 1.6.20. When comparing the > examples we've noticed the call flow is slightly different, as follows. > > > FreeSWITCH 1.8.2 > > Leg A: switch_channel.c:2249 (sofia/internal/SRC_EXT at DOMAIN:PORT) > Callstate Change DOWN -> RINGING > Leg B: sofia.c:7291 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering > state [calling][0] > Leg B: sofia.c:7291 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering > state [proceeding][180] > Leg B: sofia.c:7401 Ring-Ready sofia/internal/DST_EXT at LAN_IP:PORT! > Leg B: switch_channel.c:3354 (sofia/internal/DST_EXT at LAN_IP:PORT) > Callstate Change DOWN -> RINGING > Leg A: switch_ivr_originate.c:1246 Sending early media > Leg A: switch_channel.c:3482 (sofia/internal/SRC_EXT at DOMAIN:PORT) > Callstate Change RINGING -> EARLY > Leg A: sofia.c:7291 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering > state [early][183] > Leg B: sofia.c:7291 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering > state [completing][200] > Leg B: switch_channel.c:3482 (sofia/internal/DST_EXT at LAN_IP:PORT) > Callstate Change RINGING -> EARLY > Leg B: sofia.c:7291 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering > state [ready][200] > Leg B: sofia.c:8429 Channel [sofia/internal/DST_EXT at LAN_IP:PORT] has been > answered > Leg B: switch_channel.c:3781 (sofia/internal/DST_EXT at LAN_IP:PORT) > Callstate Change EARLY -> ACTIVE > Leg A: switch_ivr_bridge.c:766 Channel [sofia/internal/SRC_EXT at DOMAIN:PORT] > has been answered > Leg A: sofia.c:7291 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering > state [completed][200] > Leg A: switch_channel.c:3781 (sofia/internal/SRC_EXT at DOMAIN:PORT) > Callstate Change EARLY -> ACTIVE > Leg A: sofia.c:7291 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering > state [ready][200] > Leg A: switch_ivr_async.c:1500 No silence detection configured; assuming > start of speech > Leg B: sofia.c:7291 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering > state [calling][0] > Leg A: sofia.c:7291 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering > state [calling][0] > Leg A: sofia.c:7291 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering > state [ready][200] > Leg A: sofia.c:8272 Processing updated SDP > Leg B: sofia.c:7291 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering > state [ready][200] > > > FreeSWITCH 1.6.20 > > Leg A: switch_channel.c:2249 (sofia/internal/SRC_EXT at DOMAIN:PORT) > Callstate Change DOWN -> RINGING > Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering > state [calling][0] > Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering > state [proceeding][180] > Leg B: sofia.c:7192 Ring-Ready sofia/internal/DST_EXT at LAN_IP:PORT! > Leg B: switch_channel.c:3346 (sofia/internal/DST_EXT at LAN_IP:PORT) > Callstate Change DOWN -> RINGING > Leg A: switch_ivr_originate.c:1215 Sending early media > Leg A: switch_channel.c:3474 (sofia/internal/SRC_EXT at DOMAIN:PORT) > Callstate Change RINGING -> EARLY > Leg A: sofia.c:7084 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering > state [early][183] > Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering > state [completing][200] > Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering > state [ready][200] > Leg A: switch_channel.c:3773 (sofia/internal/SRC_EXT at DOMAIN:PORT) > Callstate Change EARLY -> ACTIVE > Leg A: sofia.c:7084 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering > state [completed][200] > Leg A: switch_ivr_originate.c:3705 Originate Resulted in Success: > [sofia/internal/DST_EXT at LAN_IP:PORT] > Leg B: switch_channel.c:3773 (sofia/internal/DST_EXT at LAN_IP:PORT) > Callstate Change RINGING -> ACTIVE > Leg A: sofia.c:7084 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering > state [ready][200] > Leg B: Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state > [ready][200] > Leg A: switch_ivr_async.c:1500 No silence detection configured; assuming > start of speech > Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering > state [calling][0] > Leg A: sofia.c:7084 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering > state [calling][0] > Leg A: sofia.c:7084 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering > state [ready][200] > Leg A: sofia.c:8061 Processing updated SDP > Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering > state [completing][200] > Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering > state [ready][200] > > > On 1.6.20 Leg B changes straight from RINGING to ACTIVE, but on 1.8.2 Leg > B first changes from RINGING to EARLY then EARLY to ACTIVE, not sure if > this could be related. > > > We've experimented with the following to no avail. > rtp-rewrite-timestamps > send_silence_when_idle > fsctl sync_clock > suppress_cng > ignore_early_media > > As per: > https://freeswitch.org/confluence/display/FREESWITCH/RTP+Issues > https://freeswitch.org/confluence/display/FREESWITCH/VAD+and+CNG > https://freeswitch.org/confluence/display/FREESWITCH/send_silence_when_idle > https://freeswitch.org/confluence/display/FREESWITCH/Early+Media > > > The calls are local between two extensions\endpoints on the same > FreeSWITCH instance and the same SIP profile, the SIP profiles on both > servers (1.6.20 and 1.8.2) are identical. > > > Does anyone have any ideas? > > > Thanks, > > Shaun > _________________________________________________________________________ > 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 ch.chhatra at gmail.com Sun Dec 16 04:52:36 2018 From: ch.chhatra at gmail.com (Chhorm Chhatra) Date: Sun, 16 Dec 2018 11:52:36 +0700 Subject: [Freeswitch-users] MOD_CALLCENTER cannot originate call to agent Message-ID: Hi, Currently, I am experiencing a very weird behavior of mod_callcenter like the following: 1. after 4 to 5 mins of hearing the moh-sound, it started to randomly play media from the sound/music folder 2. MOD_CALLCENTER *can never bridge any call to the agent. It mostly says the user is not registered (although it's registered). Sometimes, it says no-answer but the agent never rings.* *Please kindly help me.* -------------- next part -------------- An HTML attachment was scrubbed... URL: From ch.chhatra at gmail.com Sun Dec 16 15:48:14 2018 From: ch.chhatra at gmail.com (Chhorm Chhatra) Date: Sun, 16 Dec 2018 22:48:14 +0700 Subject: [Freeswitch-users] MOD_CALLCENTER cannot originate call to agent In-Reply-To: References: Message-ID: My case is that Freeswitch cannot originate the call to the agent (cause: RECOVERY_ON_TIMER_EXPIRE). Any help would be very appreciated. On Sun, 16 Dec 2018 at 11:52, Chhorm Chhatra wrote: > Hi, > Currently, I am experiencing a very weird behavior of mod_callcenter like > the following: > 1. after 4 to 5 mins of hearing the moh-sound, it started to randomly play > media from the sound/music folder > 2. MOD_CALLCENTER *can never bridge any call to the agent. It mostly says > the user is not registered (although it's registered). Sometimes, it says > no-answer but the agent never rings.* > > *Please kindly help me.* > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From mitch.capper at gmail.com Sun Dec 16 17:44:44 2018 From: mitch.capper at gmail.com (Mitch Capper) Date: Sun, 16 Dec 2018 09:44:44 -0800 Subject: [Freeswitch-users] FS 1.8 Docker image In-Reply-To: References: Message-ID: I run FS in docker. It is also handy for filing bug reports with exact repo so I have some dockerfiles up for example: https://freeswitch.org/jira/browse/FS-7833 full source build. ~mitch On Fri, Dec 14, 2018 at 1:23 AM Alexey Sibyakin wrote: > Hi, > > Try to install dbus into your Stretch. > > Regards, > > Alex > > On Thu, Dec 13, 2018 at 3:40 AM Sergey Safarov > wrote: > >> you can use safarov/freeswitch:1.8.2 >> image >> repo of this is located here >> https://bitbucket.org/sergey-safarov/freeswitch/commits/branch/v1.8 >> >> >> ср, 12 дек. 2018 г. в 18:14, Filipe Cunha : >> >>> Hi all, >>> >>> I'm trying to create a docker image with the latest version of >>> FreeSWITCH. But I'm not able to do it. These are the steps that I did to >>> try to do it. >>> >>> 1. Download debian stretch image >>> *docker pull debian:stretch * >>> 2. Build FreeSWITCH from source code following the guide >>> >>> in confluence step by step. I follow the production version >>> 3. The step 2 went very well, and then I follow the post-installation >>> instructions >>> >>> and this was when it stop working. I set up owners and permissions and when >>> I try to do the systemd I was not able to do it. I copied the file *cp >>> /usr/src/freeswitch/debian/freeswitch-systemd.freeswitch.service >>> /etc/systemd/system/freeswitch.service *the content of the file was not >>> exactly the same as the one in the tutorial, so I copied the content of the >>> file on the tutorial and replace it. And when I tried to do *systemctl >>> deamon-reload* I got this error message, *Failed to connect to bus: No >>> such file or directory*. After trying to find a solution online I found >>> this answer on SO >>> >>> >>> Do you guys know if there is a solution for this problem? >>> We plan to use the latest version of FreeSwitch in our production >>> environment, but we have a normal linux server to do it, and the devops >>> team got everything working fine, we only have this problem when we try to >>> use docker. >>> I'm trying to use docker so my team could test everything locally before >>> pushing something to the test environment, so far we have been deploying >>> our code to test environment to test our code 😖 >>> I found this image >>> that works fine in our local environment, but the version is 1.6.16. >>> Is there any braking change between 1.8.2 and 1.6.16? My concern it that >>> we start developing against an older version, and when we deploy to >>> production it doesn't work, or maybe the latest version have a super useful >>> command that we are not taking advantage of. >>> We control FreeSWITCH mainly by events, we use java to do it. >>> >>> Thanks for the help, >>> Filipe Cunha >>> >>> >>> >>> >>> _________________________________________________________________________ >>> 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. > > _________________________________________________________________________ > 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 sagarmalam at gmail.com Mon Dec 17 14:13:46 2018 From: sagarmalam at gmail.com (sagar malam) Date: Mon, 17 Dec 2018 19:43:46 +0530 Subject: [Freeswitch-users] Disabling 200 OK with contact headers of all the lines In-Reply-To: References: Message-ID: Thanks for reply Micheal. I dont want to disable multiple registrations.And i agree all the registrations are genuine. But my problem is that when FS responds to Register Request using 200 OK, The 200 OK has contact headers of all the registrations( of other SIP clients).As shown in below packet, there are three contact headers.I think there should be only one. ============================200 OK for register packet ============= 2018/12/08 13:36:35.426099 10.50.7.251:5070 -> 10.50.7.253:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 66.160.237.253:5060 ;branch=z9hG4bK9b46.f9b7a68125108f411537617d02bd48f8.0;received=10.50.7.253 Via: SIP/2.0/UDP 198.136.236.1:5060 ;rport=5060;received=10.50.8.1;branch=z9hG4bK9b46.e6f1aa3817a238f499aa3aafa4217425.0 Via: SIP/2.0/UDP 172.16.1.11;rport=1426;received=71.239.113.14;branch=z9hG4bK1df2302cD062D28F From: "Main Line" ;tag=22CE54C0-84BCCF23 To: ;tag=e8Fa2tND4U80D Call-ID: d66f0f0592c09746b903406f312eb0c2 CSeq: 590 REGISTER *Contact: ;expires=379* *Contact: ;expires=245* *Contact: ;expires=88* Date: Sat, 08 Dec 2018 08:06:35 GMT User-Agent: FreeSWITCH-mod_sofia/1.6.17~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: path, replaces Content-Length: 0 =========================================================== Also i have compared this behaviour with OpenSIPs and Kamailio, They send only one contact header. Thanks in advance. On Wed, Dec 12, 2018 at 3:08 AM Michael Jerris wrote: > These are all valid current registrations, there are params you can adjust > for multi reg that will replace previous registrations instead of allowing > multiple. > > On Dec 9, 2018, at 1:17 PM, sagar malam wrote: > > Hello , > > I am using FS SLA feature and it works very well.However i am facing an > issue of registrations getting dropped as explained below : > There are 3 phones registered with same extension number.Sofia is > configured with "sip-expires-max-deviation" to randomise registration > through expiry header in contact header.All the phones are Polycom. In case > of shared lines FS adds contact header of all the lines(or phones with same > extension) in 200 OK as shown below due to which all the phones are reading > expiry timer from first contact header only.So in below example,Phone re > registers after 379 seconds(first contact header) instead of 88 > seconds(third contact header) leading to registration expiry on FS. > > Reason why phones are always reading expiry from first contact header is > same Public IP(contact is re written by Proxy in front of FS) for all > three phones which is confusing phone to identify its own contact header. > Is there any way to configure FS to not send contact headers of all the > registrations but only one that belongs to the line itself ? or any other > way to fix it. > ============================200 OK for register packet ============= > > 2018/12/08 13:36:35.426099 10.50.7.251:5070 -> 10.50.7.253:5060 > SIP/2.0 200 OK > Via: SIP/2.0/UDP 66.160.237.253:5060 > ;branch=z9hG4bK9b46.f9b7a68125108f411537617d02bd48f8.0;received=10.50.7.253 > Via: SIP/2.0/UDP 198.136.236.1:5060 > ;rport=5060;received=10.50.8.1;branch=z9hG4bK9b46.e6f1aa3817a238f499aa3aafa4217425.0 > Via: SIP/2.0/UDP > 172.16.1.11;rport=1426;received=71.239.113.14;branch=z9hG4bK1df2302cD062D28F > From: "Main Line" ;tag=22CE54C0-84BCCF23 > To: ;tag=e8Fa2tND4U80D > Call-ID: d66f0f0592c09746b903406f312eb0c2 > CSeq: 590 REGISTER > Contact: < > sip:398 at 71.239.113.14:1071;alias=10.50.8.1~5060~1;x-nat=yes;pv-ip=172.16.1.12;pb-ip=71.239.113.14;pb-pt=1071;mac-address=64167f2ec274;transp > t=tcp>;expires=379 > Contact: < > sip:398 at 71.239.113.14:56478;alias=10.50.8.1~5060~1;x-nat=yes;pv-ip=172.16.1.13;pb-ip=71.239.113.14;pb-pt=56478;mac-address=64167f2ebd54;tran > ort=tcp>;expires=245 > Contact: < > sip:398 at 71.239.113.14:1426;alias=10.50.8.1~5060~1;x-nat=yes;pv-ip=172.16.1.11;pb-ip=71.239.113.14;pb-pt=1426;transport=udp;mac-address=64167 > eb0c2>;expires=88 > Date: Sat, 08 Dec 2018 08:06:35 GMT > User-Agent: FreeSWITCH-mod_sofia/1.6.17~64bit > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, > REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: path, replaces > Content-Length: 0 > > > _________________________________________________________________________ > 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 -- Thanks, Sagar -------------- next part -------------- An HTML attachment was scrubbed... URL: From bilaln018 at gmail.com Mon Dec 17 14:32:05 2018 From: bilaln018 at gmail.com (Bilal Abbasi) Date: Mon, 17 Dec 2018 19:32:05 +0500 Subject: [Freeswitch-users] [Zombie Ports Listening] In-Reply-To: <20181214185238.GA25158@darth.lan> References: <20181214185238.GA25158@darth.lan> Message-ID: Thanks for your reply, but i am really looking for disable that, FreeSWITCH developers must have a way to do that, as its very basic thing i am looking for. On Fri, Dec 14, 2018 at 11:52 PM Sebastian Kemper wrote: > On Fri, Dec 14, 2018 at 04:22:46PM +0500, Bilal Abbasi wrote: > > Hi Users, > > I just installed FS 1.8.2, I enabled the bare minimal configurations > needed > > for my platform, but i can see see two ports listening. > > I searched them and found out that its msrp ports, but i do not have any > > configurations/modules for these ports and they are still there, can > > somebody guide me how to put them down. > > > > tcp 0 0 0.0.0.0:2855 0.0.0.0:* > LISTEN > > 26243/freeswitch > > > > tcp 0 0 0.0.0.0:2856 0.0.0.0:* > LISTEN > > 26243/freeswitch > > > > > > > > I could not find any configuration where i can exclude them. > > Hello Abbasi, > > You can add /etc/freeswitch/autoload_configs/msrp.conf.xml. > > > > > > > > > > > > > > This doesn't disable msrp. But it only listens on localhost afterward. I > haven't found a way to disable msrp. > > This is not perfect IMHO. Maybe somebody else knows how to completely > turn it off. > > Kind regards, > Seb > -------------- next part -------------- An HTML attachment was scrubbed... URL: From joel at textplus.com Mon Dec 17 15:08:11 2018 From: joel at textplus.com (Joel Serrano) Date: Mon, 17 Dec 2018 07:08:11 -0800 Subject: [Freeswitch-users] [Zombie Ports Listening] In-Reply-To: References: <20181214185238.GA25158@darth.lan> Message-ID: Have you tried commenting out the module so it’s not loaded? On Mon, Dec 17, 2018 at 06:55 Bilal Abbasi wrote: > Thanks for your reply, but i am really looking for disable that, > FreeSWITCH developers must have a way to do that, as its very basic thing i > am looking for. > > On Fri, Dec 14, 2018 at 11:52 PM Sebastian Kemper > wrote: > >> On Fri, Dec 14, 2018 at 04:22:46PM +0500, Bilal Abbasi wrote: >> > Hi Users, >> > I just installed FS 1.8.2, I enabled the bare minimal configurations >> needed >> > for my platform, but i can see see two ports listening. >> > I searched them and found out that its msrp ports, but i do not have any >> > configurations/modules for these ports and they are still there, can >> > somebody guide me how to put them down. >> > >> > tcp 0 0 0.0.0.0:2855 0.0.0.0:* >> LISTEN >> > 26243/freeswitch >> > >> > tcp 0 0 0.0.0.0:2856 0.0.0.0:* >> LISTEN >> > 26243/freeswitch >> > >> > >> > >> > I could not find any configuration where i can exclude them. >> >> Hello Abbasi, >> >> You can add /etc/freeswitch/autoload_configs/msrp.conf.xml. >> >> >> >> >> >> >> >> >> >> >> >> >> >> This doesn't disable msrp. But it only listens on localhost afterward. I >> haven't found a way to disable msrp. >> >> This is not perfect IMHO. Maybe somebody else knows how to completely >> turn it off. >> >> Kind regards, >> Seb >> > _________________________________________________________________________ > 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 shaun.stokes at itec-support.co.uk Mon Dec 17 15:18:14 2018 From: shaun.stokes at itec-support.co.uk (Shaun Stokes) Date: Mon, 17 Dec 2018 15:18:14 +0000 Subject: [Freeswitch-users] FreeSWITCH 1.8.2 - 1-2 second dropped Audio\RTP at the start of a call In-Reply-To: References: <16D901F8-1A94-4FF3-AAC9-BF37E073BEE4@gmail.com> <97EE2A01-D430-421A-90B6-7CFA71DAF34B@gmail.com>, , , , Message-ID: We reverted back to FreeSWITCH 1.6.20 but when this is compiled on the Debian 9 server the problem still occurs. We had to workaround some build errors for FS 1.6.20 to compile on Debian 9 with PostgreSQL 11 but the problem was still present, as follows. -------------------- # Uninstall 1.1.0 SSL header files (libssl-dev) and install the older ones (libssl1.0-dev). apt-get install libssl1.0-dev # Fix PGSQL 11 Support In the file: /usr/src/freeswitch/srcswitch_pgsql.c On line 389, replace this: #if POSTGRESQL_MAJOR_VERSION >= 9 && POSTGRESQL_MINOR_VERSION >= 2 With: #if (POSTGRESQL_MAJOR_VERSION == 9 && POSTGRESQL_MINOR_VERSION >= 2) || POSTGRESQL_MAJOR_VERSION > 9 # Do not build mod_flite or mod_enum sed -i /usr/src/freeswitch/modules.conf -e s:'asr_tts/mod_flite:#asr_tts/mod_flite:' sed -i /usr/src/freeswitch/modules.conf -e s:'applications/mod_enum:#applications/mod_enum:' -------------------- We took the FS 1.6.20 binaries (pre-compiled) from a Debian 8 server and restored them to our Debian 9 server which resolved the issue but we had to copy some missing libs from a Debian 8 server: /usr/lib/x86_64-linux-gnu/libssl.so.1.0.0 /usr/lib/x86_64-linux-gnu/libcrypto.so.1.0.0 Given that the problem changes when the source-code is compiled on different servers we suspect this may be a package problem not specific to FreeSWITCH. This is also a problem on master, raised JIRA: https://freeswitch.org/jira/browse/FS-11572 ________________________________ From: FreeSWITCH-users on behalf of Shaun Stokes Sent: 14 December 2018 13:16:14 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] FreeSWITCH 1.8.2 - 1-2 second dropped Audio\RTP at the start of a call Correction, we had moved FreeSWITCH 1.4 (not 1.8) to Server 1 which worked without any audio delays. Upon testing FreeSWITCH 1.8 on Server 1 there is a 1-2 second delay before RTP is established once the call is answered. This is a FreeSWITCH 1.8.2 issue, not a Debian 9 specific (also occurs on Debian 8). FreeSWITCH 1.6 and 1.4 are not effected using the same configuration through-out. ________________________________ From: Shaun Stokes Sent: 14 December 2018 11:44:18 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] FreeSWITCH 1.8.2 - 1-2 second dropped Audio\RTP at the start of a call We have built two test servers side by side on the same hardware with the same configuration, as follows. Server 1: Debian 8 with FreeSWITCH 1.6.20 Server 2: Debian 9 with FreeSWITCH 1.8.2 We can replicate the 1-2 second delay on Server 2 only, whereas Server 1 provides near instant RTP in both directions upon answer. Interestingly, if we move FreeSWITCH 1.8.2 from Server 2 to Server 1 there are still no issues with delay on Server 1, the problem is only observable on the Server 2 running Debian 9 so the problem is not specifically related to FreeSWITCH 1.8.2. At this stage it seems likely the issue lies with Debian 9 or the change in packages on Debian 9. Thanks, Shaun ________________________________ From: FreeSWITCH-users on behalf of Shaun Stokes Sent: 11 December 2018 15:28:33 To: FreeSWITCH Users Help Subject: [Freeswitch-users] FreeSWITCH 1.8.2 - 1-2 second dropped Audio\RTP at the start of a call Hi All, Since we've been using FreeSWITCH 1.8.2 we've noticed that the first 1-2 seconds of Audio\RTP at the start of the call when the call is answered is now dropped\missing but this doesn't occur on 1.6.20. When comparing the examples we've noticed the call flow is slightly different, as follows. FreeSWITCH 1.8.2 Leg A: switch_channel.c:2249 (sofia/internal/SRC_EXT at DOMAIN:PORT) Callstate Change DOWN -> RINGING Leg B: sofia.c:7291 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [calling][0] Leg B: sofia.c:7291 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [proceeding][180] Leg B: sofia.c:7401 Ring-Ready sofia/internal/DST_EXT at LAN_IP:PORT! Leg B: switch_channel.c:3354 (sofia/internal/DST_EXT at LAN_IP:PORT) Callstate Change DOWN -> RINGING Leg A: switch_ivr_originate.c:1246 Sending early media Leg A: switch_channel.c:3482 (sofia/internal/SRC_EXT at DOMAIN:PORT) Callstate Change RINGING -> EARLY Leg A: sofia.c:7291 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering state [early][183] Leg B: sofia.c:7291 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [completing][200] Leg B: switch_channel.c:3482 (sofia/internal/DST_EXT at LAN_IP:PORT) Callstate Change RINGING -> EARLY Leg B: sofia.c:7291 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [ready][200] Leg B: sofia.c:8429 Channel [sofia/internal/DST_EXT at LAN_IP:PORT] has been answered Leg B: switch_channel.c:3781 (sofia/internal/DST_EXT at LAN_IP:PORT) Callstate Change EARLY -> ACTIVE Leg A: switch_ivr_bridge.c:766 Channel [sofia/internal/SRC_EXT at DOMAIN:PORT] has been answered Leg A: sofia.c:7291 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering state [completed][200] Leg A: switch_channel.c:3781 (sofia/internal/SRC_EXT at DOMAIN:PORT) Callstate Change EARLY -> ACTIVE Leg A: sofia.c:7291 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering state [ready][200] Leg A: switch_ivr_async.c:1500 No silence detection configured; assuming start of speech Leg B: sofia.c:7291 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [calling][0] Leg A: sofia.c:7291 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering state [calling][0] Leg A: sofia.c:7291 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering state [ready][200] Leg A: sofia.c:8272 Processing updated SDP Leg B: sofia.c:7291 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [ready][200] FreeSWITCH 1.6.20 Leg A: switch_channel.c:2249 (sofia/internal/SRC_EXT at DOMAIN:PORT) Callstate Change DOWN -> RINGING Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [calling][0] Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [proceeding][180] Leg B: sofia.c:7192 Ring-Ready sofia/internal/DST_EXT at LAN_IP:PORT! Leg B: switch_channel.c:3346 (sofia/internal/DST_EXT at LAN_IP:PORT) Callstate Change DOWN -> RINGING Leg A: switch_ivr_originate.c:1215 Sending early media Leg A: switch_channel.c:3474 (sofia/internal/SRC_EXT at DOMAIN:PORT) Callstate Change RINGING -> EARLY Leg A: sofia.c:7084 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering state [early][183] Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [completing][200] Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [ready][200] Leg A: switch_channel.c:3773 (sofia/internal/SRC_EXT at DOMAIN:PORT) Callstate Change EARLY -> ACTIVE Leg A: sofia.c:7084 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering state [completed][200] Leg A: switch_ivr_originate.c:3705 Originate Resulted in Success: [sofia/internal/DST_EXT at LAN_IP:PORT] Leg B: switch_channel.c:3773 (sofia/internal/DST_EXT at LAN_IP:PORT) Callstate Change RINGING -> ACTIVE Leg A: sofia.c:7084 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering state [ready][200] Leg B: Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [ready][200] Leg A: switch_ivr_async.c:1500 No silence detection configured; assuming start of speech Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [calling][0] Leg A: sofia.c:7084 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering state [calling][0] Leg A: sofia.c:7084 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering state [ready][200] Leg A: sofia.c:8061 Processing updated SDP Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [completing][200] Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [ready][200] On 1.6.20 Leg B changes straight from RINGING to ACTIVE, but on 1.8.2 Leg B first changes from RINGING to EARLY then EARLY to ACTIVE, not sure if this could be related. We've experimented with the following to no avail. rtp-rewrite-timestamps send_silence_when_idle fsctl sync_clock suppress_cng ignore_early_media As per: https://freeswitch.org/confluence/display/FREESWITCH/RTP+Issues https://freeswitch.org/confluence/display/FREESWITCH/VAD+and+CNG https://freeswitch.org/confluence/display/FREESWITCH/send_silence_when_idle https://freeswitch.org/confluence/display/FREESWITCH/Early+Media The calls are local between two extensions\endpoints on the same FreeSWITCH instance and the same SIP profile, the SIP profiles on both servers (1.6.20 and 1.8.2) are identical. Does anyone have any ideas? Thanks, Shaun -------------- next part -------------- An HTML attachment was scrubbed... URL: From me at nevian.org Mon Dec 17 16:07:32 2018 From: me at nevian.org (Serge S. Yuriev) Date: Mon, 17 Dec 2018 19:07:32 +0300 Subject: [Freeswitch-users] calls to local 311 line with two IVRs In-Reply-To: <501702402.43110.1544639656116.JavaMail.zimbra@calgaryit.com> References: <26233380.38805.1544587345787.JavaMail.zimbra@calgaryit.com> <501702402.43110.1544639656116.JavaMail.zimbra@calgaryit.com> Message-ID: <788aa736-39d2-3a90-f68d-24a29e60c5bf@nevian.org> Hi, You have to set this towards provider (buggy Nortel/Cisco/Avaya) On 12/12/2018 21:34, George wrote: > here it is: > > https://freeswitch.org/jira/secure/attachment/28046/cap20181212.pcap > > Thank You, > George > > ----- Original Message ----- > From: "Michael Jerris" > To: "freeswitch-users" > Sent: Wednesday, December 12, 2018 9:57:09 AM > Subject: Re: [Freeswitch-users] calls to local 311 line with two IVRs > > Would need to see the full sip trace to answer this. A debug log with sip trace is most useful > >> On Dec 11, 2018, at 11:02 PM, George wrote: >> >> when I call a local 311 number, the call goes through, I hear the first announcement, after with time the call is supposed to get transferred to a second IVR which should give the choices of departments, the call just goes silent and eventually gets hung on on. >> >> FreeSWITCH Version 1.8.2-3-a98a958ac3~64bit (-3-a98a958ac3 64bit) >> Debian 9.6 >> >> this is what my FS sends back to the provider: >> >> v=0 >> o=FreeSWITCH 1544570427 1544570429 IN IP4 10.185.45.238 >> s=FreeSWITCH >> c=IN IP4 10.185.45.238 >> t=0 0 >> m=audio 16506 RTP/AVP 0 101 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=sendonly >> a=ptime:20 >> >> the provider says its because the a=sendonly I am not receiving the announcements >> >> any help on this? >> >> Thank You, >> George > > > _________________________________________________________________________ > 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 > -- Serge S. Yuriev Senior VoIP engineer From gmaruzz at gmail.com Mon Dec 17 17:17:17 2018 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Mon, 17 Dec 2018 18:17:17 +0100 Subject: [Freeswitch-users] FreeSWITCH 1.8.2 - 1-2 second dropped Audio\RTP at the start of a call In-Reply-To: References: <16D901F8-1A94-4FF3-AAC9-BF37E073BEE4@gmail.com> <97EE2A01-D430-421A-90B6-7CFA71DAF34B@gmail.com> Message-ID: Seems you have a problem of rtp flow not yet established... Maybe you want to experiment establishing rtp before answering, or before bridging... On Mon, Dec 17, 2018 at 5:56 PM Faisal Hanif wrote: > Seems like your issue could be related IP Config, ICE & NAT which can > cause delay in media port identification on different servers on different > version of FS. > > On Fri, Dec 14, 2018, 6:59 PM Shaun Stokes < > shaun.stokes at itec-support.co.uk wrote: > >> Correction, we had moved FreeSWITCH 1.4 (not 1.8) to Server 1 which >> worked without any audio delays. Upon testing FreeSWITCH 1.8 on Server 1 >> there is a 1-2 second delay before RTP is established once the call is >> answered. >> >> This is a FreeSWITCH 1.8.2 issue, not a Debian 9 specific (also occurs on >> Debian 8). FreeSWITCH 1.6 and 1.4 are not effected using the same >> configuration through-out. >> ------------------------------ >> *From:* Shaun Stokes >> *Sent:* 14 December 2018 11:44:18 >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] FreeSWITCH 1.8.2 - 1-2 second dropped >> Audio\RTP at the start of a call >> >> We have built two test servers side by side on the same hardware with the >> same configuration, as follows. >> Server 1: Debian 8 with FreeSWITCH 1.6.20 >> Server 2: Debian 9 with FreeSWITCH 1.8.2 >> >> We can replicate the 1-2 second delay on Server 2 only, whereas Server 1 >> provides near instant RTP in both directions upon answer. Interestingly, if >> we move FreeSWITCH 1.8.2 from Server 2 to Server 1 there are still no >> issues with delay on Server 1, the problem is only observable on the Server >> 2 running Debian 9 so the problem is not specifically related to FreeSWITCH >> 1.8.2. >> >> At this stage it seems likely the issue lies with Debian 9 or the change >> in packages on Debian 9. >> >> Thanks, >> Shaun >> ------------------------------ >> *From:* FreeSWITCH-users >> on behalf of Shaun Stokes >> *Sent:* 11 December 2018 15:28:33 >> *To:* FreeSWITCH Users Help >> *Subject:* [Freeswitch-users] FreeSWITCH 1.8.2 - 1-2 second dropped >> Audio\RTP at the start of a call >> >> >> Hi All, >> >> >> Since we've been using FreeSWITCH 1.8.2 we've noticed that the first 1-2 >> seconds of Audio\RTP at the start of the call when the call is answered is >> now dropped\missing but this doesn't occur on 1.6.20. When comparing the >> examples we've noticed the call flow is slightly different, as follows. >> >> >> FreeSWITCH 1.8.2 >> >> Leg A: switch_channel.c:2249 (sofia/internal/SRC_EXT at DOMAIN:PORT) >> Callstate Change DOWN -> RINGING >> Leg B: sofia.c:7291 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering >> state [calling][0] >> Leg B: sofia.c:7291 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering >> state [proceeding][180] >> Leg B: sofia.c:7401 Ring-Ready sofia/internal/DST_EXT at LAN_IP:PORT! >> Leg B: switch_channel.c:3354 (sofia/internal/DST_EXT at LAN_IP:PORT) >> Callstate Change DOWN -> RINGING >> Leg A: switch_ivr_originate.c:1246 Sending early media >> Leg A: switch_channel.c:3482 (sofia/internal/SRC_EXT at DOMAIN:PORT) >> Callstate Change RINGING -> EARLY >> Leg A: sofia.c:7291 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering >> state [early][183] >> Leg B: sofia.c:7291 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering >> state [completing][200] >> Leg B: switch_channel.c:3482 (sofia/internal/DST_EXT at LAN_IP:PORT) >> Callstate Change RINGING -> EARLY >> Leg B: sofia.c:7291 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering >> state [ready][200] >> Leg B: sofia.c:8429 Channel [sofia/internal/DST_EXT at LAN_IP:PORT] has >> been answered >> Leg B: switch_channel.c:3781 (sofia/internal/DST_EXT at LAN_IP:PORT) >> Callstate Change EARLY -> ACTIVE >> Leg A: switch_ivr_bridge.c:766 Channel [sofia/internal/SRC_EXT at DOMAIN:PORT] >> has been answered >> Leg A: sofia.c:7291 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering >> state [completed][200] >> Leg A: switch_channel.c:3781 (sofia/internal/SRC_EXT at DOMAIN:PORT) >> Callstate Change EARLY -> ACTIVE >> Leg A: sofia.c:7291 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering >> state [ready][200] >> Leg A: switch_ivr_async.c:1500 No silence detection configured; assuming >> start of speech >> Leg B: sofia.c:7291 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering >> state [calling][0] >> Leg A: sofia.c:7291 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering >> state [calling][0] >> Leg A: sofia.c:7291 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering >> state [ready][200] >> Leg A: sofia.c:8272 Processing updated SDP >> Leg B: sofia.c:7291 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering >> state [ready][200] >> >> >> FreeSWITCH 1.6.20 >> >> Leg A: switch_channel.c:2249 (sofia/internal/SRC_EXT at DOMAIN:PORT) >> Callstate Change DOWN -> RINGING >> Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering >> state [calling][0] >> Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering >> state [proceeding][180] >> Leg B: sofia.c:7192 Ring-Ready sofia/internal/DST_EXT at LAN_IP:PORT! >> Leg B: switch_channel.c:3346 (sofia/internal/DST_EXT at LAN_IP:PORT) >> Callstate Change DOWN -> RINGING >> Leg A: switch_ivr_originate.c:1215 Sending early media >> Leg A: switch_channel.c:3474 (sofia/internal/SRC_EXT at DOMAIN:PORT) >> Callstate Change RINGING -> EARLY >> Leg A: sofia.c:7084 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering >> state [early][183] >> Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering >> state [completing][200] >> Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering >> state [ready][200] >> Leg A: switch_channel.c:3773 (sofia/internal/SRC_EXT at DOMAIN:PORT) >> Callstate Change EARLY -> ACTIVE >> Leg A: sofia.c:7084 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering >> state [completed][200] >> Leg A: switch_ivr_originate.c:3705 Originate Resulted in Success: >> [sofia/internal/DST_EXT at LAN_IP:PORT] >> Leg B: switch_channel.c:3773 (sofia/internal/DST_EXT at LAN_IP:PORT) >> Callstate Change RINGING -> ACTIVE >> Leg A: sofia.c:7084 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering >> state [ready][200] >> Leg B: Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state >> [ready][200] >> Leg A: switch_ivr_async.c:1500 No silence detection configured; assuming >> start of speech >> Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering >> state [calling][0] >> Leg A: sofia.c:7084 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering >> state [calling][0] >> Leg A: sofia.c:7084 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering >> state [ready][200] >> Leg A: sofia.c:8061 Processing updated SDP >> Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering >> state [completing][200] >> Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering >> state [ready][200] >> >> >> On 1.6.20 Leg B changes straight from RINGING to ACTIVE, but on 1.8.2 Leg >> B first changes from RINGING to EARLY then EARLY to ACTIVE, not sure if >> this could be related. >> >> >> We've experimented with the following to no avail. >> rtp-rewrite-timestamps >> send_silence_when_idle >> fsctl sync_clock >> suppress_cng >> ignore_early_media >> >> As per: >> https://freeswitch.org/confluence/display/FREESWITCH/RTP+Issues >> https://freeswitch.org/confluence/display/FREESWITCH/VAD+and+CNG >> >> https://freeswitch.org/confluence/display/FREESWITCH/send_silence_when_idle >> https://freeswitch.org/confluence/display/FREESWITCH/Early+Media >> >> >> The calls are local between two extensions\endpoints on the same >> FreeSWITCH instance and the same SIP profile, the SIP profiles on both >> servers (1.6.20 and 1.8.2) are identical. >> >> >> Does anyone have any ideas? >> >> >> Thanks, >> >> Shaun >> _________________________________________________________________________ >> 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 -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: From me at nevian.org Mon Dec 17 18:22:00 2018 From: me at nevian.org (Serge S. Yuriev) Date: Mon, 17 Dec 2018 21:22:00 +0300 Subject: [Freeswitch-users] FS 1.8 Docker image In-Reply-To: References: Message-ID: <13192151545070920@myt6-7734411c649e.qloud-c.yandex.net> An HTML attachment was scrubbed... URL: From sebastian_ml at gmx.net Mon Dec 17 18:41:23 2018 From: sebastian_ml at gmx.net (Sebastian Kemper) Date: Mon, 17 Dec 2018 19:41:23 +0100 Subject: [Freeswitch-users] [Zombie Ports Listening] In-Reply-To: References: <20181214185238.GA25158@darth.lan> Message-ID: <20181217184123.GA6456@darth.lan> On Mon, Dec 17, 2018 at 07:08:11AM -0800, Joel Serrano wrote: > Have you tried commenting out the module so it’s not loaded? Hello Joel, It's not a module. I checked the source in src/switch_msrp.c, it doesn't seem to have an off switch. Kind regards, Seb From mike at freeswitch.org Mon Dec 17 18:54:23 2018 From: mike at freeswitch.org (Mike Jerris) Date: Mon, 17 Dec 2018 13:54:23 -0500 Subject: [Freeswitch-users] Disabling 200 OK with contact headers of all the lines In-Reply-To: References: Message-ID: <08A105FB-0134-4ABF-B134-171C928C14F3@freeswitch.org> https://tools.ietf.org/html/rfc3261#section-10.3 8. The registrar returns a 200 (OK) response. The response MUST contain Contact header field values enumerating all current bindings. Each Contact value MUST feature an "expires" parameter indicating its expiration interval chosen by the registrar. The response SHOULD include a Date header field. > On Dec 17, 2018, at 9:13 AM, sagar malam wrote: > > Thanks for reply Micheal. > > I dont want to disable multiple registrations.And i agree all the registrations are genuine. > > But my problem is that when FS responds to Register Request using 200 OK, The 200 OK has contact headers of all the registrations( of other SIP clients).As shown in below packet, there are three contact headers.I think there should be only one. > > ============================200 OK for register packet ============= > > 2018/12/08 13:36:35.426099 10.50.7.251:5070 -> 10.50.7.253:5060 > SIP/2.0 200 OK > Via: SIP/2.0/UDP 66.160.237.253:5060;branch=z9hG4bK9b46.f9b7a68125108f411537617d02bd48f8.0;received=10.50.7.253 > Via: SIP/2.0/UDP 198.136.236.1:5060;rport=5060;received=10.50.8.1;branch=z9hG4bK9b46.e6f1aa3817a238f499aa3aafa4217425.0 > Via: SIP/2.0/UDP 172.16.1.11;rport=1426;received=71.239.113.14;branch=z9hG4bK1df2302cD062D28F > From: "Main Line" >;tag=22CE54C0-84BCCF23 > To: >;tag=e8Fa2tND4U80D > Call-ID: d66f0f0592c09746b903406f312eb0c2 > CSeq: 590 REGISTER > Contact: t=tcp>;expires=379 > Contact: ort=tcp>;expires=245 > Contact: eb0c2>;expires=88 > Date: Sat, 08 Dec 2018 08:06:35 GMT > User-Agent: FreeSWITCH-mod_sofia/1.6.17~64bit > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: path, replaces > Content-Length: 0 > > =========================================================== > > Also i have compared this behaviour with OpenSIPs and Kamailio, They send only one contact header. > > Thanks in advance. > > > > On Wed, Dec 12, 2018 at 3:08 AM Michael Jerris > wrote: > These are all valid current registrations, there are params you can adjust for multi reg that will replace previous registrations instead of allowing multiple. > >> On Dec 9, 2018, at 1:17 PM, sagar malam > wrote: >> >> Hello , >> >> I am using FS SLA feature and it works very well.However i am facing an issue of registrations getting dropped as explained below : >> There are 3 phones registered with same extension number.Sofia is configured with "sip-expires-max-deviation" to randomise registration through expiry header in contact header.All the phones are Polycom. In case of shared lines FS adds contact header of all the lines(or phones with same extension) in 200 OK as shown below due to which all the phones are reading expiry timer from first contact header only.So in below example,Phone re registers after 379 seconds(first contact header) instead of 88 seconds(third contact header) leading to registration expiry on FS. >> >> Reason why phones are always reading expiry from first contact header is same Public IP(contact is re written by Proxy in front of FS) for all three phones which is confusing phone to identify its own contact header. >> Is there any way to configure FS to not send contact headers of all the registrations but only one that belongs to the line itself ? or any other way to fix it. >> ============================200 OK for register packet ============= >> >> 2018/12/08 13:36:35.426099 10.50.7.251:5070 -> 10.50.7.253:5060 >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 66.160.237.253:5060;branch=z9hG4bK9b46.f9b7a68125108f411537617d02bd48f8.0;received=10.50.7.253 >> Via: SIP/2.0/UDP 198.136.236.1:5060;rport=5060;received=10.50.8.1;branch=z9hG4bK9b46.e6f1aa3817a238f499aa3aafa4217425.0 >> Via: SIP/2.0/UDP 172.16.1.11;rport=1426;received=71.239.113.14;branch=z9hG4bK1df2302cD062D28F >> From: "Main Line" >;tag=22CE54C0-84BCCF23 >> To: >;tag=e8Fa2tND4U80D >> Call-ID: d66f0f0592c09746b903406f312eb0c2 >> CSeq: 590 REGISTER >> Contact: >> t=tcp>;expires=379 >> Contact: >> ort=tcp>;expires=245 >> Contact: >> eb0c2>;expires=88 >> Date: Sat, 08 Dec 2018 08:06:35 GMT >> User-Agent: FreeSWITCH-mod_sofia/1.6.17~64bit >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: path, replaces >> Content-Length: 0 -------------- next part -------------- An HTML attachment was scrubbed... URL: From anthony.minessale at gmail.com Mon Dec 17 20:55:10 2018 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 17 Dec 2018 14:55:10 -0600 Subject: [Freeswitch-users] FreeSWITCH 1.8.2 - 1-2 second dropped Audio\RTP at the start of a call In-Reply-To: References: <16D901F8-1A94-4FF3-AAC9-BF37E073BEE4@gmail.com> <97EE2A01-D430-421A-90B6-7CFA71DAF34B@gmail.com> Message-ID: Sounds like a broken record but the mailing list is not the best place to report issues. Almost inevitably, questions will be asked and data will need to be collected like logs etc. This is why we already point to JIRA to file tickets. Recommendations: 1) Make sure you don't have "answer_delay" set. 2) Check the media signaling data to make sure you are not using "rtp-auto-adjust which adds a time to media establishment to correct for incorrect media IPs. 3) Get a pcap as well as debug + sip trace before reporting any issue because its always going to be the first request anyway. 4) Use JIRA, feel free to ask about the JIRA here but don't rely on 1990's listserv server to track the issue progress. sofia global siptrace on console loglevel debug fsctl debug_level 10 On Mon, Dec 17, 2018 at 2:48 PM Shaun Stokes < shaun.stokes at itec-support.co.uk> wrote: > We reverted back to FreeSWITCH 1.6.20 but when this is compiled on > the Debian 9 server the problem still occurs. > > > We had to workaround some build errors for FS 1.6.20 to compile on Debian > 9 with PostgreSQL 11 but the problem was still present, as follows. > > -------------------- > > # Uninstall 1.1.0 SSL header files (libssl-dev) and install the older ones > (libssl1.0-dev). > apt-get install libssl1.0-dev > > # Fix PGSQL 11 Support > In the file: > /usr/src/freeswitch/srcswitch_pgsql.c > On line 389, replace this: > #if POSTGRESQL_MAJOR_VERSION >= 9 && POSTGRESQL_MINOR_VERSION >= 2 > With: > #if (POSTGRESQL_MAJOR_VERSION == 9 && POSTGRESQL_MINOR_VERSION >= 2) || > POSTGRESQL_MAJOR_VERSION > 9 > > # Do not build mod_flite or mod_enum > sed -i /usr/src/freeswitch/modules.conf -e > s:'asr_tts/mod_flite:#asr_tts/mod_flite:' > sed -i /usr/src/freeswitch/modules.conf -e > s:'applications/mod_enum:#applications/mod_enum:' > -------------------- > > We took the FS 1.6.20 binaries (pre-compiled) from a Debian 8 server and > restored them to our Debian 9 server which resolved the issue but we had to > copy some missing libs from a Debian 8 server: > /usr/lib/x86_64-linux-gnu/libssl.so.1.0.0 > /usr/lib/x86_64-linux-gnu/libcrypto.so.1.0.0 > > Given that the problem changes when the source-code is compiled on > different servers we suspect this may be a package problem not specific to > FreeSWITCH. > > This is also a problem on master, raised JIRA: > https://freeswitch.org/jira/browse/FS-11572 > > ------------------------------ > *From:* FreeSWITCH-users > on behalf of Shaun Stokes > *Sent:* 14 December 2018 13:16:14 > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] FreeSWITCH 1.8.2 - 1-2 second dropped > Audio\RTP at the start of a call > > Correction, we had moved FreeSWITCH 1.4 (not 1.8) to Server 1 which worked > without any audio delays. Upon testing FreeSWITCH 1.8 on Server 1 there is > a 1-2 second delay before RTP is established once the call is answered. > > This is a FreeSWITCH 1.8.2 issue, not a Debian 9 specific (also occurs on > Debian 8). FreeSWITCH 1.6 and 1.4 are not effected using the same > configuration through-out. > ------------------------------ > *From:* Shaun Stokes > *Sent:* 14 December 2018 11:44:18 > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] FreeSWITCH 1.8.2 - 1-2 second dropped > Audio\RTP at the start of a call > > We have built two test servers side by side on the same hardware with the > same configuration, as follows. > Server 1: Debian 8 with FreeSWITCH 1.6.20 > Server 2: Debian 9 with FreeSWITCH 1.8.2 > > We can replicate the 1-2 second delay on Server 2 only, whereas Server 1 > provides near instant RTP in both directions upon answer. Interestingly, if > we move FreeSWITCH 1.8.2 from Server 2 to Server 1 there are still no > issues with delay on Server 1, the problem is only observable on the Server > 2 running Debian 9 so the problem is not specifically related to FreeSWITCH > 1.8.2. > > At this stage it seems likely the issue lies with Debian 9 or the change > in packages on Debian 9. > > Thanks, > Shaun > ------------------------------ > *From:* FreeSWITCH-users > on behalf of Shaun Stokes > *Sent:* 11 December 2018 15:28:33 > *To:* FreeSWITCH Users Help > *Subject:* [Freeswitch-users] FreeSWITCH 1.8.2 - 1-2 second dropped > Audio\RTP at the start of a call > > > Hi All, > > > Since we've been using FreeSWITCH 1.8.2 we've noticed that the first 1-2 > seconds of Audio\RTP at the start of the call when the call is answered is > now dropped\missing but this doesn't occur on 1.6.20. When comparing the > examples we've noticed the call flow is slightly different, as follows. > > > FreeSWITCH 1.8.2 > > Leg A: switch_channel.c:2249 (sofia/internal/SRC_EXT at DOMAIN:PORT) > Callstate Change DOWN -> RINGING > Leg B: sofia.c:7291 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering > state [calling][0] > Leg B: sofia.c:7291 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering > state [proceeding][180] > Leg B: sofia.c:7401 Ring-Ready sofia/internal/DST_EXT at LAN_IP:PORT! > Leg B: switch_channel.c:3354 (sofia/internal/DST_EXT at LAN_IP:PORT) > Callstate Change DOWN -> RINGING > Leg A: switch_ivr_originate.c:1246 Sending early media > Leg A: switch_channel.c:3482 (sofia/internal/SRC_EXT at DOMAIN:PORT) > Callstate Change RINGING -> EARLY > Leg A: sofia.c:7291 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering > state [early][183] > Leg B: sofia.c:7291 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering > state [completing][200] > Leg B: switch_channel.c:3482 (sofia/internal/DST_EXT at LAN_IP:PORT) > Callstate Change RINGING -> EARLY > Leg B: sofia.c:7291 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering > state [ready][200] > Leg B: sofia.c:8429 Channel [sofia/internal/DST_EXT at LAN_IP:PORT] has been > answered > Leg B: switch_channel.c:3781 (sofia/internal/DST_EXT at LAN_IP:PORT) > Callstate Change EARLY -> ACTIVE > Leg A: switch_ivr_bridge.c:766 Channel [sofia/internal/SRC_EXT at DOMAIN:PORT] > has been answered > Leg A: sofia.c:7291 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering > state [completed][200] > Leg A: switch_channel.c:3781 (sofia/internal/SRC_EXT at DOMAIN:PORT) > Callstate Change EARLY -> ACTIVE > Leg A: sofia.c:7291 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering > state [ready][200] > Leg A: switch_ivr_async.c:1500 No silence detection configured; assuming > start of speech > Leg B: sofia.c:7291 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering > state [calling][0] > Leg A: sofia.c:7291 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering > state [calling][0] > Leg A: sofia.c:7291 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering > state [ready][200] > Leg A: sofia.c:8272 Processing updated SDP > Leg B: sofia.c:7291 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering > state [ready][200] > > > FreeSWITCH 1.6.20 > > Leg A: switch_channel.c:2249 (sofia/internal/SRC_EXT at DOMAIN:PORT) > Callstate Change DOWN -> RINGING > Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering > state [calling][0] > Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering > state [proceeding][180] > Leg B: sofia.c:7192 Ring-Ready sofia/internal/DST_EXT at LAN_IP:PORT! > Leg B: switch_channel.c:3346 (sofia/internal/DST_EXT at LAN_IP:PORT) > Callstate Change DOWN -> RINGING > Leg A: switch_ivr_originate.c:1215 Sending early media > Leg A: switch_channel.c:3474 (sofia/internal/SRC_EXT at DOMAIN:PORT) > Callstate Change RINGING -> EARLY > Leg A: sofia.c:7084 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering > state [early][183] > Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering > state [completing][200] > Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering > state [ready][200] > Leg A: switch_channel.c:3773 (sofia/internal/SRC_EXT at DOMAIN:PORT) > Callstate Change EARLY -> ACTIVE > Leg A: sofia.c:7084 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering > state [completed][200] > Leg A: switch_ivr_originate.c:3705 Originate Resulted in Success: > [sofia/internal/DST_EXT at LAN_IP:PORT] > Leg B: switch_channel.c:3773 (sofia/internal/DST_EXT at LAN_IP:PORT) > Callstate Change RINGING -> ACTIVE > Leg A: sofia.c:7084 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering > state [ready][200] > Leg B: Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state > [ready][200] > Leg A: switch_ivr_async.c:1500 No silence detection configured; assuming > start of speech > Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering > state [calling][0] > Leg A: sofia.c:7084 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering > state [calling][0] > Leg A: sofia.c:7084 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering > state [ready][200] > Leg A: sofia.c:8061 Processing updated SDP > Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering > state [completing][200] > Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering > state [ready][200] > > > On 1.6.20 Leg B changes straight from RINGING to ACTIVE, but on 1.8.2 Leg > B first changes from RINGING to EARLY then EARLY to ACTIVE, not sure if > this could be related. > > > We've experimented with the following to no avail. > rtp-rewrite-timestamps > send_silence_when_idle > fsctl sync_clock > suppress_cng > ignore_early_media > > As per: > https://freeswitch.org/confluence/display/FREESWITCH/RTP+Issues > https://freeswitch.org/confluence/display/FREESWITCH/VAD+and+CNG > https://freeswitch.org/confluence/display/FREESWITCH/send_silence_when_idle > https://freeswitch.org/confluence/display/FREESWITCH/Early+Media > > > The calls are local between two extensions\endpoints on the same > FreeSWITCH instance and the same SIP profile, the SIP profiles on both > servers (1.6.20 and 1.8.2) are identical. > > > Does anyone have any ideas? > > > Thanks, > > Shaun > _________________________________________________________________________ > 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 -- Anthony Minessale II Founder, FreeSWITCH. http://freeswitch.com https://youtu.be/l_hOxzCt6X4 https://www.youtube.com/watch?v=oAxXgyx5jUw https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA -------------- next part -------------- An HTML attachment was scrubbed... URL: From sat at calgaryit.com Tue Dec 18 03:27:17 2018 From: sat at calgaryit.com (George) Date: Mon, 17 Dec 2018 20:27:17 -0700 (MST) Subject: [Freeswitch-users] calls to local 311 line with two IVRs In-Reply-To: <788aa736-39d2-3a90-f68d-24a29e60c5bf@nevian.org> References: <26233380.38805.1544587345787.JavaMail.zimbra@calgaryit.com> <501702402.43110.1544639656116.JavaMail.zimbra@calgaryit.com> <788aa736-39d2-3a90-f68d-24a29e60c5bf@nevian.org> Message-ID: <273428211.6465.1545103637501.JavaMail.zimbra@calgaryit.com> Serge, much appreciated, this worked!!!! are there any FAqs on this? Thank You, George ----- Original Message ----- From: "Serge S. Yuriev" To: "freeswitch-users" Sent: Monday, December 17, 2018 9:07:32 AM Subject: Re: [Freeswitch-users] calls to local 311 line with two IVRs Hi, You have to set this towards provider (buggy Nortel/Cisco/Avaya) On 12/12/2018 21:34, George wrote: > here it is: > > https://freeswitch.org/jira/secure/attachment/28046/cap20181212.pcap > > Thank You, > George > > ----- Original Message ----- > From: "Michael Jerris" > To: "freeswitch-users" > Sent: Wednesday, December 12, 2018 9:57:09 AM > Subject: Re: [Freeswitch-users] calls to local 311 line with two IVRs > > Would need to see the full sip trace to answer this. A debug log with sip trace is most useful > >> On Dec 11, 2018, at 11:02 PM, George wrote: >> >> when I call a local 311 number, the call goes through, I hear the first announcement, after with time the call is supposed to get transferred to a second IVR which should give the choices of departments, the call just goes silent and eventually gets hung on on. >> >> FreeSWITCH Version 1.8.2-3-a98a958ac3~64bit (-3-a98a958ac3 64bit) >> Debian 9.6 >> >> this is what my FS sends back to the provider: >> >> v=0 >> o=FreeSWITCH 1544570427 1544570429 IN IP4 10.185.45.238 >> s=FreeSWITCH >> c=IN IP4 10.185.45.238 >> t=0 0 >> m=audio 16506 RTP/AVP 0 101 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=sendonly >> a=ptime:20 >> >> the provider says its because the a=sendonly I am not receiving the announcements >> >> any help on this? >> >> Thank You, >> George > > > _________________________________________________________________________ > 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 > -- Serge S. Yuriev Senior VoIP engineer _________________________________________________________________________ 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 igor.potjevlesch at gmail.com Tue Dec 18 10:56:03 2018 From: igor.potjevlesch at gmail.com (igor.potjevlesch at gmail.com) Date: Tue, 18 Dec 2018 11:56:03 +0100 Subject: [Freeswitch-users] Relay a REGISTER Message-ID: <000f01d496c0$454fc550$cfef4ff0$@gmail.com> Hello, I'd like to relay a REGISTER received on leg A to the leg B. The idea is to have a Contact URI with the B leg IP address when the REGISTER goes out of the FS. But I didn't found how to avoid Freeswitch to handle the REGISTER locally. Regards, Igor. -------------- next part -------------- An HTML attachment was scrubbed... URL: From bilaln018 at gmail.com Tue Dec 18 16:41:09 2018 From: bilaln018 at gmail.com (Bilal Abbasi) Date: Tue, 18 Dec 2018 21:41:09 +0500 Subject: [Freeswitch-users] [Zombie Ports Listening] In-Reply-To: <20181217184123.GA6456@darth.lan> References: <20181214185238.GA25158@darth.lan> <20181217184123.GA6456@darth.lan> Message-ID: Exactly this is what i am looking for, as i cannot see any module associated with that. Regards Abbasi On Tue, 18 Dec 2018 at 2:04 AM, Sebastian Kemper wrote: > On Mon, Dec 17, 2018 at 07:08:11AM -0800, Joel Serrano wrote: > > Have you tried commenting out the module so it’s not loaded? > > Hello Joel, > > It's not a module. I checked the source in src/switch_msrp.c, it doesn't > seem to have an off switch. > > Kind regards, > Seb > > _________________________________________________________________________ > 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 julien.terrasson at gmail.com Tue Dec 18 17:49:48 2018 From: julien.terrasson at gmail.com (Julien Terrasson) Date: Tue, 18 Dec 2018 18:49:48 +0100 Subject: [Freeswitch-users] transfer B-leg to a custom extension with execute_on_answer : fails Message-ID: Hi everyone, I'm looking from advices on that issue : In the following scenario, where A & B are PSTN subscribers : STEP 1 : A call freeswitch, freeswitch answers and play an audio prompt. STEP 2 : when the prompt ends, freeswitch attempt to bridge A with B STEP 3 : B_answer. I set a no_local::execute_on_answer hook to park A and play B an audio prompt (warning message) when B_answer. But thuis is what happen when the bridge command is executed : CASE 1 : If the INVITE TO B is responded with a 183-SESSION_PROGRESS before the 200-OK, the execute_on_answer hook work as i expect (A is parked, B is played the audio prompt) CASE 2 : If the INVITE TO B is responded with a 200-OK, the execute_on_answer hook does not work as i expect (A is parked but hangup B with cause destination_out_of_order). >From the trace i can see that in CASE 1, B party is put in CS_EXCHANGE_MEDIA earlier than in CASE2 Would this be likely to create the problem ? Best Regards, Julien -------------- next part -------------- An HTML attachment was scrubbed... URL: From alex at freeswitch.com Wed Dec 19 03:55:09 2018 From: alex at freeswitch.com (Alexey Sibyakin) Date: Wed, 19 Dec 2018 12:55:09 +0900 Subject: [Freeswitch-users] Relay a REGISTER In-Reply-To: <000f01d496c0$454fc550$cfef4ff0$@gmail.com> References: <000f01d496c0$454fc550$cfef4ff0$@gmail.com> Message-ID: Hello, Try some sip proxy like Kamailio/OpenSIPs. FreeSWITCH is not a proxy. Regards, Alex On Wed, Dec 19, 2018 at 3:36 AM wrote: > Hello, > > > > I'd like to relay a REGISTER received on leg A to the leg B. The idea is > to have a Contact URI with the B leg IP address when the REGISTER goes out > of the FS. > > But I didn't found how to avoid Freeswitch to handle the REGISTER locally. > > > > Regards, > > > > Igor. > _________________________________________________________________________ > 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 Wed Dec 19 04:00:30 2018 From: alex at freeswitch.com (Alexey Sibyakin) Date: Wed, 19 Dec 2018 13:00:30 +0900 Subject: [Freeswitch-users] MOD_CALLCENTER cannot originate call to agent In-Reply-To: References: Message-ID: Hi, Usually, RECOVERY_ON_TIMER_EXPIRE is a network or signaling problem. Any siptace/pcap? Regards, Alex On Tue, Dec 18, 2018 at 1:24 AM Chhorm Chhatra wrote: > My case is that Freeswitch cannot originate the call to the agent > (cause: RECOVERY_ON_TIMER_EXPIRE). > Any help would be very appreciated. > > On Sun, 16 Dec 2018 at 11:52, Chhorm Chhatra wrote: > >> Hi, >> Currently, I am experiencing a very weird behavior of mod_callcenter like >> the following: >> 1. after 4 to 5 mins of hearing the moh-sound, it started to randomly >> play media from the sound/music folder >> 2. MOD_CALLCENTER *can never bridge any call to the agent. It mostly >> says the user is not registered (although it's registered). Sometimes, it >> says no-answer but the agent never rings.* >> >> *Please kindly help me.* >> >> _________________________________________________________________________ > 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 imfanee at gmail.com Wed Dec 19 05:51:32 2018 From: imfanee at gmail.com (Faisal Hanif) Date: Wed, 19 Dec 2018 10:51:32 +0500 Subject: [Freeswitch-users] Relay a REGISTER In-Reply-To: <000f01d496c0$454fc550$cfef4ff0$@gmail.com> References: <000f01d496c0$454fc550$cfef4ff0$@gmail.com> Message-ID: FreeSWITCH doesn't provide signalling proxy but you can achieve it in a way that create a gateway for every user, this will cause outbound gateway registration request for every incoming user registration request. Faisal On Tue, Dec 18, 2018, 11:38 PM Hello, > > > > I'd like to relay a REGISTER received on leg A to the leg B. The idea is > to have a Contact URI with the B leg IP address when the REGISTER goes out > of the FS. > > But I didn't found how to avoid Freeswitch to handle the REGISTER locally. > > > > Regards, > > > > Igor. > _________________________________________________________________________ > 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 sagarmalam at gmail.com Wed Dec 19 07:54:14 2018 From: sagarmalam at gmail.com (sagar malam) Date: Wed, 19 Dec 2018 13:24:14 +0530 Subject: [Freeswitch-users] Disabling 200 OK with contact headers of all the lines In-Reply-To: <08A105FB-0134-4ABF-B134-171C928C14F3@freeswitch.org> References: <08A105FB-0134-4ABF-B134-171C928C14F3@freeswitch.org> Message-ID: Thanks for information Mike. Yes,FS is doing right thing as per RFC.Also i check OpenSIPS and Kamailio doing same thing(i made mistake during first observation) , only difference is that it uses comma separated string of contacts in single contact header instead of separate contact headers for each registration. Looking into sofia source code, I found that FS had option to not look up in DB and return contact header(only one) as it was in register packet ( *reg-deny-binding-fetch-and-no-lookup* ).This is exactly what i was looking for and it solved my problem as well with polycom phones. But it is deprecated by FS. And not recommended by RFC. However i am curious to know what is purpose of it ? I was not able to find any specific reason for 200 OK to have contacts of all the bindings.Do you know why ? I can clearly find out below *issues* due to multiple contact headers : 1) Having multiple contact headers may confuse client to identify their own contact header and therefore may not be able to get registration expiry time provided by server.It happens with polycom phones for sure.( https://support.onsip.com/hc/en-us/articles/204028710-Polycom-Multiple-Contacts-Registration-Bug ) 2) Very big SIP packets( consider a case of 20 registrations with same AOR).This wont be any issue while using TCP or UDP with IPv4 but if we use UDP with IPv6 , too big packets will be dropped by Ethernet ports because as per IPv6 standards only applications(FS) can fragment IPv6 UDP packets. 3) For each register request FS will execute query in DB to fetch all the registrations which adds overhead on DB as well. Looking forward for your thoughts.Thanks again. On Tue, Dec 18, 2018 at 2:05 AM Mike Jerris wrote: > https://tools.ietf.org/html/rfc3261#section-10.3 > > 8. The registrar returns a 200 (OK) response. The response MUST > contain Contact header field values enumerating all current > bindings. Each Contact value MUST feature an "expires" > parameter indicating its expiration interval chosen by the > registrar. The response SHOULD include a Date header field. > > > > On Dec 17, 2018, at 9:13 AM, sagar malam wrote: > > Thanks for reply Micheal. > > I dont want to disable multiple registrations.And i agree all the > registrations are genuine. > > But my problem is that when FS responds to Register Request using 200 OK, > The 200 OK has contact headers of all the registrations( of other SIP > clients).As shown in below packet, there are three contact headers.I think > there should be only one. > > ============================200 OK for register packet ============= > > 2018/12/08 13:36:35.426099 10.50.7.251:5070 -> 10.50.7.253:5060 > SIP/2.0 200 OK > Via: SIP/2.0/UDP 66.160.237.253:5060 > ;branch=z9hG4bK9b46.f9b7a68125108f411537617d02bd48f8.0;received=10.50.7.253 > Via: SIP/2.0/UDP 198.136.236.1:5060 > ;rport=5060;received=10.50.8.1;branch=z9hG4bK9b46.e6f1aa3817a238f499aa3aafa4217425.0 > Via: SIP/2.0/UDP > 172.16.1.11;rport=1426;received=71.239.113.14;branch=z9hG4bK1df2302cD062D28F > From: "Main Line" ;tag=22CE54C0-84BCCF23 > To: ;tag=e8Fa2tND4U80D > Call-ID: d66f0f0592c09746b903406f312eb0c2 > CSeq: 590 REGISTER > *Contact: > *t=tcp>;expires=379* > *Contact: > *ort=tcp>;expires=245* > *Contact: > *eb0c2>;expires=88* > Date: Sat, 08 Dec 2018 08:06:35 GMT > User-Agent: FreeSWITCH-mod_sofia/1.6.17~64bit > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, > REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: path, replaces > Content-Length: 0 > > =========================================================== > > Also i have compared this behaviour with OpenSIPs and Kamailio, They send > only one contact header. > > Thanks in advance. > > > > On Wed, Dec 12, 2018 at 3:08 AM Michael Jerris wrote: > >> These are all valid current registrations, there are params you can >> adjust for multi reg that will replace previous registrations instead of >> allowing multiple. >> >> On Dec 9, 2018, at 1:17 PM, sagar malam wrote: >> >> Hello , >> >> I am using FS SLA feature and it works very well.However i am facing an >> issue of registrations getting dropped as explained below : >> There are 3 phones registered with same extension number.Sofia is >> configured with "sip-expires-max-deviation" to randomise registration >> through expiry header in contact header.All the phones are Polycom. In case >> of shared lines FS adds contact header of all the lines(or phones with same >> extension) in 200 OK as shown below due to which all the phones are reading >> expiry timer from first contact header only.So in below example,Phone re >> registers after 379 seconds(first contact header) instead of 88 >> seconds(third contact header) leading to registration expiry on FS. >> >> Reason why phones are always reading expiry from first contact header is >> same Public IP(contact is re written by Proxy in front of FS) for all >> three phones which is confusing phone to identify its own contact header. >> Is there any way to configure FS to not send contact headers of all the >> registrations but only one that belongs to the line itself ? or any other >> way to fix it. >> ============================200 OK for register packet ============= >> >> 2018/12/08 13:36:35.426099 10.50.7.251:5070 -> 10.50.7.253:5060 >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 66.160.237.253:5060 >> ;branch=z9hG4bK9b46.f9b7a68125108f411537617d02bd48f8.0;received=10.50.7.253 >> Via: SIP/2.0/UDP 198.136.236.1:5060 >> ;rport=5060;received=10.50.8.1;branch=z9hG4bK9b46.e6f1aa3817a238f499aa3aafa4217425.0 >> Via: SIP/2.0/UDP >> 172.16.1.11;rport=1426;received=71.239.113.14;branch=z9hG4bK1df2302cD062D28F >> From: "Main Line" ;tag=22CE54C0-84BCCF23 >> To: ;tag=e8Fa2tND4U80D >> Call-ID: d66f0f0592c09746b903406f312eb0c2 >> CSeq: 590 REGISTER >> Contact: < >> sip:398 at 71.239.113.14:1071;alias=10.50.8.1~5060~1;x-nat=yes;pv-ip=172.16.1.12;pb-ip=71.239.113.14;pb-pt=1071;mac-address=64167f2ec274;transp >> t=tcp>;expires=379 >> Contact: < >> sip:398 at 71.239.113.14:56478;alias=10.50.8.1~5060~1;x-nat=yes;pv-ip=172.16.1.13;pb-ip=71.239.113.14;pb-pt=56478;mac-address=64167f2ebd54;tran >> ort=tcp>;expires=245 >> Contact: < >> sip:398 at 71.239.113.14:1426;alias=10.50.8.1~5060~1;x-nat=yes;pv-ip=172.16.1.11;pb-ip=71.239.113.14;pb-pt=1426;transport=udp;mac-address=64167 >> eb0c2>;expires=88 >> Date: Sat, 08 Dec 2018 08:06:35 GMT >> User-Agent: FreeSWITCH-mod_sofia/1.6.17~64bit >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: path, replaces >> Content-Length: 0 >> >> > _________________________________________________________________________ > 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 -- Thanks, Sagar -------------- next part -------------- An HTML attachment was scrubbed... URL: From shaun.stokes at itec-support.co.uk Wed Dec 19 14:07:18 2018 From: shaun.stokes at itec-support.co.uk (Shaun Stokes) Date: Wed, 19 Dec 2018 14:07:18 +0000 Subject: [Freeswitch-users] FreeSWITCH 1.8.2 - 1-2 second dropped Audio\RTP at the start of a call In-Reply-To: References: <16D901F8-1A94-4FF3-AAC9-BF37E073BEE4@gmail.com> <97EE2A01-D430-421A-90B6-7CFA71DAF34B@gmail.com> , Message-ID: Thank you all for your recommendations. We raised a JIRA but it was closed as incomplete, we will look at re-opening once we've compiled the relevant logs and pcaps etc unless we reach a conclusion before hand. Some additional points: - We do not set "answer_delay". - There is still a 1-2 second delay after setting "disable_rtp_auto_adjust" on the SIP profile. We've identified this as a package issue, when FreeSWITCH is built with certain packages installed this is causing mod_sofia.so to be much smaller than it should be (5.49MB instead of 8.58MB). We're able to consistently reproduce the following when running FreeSWITCH on the same Debian 9 server using the same FreeSWITCH configuration and scripts etc: - FreeSWITCH 1.8.2 built on Debian 9 - 1-2 second audio\RTP delay on answer - FreeSWITCH 1.6.20 built on Debian 9 - 1-2 second audio\RTP delay on answer - FreeSWITCH 1.6.20 built on Debian 8 (moved binaries to Debian 9 server) - No audio\RTP delay on answer (audio\RTP is instantaneous) - FreeSWITCH 1.8.2 built on Debian 9 base (no additional packages) - No audio\RTP delay on answer (audio\RTP is instantaneous) Will post back once we've identified the root cause. Thanks, Shaun ________________________________ From: FreeSWITCH-users on behalf of Anthony Minessale Sent: 17 December 2018 20:55 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] FreeSWITCH 1.8.2 - 1-2 second dropped Audio\RTP at the start of a call Sounds like a broken record but the mailing list is not the best place to report issues. Almost inevitably, questions will be asked and data will need to be collected like logs etc. This is why we already point to JIRA to file tickets. Recommendations: 1) Make sure you don't have "answer_delay" set. 2) Check the media signaling data to make sure you are not using "rtp-auto-adjust which adds a time to media establishment to correct for incorrect media IPs. 3) Get a pcap as well as debug + sip trace before reporting any issue because its always going to be the first request anyway. 4) Use JIRA, feel free to ask about the JIRA here but don't rely on 1990's listserv server to track the issue progress. sofia global siptrace on console loglevel debug fsctl debug_level 10 On Mon, Dec 17, 2018 at 2:48 PM Shaun Stokes > wrote: We reverted back to FreeSWITCH 1.6.20 but when this is compiled on the Debian 9 server the problem still occurs. We had to workaround some build errors for FS 1.6.20 to compile on Debian 9 with PostgreSQL 11 but the problem was still present, as follows. -------------------- # Uninstall 1.1.0 SSL header files (libssl-dev) and install the older ones (libssl1.0-dev). apt-get install libssl1.0-dev # Fix PGSQL 11 Support In the file: /usr/src/freeswitch/srcswitch_pgsql.c On line 389, replace this: #if POSTGRESQL_MAJOR_VERSION >= 9 && POSTGRESQL_MINOR_VERSION >= 2 With: #if (POSTGRESQL_MAJOR_VERSION == 9 && POSTGRESQL_MINOR_VERSION >= 2) || POSTGRESQL_MAJOR_VERSION > 9 # Do not build mod_flite or mod_enum sed -i /usr/src/freeswitch/modules.conf -e s:'asr_tts/mod_flite:#asr_tts/mod_flite:' sed -i /usr/src/freeswitch/modules.conf -e s:'applications/mod_enum:#applications/mod_enum:' -------------------- We took the FS 1.6.20 binaries (pre-compiled) from a Debian 8 server and restored them to our Debian 9 server which resolved the issue but we had to copy some missing libs from a Debian 8 server: /usr/lib/x86_64-linux-gnu/libssl.so.1.0.0 /usr/lib/x86_64-linux-gnu/libcrypto.so.1.0.0 Given that the problem changes when the source-code is compiled on different servers we suspect this may be a package problem not specific to FreeSWITCH. This is also a problem on master, raised JIRA: https://freeswitch.org/jira/browse/FS-11572 ________________________________ From: FreeSWITCH-users > on behalf of Shaun Stokes > Sent: 14 December 2018 13:16:14 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] FreeSWITCH 1.8.2 - 1-2 second dropped Audio\RTP at the start of a call Correction, we had moved FreeSWITCH 1.4 (not 1.8) to Server 1 which worked without any audio delays. Upon testing FreeSWITCH 1.8 on Server 1 there is a 1-2 second delay before RTP is established once the call is answered. This is a FreeSWITCH 1.8.2 issue, not a Debian 9 specific (also occurs on Debian 8). FreeSWITCH 1.6 and 1.4 are not effected using the same configuration through-out. ________________________________ From: Shaun Stokes Sent: 14 December 2018 11:44:18 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] FreeSWITCH 1.8.2 - 1-2 second dropped Audio\RTP at the start of a call We have built two test servers side by side on the same hardware with the same configuration, as follows. Server 1: Debian 8 with FreeSWITCH 1.6.20 Server 2: Debian 9 with FreeSWITCH 1.8.2 We can replicate the 1-2 second delay on Server 2 only, whereas Server 1 provides near instant RTP in both directions upon answer. Interestingly, if we move FreeSWITCH 1.8.2 from Server 2 to Server 1 there are still no issues with delay on Server 1, the problem is only observable on the Server 2 running Debian 9 so the problem is not specifically related to FreeSWITCH 1.8.2. At this stage it seems likely the issue lies with Debian 9 or the change in packages on Debian 9. Thanks, Shaun ________________________________ From: FreeSWITCH-users > on behalf of Shaun Stokes > Sent: 11 December 2018 15:28:33 To: FreeSWITCH Users Help Subject: [Freeswitch-users] FreeSWITCH 1.8.2 - 1-2 second dropped Audio\RTP at the start of a call Hi All, Since we've been using FreeSWITCH 1.8.2 we've noticed that the first 1-2 seconds of Audio\RTP at the start of the call when the call is answered is now dropped\missing but this doesn't occur on 1.6.20. When comparing the examples we've noticed the call flow is slightly different, as follows. FreeSWITCH 1.8.2 Leg A: switch_channel.c:2249 (sofia/internal/SRC_EXT at DOMAIN:PORT) Callstate Change DOWN -> RINGING Leg B: sofia.c:7291 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [calling][0] Leg B: sofia.c:7291 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [proceeding][180] Leg B: sofia.c:7401 Ring-Ready sofia/internal/DST_EXT at LAN_IP:PORT! Leg B: switch_channel.c:3354 (sofia/internal/DST_EXT at LAN_IP:PORT) Callstate Change DOWN -> RINGING Leg A: switch_ivr_originate.c:1246 Sending early media Leg A: switch_channel.c:3482 (sofia/internal/SRC_EXT at DOMAIN:PORT) Callstate Change RINGING -> EARLY Leg A: sofia.c:7291 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering state [early][183] Leg B: sofia.c:7291 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [completing][200] Leg B: switch_channel.c:3482 (sofia/internal/DST_EXT at LAN_IP:PORT) Callstate Change RINGING -> EARLY Leg B: sofia.c:7291 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [ready][200] Leg B: sofia.c:8429 Channel [sofia/internal/DST_EXT at LAN_IP:PORT] has been answered Leg B: switch_channel.c:3781 (sofia/internal/DST_EXT at LAN_IP:PORT) Callstate Change EARLY -> ACTIVE Leg A: switch_ivr_bridge.c:766 Channel [sofia/internal/SRC_EXT at DOMAIN:PORT] has been answered Leg A: sofia.c:7291 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering state [completed][200] Leg A: switch_channel.c:3781 (sofia/internal/SRC_EXT at DOMAIN:PORT) Callstate Change EARLY -> ACTIVE Leg A: sofia.c:7291 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering state [ready][200] Leg A: switch_ivr_async.c:1500 No silence detection configured; assuming start of speech Leg B: sofia.c:7291 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [calling][0] Leg A: sofia.c:7291 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering state [calling][0] Leg A: sofia.c:7291 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering state [ready][200] Leg A: sofia.c:8272 Processing updated SDP Leg B: sofia.c:7291 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [ready][200] FreeSWITCH 1.6.20 Leg A: switch_channel.c:2249 (sofia/internal/SRC_EXT at DOMAIN:PORT) Callstate Change DOWN -> RINGING Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [calling][0] Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [proceeding][180] Leg B: sofia.c:7192 Ring-Ready sofia/internal/DST_EXT at LAN_IP:PORT! Leg B: switch_channel.c:3346 (sofia/internal/DST_EXT at LAN_IP:PORT) Callstate Change DOWN -> RINGING Leg A: switch_ivr_originate.c:1215 Sending early media Leg A: switch_channel.c:3474 (sofia/internal/SRC_EXT at DOMAIN:PORT) Callstate Change RINGING -> EARLY Leg A: sofia.c:7084 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering state [early][183] Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [completing][200] Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [ready][200] Leg A: switch_channel.c:3773 (sofia/internal/SRC_EXT at DOMAIN:PORT) Callstate Change EARLY -> ACTIVE Leg A: sofia.c:7084 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering state [completed][200] Leg A: switch_ivr_originate.c:3705 Originate Resulted in Success: [sofia/internal/DST_EXT at LAN_IP:PORT] Leg B: switch_channel.c:3773 (sofia/internal/DST_EXT at LAN_IP:PORT) Callstate Change RINGING -> ACTIVE Leg A: sofia.c:7084 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering state [ready][200] Leg B: Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [ready][200] Leg A: switch_ivr_async.c:1500 No silence detection configured; assuming start of speech Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [calling][0] Leg A: sofia.c:7084 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering state [calling][0] Leg A: sofia.c:7084 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering state [ready][200] Leg A: sofia.c:8061 Processing updated SDP Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [completing][200] Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [ready][200] On 1.6.20 Leg B changes straight from RINGING to ACTIVE, but on 1.8.2 Leg B first changes from RINGING to EARLY then EARLY to ACTIVE, not sure if this could be related. We've experimented with the following to no avail. rtp-rewrite-timestamps send_silence_when_idle fsctl sync_clock suppress_cng ignore_early_media As per: https://freeswitch.org/confluence/display/FREESWITCH/RTP+Issues https://freeswitch.org/confluence/display/FREESWITCH/VAD+and+CNG https://freeswitch.org/confluence/display/FREESWITCH/send_silence_when_idle https://freeswitch.org/confluence/display/FREESWITCH/Early+Media The calls are local between two extensions\endpoints on the same FreeSWITCH instance and the same SIP profile, the SIP profiles on both servers (1.6.20 and 1.8.2) are identical. Does anyone have any ideas? Thanks, Shaun _________________________________________________________________________ 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 -- Anthony Minessale II Founder, FreeSWITCH. http://freeswitch.com https://youtu.be/l_hOxzCt6X4 https://www.youtube.com/watch?v=oAxXgyx5jUw https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA -------------- next part -------------- An HTML attachment was scrubbed... URL: From ch.chhatra at gmail.com Wed Dec 19 15:07:50 2018 From: ch.chhatra at gmail.com (Chhorm Chhatra) Date: Wed, 19 Dec 2018 22:07:50 +0700 Subject: [Freeswitch-users] MOD_CALLCENTER cannot originate call to agent In-Reply-To: References: Message-ID: Hello, Thank you for your reply. I really appreciate it. you are right. This is clearly a NAT problem. It was resolved by configuring NAT Port Mapping from Freeswitch Server to the contactcenter agent. (In case of Freeswitch deploying in different vLAN from agents, make sure Freeswitch can reach the agents without any restriction. You might want to ask the network admin to allow this rule). On Wed, Dec 19, 2018 at 10:01 PM Alexey Sibyakin wrote: > Hi, > > Usually, RECOVERY_ON_TIMER_EXPIRE is a network or signaling problem. Any > siptace/pcap? > > Regards, > > Alex > > On Tue, Dec 18, 2018 at 1:24 AM Chhorm Chhatra > wrote: > >> My case is that Freeswitch cannot originate the call to the agent >> (cause: RECOVERY_ON_TIMER_EXPIRE). >> Any help would be very appreciated. >> >> On Sun, 16 Dec 2018 at 11:52, Chhorm Chhatra >> wrote: >> >>> Hi, >>> Currently, I am experiencing a very weird behavior of mod_callcenter >>> like the following: >>> 1. after 4 to 5 mins of hearing the moh-sound, it started to randomly >>> play media from the sound/music folder >>> 2. MOD_CALLCENTER *can never bridge any call to the agent. It mostly >>> says the user is not registered (although it's registered). Sometimes, it >>> says no-answer but the agent never rings.* >>> >>> *Please kindly help me.* >>> >>> _________________________________________________________________________ >> 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 shaun.stokes at itec-support.co.uk Wed Dec 19 21:43:36 2018 From: shaun.stokes at itec-support.co.uk (Shaun Stokes) Date: Wed, 19 Dec 2018 21:43:36 +0000 Subject: [Freeswitch-users] FreeSWITCH 1.8.2 - 1-2 second dropped Audio\RTP at the start of a call In-Reply-To: References: <16D901F8-1A94-4FF3-AAC9-BF37E073BEE4@gmail.com> <97EE2A01-D430-421A-90B6-7CFA71DAF34B@gmail.com> , , Message-ID: We've identified the cause and have now implemented a fix, this was a local issue caused by a bug in our build script which applies tweaks to mod_sofia. Thanks, Shaun ________________________________ From: FreeSWITCH-users on behalf of Shaun Stokes Sent: 19 December 2018 14:07:18 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] FreeSWITCH 1.8.2 - 1-2 second dropped Audio\RTP at the start of a call Thank you all for your recommendations. We raised a JIRA but it was closed as incomplete, we will look at re-opening once we've compiled the relevant logs and pcaps etc unless we reach a conclusion before hand. Some additional points: - We do not set "answer_delay". - There is still a 1-2 second delay after setting "disable_rtp_auto_adjust" on the SIP profile. We've identified this as a package issue, when FreeSWITCH is built with certain packages installed this is causing mod_sofia.so to be much smaller than it should be (5.49MB instead of 8.58MB). We're able to consistently reproduce the following when running FreeSWITCH on the same Debian 9 server using the same FreeSWITCH configuration and scripts etc: - FreeSWITCH 1.8.2 built on Debian 9 - 1-2 second audio\RTP delay on answer - FreeSWITCH 1.6.20 built on Debian 9 - 1-2 second audio\RTP delay on answer - FreeSWITCH 1.6.20 built on Debian 8 (moved binaries to Debian 9 server) - No audio\RTP delay on answer (audio\RTP is instantaneous) - FreeSWITCH 1.8.2 built on Debian 9 base (no additional packages) - No audio\RTP delay on answer (audio\RTP is instantaneous) Will post back once we've identified the root cause. Thanks, Shaun ________________________________ From: FreeSWITCH-users on behalf of Anthony Minessale Sent: 17 December 2018 20:55 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] FreeSWITCH 1.8.2 - 1-2 second dropped Audio\RTP at the start of a call Sounds like a broken record but the mailing list is not the best place to report issues. Almost inevitably, questions will be asked and data will need to be collected like logs etc. This is why we already point to JIRA to file tickets. Recommendations: 1) Make sure you don't have "answer_delay" set. 2) Check the media signaling data to make sure you are not using "rtp-auto-adjust which adds a time to media establishment to correct for incorrect media IPs. 3) Get a pcap as well as debug + sip trace before reporting any issue because its always going to be the first request anyway. 4) Use JIRA, feel free to ask about the JIRA here but don't rely on 1990's listserv server to track the issue progress. sofia global siptrace on console loglevel debug fsctl debug_level 10 On Mon, Dec 17, 2018 at 2:48 PM Shaun Stokes > wrote: We reverted back to FreeSWITCH 1.6.20 but when this is compiled on the Debian 9 server the problem still occurs. We had to workaround some build errors for FS 1.6.20 to compile on Debian 9 with PostgreSQL 11 but the problem was still present, as follows. -------------------- # Uninstall 1.1.0 SSL header files (libssl-dev) and install the older ones (libssl1.0-dev). apt-get install libssl1.0-dev # Fix PGSQL 11 Support In the file: /usr/src/freeswitch/srcswitch_pgsql.c On line 389, replace this: #if POSTGRESQL_MAJOR_VERSION >= 9 && POSTGRESQL_MINOR_VERSION >= 2 With: #if (POSTGRESQL_MAJOR_VERSION == 9 && POSTGRESQL_MINOR_VERSION >= 2) || POSTGRESQL_MAJOR_VERSION > 9 # Do not build mod_flite or mod_enum sed -i /usr/src/freeswitch/modules.conf -e s:'asr_tts/mod_flite:#asr_tts/mod_flite:' sed -i /usr/src/freeswitch/modules.conf -e s:'applications/mod_enum:#applications/mod_enum:' -------------------- We took the FS 1.6.20 binaries (pre-compiled) from a Debian 8 server and restored them to our Debian 9 server which resolved the issue but we had to copy some missing libs from a Debian 8 server: /usr/lib/x86_64-linux-gnu/libssl.so.1.0.0 /usr/lib/x86_64-linux-gnu/libcrypto.so.1.0.0 Given that the problem changes when the source-code is compiled on different servers we suspect this may be a package problem not specific to FreeSWITCH. This is also a problem on master, raised JIRA: https://freeswitch.org/jira/browse/FS-11572 ________________________________ From: FreeSWITCH-users > on behalf of Shaun Stokes > Sent: 14 December 2018 13:16:14 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] FreeSWITCH 1.8.2 - 1-2 second dropped Audio\RTP at the start of a call Correction, we had moved FreeSWITCH 1.4 (not 1.8) to Server 1 which worked without any audio delays. Upon testing FreeSWITCH 1.8 on Server 1 there is a 1-2 second delay before RTP is established once the call is answered. This is a FreeSWITCH 1.8.2 issue, not a Debian 9 specific (also occurs on Debian 8). FreeSWITCH 1.6 and 1.4 are not effected using the same configuration through-out. ________________________________ From: Shaun Stokes Sent: 14 December 2018 11:44:18 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] FreeSWITCH 1.8.2 - 1-2 second dropped Audio\RTP at the start of a call We have built two test servers side by side on the same hardware with the same configuration, as follows. Server 1: Debian 8 with FreeSWITCH 1.6.20 Server 2: Debian 9 with FreeSWITCH 1.8.2 We can replicate the 1-2 second delay on Server 2 only, whereas Server 1 provides near instant RTP in both directions upon answer. Interestingly, if we move FreeSWITCH 1.8.2 from Server 2 to Server 1 there are still no issues with delay on Server 1, the problem is only observable on the Server 2 running Debian 9 so the problem is not specifically related to FreeSWITCH 1.8.2. At this stage it seems likely the issue lies with Debian 9 or the change in packages on Debian 9. Thanks, Shaun ________________________________ From: FreeSWITCH-users > on behalf of Shaun Stokes > Sent: 11 December 2018 15:28:33 To: FreeSWITCH Users Help Subject: [Freeswitch-users] FreeSWITCH 1.8.2 - 1-2 second dropped Audio\RTP at the start of a call Hi All, Since we've been using FreeSWITCH 1.8.2 we've noticed that the first 1-2 seconds of Audio\RTP at the start of the call when the call is answered is now dropped\missing but this doesn't occur on 1.6.20. When comparing the examples we've noticed the call flow is slightly different, as follows. FreeSWITCH 1.8.2 Leg A: switch_channel.c:2249 (sofia/internal/SRC_EXT at DOMAIN:PORT) Callstate Change DOWN -> RINGING Leg B: sofia.c:7291 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [calling][0] Leg B: sofia.c:7291 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [proceeding][180] Leg B: sofia.c:7401 Ring-Ready sofia/internal/DST_EXT at LAN_IP:PORT! Leg B: switch_channel.c:3354 (sofia/internal/DST_EXT at LAN_IP:PORT) Callstate Change DOWN -> RINGING Leg A: switch_ivr_originate.c:1246 Sending early media Leg A: switch_channel.c:3482 (sofia/internal/SRC_EXT at DOMAIN:PORT) Callstate Change RINGING -> EARLY Leg A: sofia.c:7291 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering state [early][183] Leg B: sofia.c:7291 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [completing][200] Leg B: switch_channel.c:3482 (sofia/internal/DST_EXT at LAN_IP:PORT) Callstate Change RINGING -> EARLY Leg B: sofia.c:7291 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [ready][200] Leg B: sofia.c:8429 Channel [sofia/internal/DST_EXT at LAN_IP:PORT] has been answered Leg B: switch_channel.c:3781 (sofia/internal/DST_EXT at LAN_IP:PORT) Callstate Change EARLY -> ACTIVE Leg A: switch_ivr_bridge.c:766 Channel [sofia/internal/SRC_EXT at DOMAIN:PORT] has been answered Leg A: sofia.c:7291 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering state [completed][200] Leg A: switch_channel.c:3781 (sofia/internal/SRC_EXT at DOMAIN:PORT) Callstate Change EARLY -> ACTIVE Leg A: sofia.c:7291 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering state [ready][200] Leg A: switch_ivr_async.c:1500 No silence detection configured; assuming start of speech Leg B: sofia.c:7291 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [calling][0] Leg A: sofia.c:7291 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering state [calling][0] Leg A: sofia.c:7291 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering state [ready][200] Leg A: sofia.c:8272 Processing updated SDP Leg B: sofia.c:7291 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [ready][200] FreeSWITCH 1.6.20 Leg A: switch_channel.c:2249 (sofia/internal/SRC_EXT at DOMAIN:PORT) Callstate Change DOWN -> RINGING Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [calling][0] Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [proceeding][180] Leg B: sofia.c:7192 Ring-Ready sofia/internal/DST_EXT at LAN_IP:PORT! Leg B: switch_channel.c:3346 (sofia/internal/DST_EXT at LAN_IP:PORT) Callstate Change DOWN -> RINGING Leg A: switch_ivr_originate.c:1215 Sending early media Leg A: switch_channel.c:3474 (sofia/internal/SRC_EXT at DOMAIN:PORT) Callstate Change RINGING -> EARLY Leg A: sofia.c:7084 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering state [early][183] Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [completing][200] Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [ready][200] Leg A: switch_channel.c:3773 (sofia/internal/SRC_EXT at DOMAIN:PORT) Callstate Change EARLY -> ACTIVE Leg A: sofia.c:7084 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering state [completed][200] Leg A: switch_ivr_originate.c:3705 Originate Resulted in Success: [sofia/internal/DST_EXT at LAN_IP:PORT] Leg B: switch_channel.c:3773 (sofia/internal/DST_EXT at LAN_IP:PORT) Callstate Change RINGING -> ACTIVE Leg A: sofia.c:7084 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering state [ready][200] Leg B: Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [ready][200] Leg A: switch_ivr_async.c:1500 No silence detection configured; assuming start of speech Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [calling][0] Leg A: sofia.c:7084 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering state [calling][0] Leg A: sofia.c:7084 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering state [ready][200] Leg A: sofia.c:8061 Processing updated SDP Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [completing][200] Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [ready][200] On 1.6.20 Leg B changes straight from RINGING to ACTIVE, but on 1.8.2 Leg B first changes from RINGING to EARLY then EARLY to ACTIVE, not sure if this could be related. We've experimented with the following to no avail. rtp-rewrite-timestamps send_silence_when_idle fsctl sync_clock suppress_cng ignore_early_media As per: https://freeswitch.org/confluence/display/FREESWITCH/RTP+Issues https://freeswitch.org/confluence/display/FREESWITCH/VAD+and+CNG https://freeswitch.org/confluence/display/FREESWITCH/send_silence_when_idle https://freeswitch.org/confluence/display/FREESWITCH/Early+Media The calls are local between two extensions\endpoints on the same FreeSWITCH instance and the same SIP profile, the SIP profiles on both servers (1.6.20 and 1.8.2) are identical. Does anyone have any ideas? Thanks, Shaun _________________________________________________________________________ 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 -- Anthony Minessale II Founder, FreeSWITCH. http://freeswitch.com https://youtu.be/l_hOxzCt6X4 https://www.youtube.com/watch?v=oAxXgyx5jUw https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA -------------- next part -------------- An HTML attachment was scrubbed... URL: From imfanee at gmail.com Thu Dec 20 06:42:33 2018 From: imfanee at gmail.com (Faisal Hanif) Date: Thu, 20 Dec 2018 11:42:33 +0500 Subject: [Freeswitch-users] Conference Video Mux Option doesn't display video but Video Muted (but audio not affected) instead In-Reply-To: References: Message-ID: Hi, I have created JIRA couple of days before, https://freeswitch.org/jira/browse/FS-11574 but no activity on it yet, Do I need to do something else ? Regards, Faisal On Mon, 3 Dec 2018 at 19:45, Alexey Sibyakin wrote: > That's odd. Please issue a Jira. > https://freeswitch.org/confluence/display/FREESWITCH/Reporting+Bugs+to+JIRA > > Regards, > > Alex > > On Sun, Dec 2, 2018 at 7:40 AM Faisal Hanif wrote: > >> Hi Alex, >> >> Thanks a lot for reply. I have adopted the suggestion and installed >> FreeSWITCH 1.8.2 from package on Debian 9 but I still have the same issue, >> >> FreeSWITCH Version 1.8.2-3-a98a958ac3~64bit (-3-a98a958ac3 64bit) >> Distributor ID: Debian >> Description: Debian GNU/Linux 9.6 (stretch) >> Release: 9.6 >> Codename: stretch >> >> This error I got in logs while joining the conference, >> >> 2018-12-01 06:33:43.020210 [NOTICE] switch_core_media.c:15575 Activating >> write resampler >> 2018-12-01 06:33:43.050211 [NOTICE] switch_vpx.c:486 VPX encoder reset >> (WxH/BW) from 0x0/0 to 1280x480/1024 >> 2018-12-01 06:33:43.070188 [NOTICE] switch_vpx.c:486 VPX encoder reset >> (WxH/BW) from 1280x480/1024 to 1280x480/645 >> 2018-12-01 06:33:43.070188 [ERR] switch_vpx.c:841 VPX encode error >> 8:Invalid parameter:(null) >> >> >> I really appreciate your cooperation. >> >> Regards, >> >> Faisal >> >> On Fri, 30 Nov 2018 at 16:02, Alexey Sibyakin >> wrote: >> >>> Hi, >>> >>> Try Debian 9 and FreeSWITCH 1.8.2 from official packages. If you are >>> going to use dev version on nonsupported OS you have to handle it yourself. >>> >>> Regards, >>> >>> Alex >>> >>> On Thu, Nov 29, 2018 at 1:16 AM Faisal Hanif wrote: >>> >>>> Hi Geeks, >>>> >>>> I am trying to implement a conference in mux mode and FreeSWITCH send >>>> canvas properly but never show video on but a pic "Video Muted (but audio >>>> not affected)" pic in place of every member's video on canvas. I tried a >>>> lot with no success :( >>>> >>>> OS : Ubuntu 14.04.5 LTS trusty >>>> FreeSWITCH Version 1.9.0+git~20181120T210412Z~968c76b29c~64bit (git >>>> 968c76b 2018-11-20 21:04:12Z 64bit) >>>> >>>> My conference profile is >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>> value="tone_stream://%(200,0,500,600,700)"/> >>>> >>> value="tone_stream://%(500,0,300,200,100,50,25)"/> >>>> >>>> >>>> >>>> >>>> >>> value="audio-always|livearray-json-status"/> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>> value="/usr/local/freeswitch/conf/images/video-muted.png"/> >>>> >>> value="/usr/local/freeswitch/conf/images/video-muted.png"/> >>>> >>>> >>>> >>>> can anyone please help me. >>>> Regards, >>>> >>>> Faisal >>>> >>>> >>>> _________________________________________________________________________ >>>> 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 >> >> >> >> -- >> Regards, >> >> Faisal Hanif >> _________________________________________________________________________ >> 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 -- Regards, Faisal Hanif -------------- next part -------------- An HTML attachment was scrubbed... URL: From melekoktay at gmail.com Thu Dec 20 09:05:03 2018 From: melekoktay at gmail.com (Melek Oktay) Date: Thu, 20 Dec 2018 10:05:03 +0100 Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 150, Issue 16 In-Reply-To: References: Message-ID: *the problem isnt freeswitch, but your event consumer isnt consuming the events fast enough. you should adjust your side of the equation.the event queue is what drives everything * Hi again, I think FreeSwitch core mechanism should not be blocked when user module consuming thread is not fast enough. Per-module event queue with multi-thread consumer thread mechanism could solve this issue, or something else (i don't know). Since user can add its module to the Freeswitch, s/he could not block core of freeswitch since its wrong action i think. Regards On Sun, Dec 9, 2018 at 1:31 PM < freeswitch-users-request at lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users at lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request at lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner at lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > Today's Topics: > > 1. Re: FreeSwitch blocked (Ken Rice) > > > > ---------- Forwarded message ---------- > From: Ken Rice > To: FreeSWITCH Users Help > Cc: > Bcc: > Date: Sat, 8 Dec 2018 03:03:30 -0600 > Subject: Re: [Freeswitch-users] FreeSwitch blocked > the problem isnt freeswitch, but your event consumer isnt consuming the > events fast enough. you should adjust your side of the equation. > > the event queue is what drives everything > > Sent from my iPhone > > On Dec 7, 2018, at 01:29, Melek Oktay wrote: > > Hi, > > After deeply research about this issue, we understand *Freeswitch-Core* > mechanism and three possible solutions for fixing these issue. > > Here the link > https://stackoverflow.com/questions/53609817/freeswitch-blocked > > Actually it would be be better third solution supported by *Freeswitch*, > before deliver event to module, it create new thread for not waiting > consumer thread (event hendler) > > On Fri, Dec 22, 2017 at 8:35 AM Melek Oktay wrote: > >> >> Hi, >> >> FreeSwitch software working well in a few days (~3 - 5 days), then new >> incoming call requests are accepted since FreeSwitch is blocked !! Ongoing >> calls continue their session, their calls seems not effected, but new calls >> are not accepted. I got FreeSwitch snapshot and analyzed it in GDB. >> >> I have 601 therads & most of them are waiting >> >> Thread 0x7f16bc55f700 (LWP 28544) pthread_cond_wait@@GLIBC_2.3.2 () at >> ../nptl/sysdeps/unix/sysv/linux/x86_64/pthread_cond_wait.S:185 >> >> When i apply "*thread apply all bt*" in gdb, I see most of the threads >> try to push events into queue (*switch_queue_push *) >> >> Thread 600 (Thread 0x7f16bc55f700 (LWP 28544)): >> #0 pthread_cond_wait@@GLIBC_2.3.2 () at >> ../nptl/sysdeps/unix/sysv/linux/x86_64/pthread_cond_wait.S:185 >> #1 0x00007f180cf9b87d in apr_thread_cond_wait (cond=, >> mutex=) at locks/unix/thread_cond.c:68 >> #2 0x00007f180cf92dd0 in apr_queue_push (queue=queue at entry=0x7f180db157a8, >> data=data at entry=0x7f16d3d5ec20) at misc/apr_queue.c:166 >> #3 0x00007f180cc958fb in *switch_queue_push *(queue=*0x7f180db157a8*, >> data=data at entry=*0x7f16d3d5ec20*) at src/switch_apr.c:1134 >> #4 0x00007f180cd17850 in switch_event_queue_dispatch_event >> (eventp=0x7f16bc55ec48) at src/switch_event.c:384 >> #5 switch_event_fire_detailed (file=file at entry=0x7f180cfb07ea >> "src/switch_channel.c", func=func at entry=0x7f180cfb2ba0 <__func__.18348> >> "switch_channel_perform_set_running_state", line=line at entry=2260, >> event=event at entry=0x7f16bc55ec48, user_data=user_data at entry=0x0) at >> src/switch_event.c:1986 >> #6 0x00007f180cc9f118 in switch_channel_perform_set_running_state >> (channel=0x7f17e3e7de00, state=CS_NEW, file=0x7f180cfbc590 >> "src/switch_core_state_machine.c", func=, line=543) at >> src/switch_channel.c:2260 >> #7 0x00007f180ccc87d0 in switch_core_session_run >> (session=0x7f17e3e7fd28) at src/switch_core_state_machine.c:543 >> #8 0x00007f180ccc36de in switch_core_session_thread (thread=> out>, obj=0x7f17e3e7fd28) at src/switch_core_session.c:1629 >> #9 0x00007f180ccbf47d in switch_core_session_thread_pool_worker >> (thread=0x7f17e3e9abb0, obj=0x80) at src/switch_core_session.c:1692 >> #10 0x00007f180cfa1910 in dummy_worker (opaque=0x7f17e3e9abb0) at >> threadproc/unix/thread.c:151 >> #11 0x00007f180c1e0064 in start_thread (arg=0x7f16bc55f700) at >> pthread_create.c:309 >> #12 0x00007f180b8b862d in clone () at >> ../sysdeps/unix/sysv/linux/x86_64/clone.S:111 >> >> >> More interesting thing is below, when I look up event type, approximately >> all of them are "SWITCH_EVENT_CHANNEL_STATE" and switch_queue (i think >> sofia_module queue is used in this scenario ) *become full* !!! *nelts* >> (number of elements ) and *bounds *values are equal, and there are 553 >> (full_waiters) waiters try to push , but no body try to consume it >> (empty_waiters = 0) >> >> (gdb) print *(switch_queue_t *) *0x7f180db157a8* >> $1 = { >> data = 0x7f1805cfe038, >> nelts = 50000, >> in = 43000, >> out = 43000, >> bounds = 50000, >> full_waiters = 553, >> empty_waiters = 0, >> one_big_mutex = 0x7f180db157e8, >> not_empty = 0x7f180db15838, >> not_full = 0x7f180db15890, >> terminated = 0 >> } >> >> (gdb) print *(switch_event_t *) *0x7f16d3d5ec20* >> $1 = { >> event_id = SWITCH_EVENT_CHANNEL_STATE, >> priority = SWITCH_PRIORITY_NORMAL, >> owner = 0x0, >> subclass_name = 0x0, >> headers = 0x7f16d3d5f750, >> last_header = 0x7f16d3d601d0, >> body = 0x0, >> bind_user_data = 0x0, >> event_user_data = 0x0, >> key = 0, >> next = 0x0, >> flags = 0 >> } >> >> >> Why i am gonna getting this state? >> >> Any thoughts, tips, tricks would be much appreciated. >> >> Regards, >> >> Angel >> >> _________________________________________________________________________ > 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 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From melekoktay at gmail.com Thu Dec 20 12:34:02 2018 From: melekoktay at gmail.com (Melek Oktay) Date: Thu, 20 Dec 2018 13:34:02 +0100 Subject: [Freeswitch-users] FreeSwitch blocked In-Reply-To: References: Message-ID: On Thu, Dec 20, 2018 at 10:05 AM Melek Oktay wrote: > > > *the problem isnt freeswitch, but your event consumer isnt consuming the > events fast enough. you should adjust your side of the equation.the event > queue is what drives everything * > > Hi again, > > I think FreeSwitch core mechanism should not be blocked when user module > consuming thread is not fast enough. Per-module event queue with > multi-thread consumer thread mechanism could solve this issue, or something > else (i don't know). > Since user can add its module to the Freeswitch, s/he could not block > core of freeswitch since its wrong action i think. > > Regards > > On Sun, Dec 9, 2018 at 1:31 PM < > freeswitch-users-request at lists.freeswitch.org> wrote: > >> Send FreeSWITCH-users mailing list submissions to >> freeswitch-users at lists.freeswitch.org >> >> To subscribe or unsubscribe via the World Wide Web, visit >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> or, via email, send a message with subject or body 'help' to >> freeswitch-users-request at lists.freeswitch.org >> >> You can reach the person managing the list at >> freeswitch-users-owner at lists.freeswitch.org >> >> When replying, please edit your Subject line so it is more specific >> than "Re: Contents of FreeSWITCH-users digest..." >> Today's Topics: >> >> 1. Re: FreeSwitch blocked (Ken Rice) >> >> >> >> ---------- Forwarded message ---------- >> From: Ken Rice >> To: FreeSWITCH Users Help >> Cc: >> Bcc: >> Date: Sat, 8 Dec 2018 03:03:30 -0600 >> Subject: Re: [Freeswitch-users] FreeSwitch blocked >> the problem isnt freeswitch, but your event consumer isnt consuming the >> events fast enough. you should adjust your side of the equation. >> >> the event queue is what drives everything >> >> Sent from my iPhone >> >> On Dec 7, 2018, at 01:29, Melek Oktay wrote: >> >> Hi, >> >> After deeply research about this issue, we understand *Freeswitch-Core* >> mechanism and three possible solutions for fixing these issue. >> >> Here the link >> https://stackoverflow.com/questions/53609817/freeswitch-blocked >> >> Actually it would be be better third solution supported by *Freeswitch*, >> before deliver event to module, it create new thread for not waiting >> consumer thread (event hendler) >> >> On Fri, Dec 22, 2017 at 8:35 AM Melek Oktay wrote: >> >>> >>> Hi, >>> >>> FreeSwitch software working well in a few days (~3 - 5 days), then new >>> incoming call requests are accepted since FreeSwitch is blocked !! Ongoing >>> calls continue their session, their calls seems not effected, but new calls >>> are not accepted. I got FreeSwitch snapshot and analyzed it in GDB. >>> >>> I have 601 therads & most of them are waiting >>> >>> Thread 0x7f16bc55f700 (LWP 28544) pthread_cond_wait@@GLIBC_2.3.2 () at >>> ../nptl/sysdeps/unix/sysv/linux/x86_64/pthread_cond_wait.S:185 >>> >>> When i apply "*thread apply all bt*" in gdb, I see most of the threads >>> try to push events into queue (*switch_queue_push *) >>> >>> Thread 600 (Thread 0x7f16bc55f700 (LWP 28544)): >>> #0 pthread_cond_wait@@GLIBC_2.3.2 () at >>> ../nptl/sysdeps/unix/sysv/linux/x86_64/pthread_cond_wait.S:185 >>> #1 0x00007f180cf9b87d in apr_thread_cond_wait (cond=, >>> mutex=) at locks/unix/thread_cond.c:68 >>> #2 0x00007f180cf92dd0 in apr_queue_push (queue=queue at entry=0x7f180db157a8, >>> data=data at entry=0x7f16d3d5ec20) at misc/apr_queue.c:166 >>> #3 0x00007f180cc958fb in *switch_queue_push *(queue=*0x7f180db157a8*, >>> data=data at entry=*0x7f16d3d5ec20*) at src/switch_apr.c:1134 >>> #4 0x00007f180cd17850 in switch_event_queue_dispatch_event >>> (eventp=0x7f16bc55ec48) at src/switch_event.c:384 >>> #5 switch_event_fire_detailed (file=file at entry=0x7f180cfb07ea >>> "src/switch_channel.c", func=func at entry=0x7f180cfb2ba0 <__func__.18348> >>> "switch_channel_perform_set_running_state", line=line at entry=2260, >>> event=event at entry=0x7f16bc55ec48, user_data=user_data at entry=0x0) at >>> src/switch_event.c:1986 >>> #6 0x00007f180cc9f118 in switch_channel_perform_set_running_state >>> (channel=0x7f17e3e7de00, state=CS_NEW, file=0x7f180cfbc590 >>> "src/switch_core_state_machine.c", func=, line=543) at >>> src/switch_channel.c:2260 >>> #7 0x00007f180ccc87d0 in switch_core_session_run >>> (session=0x7f17e3e7fd28) at src/switch_core_state_machine.c:543 >>> #8 0x00007f180ccc36de in switch_core_session_thread (thread=>> out>, obj=0x7f17e3e7fd28) at src/switch_core_session.c:1629 >>> #9 0x00007f180ccbf47d in switch_core_session_thread_pool_worker >>> (thread=0x7f17e3e9abb0, obj=0x80) at src/switch_core_session.c:1692 >>> #10 0x00007f180cfa1910 in dummy_worker (opaque=0x7f17e3e9abb0) at >>> threadproc/unix/thread.c:151 >>> #11 0x00007f180c1e0064 in start_thread (arg=0x7f16bc55f700) at >>> pthread_create.c:309 >>> #12 0x00007f180b8b862d in clone () at >>> ../sysdeps/unix/sysv/linux/x86_64/clone.S:111 >>> >>> >>> More interesting thing is below, when I look up event type, >>> approximately all of them are "SWITCH_EVENT_CHANNEL_STATE" and >>> switch_queue (i think sofia_module queue is used in this scenario ) *become >>> full* !!! *nelts* (number of elements ) and *bounds *values are equal, >>> and there are 553 (full_waiters) waiters try to push , but no body try to >>> consume it (empty_waiters = 0) >>> >>> (gdb) print *(switch_queue_t *) *0x7f180db157a8* >>> $1 = { >>> data = 0x7f1805cfe038, >>> nelts = 50000, >>> in = 43000, >>> out = 43000, >>> bounds = 50000, >>> full_waiters = 553, >>> empty_waiters = 0, >>> one_big_mutex = 0x7f180db157e8, >>> not_empty = 0x7f180db15838, >>> not_full = 0x7f180db15890, >>> terminated = 0 >>> } >>> >>> (gdb) print *(switch_event_t *) *0x7f16d3d5ec20* >>> $1 = { >>> event_id = SWITCH_EVENT_CHANNEL_STATE, >>> priority = SWITCH_PRIORITY_NORMAL, >>> owner = 0x0, >>> subclass_name = 0x0, >>> headers = 0x7f16d3d5f750, >>> last_header = 0x7f16d3d601d0, >>> body = 0x0, >>> bind_user_data = 0x0, >>> event_user_data = 0x0, >>> key = 0, >>> next = 0x0, >>> flags = 0 >>> } >>> >>> >>> Why i am gonna getting this state? >>> >>> Any thoughts, tips, tricks would be much appreciated. >>> >>> Regards, >>> >>> Angel >>> >>> _________________________________________________________________________ >> 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 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From shaun.stokes at itec-support.co.uk Thu Dec 20 10:46:28 2018 From: shaun.stokes at itec-support.co.uk (Shaun Stokes) Date: Thu, 20 Dec 2018 10:46:28 +0000 Subject: [Freeswitch-users] Relay a REGISTER In-Reply-To: <000f01d496c0$454fc550$cfef4ff0$@gmail.com> References: <000f01d496c0$454fc550$cfef4ff0$@gmail.com> Message-ID: If you want to achieve this using FreeSWITCH you would likely need to modify the source-code and\or create a new module to control this functionality, it wouldn't be a simple task. You're much better off using a SIP Proxy such as Kamailio which allows you to control the flow of the SIP packets your-self through the configuration, although it can be a bit of a learning curve for those not familiar with Kamailio. Shaun ________________________________ From: FreeSWITCH-users on behalf of igor.potjevlesch at gmail.com Sent: 18 December 2018 10:56:03 To: 'FreeSWITCH Users Help' Subject: [Freeswitch-users] Relay a REGISTER Hello, I'd like to relay a REGISTER received on leg A to the leg B. The idea is to have a Contact URI with the B leg IP address when the REGISTER goes out of the FS. But I didn't found how to avoid Freeswitch to handle the REGISTER locally. Regards, Igor. -------------- next part -------------- An HTML attachment was scrubbed... URL: From filipe at redseven.tech Thu Dec 20 10:38:58 2018 From: filipe at redseven.tech (Filipe Cunha) Date: Thu, 20 Dec 2018 10:38:58 +0000 Subject: [Freeswitch-users] FS 1.8 Docker image In-Reply-To: <13192151545070920@myt6-7734411c649e.qloud-c.yandex.net> References: <13192151545070920@myt6-7734411c649e.qloud-c.yandex.net> Message-ID: I end up downloading safarov/freeswitch:1.8.2 image, installed my custom configuration and everything is working fine. Thanks for the help guys. On Mon, Dec 17, 2018 at 9:01 PM Serge S. Yuriev wrote: > Hi > > I'm afraid your pastebin no more available. > Returned 404 > > -- > Wbr, Serge via mobile > > 17.12.2018, 17:55, "Mitch Capper" : > > I run FS in docker. It is also handy for filing bug reports with exact > repo so I have some dockerfiles up for example: > https://freeswitch.org/jira/browse/FS-7833 > full source build. > > > ~mitch > > > On Fri, Dec 14, 2018 at 1:23 AM Alexey Sibyakin > wrote: > > Hi, > > Try to install dbus into your Stretch. > > Regards, > > Alex > > On Thu, Dec 13, 2018 at 3:40 AM Sergey Safarov > wrote: > > you can use safarov/freeswitch:1.8.2 > image > repo of this is located here > https://bitbucket.org/sergey-safarov/freeswitch/commits/branch/v1.8 > > > ср, 12 дек. 2018 г. в 18:14, Filipe Cunha : > > Hi all, > > I'm trying to create a docker image with the latest version of FreeSWITCH. > But I'm not able to do it. These are the steps that I did to try to do it. > > 1. Download debian stretch image > *docker pull debian:stretch * > 2. Build FreeSWITCH from source code following the guide > > in confluence step by step. I follow the production version > 3. The step 2 went very well, and then I follow the post-installation > instructions > > and this was when it stop working. I set up owners and permissions and when > I try to do the systemd I was not able to do it. I copied the file *cp > /usr/src/freeswitch/debian/freeswitch-systemd.freeswitch.service > /etc/systemd/system/freeswitch.service *the content of the file was not > exactly the same as the one in the tutorial, so I copied the content of the > file on the tutorial and replace it. And when I tried to do *systemctl > deamon-reload* I got this error message, *Failed to connect to bus: No > such file or directory*. After trying to find a solution online I found > this answer on SO > > > Do you guys know if there is a solution for this problem? > We plan to use the latest version of FreeSwitch in our production > environment, but we have a normal linux server to do it, and the devops > team got everything working fine, we only have this problem when we try to > use docker. > I'm trying to use docker so my team could test everything locally before > pushing something to the test environment, so far we have been deploying > our code to test environment to test our code 😖 > I found this image > that works fine in our local environment, but the version is 1.6.16. > Is there any braking change between 1.8.2 and 1.6.16? My concern it that > we start developing against an older version, and when we deploy to > production it doesn't work, or maybe the latest version have a super useful > command that we are not taking advantage of. > We control FreeSWITCH mainly by events, we use java to do it. > > Thanks for the help, > Filipe Cunha > > > > > _________________________________________________________________________ > 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. > > _________________________________________________________________________ > 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 -- RED SEVEN TECHNOLOGY GROUP Filipe Cunha Software Engineer (+44)07753990966 filipe at redseven.tech -------------- next part -------------- An HTML attachment was scrubbed... URL: From covici at ccs.covici.com Thu Dec 20 12:22:00 2018 From: covici at ccs.covici.com (John Covici) Date: Thu, 20 Dec 2018 07:22:00 -0500 Subject: [Freeswitch-users] FS 1.8 Docker image In-Reply-To: References: <13192151545070920@myt6-7734411c649e.qloud-c.yandex.net> Message-ID: I can't get into the container once I download the image, I tried to exec it with /bin/bash, but that seems not to exist. I wanted to start ssh, so I could copy some files in there and maybe make other changes, but no joy. On Thu, 20 Dec 2018 05:38:58 -0500, Filipe Cunha wrote: > > [1 ] > [1.1 ] > [1.2 ] > I end up downloading safarov/freeswitch:1.8.2 image, installed my custom configuration and everything is working fine. > Thanks for the help guys. > > On Mon, Dec 17, 2018 at 9:01 PM Serge S. Yuriev wrote: > > Hi > > I'm afraid your pastebin no more available. > Returned 404 > > -- > Wbr, Serge via mobile > > 17.12.2018, 17:55, "Mitch Capper" : > > I run FS in docker. It is also handy for filing bug reports with exact repo so I have some dockerfiles up for example: https://freeswitch.org/jira/browse/FS-7833 > full source build. > > ~mitch > > On Fri, Dec 14, 2018 at 1:23 AM Alexey Sibyakin wrote: > > Hi, > > Try to install dbus into your Stretch. > > Regards, > > Alex > > On Thu, Dec 13, 2018 at 3:40 AM Sergey Safarov wrote: > > you can use safarov/freeswitch:1.8.2 image > repo of this is located here https://bitbucket.org/sergey-safarov/freeswitch/commits/branch/v1.8 > > ср, 12 дек. 2018 г. в 18:14, Filipe Cunha : > > Hi all, > > I'm trying to create a docker image with the latest version of FreeSWITCH. But I'm not able to do it. These are the steps that I did to try to do it. > > 1. Download debian stretch image docker pull debian:stretch > 2. Build FreeSWITCH from source code following the guide in confluence step by step. I follow the production version > 3. The step 2 went very well, and then I follow the post-installation instructions and this was when it stop working. I set up owners and permissions and when I try to do the systemd I was not able to do it. I copied > the file cp /usr/src/freeswitch/debian/freeswitch-systemd.freeswitch.service /etc/systemd/system/freeswitch.service the content of the file was not exactly the same as the one in the tutorial, so I copied the content > of the file on the tutorial and replace it. And when I tried to do systemctl deamon-reload I got this error message, Failed to connect to bus: No such file or directory. After trying to find a solution online I found > this answer on SO > > Do you guys know if there is a solution for this problem? > We plan to use the latest version of FreeSwitch in our production environment, but we have a normal linux server to do it, and the devops team got everything working fine, we only have this problem when we try to use > docker. > I'm trying to use docker so my team could test everything locally before pushing something to the test environment, so far we have been deploying our code to test environment to test our code 😖 > I found this image that works fine in our local environment, but the version is 1.6.16. > Is there any braking change between 1.8.2 and 1.6.16? My concern it that we start developing against an older version, and when we deploy to production it doesn't work, or maybe the latest version have a super useful > command that we are not taking advantage of. > We control FreeSWITCH mainly by events, we use java to do it. > > Thanks for the help, > Filipe Cunha > > > _________________________________________________________________________ > 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. > > _________________________________________________________________________ > 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 > > -- > > * RED SEVEN TECHNOLOGY GROUP > > Filipe Cunha > > Software Engineer > > * (+44)07753990966 > > * filipe at redseven.tech > * > * > * > [2 ] > _________________________________________________________________________ > 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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici wb2una covici at ccs.covici.com From mjlopez at smartic.es Thu Dec 20 12:34:26 2018 From: mjlopez at smartic.es (=?UTF-8?Q?Miguel_Jes=C3=BAs_L=C3=B3pez_Valverde?=) Date: Thu, 20 Dec 2018 13:34:26 +0100 Subject: [Freeswitch-users] usernames whith special simbols cant recive calls on FS 1.8 In-Reply-To: <8D79C314-B2F2-4CE9-A854-C4AEE2F4DFE5@jerris.com> References: <00c001d4778c$67f9add0$37ed0970$@smartic.es> <8D79C314-B2F2-4CE9-A854-C4AEE2F4DFE5@jerris.com> Message-ID: <006e01d49860$593f73c0$0bbe5b40$@smartic.es> Hi all As you indicated to me through Slak and in this email, I keep periodic inquiries through Slak about the status of this incident, but I do not get answer in any way after the last one given in this email. I have also consulted about the following: Is it possible to recompile FS18 in its version 1.8.2 on my own, eliminating the burden of the patch that generates this problem or should I wait for a new version 1.8.3 with it corrected ?. In case it can be recompiled by me, could you tell me in a basic way how to do it? But I'm also not having answers through Slak. For this I return to send you these queries via email, as it is important for me to find a solution to this issue to be able to update the current FS1.6 servers that I have available to FS1.8 Thank you again and greetings. Miguel J. López. De: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] En nombre de Michael Jerris Enviado el: jueves, 29 de noviembre de 2018 21:46 Para: FreeSWITCH Users Help Asunto: Re: [Freeswitch-users] usernames whith special simbols cant recive calls on FS 1.8 This issue is cause by the patch adding in: https://freeswitch.org/jira/browse/FS-9791 And is specific to calling sip registered endpoints with ~ in the username. I’ve reached out to the original author of that patch, and if they don’t respond with a fix I’ll have to revert that patch. Please remind me if I dont follow up on that issue soon by commenting on that jira. On Nov 8, 2018, at 12:56 PM, Miguel Jesús López Valverde > wrote: Hello again guys: I'm sending this question in case someone can tell me how to solve the following problem. We currently have a production platform working with FS 1.6 where users have historically been created in the form username~companyname like usernames and the incoming calls to these users work correctly. Also, a request on console like “sofia_contact nameusername~companyname at domain” returns a right result. Now I am considering the migration of these platforms to FS 1.8 and, for this, doing an assembly of initial development. Here I have been able to verify that although these users register correctly, they can issue outgoing calls and appear correctly listed in queries of type “sofia status profile internal reg”, but these users can’t receive incoming calls, in console the information " Cannot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED] " is always obtained and when I launch the query “sofia_contact nameusername~companyname at domain” in console, I get the result “error/user_not_registered”; I only have to change the value ~ for example by a dot and the incoming calls works correctly and also the query “sofia_contact nameusername.companyname at domain ” return a right result. I guess I need to adjust the configuration of some library of FS 1.8 and recompile it to get back an identical operation to version 1.6, can someone help me in indicating which library or with what adjustment I could get this operation again? Thank you very much and best regards!! --- El software de antivirus Avast ha analizado este correo electrónico en busca de virus. https://www.avast.com/antivirus -------------- next part -------------- An HTML attachment was scrubbed... URL: From filipe at redseven.tech Thu Dec 20 15:16:03 2018 From: filipe at redseven.tech (Filipe Cunha) Date: Thu, 20 Dec 2018 15:16:03 +0000 Subject: [Freeswitch-users] FS 1.8 Docker image In-Reply-To: References: <13192151545070920@myt6-7734411c649e.qloud-c.yandex.net> Message-ID: This is what I did. *docker pull safarov/freeswitch* *docker volume create --name freeswitch-soundsdocker run --net=host --name freeswitch -e SOUND_RATES=8000:16000 -e SOUND_TYPES=music:en-us-callie -v freeswitch-sounds:/usr/share/freeswitch/sounds -v /etc/freeswitch/:/etc/freeswitch safarov/freeswitch* *docker exec -it freeswitch /bin/sh* *cd usr/bin/* *and you get inside the correct folder On Thu, Dec 20, 2018 at 2:43 PM John Covici wrote: > I can't get into the container once I download the image, I tried to > exec it with /bin/bash, but that seems not to exist. I wanted to > start ssh, so I could copy some files in there and maybe make other > changes, but no joy. > > On Thu, 20 Dec 2018 05:38:58 -0500, > Filipe Cunha wrote: > > > > [1 ] > > [1.1 ] > > [1.2 ] > > I end up downloading safarov/freeswitch:1.8.2 image, installed my custom > configuration and everything is working fine. > > Thanks for the help guys. > > > > On Mon, Dec 17, 2018 at 9:01 PM Serge S. Yuriev wrote: > > > > Hi > > > > I'm afraid your pastebin no more available. > > Returned 404 > > > > -- > > Wbr, Serge via mobile > > > > 17.12.2018, 17:55, "Mitch Capper" : > > > > I run FS in docker. It is also handy for filing bug reports with > exact repo so I have some dockerfiles up for example: > https://freeswitch.org/jira/browse/FS-7833 > > full source build. > > > > ~mitch > > > > On Fri, Dec 14, 2018 at 1:23 AM Alexey Sibyakin > wrote: > > > > Hi, > > > > Try to install dbus into your Stretch. > > > > Regards, > > > > Alex > > > > On Thu, Dec 13, 2018 at 3:40 AM Sergey Safarov > wrote: > > > > you can use safarov/freeswitch:1.8.2 image > > repo of this is located here > https://bitbucket.org/sergey-safarov/freeswitch/commits/branch/v1.8 > > > > ср, 12 дек. 2018 г. в 18:14, Filipe Cunha : > > > > Hi all, > > > > I'm trying to create a docker image with the latest version of > FreeSWITCH. But I'm not able to do it. These are the steps that I did to > try to do it. > > > > 1. Download debian stretch image docker pull debian:stretch > > 2. Build FreeSWITCH from source code following the guide in confluence > step by step. I follow the production version > > 3. The step 2 went very well, and then I follow the post-installation > instructions and this was when it stop working. I set up owners and > permissions and when I try to do the systemd I was not able to do it. I > copied > > the file cp > /usr/src/freeswitch/debian/freeswitch-systemd.freeswitch.service > /etc/systemd/system/freeswitch.service the content of the file was not > exactly the same as the one in the tutorial, so I copied the content > > of the file on the tutorial and replace it. And when I tried to do > systemctl deamon-reload I got this error message, Failed to connect to bus: > No such file or directory. After trying to find a solution online I found > > this answer on SO > > > > Do you guys know if there is a solution for this problem? > > We plan to use the latest version of FreeSwitch in our production > environment, but we have a normal linux server to do it, and the devops > team got everything working fine, we only have this problem when we try to > use > > docker. > > I'm trying to use docker so my team could test everything locally > before pushing something to the test environment, so far we have been > deploying our code to test environment to test our code 😖 > > I found this image that works fine in our local environment, but the > version is 1.6.16. > > Is there any braking change between 1.8.2 and 1.6.16? My concern it > that we start developing against an older version, and when we deploy to > production it doesn't work, or maybe the latest version have a super useful > > command that we are not taking advantage of. > > We control FreeSWITCH mainly by events, we use java to do it. > > > > Thanks for the help, > > Filipe Cunha > > > > > > > _________________________________________________________________________ > > 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. > > > > > _________________________________________________________________________ > > 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 > > > > -- > > > > * RED SEVEN TECHNOLOGY GROUP > > > > Filipe Cunha > > > > Software Engineer > > > > * (+44)07753990966 > > > > * filipe at redseven.tech > > * > > * > > * > > [2 ] > > _________________________________________________________________________ > > 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 > > -- > Your life is like a penny. You're going to lose it. The question is: > How do > you spend it? > > John Covici wb2una > covici at ccs.covici.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 -- RED SEVEN TECHNOLOGY GROUP Filipe Cunha Software Engineer (+44)07753990966 filipe at redseven.tech -------------- next part -------------- An HTML attachment was scrubbed... URL: From joel at textplus.com Thu Dec 20 15:21:33 2018 From: joel at textplus.com (Joel Serrano) Date: Thu, 20 Dec 2018 07:21:33 -0800 Subject: [Freeswitch-users] FreeSWITCH 1.8.2 - 1-2 second dropped Audio\RTP at the start of a call In-Reply-To: References: <16D901F8-1A94-4FF3-AAC9-BF37E073BEE4@gmail.com> <97EE2A01-D430-421A-90B6-7CFA71DAF34B@gmail.com> Message-ID: Hi Shaun, Can you share with the community what you found out? I’m curious what can be the cause for that extra 2-3s delay in RTP On Thu, Dec 20, 2018 at 01:52 Shaun Stokes wrote: > We've identified the cause and have now implemented a fix, this was a > local issue caused by a bug in our build script which applies tweaks to > mod_sofia. > > > Thanks, > > Shaun > ------------------------------ > *From:* FreeSWITCH-users > on behalf of Shaun Stokes > *Sent:* 19 December 2018 14:07:18 > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] FreeSWITCH 1.8.2 - 1-2 second dropped > Audio\RTP at the start of a call > > > Thank you all for your recommendations. We raised a JIRA but it was closed > as incomplete, we will look at re-opening once we've compiled the relevant > logs and pcaps etc unless we reach a conclusion before hand. > > > Some additional points: > > - We do not set "answer_delay". > > - There is still a 1-2 second delay after > setting "disable_rtp_auto_adjust" on the SIP profile. > > > We've identified this as a package issue, when FreeSWITCH is built with > certain packages installed this is causing mod_sofia.so to be much smaller > than it should be (5.49MB instead of 8.58MB). > > > We're able to consistently reproduce the following when running > FreeSWITCH on the same Debian 9 server using the same FreeSWITCH configuration > and scripts etc: > > - FreeSWITCH 1.8.2 built on Debian 9 - 1-2 second audio\RTP delay on > answer > > - FreeSWITCH 1.6.20 built on Debian 9 - 1-2 second audio\RTP delay on > answer > > - FreeSWITCH 1.6.20 built on Debian 8 (moved binaries to Debian 9 server) > - No audio\RTP delay on answer (audio\RTP is instantaneous) > > - FreeSWITCH 1.8.2 built on Debian 9 base (no additional packages) - No > audio\RTP delay on answer (audio\RTP is instantaneous) > > > Will post back once we've identified the root cause. > > > Thanks, > > Shaun > ------------------------------ > *From:* FreeSWITCH-users > on behalf of Anthony Minessale > *Sent:* 17 December 2018 20:55 > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] FreeSWITCH 1.8.2 - 1-2 second dropped > Audio\RTP at the start of a call > > Sounds like a broken record but the mailing list is not the best place to > report issues. Almost inevitably, questions will be asked and data will > need to be collected like logs etc. > This is why we already point to JIRA to file tickets. > > Recommendations: > > 1) Make sure you don't have "answer_delay" set. > 2) Check the media signaling data to make sure you are not using > "rtp-auto-adjust which adds a time to media establishment to correct for > incorrect media IPs. > 3) Get a pcap as well as debug + sip trace before reporting any issue > because its always going to be the first request anyway. > 4) Use JIRA, feel free to ask about the JIRA here but don't rely on 1990's > listserv server to track the issue progress. > > sofia global siptrace on > console loglevel debug > fsctl debug_level 10 > > > > > On Mon, Dec 17, 2018 at 2:48 PM Shaun Stokes < > shaun.stokes at itec-support.co.uk> wrote: > > We reverted back to FreeSWITCH 1.6.20 but when this is compiled on > the Debian 9 server the problem still occurs. > > > We had to workaround some build errors for FS 1.6.20 to compile on Debian > 9 with PostgreSQL 11 but the problem was still present, as follows. > > -------------------- > > # Uninstall 1.1.0 SSL header files (libssl-dev) and install the older ones > (libssl1.0-dev). > apt-get install libssl1.0-dev > > # Fix PGSQL 11 Support > In the file: > /usr/src/freeswitch/srcswitch_pgsql.c > On line 389, replace this: > #if POSTGRESQL_MAJOR_VERSION >= 9 && POSTGRESQL_MINOR_VERSION >= 2 > With: > #if (POSTGRESQL_MAJOR_VERSION == 9 && POSTGRESQL_MINOR_VERSION >= 2) || > POSTGRESQL_MAJOR_VERSION > 9 > > # Do not build mod_flite or mod_enum > sed -i /usr/src/freeswitch/modules.conf -e > s:'asr_tts/mod_flite:#asr_tts/mod_flite:' > sed -i /usr/src/freeswitch/modules.conf -e > s:'applications/mod_enum:#applications/mod_enum:' > -------------------- > > We took the FS 1.6.20 binaries (pre-compiled) from a Debian 8 server and > restored them to our Debian 9 server which resolved the issue but we had to > copy some missing libs from a Debian 8 server: > /usr/lib/x86_64-linux-gnu/libssl.so.1.0.0 > /usr/lib/x86_64-linux-gnu/libcrypto.so.1.0.0 > > Given that the problem changes when the source-code is compiled on > different servers we suspect this may be a package problem not specific to > FreeSWITCH. > > This is also a problem on master, raised JIRA: > https://freeswitch.org/jira/browse/FS-11572 > > ------------------------------ > *From:* FreeSWITCH-users > on behalf of Shaun Stokes > *Sent:* 14 December 2018 13:16:14 > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] FreeSWITCH 1.8.2 - 1-2 second dropped > Audio\RTP at the start of a call > > Correction, we had moved FreeSWITCH 1.4 (not 1.8) to Server 1 which worked > without any audio delays. Upon testing FreeSWITCH 1.8 on Server 1 there is > a 1-2 second delay before RTP is established once the call is answered. > > This is a FreeSWITCH 1.8.2 issue, not a Debian 9 specific (also occurs on > Debian 8). FreeSWITCH 1.6 and 1.4 are not effected using the same > configuration through-out. > ------------------------------ > *From:* Shaun Stokes > *Sent:* 14 December 2018 11:44:18 > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] FreeSWITCH 1.8.2 - 1-2 second dropped > Audio\RTP at the start of a call > > We have built two test servers side by side on the same hardware with the > same configuration, as follows. > Server 1: Debian 8 with FreeSWITCH 1.6.20 > Server 2: Debian 9 with FreeSWITCH 1.8.2 > > We can replicate the 1-2 second delay on Server 2 only, whereas Server 1 > provides near instant RTP in both directions upon answer. Interestingly, if > we move FreeSWITCH 1.8.2 from Server 2 to Server 1 there are still no > issues with delay on Server 1, the problem is only observable on the Server > 2 running Debian 9 so the problem is not specifically related to FreeSWITCH > 1.8.2. > > At this stage it seems likely the issue lies with Debian 9 or the change > in packages on Debian 9. > > Thanks, > Shaun > ------------------------------ > *From:* FreeSWITCH-users > on behalf of Shaun Stokes > *Sent:* 11 December 2018 15:28:33 > *To:* FreeSWITCH Users Help > *Subject:* [Freeswitch-users] FreeSWITCH 1.8.2 - 1-2 second dropped > Audio\RTP at the start of a call > > > Hi All, > > > Since we've been using FreeSWITCH 1.8.2 we've noticed that the first 1-2 > seconds of Audio\RTP at the start of the call when the call is answered is > now dropped\missing but this doesn't occur on 1.6.20. When comparing the > examples we've noticed the call flow is slightly different, as follows. > > > FreeSWITCH 1.8.2 > > Leg A: switch_channel.c:2249 (sofia/internal/SRC_EXT at DOMAIN:PORT) > Callstate Change DOWN -> RINGING > Leg B: sofia.c:7291 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering > state [calling][0] > Leg B: sofia.c:7291 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering > state [proceeding][180] > Leg B: sofia.c:7401 Ring-Ready sofia/internal/DST_EXT at LAN_IP:PORT! > Leg B: switch_channel.c:3354 (sofia/internal/DST_EXT at LAN_IP:PORT) > Callstate Change DOWN -> RINGING > Leg A: switch_ivr_originate.c:1246 Sending early media > Leg A: switch_channel.c:3482 (sofia/internal/SRC_EXT at DOMAIN:PORT) > Callstate Change RINGING -> EARLY > Leg A: sofia.c:7291 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering > state [early][183] > Leg B: sofia.c:7291 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering > state [completing][200] > Leg B: switch_channel.c:3482 (sofia/internal/DST_EXT at LAN_IP:PORT) > Callstate Change RINGING -> EARLY > Leg B: sofia.c:7291 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering > state [ready][200] > Leg B: sofia.c:8429 Channel [sofia/internal/DST_EXT at LAN_IP:PORT] has been > answered > Leg B: switch_channel.c:3781 (sofia/internal/DST_EXT at LAN_IP:PORT) > Callstate Change EARLY -> ACTIVE > Leg A: switch_ivr_bridge.c:766 Channel [sofia/internal/SRC_EXT at DOMAIN:PORT] > has been answered > Leg A: sofia.c:7291 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering > state [completed][200] > Leg A: switch_channel.c:3781 (sofia/internal/SRC_EXT at DOMAIN:PORT) > Callstate Change EARLY -> ACTIVE > Leg A: sofia.c:7291 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering > state [ready][200] > Leg A: switch_ivr_async.c:1500 No silence detection configured; assuming > start of speech > Leg B: sofia.c:7291 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering > state [calling][0] > Leg A: sofia.c:7291 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering > state [calling][0] > Leg A: sofia.c:7291 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering > state [ready][200] > Leg A: sofia.c:8272 Processing updated SDP > Leg B: sofia.c:7291 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering > state [ready][200] > > > FreeSWITCH 1.6.20 > > Leg A: switch_channel.c:2249 (sofia/internal/SRC_EXT at DOMAIN:PORT) > Callstate Change DOWN -> RINGING > Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering > state [calling][0] > Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering > state [proceeding][180] > Leg B: sofia.c:7192 Ring-Ready sofia/internal/DST_EXT at LAN_IP:PORT! > Leg B: switch_channel.c:3346 (sofia/internal/DST_EXT at LAN_IP:PORT) > Callstate Change DOWN -> RINGING > Leg A: switch_ivr_originate.c:1215 Sending early media > Leg A: switch_channel.c:3474 (sofia/internal/SRC_EXT at DOMAIN:PORT) > Callstate Change RINGING -> EARLY > Leg A: sofia.c:7084 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering > state [early][183] > Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering > state [completing][200] > Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering > state [ready][200] > Leg A: switch_channel.c:3773 (sofia/internal/SRC_EXT at DOMAIN:PORT) > Callstate Change EARLY -> ACTIVE > Leg A: sofia.c:7084 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering > state [completed][200] > Leg A: switch_ivr_originate.c:3705 Originate Resulted in Success: > [sofia/internal/DST_EXT at LAN_IP:PORT] > Leg B: switch_channel.c:3773 (sofia/internal/DST_EXT at LAN_IP:PORT) > Callstate Change RINGING -> ACTIVE > Leg A: sofia.c:7084 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering > state [ready][200] > Leg B: Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state > [ready][200] > Leg A: switch_ivr_async.c:1500 No silence detection configured; assuming > start of speech > Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering > state [calling][0] > Leg A: sofia.c:7084 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering > state [calling][0] > Leg A: sofia.c:7084 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering > state [ready][200] > Leg A: sofia.c:8061 Processing updated SDP > Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering > state [completing][200] > Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering > state [ready][200] > > > On 1.6.20 Leg B changes straight from RINGING to ACTIVE, but on 1.8.2 Leg > B first changes from RINGING to EARLY then EARLY to ACTIVE, not sure if > this could be related. > > > We've experimented with the following to no avail. > rtp-rewrite-timestamps > send_silence_when_idle > fsctl sync_clock > suppress_cng > ignore_early_media > > As per: > https://freeswitch.org/confluence/display/FREESWITCH/RTP+Issues > https://freeswitch.org/confluence/display/FREESWITCH/VAD+and+CNG > https://freeswitch.org/confluence/display/FREESWITCH/send_silence_when_idle > https://freeswitch.org/confluence/display/FREESWITCH/Early+Media > > > The calls are local between two extensions\endpoints on the same > FreeSWITCH instance and the same SIP profile, the SIP profiles on both > servers (1.6.20 and 1.8.2) are identical. > > > Does anyone have any ideas? > > > Thanks, > > Shaun > _________________________________________________________________________ > 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 > > > > -- > Anthony Minessale II > Founder, FreeSWITCH. > http://freeswitch.com > > > https://youtu.be/l_hOxzCt6X4 > https://www.youtube.com/watch?v=oAxXgyx5jUw > https://www.youtube.com/watch?v=9XXgW34t40s > https://www.youtube.com/watch?v=NLaDpGQuZDA > _________________________________________________________________________ > 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 s.safarov at gmail.com Thu Dec 20 16:56:41 2018 From: s.safarov at gmail.com (Sergey Safarov) Date: Thu, 20 Dec 2018 19:56:41 +0300 Subject: [Freeswitch-users] FS 1.8 Docker image In-Reply-To: References: <13192151545070920@myt6-7734411c649e.qloud-c.yandex.net> Message-ID: image is not have bash but exist "sh" (busybox). to copy files you can use docker command docker cp Or mount required folders into image like docker run -d -v /var/lib/my_service_folder:/var/lib/my_service_folder safarov/freeswitch:1.8.2 чт, 20 дек. 2018 г. в 18:07, John Covici : > I can't get into the container once I download the image, I tried to > exec it with /bin/bash, but that seems not to exist. I wanted to > start ssh, so I could copy some files in there and maybe make other > changes, but no joy. > > On Thu, 20 Dec 2018 05:38:58 -0500, > Filipe Cunha wrote: > > > > [1 ] > > [1.1 ] > > [1.2 ] > > I end up downloading safarov/freeswitch:1.8.2 image, installed my custom > configuration and everything is working fine. > > Thanks for the help guys. > > > > On Mon, Dec 17, 2018 at 9:01 PM Serge S. Yuriev wrote: > > > > Hi > > > > I'm afraid your pastebin no more available. > > Returned 404 > > > > -- > > Wbr, Serge via mobile > > > > 17.12.2018, 17:55, "Mitch Capper" : > > > > I run FS in docker. It is also handy for filing bug reports with > exact repo so I have some dockerfiles up for example: > https://freeswitch.org/jira/browse/FS-7833 > > full source build. > > > > ~mitch > > > > On Fri, Dec 14, 2018 at 1:23 AM Alexey Sibyakin > wrote: > > > > Hi, > > > > Try to install dbus into your Stretch. > > > > Regards, > > > > Alex > > > > On Thu, Dec 13, 2018 at 3:40 AM Sergey Safarov > wrote: > > > > you can use safarov/freeswitch:1.8.2 image > > repo of this is located here > https://bitbucket.org/sergey-safarov/freeswitch/commits/branch/v1.8 > > > > ср, 12 дек. 2018 г. в 18:14, Filipe Cunha : > > > > Hi all, > > > > I'm trying to create a docker image with the latest version of > FreeSWITCH. But I'm not able to do it. These are the steps that I did to > try to do it. > > > > 1. Download debian stretch image docker pull debian:stretch > > 2. Build FreeSWITCH from source code following the guide in confluence > step by step. I follow the production version > > 3. The step 2 went very well, and then I follow the post-installation > instructions and this was when it stop working. I set up owners and > permissions and when I try to do the systemd I was not able to do it. I > copied > > the file cp > /usr/src/freeswitch/debian/freeswitch-systemd.freeswitch.service > /etc/systemd/system/freeswitch.service the content of the file was not > exactly the same as the one in the tutorial, so I copied the content > > of the file on the tutorial and replace it. And when I tried to do > systemctl deamon-reload I got this error message, Failed to connect to bus: > No such file or directory. After trying to find a solution online I found > > this answer on SO > > > > Do you guys know if there is a solution for this problem? > > We plan to use the latest version of FreeSwitch in our production > environment, but we have a normal linux server to do it, and the devops > team got everything working fine, we only have this problem when we try to > use > > docker. > > I'm trying to use docker so my team could test everything locally > before pushing something to the test environment, so far we have been > deploying our code to test environment to test our code 😖 > > I found this image that works fine in our local environment, but the > version is 1.6.16. > > Is there any braking change between 1.8.2 and 1.6.16? My concern it > that we start developing against an older version, and when we deploy to > production it doesn't work, or maybe the latest version have a super useful > > command that we are not taking advantage of. > > We control FreeSWITCH mainly by events, we use java to do it. > > > > Thanks for the help, > > Filipe Cunha > > > > > > > _________________________________________________________________________ > > 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. > > > > > _________________________________________________________________________ > > 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 > > > > -- > > > > * RED SEVEN TECHNOLOGY GROUP > > > > Filipe Cunha > > > > Software Engineer > > > > * (+44)07753990966 <+44%207753%20990966> > > > > * filipe at redseven.tech > > * > > * > > * > > [2 ] > > _________________________________________________________________________ > > 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 > > -- > Your life is like a penny. You're going to lose it. The question is: > How do > you spend it? > > John Covici wb2una > covici at ccs.covici.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 shaun.stokes at itec-support.co.uk Thu Dec 20 17:45:44 2018 From: shaun.stokes at itec-support.co.uk (Shaun Stokes) Date: Thu, 20 Dec 2018 17:45:44 +0000 Subject: [Freeswitch-users] FreeSWITCH 1.8.2 - 1-2 second dropped Audio\RTP at the start of a call In-Reply-To: References: <16D901F8-1A94-4FF3-AAC9-BF37E073BEE4@gmail.com> <97EE2A01-D430-421A-90B6-7CFA71DAF34B@gmail.com> , Message-ID: Hi Joel, This was caused by a 1 second sleep timer in mod_sofia.c which should have only been applied under certain conditions. sed -i /usr/src/freeswitch/src/mod/endpoints/mod_sofia/mod_sofia.c -z -e s~'\d9\d9\d9\d9\d9\d105\d102\d32\d40\d122\d115\d116\d114\d40\d116\d101\d99\d104\d95\d112\d118\d116\d45\d62\d108\d97\d115\d116\d95\d115\d101\d110\d116\d95\d99\d97\d108\d108\d101\d101\d95\d105\d100\d95\d110\d97\d109\d101\d41\d32\d124\d124\d32\d115\d116\d114\d99\d109\d112\d40\d116\d101\d99\d104\d95\d112\d118\d116\d45\d62\d108\d97\d115\d116\d95\d115\d101\d110\d116\d95\d99\d97\d108\d108\d101\d101\d95\d105\d100\d95\d110\d97\d109\d101\d44\d32\d110\d97\d109\d101\d41\d32\d124\d124\d10\d9\d9\d9\d9\d9\d9\d122\d115\d116\d114\d40\d116\d101\d99\d104\d95\d112\d118\d116\d45\d62\d108\d97\d115\d116\d95\d115\d101\d110\d116\d95\d99\d97\d108\d108\d101\d101\d95\d105\d100\d95\d110\d117\d109\d98\d101\d114\d41\d32\d124\d124\d32\d115\d116\d114\d99\d109\d112\d40\d116\d101\d99\d104\d95\d112\d118\d116\d45\d62\d108\d97\d115\d116\d95\d115\d101\d110\d116\d95\d99\d97\d108\d108\d101\d101\d95\d105\d100\d95\d110\d117\d109\d98\d101\d114\d44\d32\d110\d117\d109\d98\d101\d114\d41\d41\d32\d123~\d9\d9\d9\d9\d9\d117\d115\d108\d101\d101\d112\d40\d49\d48\d48\d48\d48\d48\d48\d41\d59\d10\d9\d9\d9\d9\d9\d105\d102\d32\d40\d115\d119\d105\d116\d99\d104\d95\d99\d104\d97\d110\d110\d101\d108\d95\d116\d101\d115\d116\d95\d102\d108\d97\d103\d40\d99\d104\d97\d110\d110\d101\d108\d44\d32\d67\d70\d95\d65\d78\d83\d87\d69\d82\d69\d68\d41\d32\d38\d38\d32\d40\d122\d115\d116\d114\d40\d116\d101\d99\d104\d95\d112\d118\d116\d45\d62\d108\d97\d115\d116\d95\d115\d101\d110\d116\d95\d99\d97\d108\d108\d101\d101\d95\d105\d100\d95\d110\d97\d109\d101\d41\d32\d124\d124\d32\d115\d116\d114\d99\d109\d112\d40\d116\d101\d99\d104\d95\d112\d118\d116\d45\d62\d108\d97\d115\d116\d95\d115\d101\d110\d116\d95\d99\d97\d108\d108\d101\d101\d95\d105\d100\d95\d110\d97\d109\d101\d44\d32\d110\d97\d109\d101\d41\d32\d124\d124\d10\d9\d9\d9\d9\d9\d9\d122\d115\d116\d114\d40\d116\d101\d99\d104\d95\d112\d118\d116\d45\d62\d108\d97\d115\d116\d95\d115\d101\d110\d116\d95\d99\d97\d108\d108\d101\d101\d95\d105\d100\d95\d110\d117\d109\d98\d101\d114\d41\d32\d124\d124\d32\d115\d116\d114\d99\d109\d112\d40\d116\d101\d99\d104\d95\d112\d118\d116\d45\d62\d108\d97\d115\d116\d95\d115\d101\d110\d116\d95\d99\d97\d108\d108\d101\d101\d95\d105\d100\d95\d110\d117\d109\d98\d101\d114\d44\d32\d110\d117\d109\d98\d101\d114\d41\d41\d41\d32\d123~' Essentially this had been replaced in mod_sofia.c: if (zstr(tech_pvt->last_sent_callee_id_name) || strcmp(tech_pvt->last_sent_callee_id_name, name) || zstr(tech_pvt->last_sent_callee_id_number) || strcmp(tech_pvt->last_sent_callee_id_number, number)) { With this: usleep(1000000); if (switch_channel_test_flag(channel, CF_ANSWERED) && (zstr(tech_pvt->last_sent_callee_id_name) || strcmp(tech_pvt->last_sent_callee_id_name, name) || zstr(tech_pvt->last_sent_callee_id_number) || strcmp(tech_pvt->last_sent_callee_id_number, number))) { Thanks, Shaun Get Outlook for iOS ________________________________ From: FreeSWITCH-users on behalf of Joel Serrano Sent: Thursday, December 20, 2018 17:11 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] FreeSWITCH 1.8.2 - 1-2 second dropped Audio\RTP at the start of a call Hi Shaun, Can you share with the community what you found out? I’m curious what can be the cause for that extra 2-3s delay in RTP On Thu, Dec 20, 2018 at 01:52 Shaun Stokes > wrote: We've identified the cause and have now implemented a fix, this was a local issue caused by a bug in our build script which applies tweaks to mod_sofia. Thanks, Shaun ________________________________ From: FreeSWITCH-users > on behalf of Shaun Stokes > Sent: 19 December 2018 14:07:18 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] FreeSWITCH 1.8.2 - 1-2 second dropped Audio\RTP at the start of a call Thank you all for your recommendations. We raised a JIRA but it was closed as incomplete, we will look at re-opening once we've compiled the relevant logs and pcaps etc unless we reach a conclusion before hand. Some additional points: - We do not set "answer_delay". - There is still a 1-2 second delay after setting "disable_rtp_auto_adjust" on the SIP profile. We've identified this as a package issue, when FreeSWITCH is built with certain packages installed this is causing mod_sofia.so to be much smaller than it should be (5.49MB instead of 8.58MB). We're able to consistently reproduce the followingwhen running FreeSWITCH on the same Debian 9 server using the sameFreeSWITCH configuration and scripts etc: - FreeSWITCH 1.8.2 built on Debian 9 - 1-2 second audio\RTP delay on answer - FreeSWITCH 1.6.20 built on Debian 9 - 1-2 second audio\RTP delay on answer - FreeSWITCH 1.6.20 built on Debian 8 (moved binaries to Debian 9 server) - No audio\RTP delay on answer (audio\RTP isinstantaneous) - FreeSWITCH 1.8.2 built on Debian 9 base (no additional packages) - No audio\RTP delay on answer (audio\RTP is instantaneous) Will post back once we've identified the root cause. Thanks, Shaun ________________________________ From: FreeSWITCH-users > on behalf of Anthony Minessale > Sent: 17 December 2018 20:55 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] FreeSWITCH 1.8.2 - 1-2 second dropped Audio\RTP at the start of a call Sounds like a broken record but the mailing list is not the best place to report issues. Almost inevitably, questions will be asked and data will need to be collected like logs etc. This is why we already point to JIRA to file tickets. Recommendations: 1) Make sure you don't have "answer_delay" set. 2) Check the media signaling data to make sure you are not using "rtp-auto-adjust which adds a time to media establishment to correct for incorrect media IPs. 3) Get a pcap as well as debug + sip trace before reporting any issue because its always going to be the first request anyway. 4) Use JIRA, feel free to ask about the JIRA here but don't rely on 1990's listserv server to track the issue progress. sofia global siptrace on console loglevel debug fsctl debug_level 10 On Mon, Dec 17, 2018 at 2:48 PM Shaun Stokes > wrote: We reverted back to FreeSWITCH 1.6.20 but when this is compiled on the Debian 9 server the problem still occurs. We had to workaround some build errors for FS 1.6.20 to compile on Debian 9 with PostgreSQL 11 but the problem was still present, as follows. -------------------- # Uninstall 1.1.0 SSL header files (libssl-dev) and install the older ones (libssl1.0-dev). apt-get install libssl1.0-dev # Fix PGSQL 11 Support In the file: /usr/src/freeswitch/srcswitch_pgsql.c On line 389, replace this: #if POSTGRESQL_MAJOR_VERSION >= 9 && POSTGRESQL_MINOR_VERSION >= 2 With: #if (POSTGRESQL_MAJOR_VERSION == 9 && POSTGRESQL_MINOR_VERSION >= 2) || POSTGRESQL_MAJOR_VERSION > 9 # Do not build mod_flite or mod_enum sed -i /usr/src/freeswitch/modules.conf -e s:'asr_tts/mod_flite:#asr_tts/mod_flite:' sed -i /usr/src/freeswitch/modules.conf -e s:'applications/mod_enum:#applications/mod_enum:' -------------------- We took the FS 1.6.20 binaries (pre-compiled) from a Debian 8 server and restored them to our Debian 9 server which resolved the issue but we had to copy some missing libs from a Debian 8 server: /usr/lib/x86_64-linux-gnu/libssl.so.1.0.0 /usr/lib/x86_64-linux-gnu/libcrypto.so.1.0.0 Given that the problem changes when the source-code is compiled on different servers we suspect this may be a package problem not specific to FreeSWITCH. This is also a problem on master, raised JIRA: https://freeswitch.org/jira/browse/FS-11572 ________________________________ From: FreeSWITCH-users > on behalf of Shaun Stokes > Sent: 14 December 2018 13:16:14 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] FreeSWITCH 1.8.2 - 1-2 second dropped Audio\RTP at the start of a call Correction, we had moved FreeSWITCH 1.4 (not 1.8) to Server 1 which worked without any audio delays. Upon testing FreeSWITCH 1.8 on Server 1 there is a 1-2 second delay before RTP is established once the call is answered. This is a FreeSWITCH 1.8.2 issue, not a Debian 9 specific (also occurs on Debian 8). FreeSWITCH 1.6 and 1.4 are not effected using the same configuration through-out. ________________________________ From: Shaun Stokes Sent: 14 December 2018 11:44:18 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] FreeSWITCH 1.8.2 - 1-2 second dropped Audio\RTP at the start of a call We have built two test servers side by side on the same hardware with the same configuration, as follows. Server 1: Debian 8 with FreeSWITCH 1.6.20 Server 2: Debian 9 with FreeSWITCH 1.8.2 We can replicate the 1-2 second delay on Server 2 only, whereas Server 1 provides near instant RTP in both directions upon answer. Interestingly, if we move FreeSWITCH 1.8.2 from Server 2 to Server 1 there are still no issues with delay on Server 1, the problem is only observable on the Server 2 running Debian 9 so the problem is not specifically related to FreeSWITCH 1.8.2. At this stage it seems likely the issue lies with Debian 9 or the change in packages on Debian 9. Thanks, Shaun ________________________________ From: FreeSWITCH-users > on behalf of Shaun Stokes > Sent: 11 December 2018 15:28:33 To: FreeSWITCH Users Help Subject: [Freeswitch-users] FreeSWITCH 1.8.2 - 1-2 second dropped Audio\RTP at the start of a call Hi All, Since we've been using FreeSWITCH 1.8.2 we've noticed that the first 1-2 seconds of Audio\RTP at the start of the call when the call is answered is now dropped\missing but this doesn't occur on 1.6.20. When comparing the examples we've noticed the call flow is slightly different, as follows. FreeSWITCH 1.8.2 Leg A: switch_channel.c:2249 (sofia/internal/SRC_EXT at DOMAIN:PORT) Callstate Change DOWN -> RINGING Leg B: sofia.c:7291 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [calling][0] Leg B: sofia.c:7291 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [proceeding][180] Leg B: sofia.c:7401 Ring-Ready sofia/internal/DST_EXT at LAN_IP:PORT! Leg B: switch_channel.c:3354 (sofia/internal/DST_EXT at LAN_IP:PORT) Callstate Change DOWN -> RINGING Leg A: switch_ivr_originate.c:1246 Sending early media Leg A: switch_channel.c:3482 (sofia/internal/SRC_EXT at DOMAIN:PORT) Callstate Change RINGING -> EARLY Leg A: sofia.c:7291 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering state [early][183] Leg B: sofia.c:7291 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [completing][200] Leg B: switch_channel.c:3482 (sofia/internal/DST_EXT at LAN_IP:PORT) Callstate Change RINGING -> EARLY Leg B: sofia.c:7291 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [ready][200] Leg B: sofia.c:8429 Channel [sofia/internal/DST_EXT at LAN_IP:PORT] has been answered Leg B: switch_channel.c:3781 (sofia/internal/DST_EXT at LAN_IP:PORT) Callstate Change EARLY -> ACTIVE Leg A: switch_ivr_bridge.c:766 Channel [sofia/internal/SRC_EXT at DOMAIN:PORT] has been answered Leg A: sofia.c:7291 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering state [completed][200] Leg A: switch_channel.c:3781 (sofia/internal/SRC_EXT at DOMAIN:PORT) Callstate Change EARLY -> ACTIVE Leg A: sofia.c:7291 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering state [ready][200] Leg A: switch_ivr_async.c:1500 No silence detection configured; assuming start of speech Leg B: sofia.c:7291 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [calling][0] Leg A: sofia.c:7291 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering state [calling][0] Leg A: sofia.c:7291 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering state [ready][200] Leg A: sofia.c:8272 Processing updated SDP Leg B: sofia.c:7291 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [ready][200] FreeSWITCH 1.6.20 Leg A: switch_channel.c:2249 (sofia/internal/SRC_EXT at DOMAIN:PORT) Callstate Change DOWN -> RINGING Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [calling][0] Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [proceeding][180] Leg B: sofia.c:7192 Ring-Ready sofia/internal/DST_EXT at LAN_IP:PORT! Leg B: switch_channel.c:3346 (sofia/internal/DST_EXT at LAN_IP:PORT) Callstate Change DOWN -> RINGING Leg A: switch_ivr_originate.c:1215 Sending early media Leg A: switch_channel.c:3474 (sofia/internal/SRC_EXT at DOMAIN:PORT) Callstate Change RINGING -> EARLY Leg A: sofia.c:7084 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering state [early][183] Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [completing][200] Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [ready][200] Leg A: switch_channel.c:3773 (sofia/internal/SRC_EXT at DOMAIN:PORT) Callstate Change EARLY -> ACTIVE Leg A: sofia.c:7084 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering state [completed][200] Leg A: switch_ivr_originate.c:3705 Originate Resulted in Success: [sofia/internal/DST_EXT at LAN_IP:PORT] Leg B: switch_channel.c:3773 (sofia/internal/DST_EXT at LAN_IP:PORT) Callstate Change RINGING -> ACTIVE Leg A: sofia.c:7084 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering state [ready][200] Leg B: Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [ready][200] Leg A: switch_ivr_async.c:1500 No silence detection configured; assuming start of speech Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [calling][0] Leg A: sofia.c:7084 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering state [calling][0] Leg A: sofia.c:7084 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering state [ready][200] Leg A: sofia.c:8061 Processing updated SDP Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [completing][200] Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state [ready][200] On 1.6.20 Leg B changes straight from RINGING to ACTIVE, but on 1.8.2 Leg B first changes from RINGING to EARLY then EARLY to ACTIVE, not sure if this could be related. We've experimented with the following to no avail. rtp-rewrite-timestamps send_silence_when_idle fsctl sync_clock suppress_cng ignore_early_media As per: https://freeswitch.org/confluence/display/FREESWITCH/RTP+Issues https://freeswitch.org/confluence/display/FREESWITCH/VAD+and+CNG https://freeswitch.org/confluence/display/FREESWITCH/send_silence_when_idle https://freeswitch.org/confluence/display/FREESWITCH/Early+Media The calls are local between two extensions\endpoints on the same FreeSWITCH instance and the same SIP profile, the SIP profiles on both servers (1.6.20 and 1.8.2) are identical. Does anyone have any ideas? Thanks, Shaun _________________________________________________________________________ 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 -- Anthony Minessale II Founder, FreeSWITCH. http://freeswitch.com https://youtu.be/l_hOxzCt6X4 https://www.youtube.com/watch?v=oAxXgyx5jUw https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA _________________________________________________________________________ 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 covici at ccs.covici.com Thu Dec 20 18:40:52 2018 From: covici at ccs.covici.com (John Covici) Date: Thu, 20 Dec 2018 13:40:52 -0500 Subject: [Freeswitch-users] FS 1.8 Docker image In-Reply-To: References: <13192151545070920@myt6-7734411c649e.qloud-c.yandex.net> Message-ID: /Thanks, I am new to docker, so this helps. On Thu, 20 Dec 2018 11:56:41 -0500, Sergey Safarov wrote: > > [1 ] > [1.1 ] > [1.2 ] > image is not have bash > but exist "sh" (busybox). > > to copy files you can use docker command > docker cp > > Or mount required folders into image like > > docker run -d -v /var/lib/my_service_folder:/var/lib/my_service_folder safarov/freeswitch:1.8.2 > > чт, 20 дек. 2018 г. в 18:07, John Covici : > > I can't get into the container once I download the image, I tried to > exec it with /bin/bash, but that seems not to exist. I wanted to > start ssh, so I could copy some files in there and maybe make other > changes, but no joy. > > On Thu, 20 Dec 2018 05:38:58 -0500, > Filipe Cunha wrote: > > > > [1 ] > > [1.1 ] > > [1.2 ] > > I end up downloading safarov/freeswitch:1.8.2 image, installed my custom configuration and everything is working fine. > > Thanks for the help guys. > > > > On Mon, Dec 17, 2018 at 9:01 PM Serge S. Yuriev wrote: > > > > Hi > > > > I'm afraid your pastebin no more available. > > Returned 404 > > > > -- > > Wbr, Serge via mobile > > > > 17.12.2018, 17:55, "Mitch Capper" : > > > > I run FS in docker. It is also handy for filing bug reports with exact repo so I have some dockerfiles up for example: https://freeswitch.org/jira/browse/FS-7833 > > full source build. > > > > ~mitch > > > > On Fri, Dec 14, 2018 at 1:23 AM Alexey Sibyakin wrote: > > > > Hi, > > > > Try to install dbus into your Stretch. > > > > Regards, > > > > Alex > > > > On Thu, Dec 13, 2018 at 3:40 AM Sergey Safarov wrote: > > > > you can use safarov/freeswitch:1.8.2 image > > repo of this is located here https://bitbucket.org/sergey-safarov/freeswitch/commits/branch/v1.8 > > > > ср, 12 дек. 2018 г. в 18:14, Filipe Cunha : > > > > Hi all, > > > > I'm trying to create a docker image with the latest version of FreeSWITCH. But I'm not able to do it. These are the steps that I did to try to do it. > > > > 1. Download debian stretch image docker pull debian:stretch > > 2. Build FreeSWITCH from source code following the guide in confluence step by step. I follow the production version > > 3. The step 2 went very well, and then I follow the post-installation instructions and this was when it stop working. I set up owners and permissions and when I try to do the systemd I was not able to do it. I copied > > the file cp /usr/src/freeswitch/debian/freeswitch-systemd.freeswitch.service /etc/systemd/system/freeswitch.service the content of the file was not exactly the same as the one in the tutorial, so I copied the content > > of the file on the tutorial and replace it. And when I tried to do systemctl deamon-reload I got this error message, Failed to connect to bus: No such file or directory. After trying to find a solution online I found > > this answer on SO > > > > Do you guys know if there is a solution for this problem? > > We plan to use the latest version of FreeSwitch in our production environment, but we have a normal linux server to do it, and the devops team got everything working fine, we only have this problem when we try to use > > docker. > > I'm trying to use docker so my team could test everything locally before pushing something to the test environment, so far we have been deploying our code to test environment to test our code 😖 > > I found this image that works fine in our local environment, but the version is 1.6.16. > > Is there any braking change between 1.8.2 and 1.6.16? My concern it that we start developing against an older version, and when we deploy to production it doesn't work, or maybe the latest version have a super useful > > command that we are not taking advantage of. > > We control FreeSWITCH mainly by events, we use java to do it. > > > > Thanks for the help, > > Filipe Cunha > > > > > > _________________________________________________________________________ > > 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. > > > > _________________________________________________________________________ > > 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 > > > > -- > > > > * RED SEVEN TECHNOLOGY GROUP > > > > Filipe Cunha > > > > Software Engineer > > > > * (+44)07753990966 > > > > * filipe at redseven.tech > > * > > * > > * > > [2 ] > > _________________________________________________________________________ > > 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 > > -- > Your life is like a penny. You're going to lose it. The question is: > How do > you spend it? > > John Covici wb2una > covici at ccs.covici.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 > [2 ] > _________________________________________________________________________ > 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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici wb2una covici at ccs.covici.com From joel at textplus.com Thu Dec 20 19:18:44 2018 From: joel at textplus.com (Joel Serrano) Date: Thu, 20 Dec 2018 11:18:44 -0800 Subject: [Freeswitch-users] usernames whith special simbols cant recive calls on FS 1.8 In-Reply-To: <006e01d49860$593f73c0$0bbe5b40$@smartic.es> References: <00c001d4778c$67f9add0$37ed0970$@smartic.es> <8D79C314-B2F2-4CE9-A854-C4AEE2F4DFE5@jerris.com> <006e01d49860$593f73c0$0bbe5b40$@smartic.es> Message-ID: Did you open a JIRA ticket about this? If you didn’t I suggest you do that.. On Thu, Dec 20, 2018 at 05:59 Miguel Jesús López Valverde < mjlopez at smartic.es> wrote: > Hi all > > > > As you indicated to me through Slak and in this email, I keep periodic > inquiries through Slak about the status of this incident, but I do not get > answer in any way after the last one given in this email. > > > > I have also consulted about the following: > > Is it possible to recompile FS18 in its version 1.8.2 on my own, > eliminating the burden of the patch that generates this problem or should I > wait for a new version 1.8.3 with it corrected ?. In case it can be > recompiled by me, could you tell me in a basic way how to do it? > > > > But I'm also not having answers through Slak. For this I return to send > you these queries via email, as it is important for me to find a solution > to this issue to be able to update the current FS1.6 servers that I have > available to FS1.8 > > > > Thank you again and greetings. > > > > Miguel J. López. > > > > > > *De:* FreeSWITCH-users [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *En nombre de *Michael > Jerris > *Enviado el:* jueves, 29 de noviembre de 2018 21:46 > *Para:* FreeSWITCH Users Help > *Asunto:* Re: [Freeswitch-users] usernames whith special simbols cant > recive calls on FS 1.8 > > > > This issue is cause by the patch adding in: > > > > https://freeswitch.org/jira/browse/FS-9791 > > > > And is specific to calling sip registered endpoints with ~ in the > username. I’ve reached out to the original author of that patch, and if > they don’t respond with a fix I’ll have to revert that patch. Please > remind me if I dont follow up on that issue soon by commenting on that jira. > > > > On Nov 8, 2018, at 12:56 PM, Miguel Jesús López Valverde < > mjlopez at smartic.es> wrote: > > > > Hello again guys: > > > > I'm sending this question in case someone can tell me how to solve the > following problem. > > > > We currently have a production platform working with FS 1.6 where users > have historically been created in the form username~companyname like > usernames and the incoming calls to these users work correctly. Also, a > request on console like “sofia_contact nameusername~companyname at domain” > returns a right result. > > > > Now I am considering the migration of these platforms to FS 1.8 and, for > this, doing an assembly of initial development. Here I have been able to > verify that although these users register correctly, they can issue > outgoing calls and appear correctly listed in queries of type “sofia status > profile internal reg”, but these users can’t receive incoming calls, in > console the information " Cannot create outgoing channel of type [error] > cause: [USER_NOT_REGISTERED] " is always obtained and when I launch the > query “sofia_contact nameusername~companyname at domain” in console, I get > the result “error/user_not_registered”; I only have to change the value ~ > for example by a dot and the incoming calls works correctly and also the > query “sofia_contact nameusername.companyname at domain” return a right > result. > > > > I guess I need to adjust the configuration of some library of FS 1.8 and > recompile it to get back an identical operation to version 1.6, can someone > help me in indicating which library or with what adjustment I could get > this operation again? > > > > Thank you very much and best regards!! > > > > > > > Libre > de virus. www.avast.com > > <#m_4554921039687379035_DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2> > _________________________________________________________________________ > 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 sozturk at gmail.com Fri Dec 21 07:58:08 2018 From: sozturk at gmail.com (srdrztrk) Date: Fri, 21 Dec 2018 10:58:08 +0300 Subject: [Freeswitch-users] [Zombie Ports Listening] In-Reply-To: References: <20181214185238.GA25158@darth.lan> <20181217184123.GA6456@darth.lan> Message-ID: Has anyone got a solution for this problem? Regards. On Tue, 18 Dec 2018 at 21:42, Bilal Abbasi wrote: > Exactly this is what i am looking for, as i cannot see any module > associated with that. > > Regards > Abbasi > > On Tue, 18 Dec 2018 at 2:04 AM, Sebastian Kemper > wrote: > >> On Mon, Dec 17, 2018 at 07:08:11AM -0800, Joel Serrano wrote: >> > Have you tried commenting out the module so it’s not loaded? >> >> Hello Joel, >> >> It's not a module. I checked the source in src/switch_msrp.c, it doesn't >> seem to have an off switch. >> >> Kind regards, >> Seb >> >> _________________________________________________________________________ >> 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 igor.potjevlesch at gmail.com Fri Dec 21 08:54:58 2018 From: igor.potjevlesch at gmail.com (igor.potjevlesch at gmail.com) Date: Fri, 21 Dec 2018 09:54:58 +0100 Subject: [Freeswitch-users] Relay a REGISTER In-Reply-To: References: <000f01d496c0$454fc550$cfef4ff0$@gmail.com> Message-ID: <002201d4990a$da62fcb0$8f28f610$@gmail.com> Hello, Thank you for your replies. The thing is that I already use a Kamailio but the idea is to put the Freeswitch in front of the customers. But I understand that it's complicated as per your comments. Regards, Igor. De : FreeSWITCH-users De la part de Shaun Stokes Envoyé : jeudi 20 décembre 2018 11:46 À : 'FreeSWITCH Users Help' Objet : Re: [Freeswitch-users] Relay a REGISTER If you want to achieve this using FreeSWITCH you would likely need to modify the source-code and\or create a new module to control this functionality, it wouldn't be a simple task. You're much better off using a SIP Proxy such as Kamailio which allows you to control the flow of the SIP packets your-self through the configuration, although it can be a bit of a learning curve for those not familiar with Kamailio. Shaun _____ From: FreeSWITCH-users < freeswitch-users-bounces at lists.freeswitch.org> on behalf of igor.potjevlesch at gmail.com < igor.potjevlesch at gmail.com> Sent: 18 December 2018 10:56:03 To: 'FreeSWITCH Users Help' Subject: [Freeswitch-users] Relay a REGISTER Hello, I'd like to relay a REGISTER received on leg A to the leg B. The idea is to have a Contact URI with the B leg IP address when the REGISTER goes out of the FS. But I didn't found how to avoid Freeswitch to handle the REGISTER locally. Regards, Igor. -------------- next part -------------- An HTML attachment was scrubbed... URL: From andrew.keil at visytel.com Sat Dec 22 04:29:07 2018 From: andrew.keil at visytel.com (Andrew Keil) Date: Sat, 22 Dec 2018 04:29:07 +0000 Subject: [Freeswitch-users] Merry Christmas and Happy New Year Message-ID: To the FreeSWITCH team, I just wanted to pass on my regards to the entire FreeSWITCH team. Merry Christmas and Happy New Year! It has been a great day for Visytel's solution, running on top of FreeSWITCH 1.8.2 (on Windows Server 2016). Today (Live on TV) a major charity donation service in the UK raised over GBP 120,000.00 using my solution (ie. transactions done over the phone using DTMF) successfully with no errors (all in just 3 hours). Fully PCI SAQ D compliant. Can't ask for more! Warmest Regards, Andrew Keil Visytel Pty Ltd https://www.visytel.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From alaadin.abd at gmail.com Sat Dec 22 17:26:57 2018 From: alaadin.abd at gmail.com (Alaadin Abdurrahman) Date: Sat, 22 Dec 2018 19:26:57 +0200 Subject: [Freeswitch-users] FreeSWITCH library & dependency for Debian 8 Message-ID: Hello, i am newbie to freeswitch, i have installed freeswitch following the installation instruction on the FreeSWITCH 1.8 book everything works great. what i am trying to a accomplish is to install it from source code as other linux application so i would like to have a list of all the library dependency for Debian 8 i have seen to many tutorial on the internet but i would like to have list written by the freeswitch developers. best regards -------------- next part -------------- An HTML attachment was scrubbed... URL: From imfanee at gmail.com Mon Dec 24 03:10:25 2018 From: imfanee at gmail.com (Faisal Hanif) Date: Mon, 24 Dec 2018 08:10:25 +0500 Subject: [Freeswitch-users] FreeSWITCH library & dependency for Debian 8 In-Reply-To: References: Message-ID: To be honestly you can't have a list having everything but basic one listed in online document while you need to identify from errors output and install yourself while compiling . On Mon, Dec 24, 2018, 1:12 AM Alaadin Abdurrahman Hello, > i am newbie to freeswitch, i have installed freeswitch following the > installation instruction on the FreeSWITCH 1.8 book everything works great. > what i am trying to a accomplish is to install it from source code as > other linux application so i would like to have a list of all the library > dependency for Debian 8 i have seen to many tutorial on the internet but i > would like to have list written by the freeswitch developers. > > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: From anthony.minessale at gmail.com Mon Dec 24 05:44:00 2018 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 23 Dec 2018 23:44:00 -0600 Subject: [Freeswitch-users] Merry Christmas and Happy New Year In-Reply-To: References: Message-ID: Thanks! Best wishes to you too and all the rest of the community! On Sun, Dec 23, 2018 at 2:39 PM Andrew Keil wrote: > To the FreeSWITCH team, > > > > I just wanted to pass on my regards to the entire FreeSWITCH team. Merry > Christmas and Happy New Year! > > > > It has been a great day for Visytel’s solution, running on top of > FreeSWITCH 1.8.2 (on Windows Server 2016). Today (Live on TV) a major > charity donation service in the UK raised over GBP 120,000.00 using my > solution (ie. transactions done over the phone using DTMF) successfully > with no errors (all in just 3 hours). Fully PCI SAQ D compliant. Can’t > ask for more! > > > > Warmest Regards, > > > > Andrew Keil > > *Visytel Pty Ltd* > > https://www.visytel.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 -- Anthony Minessale II Founder, FreeSWITCH. http://freeswitch.com https://youtu.be/l_hOxzCt6X4 https://www.youtube.com/watch?v=oAxXgyx5jUw https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA -------------- next part -------------- An HTML attachment was scrubbed... URL: From mj310792 at gmail.com Mon Dec 24 06:20:22 2018 From: mj310792 at gmail.com (mittali jangid) Date: Mon, 24 Dec 2018 11:50:22 +0530 Subject: [Freeswitch-users] Presence not working on resubscribe Message-ID: Hi, I am working on presence in polycom phones. I have BLF key for fifo-orbits. Presence is working fine when call is parked in orbit. But if phone has lower subscription-expiry and phone resubscribes, then freeswitch responds with "Terminated" state notify. Same is happening for user-directory extension number set as BLF in polycom phones. Also if i add BLF for either fifo-orbit or extension when any call is parked or extension is attending call respectively, presence is not working as per expectation. I verified this in latest freeswitch version 1.8.2 and also Master pull. I would like to know why it is behaving this way. Below link also faced similar issue, but it is still not resolved : http://lists.freeswitch.org/pipermail/freeswitch-users/2013-October/100598.html Your help is much appreciated. Let me know what information is required from my end. Thanks and regards, Mittali Jangid -------------- next part -------------- An HTML attachment was scrubbed... URL: From alaadin.abd at gmail.com Mon Dec 24 11:00:27 2018 From: alaadin.abd at gmail.com (Alaadin Abdurrahman) Date: Mon, 24 Dec 2018 13:00:27 +0200 Subject: [Freeswitch-users] FreeSWITCH library & dependency for Debian 8 In-Reply-To: References: Message-ID: Thank you for your reply On Mon, Dec 24, 2018 at 12:43 PM Faisal Hanif wrote: > To be honestly you can't have a list having everything but basic one > listed in online document while you need to identify from errors output and > install yourself while compiling . > > On Mon, Dec 24, 2018, 1:12 AM Alaadin Abdurrahman wrote: > >> Hello, >> i am newbie to freeswitch, i have installed freeswitch following the >> installation instruction on the FreeSWITCH 1.8 book everything works great. >> what i am trying to a accomplish is to install it from source code as >> other linux application so i would like to have a list of all the library >> dependency for Debian 8 i have seen to many tutorial on the internet but i >> would like to have list written by the freeswitch developers. >> >> 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 > > _________________________________________________________________________ > 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 bilaln018 at gmail.com Mon Dec 24 15:19:08 2018 From: bilaln018 at gmail.com (Bilal Abbasi) Date: Mon, 24 Dec 2018 20:19:08 +0500 Subject: [Freeswitch-users] [Zombie Ports Listening] In-Reply-To: References: <20181214185238.GA25158@darth.lan> <20181217184123.GA6456@darth.lan> Message-ID: I remember those days when Brain West atleast look at each mail. I wish i can get those days back. Regards Abbasi On Fri, 21 Dec 2018 at 9:55 PM, srdrztrk wrote: > Has anyone got a solution for this problem? > > Regards. > > On Tue, 18 Dec 2018 at 21:42, Bilal Abbasi wrote: > >> Exactly this is what i am looking for, as i cannot see any module >> associated with that. >> >> Regards >> Abbasi >> >> On Tue, 18 Dec 2018 at 2:04 AM, Sebastian Kemper >> wrote: >> >>> On Mon, Dec 17, 2018 at 07:08:11AM -0800, Joel Serrano wrote: >>> > Have you tried commenting out the module so it’s not loaded? >>> >>> Hello Joel, >>> >>> It's not a module. I checked the source in src/switch_msrp.c, it doesn't >>> seem to have an off switch. >>> >>> Kind regards, >>> Seb >>> >>> _________________________________________________________________________ >>> 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 gmaruzz at gmail.com Wed Dec 26 12:31:13 2018 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Wed, 26 Dec 2018 13:31:13 +0100 Subject: [Freeswitch-users] [Zombie Ports Listening] In-Reply-To: References: <20181214185238.GA25158@darth.lan> <20181217184123.GA6456@darth.lan> Message-ID: Bilal, If listening on those ports bothers you, block those ports via iptables. Yes, #metoo want my head full of hair back. Happy New Year to you and Brian -giovanni On Wed, Dec 26, 2018, 13:07 Bilal Abbasi I remember those days when Brain West atleast look at each mail. > I wish i can get those days back. > > Regards > Abbasi > > On Fri, 21 Dec 2018 at 9:55 PM, srdrztrk wrote: > >> Has anyone got a solution for this problem? >> >> Regards. >> >> On Tue, 18 Dec 2018 at 21:42, Bilal Abbasi wrote: >> >>> Exactly this is what i am looking for, as i cannot see any module >>> associated with that. >>> >>> Regards >>> Abbasi >>> >>> On Tue, 18 Dec 2018 at 2:04 AM, Sebastian Kemper >>> wrote: >>> >>>> On Mon, Dec 17, 2018 at 07:08:11AM -0800, Joel Serrano wrote: >>>> > Have you tried commenting out the module so it’s not loaded? >>>> >>>> Hello Joel, >>>> >>>> It's not a module. I checked the source in src/switch_msrp.c, it doesn't >>>> seem to have an off switch. >>>> >>>> Kind regards, >>>> Seb >>>> >>>> >>>> _________________________________________________________________________ >>>> 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 brian at freeswitch.com Wed Dec 26 13:57:45 2018 From: brian at freeswitch.com (Brian West) Date: Wed, 26 Dec 2018 08:57:45 -0500 Subject: [Freeswitch-users] [Zombie Ports Listening] In-Reply-To: References: <20181214185238.GA25158@darth.lan> <20181217184123.GA6456@darth.lan> Message-ID: I do look at emails, also this should be reported to JIRA please. /b On Wed, Dec 26, 2018 at 6:32 AM Bilal Abbasi wrote: > I remember those days when Brain West atleast look at each mail. > I wish i can get those days back. > > Regards > Abbasi > > On Fri, 21 Dec 2018 at 9:55 PM, srdrztrk wrote: > >> Has anyone got a solution for this problem? >> >> Regards. >> >> On Tue, 18 Dec 2018 at 21:42, Bilal Abbasi wrote: >> >>> Exactly this is what i am looking for, as i cannot see any module >>> associated with that. >>> >>> Regards >>> Abbasi >>> >>> On Tue, 18 Dec 2018 at 2:04 AM, Sebastian Kemper >>> wrote: >>> >>>> On Mon, Dec 17, 2018 at 07:08:11AM -0800, Joel Serrano wrote: >>>> > Have you tried commenting out the module so it’s not loaded? >>>> >>>> Hello Joel, >>>> >>>> It's not a module. I checked the source in src/switch_msrp.c, it doesn't >>>> seem to have an off switch. >>>> >>>> Kind regards, >>>> Seb >>>> >>>> >>>> _________________________________________________________________________ >>>> 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 -- Brian West | Co-founder and Developer Need Commercial support? email sales at freeswitch.com FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 Email: brian at freeswitch.com Mobile: 918-424-9378 Website: https://www.FreeSWITCH.com [image: https://www.facebook.com/signalwireinc?src=email] [image: https://twitter.com/freeswitch] -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at freeswitch.com Wed Dec 26 15:29:05 2018 From: mike at freeswitch.com (Mike Jerris) Date: Wed, 26 Dec 2018 10:29:05 -0500 Subject: [Freeswitch-users] FreeSWITCH library & dependency for Debian 8 In-Reply-To: References: Message-ID: Take at look at what is in the debian repository. Those are the deps we need for what we build, and some override packages to fix issues in system packages. On Sun, Dec 23, 2018 at 3:12 PM Alaadin Abdurrahman wrote: > Hello, > i am newbie to freeswitch, i have installed freeswitch following the > installation instruction on the FreeSWITCH 1.8 book everything works great. > what i am trying to a accomplish is to install it from source code as > other linux application so i would like to have a list of all the library > dependency for Debian 8 i have seen to many tutorial on the internet but i > would like to have list written by the freeswitch developers. > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at freeswitch.com Wed Dec 26 15:31:45 2018 From: mike at freeswitch.com (Mike Jerris) Date: Wed, 26 Dec 2018 10:31:45 -0500 Subject: [Freeswitch-users] [Zombie Ports Listening] In-Reply-To: References: <20181214185238.GA25158@darth.lan> <20181217184123.GA6456@darth.lan> Message-ID: There is not a solution for this problem currently, it needs to be fixed in code. I saw this last week but we haven’t done a fix for it yet, someone should file a jira for this issue. On Wed, Dec 26, 2018 at 10:26 AM Brian West wrote: > I do look at emails, also this should be reported to JIRA please. > > /b > > On Wed, Dec 26, 2018 at 6:32 AM Bilal Abbasi wrote: > >> I remember those days when Brain West atleast look at each mail. >> I wish i can get those days back. >> >> Regards >> Abbasi >> >> On Fri, 21 Dec 2018 at 9:55 PM, srdrztrk wrote: >> >>> Has anyone got a solution for this problem? >>> >>> Regards. >>> >>> On Tue, 18 Dec 2018 at 21:42, Bilal Abbasi wrote: >>> >>>> Exactly this is what i am looking for, as i cannot see any module >>>> associated with that. >>>> >>>> Regards >>>> Abbasi >>>> >>>> On Tue, 18 Dec 2018 at 2:04 AM, Sebastian Kemper >>>> wrote: >>>> >>>>> On Mon, Dec 17, 2018 at 07:08:11AM -0800, Joel Serrano wrote: >>>>> > Have you tried commenting out the module so it’s not loaded? >>>>> >>>>> Hello Joel, >>>>> >>>>> It's not a module. I checked the source in src/switch_msrp.c, it >>>>> doesn't >>>>> seem to have an off switch. >>>>> >>>> -------------- next part -------------- An HTML attachment was scrubbed... URL: From bilaln018 at gmail.com Thu Dec 27 06:14:12 2018 From: bilaln018 at gmail.com (Bilal Abbasi) Date: Thu, 27 Dec 2018 11:14:12 +0500 Subject: [Freeswitch-users] [Zombie Ports Listening] In-Reply-To: References: <20181214185238.GA25158@darth.lan> <20181217184123.GA6456@darth.lan> Message-ID: Thanks alot brian, for being here ;) Thanks all Regards Abbasi On Wed, 26 Dec 2018 at 8:44 PM, Brian West wrote: > I do look at emails, also this should be reported to JIRA please. > > /b > > On Wed, Dec 26, 2018 at 6:32 AM Bilal Abbasi wrote: > >> I remember those days when Brain West atleast look at each mail. >> I wish i can get those days back. >> >> Regards >> Abbasi >> >> On Fri, 21 Dec 2018 at 9:55 PM, srdrztrk wrote: >> >>> Has anyone got a solution for this problem? >>> >>> Regards. >>> >>> On Tue, 18 Dec 2018 at 21:42, Bilal Abbasi wrote: >>> >>>> Exactly this is what i am looking for, as i cannot see any module >>>> associated with that. >>>> >>>> Regards >>>> Abbasi >>>> >>>> On Tue, 18 Dec 2018 at 2:04 AM, Sebastian Kemper >>>> wrote: >>>> >>>>> On Mon, Dec 17, 2018 at 07:08:11AM -0800, Joel Serrano wrote: >>>>> > Have you tried commenting out the module so it’s not loaded? >>>>> >>>>> Hello Joel, >>>>> >>>>> It's not a module. I checked the source in src/switch_msrp.c, it >>>>> doesn't >>>>> seem to have an off switch. >>>>> >>>>> Kind regards, >>>>> Seb >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> 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 > > -- > > Brian West | Co-founder and Developer > > Need Commercial support? email sales at freeswitch.com > > FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 > > > Email: brian at freeswitch.com > > Mobile: 918-424-9378 > > Website: https://www.FreeSWITCH.com > > [image: https://www.facebook.com/signalwireinc?src=email] > [image: > https://twitter.com/freeswitch] > _________________________________________________________________________ > 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 imfanee at gmail.com Thu Dec 27 07:48:27 2018 From: imfanee at gmail.com (Faisal Hanif) Date: Thu, 27 Dec 2018 12:48:27 +0500 Subject: [Freeswitch-users] Conference Video Mux Option doesn't display video but Video Muted (but audio not affected) instead In-Reply-To: References: Message-ID: Hi #Alex / #Brain / #Mike, I had created JIAR (https://freeswitch.org/jira/browse/FS-11574) about video mux issue but I see no activity there, Do I need to do something on it to get it noticed? If yes please guide me so. Regards, Faisal On Thu, 20 Dec 2018 at 11:42, Faisal Hanif wrote: > Hi, > > I have created JIRA couple of days before, > > https://freeswitch.org/jira/browse/FS-11574 > > but no activity on it yet, Do I need to do something else ? > > Regards, > > Faisal > > On Mon, 3 Dec 2018 at 19:45, Alexey Sibyakin wrote: > >> That's odd. Please issue a Jira. >> >> https://freeswitch.org/confluence/display/FREESWITCH/Reporting+Bugs+to+JIRA >> >> Regards, >> >> Alex >> >> On Sun, Dec 2, 2018 at 7:40 AM Faisal Hanif wrote: >> >>> Hi Alex, >>> >>> Thanks a lot for reply. I have adopted the suggestion and installed >>> FreeSWITCH 1.8.2 from package on Debian 9 but I still have the same issue, >>> >>> FreeSWITCH Version 1.8.2-3-a98a958ac3~64bit (-3-a98a958ac3 64bit) >>> Distributor ID: Debian >>> Description: Debian GNU/Linux 9.6 (stretch) >>> Release: 9.6 >>> Codename: stretch >>> >>> This error I got in logs while joining the conference, >>> >>> 2018-12-01 06:33:43.020210 [NOTICE] switch_core_media.c:15575 Activating >>> write resampler >>> 2018-12-01 06:33:43.050211 [NOTICE] switch_vpx.c:486 VPX encoder reset >>> (WxH/BW) from 0x0/0 to 1280x480/1024 >>> 2018-12-01 06:33:43.070188 [NOTICE] switch_vpx.c:486 VPX encoder reset >>> (WxH/BW) from 1280x480/1024 to 1280x480/645 >>> 2018-12-01 06:33:43.070188 [ERR] switch_vpx.c:841 VPX encode error >>> 8:Invalid parameter:(null) >>> >>> >>> I really appreciate your cooperation. >>> >>> Regards, >>> >>> Faisal >>> >>> On Fri, 30 Nov 2018 at 16:02, Alexey Sibyakin >>> wrote: >>> >>>> Hi, >>>> >>>> Try Debian 9 and FreeSWITCH 1.8.2 from official packages. If you are >>>> going to use dev version on nonsupported OS you have to handle it yourself. >>>> >>>> Regards, >>>> >>>> Alex >>>> >>>> On Thu, Nov 29, 2018 at 1:16 AM Faisal Hanif wrote: >>>> >>>>> Hi Geeks, >>>>> >>>>> I am trying to implement a conference in mux mode and FreeSWITCH send >>>>> canvas properly but never show video on but a pic "Video Muted (but audio >>>>> not affected)" pic in place of every member's video on canvas. I tried a >>>>> lot with no success :( >>>>> >>>>> OS : Ubuntu 14.04.5 LTS trusty >>>>> FreeSWITCH Version 1.9.0+git~20181120T210412Z~968c76b29c~64bit (git >>>>> 968c76b 2018-11-20 21:04:12Z 64bit) >>>>> >>>>> My conference profile is >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>> value="tone_stream://%(200,0,500,600,700)"/> >>>>> >>>> value="tone_stream://%(500,0,300,200,100,50,25)"/> >>>>> >>>>> >>>>> >>>>> >>>>> >>>> value="audio-always|livearray-json-status"/> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>> value="/usr/local/freeswitch/conf/images/video-muted.png"/> >>>>> >>>> value="/usr/local/freeswitch/conf/images/video-muted.png"/> >>>>> >>>>> >>>>> >>>>> can anyone please help me. >>>>> Regards, >>>>> >>>>> Faisal >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> 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 >>> >>> >>> >>> -- >>> Regards, >>> >>> Faisal Hanif >>> _________________________________________________________________________ >>> 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 > > > > -- > Regards, > > Faisal Hanif > -- Regards, Faisal Hanif -------------- next part -------------- An HTML attachment was scrubbed... URL: From joelists at tm.net.uk Fri Dec 28 16:05:28 2018 From: joelists at tm.net.uk (Joseph Waite) Date: Fri, 28 Dec 2018 16:05:28 +0000 Subject: [Freeswitch-users] Controlling ptime in INVITE Message-ID: Evening All. I am having an issue, currently with Cisco SPA phones, where they are trying to use a prime of 30, however one of our providers does not support this and fails the call. Customer does not notice as call fails over to another provider as it should, however the failover provider is more expensive so not ideal for us. We don’t notice this unless we examine cdr logs. Is there a way, either to tell freeswitch to send a response to the phone negotiating time of 20 or to refuse the call if it comes in with time of 30? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: From alaadin.abd at gmail.com Fri Dec 28 16:29:01 2018 From: alaadin.abd at gmail.com (Alaadin Abdurrahman) Date: Fri, 28 Dec 2018 18:29:01 +0200 Subject: [Freeswitch-users] FreeSWITCH library & dependency for Debian 8 In-Reply-To: References: Message-ID: Thanks, I will take a look On Fri, Dec 28, 2018 at 6:26 PM Mike Jerris wrote: > Take at look at what is in the debian repository. Those are the deps we > need for what we build, and some override packages to fix issues in system > packages. > > On Sun, Dec 23, 2018 at 3:12 PM Alaadin Abdurrahman > wrote: > >> Hello, >> i am newbie to freeswitch, i have installed freeswitch following the >> installation instruction on the FreeSWITCH 1.8 book everything works great. >> what i am trying to a accomplish is to install it from source code as >> other linux application so i would like to have a list of all the library >> dependency for Debian 8 i have seen to many tutorial on the internet but i >> would like to have list written by the freeswitch developers. >> >> _________________________________________________________________________ > 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 social at bohboh.info Fri Dec 28 20:24:33 2018 From: social at bohboh.info (Social Boh) Date: Fri, 28 Dec 2018 15:24:33 -0500 Subject: [Freeswitch-users] Controlling ptime in INVITE In-Reply-To: References: Message-ID: <7faf9064-24a8-12ac-3360-bd3de5205a10@bohboh.info> Hello, Are you tried to use *absolute_codec_string* in the outbound block of your dialplan? like: Regards --- I'm SoCIaL, MayBe El 28/12/2018 a las 11:05, Joseph Waite escribió: > Evening All. > > I am having an issue, currently with Cisco SPA phones, where they are > trying to use a prime of 30, however one of our providers does not > support this and fails the call. > > Customer does not notice as call fails over to another provider as it > should, however the failover provider is more expensive so not ideal > for us. > > > We don’t notice this unless we examine cdr logs. > > Is there a way, either to tell freeswitch to send a response to the > phone negotiating time of 20 or to refuse the call if it comes in with > time of 30? > > 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 mike at freeswitch.org Fri Dec 28 20:47:26 2018 From: mike at freeswitch.org (Mike Jerris) Date: Fri, 28 Dec 2018 15:47:26 -0500 Subject: [Freeswitch-users] FreeSWITCH library & dependency for Debian 8 In-Reply-To: References: Message-ID: <1A37E7A2-67B7-4951-BA13-2B7F49E02CD4@freeswitch.org> Also this doc has all the info of how to install the required deps: https://freeswitch.org/confluence/display/FREESWITCH/Debian+8+Jessie > On Dec 28, 2018, at 11:29 AM, Alaadin Abdurrahman wrote: > > Thanks, I will take a look > > On Fri, Dec 28, 2018 at 6:26 PM Mike Jerris > wrote: > Take at look at what is in the debian repository. Those are the deps we need for what we build, and some override packages to fix issues in system packages. > > On Sun, Dec 23, 2018 at 3:12 PM Alaadin Abdurrahman > wrote: > Hello, > i am newbie to freeswitch, i have installed freeswitch following the installation instruction on the FreeSWITCH 1.8 book everything works great. > what i am trying to a accomplish is to install it from source code as other linux application so i would like to have a list of all the library dependency for Debian 8 i have seen to many tutorial on the internet but i would like to have list written by the freeswitch developers -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Fri Dec 28 21:12:27 2018 From: david.villasmil.work at gmail.com (David Villasmil) Date: Fri, 28 Dec 2018 22:12:27 +0100 Subject: [Freeswitch-users] Module's architecture image Message-ID: Hello all, Years ago, Anthony sent out an image explaining FS' module architecture. I had it somewhere but can't find it now... Anyone got it? Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 -------------- next part -------------- An HTML attachment was scrubbed... URL: From joelists at tm.net.uk Fri Dec 28 22:06:52 2018 From: joelists at tm.net.uk (Joseph Waite) Date: Fri, 28 Dec 2018 22:06:52 +0000 Subject: [Freeswitch-users] Controlling ptime in INVITE In-Reply-To: <7faf9064-24a8-12ac-3360-bd3de5205a10@bohboh.info> References: <7faf9064-24a8-12ac-3360-bd3de5205a10@bohboh.info> Message-ID: <0D1C4B76-8A06-4AB5-9E38-3B6585B78427@tm.net.uk> I I haven’t. Is there a way to set this in the inbound allowed codec?? Joe Waite > On 28 Dec 2018, at 20:24, Social Boh wrote: > > Hello, > > Are you tried to use absolute_codec_string in the outbound block of your dialplan? > > like: > > > > Regards > > --- > I'm SoCIaL, MayBe >> El 28/12/2018 a las 11:05, Joseph Waite escribió: >> Evening All. >> >> I am having an issue, currently with Cisco SPA phones, where they are trying to use a prime of 30, however one of our providers does not support this and fails the call. >> >> Customer does not notice as call fails over to another provider as it should, however the failover provider is more expensive so not ideal for us. >> >> >> We don’t notice this unless we examine cdr logs. >> >> Is there a way, either to tell freeswitch to send a response to the phone negotiating time of 20 or to refuse the call if it comes in with time of 30? >> >> 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: From s.safarov at gmail.com Sat Dec 29 04:13:21 2018 From: s.safarov at gmail.com (Sergey Safarov) Date: Sat, 29 Dec 2018 07:13:21 +0300 Subject: [Freeswitch-users] [Zombie Ports Listening] In-Reply-To: References: <20181214185238.GA25158@darth.lan> <20181217184123.GA6456@darth.lan> Message-ID: Will it turn off MSRP support via a codec string or another global variable or profile parameter? пт, 28 дек. 2018 г. в 20:06, Mike Jerris : > There is not a solution for this problem currently, it needs to be fixed > in code. I saw this last week but we haven’t done a fix for it yet, > someone should file a jira for this issue. > > On Wed, Dec 26, 2018 at 10:26 AM Brian West wrote: > >> I do look at emails, also this should be reported to JIRA please. >> >> /b >> >> On Wed, Dec 26, 2018 at 6:32 AM Bilal Abbasi wrote: >> >>> I remember those days when Brain West atleast look at each mail. >>> I wish i can get those days back. >>> >>> Regards >>> Abbasi >>> >>> On Fri, 21 Dec 2018 at 9:55 PM, srdrztrk wrote: >>> >>>> Has anyone got a solution for this problem? >>>> >>>> Regards. >>>> >>>> On Tue, 18 Dec 2018 at 21:42, Bilal Abbasi wrote: >>>> >>>>> Exactly this is what i am looking for, as i cannot see any module >>>>> associated with that. >>>>> >>>>> Regards >>>>> Abbasi >>>>> >>>>> On Tue, 18 Dec 2018 at 2:04 AM, Sebastian Kemper >>>>> wrote: >>>>> >>>>>> On Mon, Dec 17, 2018 at 07:08:11AM -0800, Joel Serrano wrote: >>>>>> > Have you tried commenting out the module so it’s not loaded? >>>>>> >>>>>> Hello Joel, >>>>>> >>>>>> It's not a module. I checked the source in src/switch_msrp.c, it >>>>>> doesn't >>>>>> seem to have an off switch. >>>>>> >>>>> > _________________________________________________________________________ > 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 bilaln018 at gmail.com Sat Dec 29 10:19:44 2018 From: bilaln018 at gmail.com (Bilal Abbasi) Date: Sat, 29 Dec 2018 15:19:44 +0500 Subject: [Freeswitch-users] [Zombie Ports Listening] In-Reply-To: References: <20181214185238.GA25158@darth.lan> <20181217184123.GA6456@darth.lan> Message-ID: I will do that shortly. Regards Abbasi On Fri, 28 Dec 2018 at 10:19 PM, Mike Jerris wrote: > There is not a solution for this problem currently, it needs to be fixed > in code. I saw this last week but we haven’t done a fix for it yet, > someone should file a jira for this issue. > > On Wed, Dec 26, 2018 at 10:26 AM Brian West wrote: > >> I do look at emails, also this should be reported to JIRA please. >> >> /b >> >> On Wed, Dec 26, 2018 at 6:32 AM Bilal Abbasi wrote: >> >>> I remember those days when Brain West atleast look at each mail. >>> I wish i can get those days back. >>> >>> Regards >>> Abbasi >>> >>> On Fri, 21 Dec 2018 at 9:55 PM, srdrztrk wrote: >>> >>>> Has anyone got a solution for this problem? >>>> >>>> Regards. >>>> >>>> On Tue, 18 Dec 2018 at 21:42, Bilal Abbasi wrote: >>>> >>>>> Exactly this is what i am looking for, as i cannot see any module >>>>> associated with that. >>>>> >>>>> Regards >>>>> Abbasi >>>>> >>>>> On Tue, 18 Dec 2018 at 2:04 AM, Sebastian Kemper >>>>> wrote: >>>>> >>>>>> On Mon, Dec 17, 2018 at 07:08:11AM -0800, Joel Serrano wrote: >>>>>> > Have you tried commenting out the module so it’s not loaded? >>>>>> >>>>>> Hello Joel, >>>>>> >>>>>> It's not a module. I checked the source in src/switch_msrp.c, it >>>>>> doesn't >>>>>> seem to have an off switch. >>>>>> >>>>> > _________________________________________________________________________ > 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 abaci64 at gmail.com Sun Dec 30 18:42:49 2018 From: abaci64 at gmail.com (Abaci B) Date: Sun, 30 Dec 2018 13:42:49 -0500 Subject: [Freeswitch-users] hangup after 500 seconds of recording Message-ID: Hi, I noticed recently that when recording in freeswitch from a Verizon Wireless 4G LTE phone the call will hangup after 500 seconds of recording, I would guess they have a new media timeout detection system that ignores silence. I tried using setting "record_waste_resources=true" which usually solves the problem, but that didn't help, was wondering if anyone else experiences this issue or knows of a solution to the problem. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: From bjordan at e-teleco.com Mon Dec 31 00:46:32 2018 From: bjordan at e-teleco.com (bjordan at e-teleco.com) Date: Mon, 31 Dec 2018 00:46:32 +0000 Subject: [Freeswitch-users] Module's architecture image In-Reply-To: References: Message-ID: I do not know the specific image you are talking about but could it be this one? [https://www.packtpub.com/sites/default/files/Article-Images/1004_01_01.png] From: FreeSWITCH-users On Behalf Of David Villasmil Sent: Friday, December 28, 2018 1:12 PM To: FreeSWITCH Users Help Subject: [Freeswitch-users] Module's architecture image Hello all, Years ago, Anthony sent out an image explaining FS' module architecture. I had it somewhere but can't find it now... Anyone got it? Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.png Type: image/png Size: 325287 bytes Desc: image001.png URL: From bjordan at e-teleco.com Mon Dec 31 01:11:02 2018 From: bjordan at e-teleco.com (bjordan at e-teleco.com) Date: Mon, 31 Dec 2018 01:11:02 +0000 Subject: [Freeswitch-users] Controlling ptime in INVITE In-Reply-To: <0D1C4B76-8A06-4AB5-9E38-3B6585B78427@tm.net.uk> References: <7faf9064-24a8-12ac-3360-bd3de5205a10@bohboh.info> <0D1C4B76-8A06-4AB5-9E38-3B6585B78427@tm.net.uk> Message-ID: I am not familiar if you can mismatch ptimes on A and B legs (I don’t see why it wouldn’t work though I would assume FS would transcode appropriately) but if you set the param on the bridge to that provider it could solve the issue. To answer your question you could set it in on the inbound allowed codec by setting the global_codec_prefs in vars.xml, I think you can do the same syntax with the @20i but if you do that in the inbound codecs and the Cisco phone doesn’t renegotiate with your specified ptime it seems possible the call could be lost since there is no valid codec? Maybe someone else could chime in who has more experience with Cisco phones and their behavior. You could also set it up on the sofia sip profile if you don’t want to do it globally. https://freeswitch.org/confluence/display/FREESWITCH/Codecs+and+Media#CodecsandMedia-sofia.conf.xmlfile Thanks, Branden Jordan From: FreeSWITCH-users On Behalf Of Joseph Waite Sent: Friday, December 28, 2018 2:07 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Controlling ptime in INVITE I I haven’t. Is there a way to set this in the inbound allowed codec?? Joe Waite On 28 Dec 2018, at 20:24, Social Boh > wrote: Hello, Are you tried to use absolute_codec_string in the outbound block of your dialplan? like: Regards --- I'm SoCIaL, MayBe El 28/12/2018 a las 11:05, Joseph Waite escribió: Evening All. I am having an issue, currently with Cisco SPA phones, where they are trying to use a prime of 30, however one of our providers does not support this and fails the call. Customer does not notice as call fails over to another provider as it should, however the failover provider is more expensive so not ideal for us. We don’t notice this unless we examine cdr logs. Is there a way, either to tell freeswitch to send a response to the phone negotiating time of 20 or to refuse the call if it comes in with time of 30? 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: From sagarmalam at gmail.com Mon Dec 31 06:29:35 2018 From: sagarmalam at gmail.com (sagar malam) Date: Mon, 31 Dec 2018 11:59:35 +0530 Subject: [Freeswitch-users] Disabling 200 OK with contact headers of all the lines In-Reply-To: References: <08A105FB-0134-4ABF-B134-171C928C14F3@freeswitch.org> Message-ID: Micheal, Happy New Year ! Your thoughts on this is very much appreciated.Thanks On Wed, Dec 19, 2018 at 1:24 PM sagar malam wrote: > Thanks for information Mike. > > Yes,FS is doing right thing as per RFC.Also i check OpenSIPS and Kamailio > doing same thing(i made mistake during first observation) , only difference > is that it uses comma separated string of contacts in single contact header > instead of separate contact headers for each registration. > > Looking into sofia source code, I found that FS had option to not look up > in DB and return contact header(only one) as it was in register packet ( > *reg-deny-binding-fetch-and-no-lookup* ).This is exactly what i was > looking for and it solved my problem as well with polycom phones. But it is > deprecated by FS. And not recommended by RFC. > > However i am curious to know what is purpose of it ? I was not able to > find any specific reason for 200 OK to have contacts of all the bindings.Do > you know why ? > > I can clearly find out below *issues* due to multiple contact headers : > 1) Having multiple contact headers may confuse client to identify their > own contact header and therefore may not be able to get registration expiry > time provided by server.It happens with polycom phones for sure.( > https://support.onsip.com/hc/en-us/articles/204028710-Polycom-Multiple-Contacts-Registration-Bug > ) > 2) Very big SIP packets( consider a case of 20 registrations with same > AOR).This wont be any issue while using TCP or UDP with IPv4 but if we use > UDP with IPv6 , too big packets will be dropped by Ethernet ports because > as per IPv6 standards only applications(FS) can fragment IPv6 UDP packets. > 3) For each register request FS will execute query in DB to fetch all the > registrations which adds overhead on DB as well. > > Looking forward for your thoughts.Thanks again. > > On Tue, Dec 18, 2018 at 2:05 AM Mike Jerris wrote: > >> https://tools.ietf.org/html/rfc3261#section-10.3 >> >> 8. The registrar returns a 200 (OK) response. The response MUST >> contain Contact header field values enumerating all current >> bindings. Each Contact value MUST feature an "expires" >> parameter indicating its expiration interval chosen by the >> registrar. The response SHOULD include a Date header field. >> >> >> >> On Dec 17, 2018, at 9:13 AM, sagar malam wrote: >> >> Thanks for reply Micheal. >> >> I dont want to disable multiple registrations.And i agree all the >> registrations are genuine. >> >> But my problem is that when FS responds to Register Request using 200 OK, >> The 200 OK has contact headers of all the registrations( of other SIP >> clients).As shown in below packet, there are three contact headers.I think >> there should be only one. >> >> ============================200 OK for register packet ============= >> >> 2018/12/08 13:36:35.426099 10.50.7.251:5070 -> 10.50.7.253:5060 >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 66.160.237.253:5060 >> ;branch=z9hG4bK9b46.f9b7a68125108f411537617d02bd48f8.0;received=10.50.7.253 >> Via: SIP/2.0/UDP 198.136.236.1:5060 >> ;rport=5060;received=10.50.8.1;branch=z9hG4bK9b46.e6f1aa3817a238f499aa3aafa4217425.0 >> Via: SIP/2.0/UDP >> 172.16.1.11;rport=1426;received=71.239.113.14;branch=z9hG4bK1df2302cD062D28F >> From: "Main Line" ;tag=22CE54C0-84BCCF23 >> To: ;tag=e8Fa2tND4U80D >> Call-ID: d66f0f0592c09746b903406f312eb0c2 >> CSeq: 590 REGISTER >> *Contact: >> > *t=tcp>;expires=379* >> *Contact: >> > *ort=tcp>;expires=245* >> *Contact: >> > *eb0c2>;expires=88* >> Date: Sat, 08 Dec 2018 08:06:35 GMT >> User-Agent: FreeSWITCH-mod_sofia/1.6.17~64bit >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: path, replaces >> Content-Length: 0 >> >> =========================================================== >> >> Also i have compared this behaviour with OpenSIPs and Kamailio, They send >> only one contact header. >> >> Thanks in advance. >> >> >> >> On Wed, Dec 12, 2018 at 3:08 AM Michael Jerris wrote: >> >>> These are all valid current registrations, there are params you can >>> adjust for multi reg that will replace previous registrations instead of >>> allowing multiple. >>> >>> On Dec 9, 2018, at 1:17 PM, sagar malam wrote: >>> >>> Hello , >>> >>> I am using FS SLA feature and it works very well.However i am facing an >>> issue of registrations getting dropped as explained below : >>> There are 3 phones registered with same extension number.Sofia is >>> configured with "sip-expires-max-deviation" to randomise registration >>> through expiry header in contact header.All the phones are Polycom. In case >>> of shared lines FS adds contact header of all the lines(or phones with same >>> extension) in 200 OK as shown below due to which all the phones are reading >>> expiry timer from first contact header only.So in below example,Phone re >>> registers after 379 seconds(first contact header) instead of 88 >>> seconds(third contact header) leading to registration expiry on FS. >>> >>> Reason why phones are always reading expiry from first contact header is >>> same Public IP(contact is re written by Proxy in front of FS) for all >>> three phones which is confusing phone to identify its own contact header. >>> Is there any way to configure FS to not send contact headers of all the >>> registrations but only one that belongs to the line itself ? or any other >>> way to fix it. >>> ============================200 OK for register packet ============= >>> >>> 2018/12/08 13:36:35.426099 10.50.7.251:5070 -> 10.50.7.253:5060 >>> SIP/2.0 200 OK >>> Via: SIP/2.0/UDP 66.160.237.253:5060 >>> ;branch=z9hG4bK9b46.f9b7a68125108f411537617d02bd48f8.0;received=10.50.7.253 >>> Via: SIP/2.0/UDP 198.136.236.1:5060 >>> ;rport=5060;received=10.50.8.1;branch=z9hG4bK9b46.e6f1aa3817a238f499aa3aafa4217425.0 >>> Via: SIP/2.0/UDP >>> 172.16.1.11;rport=1426;received=71.239.113.14;branch=z9hG4bK1df2302cD062D28F >>> From: "Main Line" ;tag=22CE54C0-84BCCF23 >>> To: ;tag=e8Fa2tND4U80D >>> Call-ID: d66f0f0592c09746b903406f312eb0c2 >>> CSeq: 590 REGISTER >>> Contact: < >>> sip:398 at 71.239.113.14:1071;alias=10.50.8.1~5060~1;x-nat=yes;pv-ip=172.16.1.12;pb-ip=71.239.113.14;pb-pt=1071;mac-address=64167f2ec274;transp >>> t=tcp>;expires=379 >>> Contact: < >>> sip:398 at 71.239.113.14:56478;alias=10.50.8.1~5060~1;x-nat=yes;pv-ip=172.16.1.13;pb-ip=71.239.113.14;pb-pt=56478;mac-address=64167f2ebd54;tran >>> ort=tcp>;expires=245 >>> Contact: < >>> sip:398 at 71.239.113.14:1426;alias=10.50.8.1~5060~1;x-nat=yes;pv-ip=172.16.1.11;pb-ip=71.239.113.14;pb-pt=1426;transport=udp;mac-address=64167 >>> eb0c2>;expires=88 >>> Date: Sat, 08 Dec 2018 08:06:35 GMT >>> User-Agent: FreeSWITCH-mod_sofia/1.6.17~64bit >>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, >>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >>> Supported: path, replaces >>> Content-Length: 0 >>> >>> >> _________________________________________________________________________ >> 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 > > > > -- > Thanks, > > Sagar > -- Thanks, Sagar -------------- next part -------------- An HTML attachment was scrubbed... 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